From freeswitch-list at puzzled.xs4all.nl Thu Mar 1 00:14:25 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 29 Feb 2012 22:14:25 +0100 Subject: [Freeswitch-users] mod perl prob - ExtUtils/Embed.pm In-Reply-To: References: <4F4E7005.8090502@puzzled.xs4all.nl> Message-ID: <4F4E9531.1050209@puzzled.xs4all.nl> On 29-02-12 20:21, spyros papadopoulos wrote: > Thank you. The problem was resolved by explicitly including the > perl-ExtUtils-Embed package > thanks again, My pleasure but the output from the command I gave you basically tells you that you should install perl-devel since that seems to be missing: $ yum provides "*/EXTERN.h" Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile 4:perl-devel-5.10.1-119.el6_1.1.i686 : Header files for use in perl development Repo : base-local Matched from: Filename : /usr/lib/perl5/CORE/EXTERN.h But if including perl-ExtUtils-Embed works for you then good for you :) Regards, Patrick From anita.hall at simmortel.com Thu Mar 1 00:41:51 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Thu, 1 Mar 2012 03:11:51 +0530 Subject: [Freeswitch-users] 401 Unauthorized in iptel external gateway In-Reply-To: <1067670332.26333.1330516671550.JavaMail.root@mailserver.edistar.com> References: <1067670332.26333.1330516671550.JavaMail.root@mailserver.edistar.com> Message-ID: Yes, Denis that would be useful as well. For instance, does FS send 2 REGISTER requests even then? What are the addresses in sent and received SIP messages ? Moreover, you could remove FreeSWITCH from the equation altogether and try using Sofia SIP as a stand-alone client. http://sofia-sip.sourceforge.net/ regards, Anita On Wed, Feb 29, 2012 at 5:27 PM, Denis Gasparin wrote: > > Do you mean to do a sip trace of a working call to the iptel account using > a sip phone registered to freeswitch (without the gateway enabled)? > > Please let me know so I'll provide you the correct data. > > Thank you > > Denis > > ----- Messaggio originale ----- > > > Da: "Anita Hall" > A: "FreeSWITCH Users Help" > Inviato: Mercoled?, 29 febbraio 2012 12:27:24 > Oggetto: Re: [Freeswitch-users] 401 Unauthorized in iptel external gateway > > I could figure out this much. > > In the second case (not working with correct config), FS is not sending > credentials because perhaps there is a mismatch in "Via" in request and > response. In the first case (working with incorrect config), there is no > such mismatch in "Via" > > Request: Via: SIP/2.0/UDP 80.116.149.239:5080 > Response: Via: SIP/2.0/UDP 192.168.1.9:5080:5080 > > Why is the port 5080 twice in Response ? > > We need to see if this is the case using Sofia SIP client as well. > > regards, > Anita > > > > > On Wed, Feb 29, 2012 at 1:50 PM, Denis Gasparin < > denis.gasparin at edistar.com > wrote: > > > > I forgot to say that freeswitch uses UPnP to do nat-traversal. > > Any tip after reading the SIP trace? > > > Thank you > Denis > > > ----- Messaggio originale ----- > > Da: "Denis Gasparin" < denis.gasparin at edistar.com > > > > A: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > > > Inviato: Marted?, 28 febbraio 2012 12:11:00 > > > > Oggetto: Re: [Freeswitch-users] 401 Unauthorized in iptel external > gateway > > Hi Michael. > > > > > ============================================================================ > > SIP trace of working but wrong configuration > > > ============================================================================ > > > > > > > > > > > > > > send 649 bytes to udp/[217.9.36.145]:5060 at 06:39:58.756531: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 192.168.1.9:5080;rport;branch=z9hG4bKH30c2j9mZ70jr > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=HS8t79j0cUccQ > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: 1ab7323f-3052-4c40-8b27-d329ebeb2d17 > > CSeq: 24869535 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 713 bytes from udp/[217.9.36.145]:5060 at 06:39:58.853547: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080;rport=27905;branch=z9hG4bKH30c2j9mZ70jr > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=HS8t79j0cUccQ > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-0631 > > Call-ID: 1ab7323f-3052-4c40-8b27-d329ebeb2d17 > > CSeq: 24869535 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x4Rk9MeAoHmuhZYnrVV44Ld9MXbGD7" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=865 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > send 854 bytes to udp/[217.9.36.145]:5060 at 06:39:58.854065: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 192.168.1.9:5080;rport;branch=z9hG4bKjct53DtrvgQ5K > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=HS8t79j0cUccQ > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: 1ab7323f-3052-4c40-8b27-d329ebeb2d17 > > CSeq: 24869536 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Authorization: Digest username="MY_IPTEL_ACCOUNT", realm=" iptel.org ", > > nonce="T0x4Rk9MeAoHmuhZYnrVV44Ld9MXbGD7", algorithm=MD5, > > uri="sip: sip.iptel.org ;transport=udp", > > response="b78f8f127a20a7ade3ed8a9041162e89" > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 694 bytes from udp/[217.9.36.145]:5060 at 06:39:58.956413: > > ------------------------------------------------------------------------ > > SIP/2.0 200 OK > > Via: SIP/2.0/UDP > > 192.168.1.9:5080;rport=27905;branch=z9hG4bKjct53DtrvgQ5K > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=HS8t79j0cUccQ > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=5697f7a671feb5d42dc9ea510e728105.2ab2 > > Call-ID: 1ab7323f-3052-4c40-8b27-d329ebeb2d17 > > CSeq: 24869536 REGISTER > > Expires: 600 > > Min-Expires: 240 > > Contact: > > ;expires=600 > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=864 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > > ============================================================================ > > SIP trace of not working but correct configuration > > > ============================================================================ > > > > > > > > > > > > > > ==> external_rtp_ip and external_sip_ip are set to my public ip > > (80.116.149.239) in vars.xml. > > > > send 655 bytes to udp/[217.9.36.145]:5060 at 06:42:20.692370: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:20.792006: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x41k9MeJptTRxwtIC32TRVrt1Ou+pd" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=869 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > send 655 bytes to udp/[217.9.36.145]:5060 at 06:42:21.692837: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:21.793018: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x41k9MeJptTRxwtIC32TRVrt1Ou+pd" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=869 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > send 655 bytes to udp/[217.9.36.145]:5060 at 06:42:23.692835: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:23.790966: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x41k9MeJptTRxwtIC32TRVrt1Ou+pd" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=869 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > freeswitch at dhcppc7> > > freeswitch at dhcppc7> > > freeswitch at dhcppc7> send 655 bytes to udp/[217.9.36.145]:5060 at > > 06:42:27.692903: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:27.792597: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x43U9MeKEIS4uLX1oGSBQjlCJ7vtme" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=864 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > freeswitch at dhcppc7> > > freeswitch at dhcppc7> send 655 bytes to udp/[217.9.36.145]:5060 at > > 06:42:31.692890: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:31.793652: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x43U9MeKEIS4uLX1oGSBQjlCJ7vtme" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=864 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > freeswitch at dhcppc7> send 655 bytes to udp/[217.9.36.145]:5060 at > > 06:42:35.692896: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:35.794678: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x45U9MeKkNgVsi09O8xEmjpBaZdLGh" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=864 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > freeswitch at dhcppc7> send 655 bytes to udp/[217.9.36.145]:5060 at > > 06:42:39.692893: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:39.794542: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x45U9MeKkNgVsi09O8xEmjpBaZdLGh" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=864 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > send 655 bytes to udp/[217.9.36.145]:5060 at 06:42:43.692953: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:43.794321: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x47U9MeLEkEcREBIX7+GErgu7sZafW" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=865 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > freeswitch at dhcppc7> send 655 bytes to udp/[217.9.36.145]:5060 at > > 06:42:47.692951: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:47.793749: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x47U9MeLEkEcREBIX7+GErgu7sZafW" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=865 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > freeswitch at dhcppc7> send 655 bytes to udp/[217.9.36.145]:5060 at > > 06:42:51.692916: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:51.794179: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x49U9MeLlULFCzzoULNr3vTAKpPBHQ" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=863 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > freeswitch at dhcppc7> 2012-02-28 07:42:52.679130 [ERR] sofia_reg.c:1962 > > iptel Registration Failed with status Request Timeout [408]. failure > > #1 > > freeswitch at dhcppc7> 2012-02-28 07:42:54.619987 [WARNING] > > sofia_reg.c:474 iptel Failed Registration [0], setting retry to 60 > > seconds. > > > > > > Thank you > > > > Denis > > > > ----- Messaggio originale ----- > > > Da: "Michael Collins" < msc at freeswitch.org > > > > A: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > > > > Inviato: Marted?, 28 febbraio 2012 1:15:43 > > > Oggetto: Re: [Freeswitch-users] 401 Unauthorized in iptel external > > > gateway > > > SIP trace would probably reveal why. > > > -MC > > > > > > > > > On Mon, Feb 27, 2012 at 3:53 PM, Denis Gasparin < > > > denis.gasparin at edistar.com > wrote: > > > > > > > > > > > > Sorry but I don't understand. I agree with you that those values are > > > invalid but using those values registration and rtp work. > > > > > > The (correct, I hope) configuration I should use according to the > > > wiki > > > in sip_profiles/external.xml: > > > > > > > > > > > > > > > > > > > > > external_rtp_ip and external_sip_ip are set to my public ip in > > > vars.xml. > > > With this configuration registration does not work. > > > > > > But with this (wrong) configuration registration and communication > > > with iptel works fine: > > > > > > > > > > > > > > value="**invalid_value_or_my_private_ip_address**"/> > > > > > value="**invalid_value_or_my_private_ip_address**"/> > > > > > > > > > Why? > > > > > > Thank you, > > > Denis > > > > > > ----- Messaggio originale ----- > > > > Da: "Michael Collins" < msc at freeswitch.org > > > > > A: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > > > > > > > > > Inviato: Luned?, 27 febbraio 2012 23:30:33 > > > > Oggetto: Re: [Freeswitch-users] 401 Unauthorized in iptel external > > > > gateway > > > > On Mon, Feb 27, 2012 at 2:08 PM, Brian West < brian at freeswitch.org > > > > > > > > > wrote: > > > > > > > > > > > > > > > > That would be because these values are INVALID. > > > > > > > > > > > > /b > > > > > > > > > > > > Indeed. Check this info for suggestions on what can go there: > > > > > > > > http://wiki.freeswitch.org/wiki/Sofia.conf.xml#ext-rtp-ip > > > > > > > > -MC > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/526b8710/attachment-0001.html From msc at freeswitch.org Thu Mar 1 00:56:31 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 Feb 2012 13:56:31 -0800 Subject: [Freeswitch-users] 401 Unauthorized in iptel external gateway In-Reply-To: <1067670332.26333.1330516671550.JavaMail.root@mailserver.edistar.com> References: <1067670332.26333.1330516671550.JavaMail.root@mailserver.edistar.com> Message-ID: And use pastebin.freeswitch.org so that you don't overwhelm my mail client. :) -MC On Wed, Feb 29, 2012 at 3:57 AM, Denis Gasparin wrote: > > Do you mean to do a sip trace of a working call to the iptel account using > a sip phone registered to freeswitch (without the gateway enabled)? > > Please let me know so I'll provide you the correct data. > > Thank you > > Denis > > ----- Messaggio originale ----- > > > Da: "Anita Hall" > A: "FreeSWITCH Users Help" > Inviato: Mercoled?, 29 febbraio 2012 12:27:24 > Oggetto: Re: [Freeswitch-users] 401 Unauthorized in iptel external gateway > > I could figure out this much. > > In the second case (not working with correct config), FS is not sending > credentials because perhaps there is a mismatch in "Via" in request and > response. In the first case (working with incorrect config), there is no > such mismatch in "Via" > > Request: Via: SIP/2.0/UDP 80.116.149.239:5080 > Response: Via: SIP/2.0/UDP 192.168.1.9:5080:5080 > > Why is the port 5080 twice in Response ? > > We need to see if this is the case using Sofia SIP client as well. > > regards, > Anita > > > > > On Wed, Feb 29, 2012 at 1:50 PM, Denis Gasparin < > denis.gasparin at edistar.com > wrote: > > > > I forgot to say that freeswitch uses UPnP to do nat-traversal. > > Any tip after reading the SIP trace? > > > Thank you > Denis > > > ----- Messaggio originale ----- > > Da: "Denis Gasparin" < denis.gasparin at edistar.com > > > > A: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > > > Inviato: Marted?, 28 febbraio 2012 12:11:00 > > > > Oggetto: Re: [Freeswitch-users] 401 Unauthorized in iptel external > gateway > > Hi Michael. > > > > > ============================================================================ > > SIP trace of working but wrong configuration > > > ============================================================================ > > > > > > > > > > > > > > send 649 bytes to udp/[217.9.36.145]:5060 at 06:39:58.756531: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 192.168.1.9:5080;rport;branch=z9hG4bKH30c2j9mZ70jr > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=HS8t79j0cUccQ > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: 1ab7323f-3052-4c40-8b27-d329ebeb2d17 > > CSeq: 24869535 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 713 bytes from udp/[217.9.36.145]:5060 at 06:39:58.853547: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080;rport=27905;branch=z9hG4bKH30c2j9mZ70jr > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=HS8t79j0cUccQ > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-0631 > > Call-ID: 1ab7323f-3052-4c40-8b27-d329ebeb2d17 > > CSeq: 24869535 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x4Rk9MeAoHmuhZYnrVV44Ld9MXbGD7" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=865 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > send 854 bytes to udp/[217.9.36.145]:5060 at 06:39:58.854065: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 192.168.1.9:5080;rport;branch=z9hG4bKjct53DtrvgQ5K > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=HS8t79j0cUccQ > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: 1ab7323f-3052-4c40-8b27-d329ebeb2d17 > > CSeq: 24869536 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Authorization: Digest username="MY_IPTEL_ACCOUNT", realm=" iptel.org ", > > nonce="T0x4Rk9MeAoHmuhZYnrVV44Ld9MXbGD7", algorithm=MD5, > > uri="sip: sip.iptel.org ;transport=udp", > > response="b78f8f127a20a7ade3ed8a9041162e89" > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 694 bytes from udp/[217.9.36.145]:5060 at 06:39:58.956413: > > ------------------------------------------------------------------------ > > SIP/2.0 200 OK > > Via: SIP/2.0/UDP > > 192.168.1.9:5080;rport=27905;branch=z9hG4bKjct53DtrvgQ5K > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=HS8t79j0cUccQ > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=5697f7a671feb5d42dc9ea510e728105.2ab2 > > Call-ID: 1ab7323f-3052-4c40-8b27-d329ebeb2d17 > > CSeq: 24869536 REGISTER > > Expires: 600 > > Min-Expires: 240 > > Contact: > > ;expires=600 > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=864 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > > ============================================================================ > > SIP trace of not working but correct configuration > > > ============================================================================ > > > > > > > > > > > > > > ==> external_rtp_ip and external_sip_ip are set to my public ip > > (80.116.149.239) in vars.xml. > > > > send 655 bytes to udp/[217.9.36.145]:5060 at 06:42:20.692370: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:20.792006: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x41k9MeJptTRxwtIC32TRVrt1Ou+pd" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=869 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > send 655 bytes to udp/[217.9.36.145]:5060 at 06:42:21.692837: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:21.793018: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x41k9MeJptTRxwtIC32TRVrt1Ou+pd" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=869 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > send 655 bytes to udp/[217.9.36.145]:5060 at 06:42:23.692835: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:23.790966: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x41k9MeJptTRxwtIC32TRVrt1Ou+pd" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=869 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > freeswitch at dhcppc7> > > freeswitch at dhcppc7> > > freeswitch at dhcppc7> send 655 bytes to udp/[217.9.36.145]:5060 at > > 06:42:27.692903: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:27.792597: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x43U9MeKEIS4uLX1oGSBQjlCJ7vtme" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=864 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > freeswitch at dhcppc7> > > freeswitch at dhcppc7> send 655 bytes to udp/[217.9.36.145]:5060 at > > 06:42:31.692890: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:31.793652: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x43U9MeKEIS4uLX1oGSBQjlCJ7vtme" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=864 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > freeswitch at dhcppc7> send 655 bytes to udp/[217.9.36.145]:5060 at > > 06:42:35.692896: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:35.794678: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x45U9MeKkNgVsi09O8xEmjpBaZdLGh" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=864 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > freeswitch at dhcppc7> send 655 bytes to udp/[217.9.36.145]:5060 at > > 06:42:39.692893: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:39.794542: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x45U9MeKkNgVsi09O8xEmjpBaZdLGh" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=864 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > send 655 bytes to udp/[217.9.36.145]:5060 at 06:42:43.692953: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:43.794321: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x47U9MeLEkEcREBIX7+GErgu7sZafW" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=865 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > freeswitch at dhcppc7> send 655 bytes to udp/[217.9.36.145]:5060 at > > 06:42:47.692951: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:47.793749: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x47U9MeLEkEcREBIX7+GErgu7sZafW" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=865 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > freeswitch at dhcppc7> send 655 bytes to udp/[217.9.36.145]:5060 at > > 06:42:51.692916: > > ------------------------------------------------------------------------ > > REGISTER sip: sip.iptel.org ;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 80.116.149.239:5080;rport;branch=z9hG4bKN0cSvF0B4Q24D > > Max-Forwards: 70 > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp> > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Contact: > > Expires: 3600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 > > 14-15-32 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > > REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > recv 718 bytes from udp/[217.9.36.145]:5060 at 06:42:51.794179: > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > > 192.168.1.9:5080:5080;rport=27905;branch=z9hG4bKN0cSvF0B4Q24D > > From: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org ;transport=udp>;tag=Ntyatg8t0eXvN > > To: > > < sip:MY_IPTEL_ACCOUNT at sip.iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-d9e7 > > Call-ID: e83d7bd7-f358-44c0-95d9-9ec9069c9711 > > CSeq: 24869606 REGISTER > > Expires: 600 > > Min-Expires: 240 > > WWW-Authenticate: Digest realm=" iptel.org ", > > nonce="T0x49U9MeLlULFCzzoULNr3vTAKpPBHQ" > > Server: ser (3.3.0-dev0 (i386/linux)) > > Content-Length: 0 > > Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=863 > > req_src_ip=80.116.149.239 req_src_port=27905 > > in_uri=sip: sip.iptel.org ;transport=udp > > out_uri=sip: sip.iptel.org ;transport=udp via_cnt==1" > > > > ------------------------------------------------------------------------ > > > > freeswitch at dhcppc7> 2012-02-28 07:42:52.679130 [ERR] sofia_reg.c:1962 > > iptel Registration Failed with status Request Timeout [408]. failure > > #1 > > freeswitch at dhcppc7> 2012-02-28 07:42:54.619987 [WARNING] > > sofia_reg.c:474 iptel Failed Registration [0], setting retry to 60 > > seconds. > > > > > > Thank you > > > > Denis > > > > ----- Messaggio originale ----- > > > Da: "Michael Collins" < msc at freeswitch.org > > > > A: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > > > > Inviato: Marted?, 28 febbraio 2012 1:15:43 > > > Oggetto: Re: [Freeswitch-users] 401 Unauthorized in iptel external > > > gateway > > > SIP trace would probably reveal why. > > > -MC > > > > > > > > > On Mon, Feb 27, 2012 at 3:53 PM, Denis Gasparin < > > > denis.gasparin at edistar.com > wrote: > > > > > > > > > > > > Sorry but I don't understand. I agree with you that those values are > > > invalid but using those values registration and rtp work. > > > > > > The (correct, I hope) configuration I should use according to the > > > wiki > > > in sip_profiles/external.xml: > > > > > > > > > > > > > > > > > > > > > external_rtp_ip and external_sip_ip are set to my public ip in > > > vars.xml. > > > With this configuration registration does not work. > > > > > > But with this (wrong) configuration registration and communication > > > with iptel works fine: > > > > > > > > > > > > > > value="**invalid_value_or_my_private_ip_address**"/> > > > > > value="**invalid_value_or_my_private_ip_address**"/> > > > > > > > > > Why? > > > > > > Thank you, > > > Denis > > > > > > ----- Messaggio originale ----- > > > > Da: "Michael Collins" < msc at freeswitch.org > > > > > A: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > > > > > > > > > Inviato: Luned?, 27 febbraio 2012 23:30:33 > > > > Oggetto: Re: [Freeswitch-users] 401 Unauthorized in iptel external > > > > gateway > > > > On Mon, Feb 27, 2012 at 2:08 PM, Brian West < brian at freeswitch.org > > > > > > > > > wrote: > > > > > > > > > > > > > > > > That would be because these values are INVALID. > > > > > > > > > > > > /b > > > > > > > > > > > > Indeed. Check this info for suggestions on what can go there: > > > > > > > > http://wiki.freeswitch.org/wiki/Sofia.conf.xml#ext-rtp-ip > > > > > > > > -MC > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120229/5b39d460/attachment-0001.html From msc at freeswitch.org Thu Mar 1 01:04:25 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 Feb 2012 14:04:25 -0800 Subject: [Freeswitch-users] Test a dialplan from fs_cli In-Reply-To: References: Message-ID: On Wed, Feb 29, 2012 at 8:17 AM, Brian Foster wrote: > I'm interested in an example as well; doesn't seem to be documented well > on the wiki. > > -BDF > How about on page 86 of the FreeSWITCH bridge book? :D -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120229/f8af3d87/attachment.html From msc at freeswitch.org Thu Mar 1 01:13:59 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 Feb 2012 14:13:59 -0800 Subject: [Freeswitch-users] attended transfer cdr In-Reply-To: <4F4DCCDB.8020405@softnet.si> References: <4F4B8A64.9000705@softnet.si> <4F4DCCDB.8020405@softnet.si> Message-ID: > > thanks for that. Now I have to somehow filter calls, so that I will not > have duplicated values. Any idea:)? > Very carefully? :P I don't have any words of wisdom other than to make a lot of test calls in a controlled environment and make note of what kinds of CDRs you get. You will notice that you get multiple CDRs for various kinds of activity. And as always, the XML CDRs have more information, including call flow, so if you learn how to parse those you will be doing yourself a huge favor. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120229/667e4f5e/attachment.html From bob.mccarthy at experient.com Thu Mar 1 01:18:37 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Wed, 29 Feb 2012 15:18:37 -0700 Subject: [Freeswitch-users] CallerID changing after Bridge using SLA on Polycom Phones In-Reply-To: <5003D7D3E06F514E8C682F18D223265C05121CD440@AZWSMS03.azwarranty.int> References: <001301ccf6ca$b15bec80$1413c580$@mccarthy@experient.com> <5003D7D3E06F514E8C682F18D223265C05121CD440@AZWSMS03.azwarranty.int> Message-ID: <00dc01ccf730$16b763b0$44262b10$@mccarthy@experient.com> worked like a champ! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Weigel, Stefan Sent: Wednesday, February 29, 2012 8:17 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] CallerID changing after Bridge using SLA on Polycom Phones Hi, that's working for me. [..] [..] then do the bridge. Best regards, Stefan Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Bob McCarthy Gesendet: Mittwoch, 29. Februar 2012 11:13 An: freeswitch-users at lists.freeswitch.org Cc: bill.oneil at experient.com Betreff: [Freeswitch-users] CallerID changing after Bridge using SLA on Polycom Phones I am having an issue similar to that experienced by Wellie Chao in his email "Caller ID on inbound calls on Polycom" 2010/4/22. The call is from an Audiocodes gateway with a dial string of *NXXXXXX# CallerID number sent is 101 which corresponds to the FXS port it came from, there is no CallerID Name. I modify the CallerID name and number in the dialplan before the bridge. On Ring the CallerID name and number are what I have changed it to. On Bridge, 80% of the time the CallerID name and number follow thru, 20% of the time it changes to what Leg A had originally before I used the export application.( name is blank and number is 101). if I put the call on Hold it reverts back to what I expected. I have tried to use set, moved the import after the bridge, ignore_display_updates=true. Any Ideas? On a further note, when phone 2 presses the shared line it shows the same CallerID name and number as phone 1. When you press BargeIn both the CallerID name and number change to 101. i.e. name is 101 number is 101 on both phones. this happens every time, is there a way to stop this behavior as well ? Thanks Bob McCarthy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120229/83a55a4e/attachment.html From abaci64 at gmail.com Thu Mar 1 01:30:39 2012 From: abaci64 at gmail.com (a b) Date: Wed, 29 Feb 2012 17:30:39 -0500 Subject: [Freeswitch-users] Segmentation fault Message-ID: Hi, I just noticed that if I bridge a call to a user that exists in the directory but doesn't have the dial-string param set, FreeSWITCH will crash. this is happening on git from yesterday. can someone confirm if this is a bug or something in my setup? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120229/91b86530/attachment.html From msc at freeswitch.org Thu Mar 1 02:12:08 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 Feb 2012 15:12:08 -0800 Subject: [Freeswitch-users] Segmentation fault In-Reply-To: References: Message-ID: Abaci, Thanks for the information. It would help us tremendously if you could go over to jira.freeswitch.org and file a bug report - that will help us keep track of the issue. Once you create the ticket the devs will be notified and if we need assistance with testing they'll put the word out. Thanks, MC On Wed, Feb 29, 2012 at 2:30 PM, a b wrote: > Hi, > I just noticed that if I bridge a call to a user that exists in the > directory but doesn't have the dial-string param set, FreeSWITCH will > crash. this is happening on git from yesterday. can someone confirm if this > is a bug or something in my setup? > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120229/2a4ef13e/attachment-0001.html From anthony.minessale at gmail.com Thu Mar 1 02:13:39 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 29 Feb 2012 17:13:39 -0600 Subject: [Freeswitch-users] Segmentation fault In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Getting_a_Backtrace On Wed, Feb 29, 2012 at 4:30 PM, a b wrote: > Hi, > I just noticed that if I bridge a call to a user that exists in the > directory but doesn't have the dial-string param set, FreeSWITCH will crash. > this is happening on git from yesterday. can?someone?confirm if this is a > bug or something in my setup? > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From mike at jerris.com Thu Mar 1 02:58:18 2012 From: mike at jerris.com (Michael Jerris) Date: Wed, 29 Feb 2012 18:58:18 -0500 Subject: [Freeswitch-users] fail to build git version of freeswitch In-Reply-To: <4F4E7286.2020009@puzzled.xs4all.nl> References: <4F4B70F5.2090009@gmail.com> <4F4BFE2F.9080108@privatedemail.net> <4F4CDDB6.5050706@privatedemail.net> <1330442150333-7325765.post@n2.nabble.com> <4F4D7A5D.5080907@privatedemail.net> <1330489081107-7328048.post@n2.nabble.com> <4F4E47F3.9010300@privatedemail.net> <4F4E543D.2050700@privatedemail.net> <4F4E59C6.9050409@privatedemail.net> <4F4E7286.2020009@puzzled.xs4all.nl> Message-ID: This message thread has been closed. Please continue any other technical discussion of this issue on jira. Mike On Feb 29, 2012, at 1:46 PM, Patrick Lists wrote: > On 29-02-12 18:00, Josh wrote: >> >>> That's my point. No need to bring it on to the mailing list, where now >>> you are spreading information on a specific bug report in several >>> different places. Now devs have to spend MORE time trying to fit all >>> the pieces together. Get it? >> The FS devs don't need to piece anything "together from several >> different places" simply because all the information they need to make a >> decision is already there - on JIRA, and has been since November 2011 by >> the look of things. If this issue was resolved in a timely manner, then >> there won't be multiple reports of it and it is a dead cert I won't be >> wasting any of my time on this here with you. > > Would you mind toning it done a bit and be a bit more courteous? Frankly > I perceive your comments as increasingly rude, especially since you have > paid ZERO for this fabulous piece of software and yet you feel it is > appropriate to mention again and again that - in your opinion - the > developers are slow and don't move forward as fast as you seem to demand > they should. You can always ask your money back and write your own > FreeSWITCH... > > Regards, > Patrick From msc at freeswitch.org Thu Mar 1 03:00:37 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 Feb 2012 16:00:37 -0800 Subject: [Freeswitch-users] fail to build git version of freeswitch In-Reply-To: <1330489081107-7328048.post@n2.nabble.com> References: <4F4B70F5.2090009@gmail.com> <4F4BFE2F.9080108@privatedemail.net> <4F4CDDB6.5050706@privatedemail.net> <1330442150333-7325765.post@n2.nabble.com> <4F4D7A5D.5080907@privatedemail.net> <1330489081107-7328048.post@n2.nabble.com> Message-ID: Die thread die! *STAB* *STAB* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120229/1a19d5ff/attachment.html From msc at freeswitch.org Thu Mar 1 03:38:19 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 Feb 2012 16:38:19 -0800 Subject: [Freeswitch-users] Mailing List Reminders Message-ID: Hello All! It seems we've had a bit of drama on the mailing list (ML) lately so I thought it would be good to take this opportunity to send out a few reminders: - This is a public list, so anything you say will be forever etched into the annals of Internet lore. Please think before you click Send. - This is a public list, so be aware that others may say things that are offensive to you personally but not to the subscribers in general. Please be ready to "deal with it" on occasion. - This list is moderated to a degree. We keep only a loose reign on the users here. However, if after several warnings a user repeatedly engages in trollish behavior then he or she WILL get moderated. Those who are incorrigibly trollish will be banned. I appreciate that most of our readers don't need these reminders, so thank you for indulging us. Keep up the good work! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120229/b9ac040e/attachment.html From david at styleflare.com Thu Mar 1 03:53:46 2012 From: david at styleflare.com (David J) Date: Wed, 29 Feb 2012 19:53:46 -0500 Subject: [Freeswitch-users] Mailing List Reminders In-Reply-To: References: Message-ID: S--w you Collins! (Only kidding) On Feb 29, 2012 7:39 PM, "Michael Collins" wrote: > Hello All! > > It seems we've had a bit of drama on the mailing list (ML) lately so I > thought it would be good to take this opportunity to send out a few > reminders: > > > - This is a public list, so anything you say will be forever etched > into the annals of Internet lore. Please think before you click Send. > - This is a public list, so be aware that others may say things > that are offensive to you personally but not to the subscribers in general. > Please be ready to "deal with it" on occasion. > - This list is moderated to a degree. We keep only a loose reign on > the users here. However, if after several warnings a user repeatedly > engages in trollish behavior then he or she WILL get moderated. Those who > are incorrigibly trollish will be banned. > > > I appreciate that most of our readers don't need these reminders, so thank > you for indulging us. Keep up the good work! > > -Michael > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120229/2a2e2330/attachment.html From wagnerspi at gmail.com Thu Mar 1 04:31:51 2012 From: wagnerspi at gmail.com (Wagner) Date: Wed, 29 Feb 2012 22:31:51 -0300 Subject: [Freeswitch-users] web application to call to fs Message-ID: hello , is there any Web application that i could use to let a user call to my ivr through my website? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120229/3c9cc3bd/attachment.html From hi-tecc at hotmail.com Thu Mar 1 03:54:39 2012 From: hi-tecc at hotmail.com (DP .) Date: Wed, 29 Feb 2012 19:54:39 -0500 Subject: [Freeswitch-users] nibble_bill (and limit) on an external transfer from IVR Message-ID: Hi List, I'm currently unable to get nibble_bill (and mod_limit) to bill on a call that arrives at an IVR, dials an extension (105) and is transferred (internally or externally) directly from the IVR. The dialed extension is simply: and the DID has the following variables: I've tried set, export, and bridge_export on both nibble_rate and nibble_account. However the cash value is always returned as nan once the call is actually transferred externally via the IVR. I've modified the transfer section above to include an internal extension (just for testing) and got the same result. DB error:2012-02-29 19:06:46.808298 [ERR] switch_odbc.c:494 ERR: [UPDATE accounts SET cash=cash-nan WHERE id='5'] The call still connects just fine, just the billing portion that stops. Dialing registered local extensions from the IVR also connect and bill just fine. However we would like to have an extension off the IVR that dials external numbers to be billed accordingly. -Damian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120229/698f7fec/attachment-0001.html From msc at freeswitch.org Thu Mar 1 04:38:06 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 Feb 2012 17:38:06 -0800 Subject: [Freeswitch-users] web application to call to fs In-Reply-To: References: Message-ID: Yep, if you don't mind using the flex client and mod_rtmp. Check it out: http://wiki.freeswitch.org/wiki/Mod_rtmp Be sure to see the note about where to get the flex client... -MC On Wed, Feb 29, 2012 at 5:31 PM, Wagner wrote: > hello , > > is there any Web application that i could use to let a user call to my ivr > through my website? > > thanks > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120229/e5899d5b/attachment.html From eli at netspectrum.com Thu Mar 1 05:01:35 2012 From: eli at netspectrum.com (Erjian Li) Date: Thu, 1 Mar 2012 10:01:35 +0800 Subject: [Freeswitch-users] a question about freeswitch conference caller-controls Message-ID: Hi All, I have a querstion about freeswitch conference caller controls. If anyone can give some hint/help, I'm very appreciated. My freeswitch server has been connected to a SIP provider's server (IDT server), and can dial out to normal cellphone numbers. Now I want to start a conference call using "conference dial ..." command, when the destination phones join the conference, the phone callers can't control the conference by pressing the keys specified in section of conference.conf.xml. (In other words, if I press key '0', it can't mute myself; press '#', it can't hang up, etc. I use the default caller-controls group.) I want to know in this situation, whether the cellphone's key tone can't be transferred to freeswitch server? Can freeswitch server only receive DTMF encapsulated in RTP packet transferred over IP? I can control the conference when I use X-Lite as client, and freeswitch outputs following log when I press key '0' in X-Lite interface: ================================== freeswitch at eli-desktop> 2012-02-29 17:02:59.349272 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 0:960 2012-02-29 17:02:59.367880 [DEBUG] mod_conference.c:2919 Queueing file '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav' for play ================================= In the case of cellphone, when users press key, the DTMF is generated and transferred via PSTN channel to the IDT server, is this correct? and if so, is it up to IDT server to encapsulate the DTMF in RTP packets and send it to freeswitch server? -- Best Regards Erjian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/3ed6a422/attachment.html From bob.mccarthy at experient.com Thu Mar 1 05:37:13 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Wed, 29 Feb 2012 19:37:13 -0700 Subject: [Freeswitch-users] CallerID changing after Bridge using SLA on Polycom Phones In-Reply-To: <00dc01ccf730$16b763b0$44262b10$@mccarthy@experient.com> References: <001301ccf6ca$b15bec80$1413c580$@mccarthy@experient.com> <5003D7D3E06F514E8C682F18D223265C05121CD440@AZWSMS03.azwarranty.int> <00dc01ccf730$16b763b0$44262b10$@mccarthy@experient.com> Message-ID: <000601ccf754$36f4d3f0$a4de7bd0$@mccarthy@experient.com> spoke too soon, with the callerid name and number work perfectly, however when phone 2 barges in both phones display's the shared line data. with the callerid name is always correct but 20% of the time the Callerid number is blank. placing the call on hold or when phone 2 barges in restores the callerid number. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob McCarthy Sent: Wednesday, February 29, 2012 3:19 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] CallerID changing after Bridge using SLA on Polycom Phones worked like a champ! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Weigel, Stefan Sent: Wednesday, February 29, 2012 8:17 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] CallerID changing after Bridge using SLA on Polycom Phones Hi, that's working for me. [..] [..] then do the bridge. Best regards, Stefan Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Bob McCarthy Gesendet: Mittwoch, 29. Februar 2012 11:13 An: freeswitch-users at lists.freeswitch.org Cc: bill.oneil at experient.com Betreff: [Freeswitch-users] CallerID changing after Bridge using SLA on Polycom Phones I am having an issue similar to that experienced by Wellie Chao in his email "Caller ID on inbound calls on Polycom" 2010/4/22. The call is from an Audiocodes gateway with a dial string of *NXXXXXX# CallerID number sent is 101 which corresponds to the FXS port it came from, there is no CallerID Name. I modify the CallerID name and number in the dialplan before the bridge. On Ring the CallerID name and number are what I have changed it to. On Bridge, 80% of the time the CallerID name and number follow thru, 20% of the time it changes to what Leg A had originally before I used the export application.( name is blank and number is 101). if I put the call on Hold it reverts back to what I expected. I have tried to use set, moved the import after the bridge, ignore_display_updates=true. Any Ideas? On a further note, when phone 2 presses the shared line it shows the same CallerID name and number as phone 1. When you press BargeIn both the CallerID name and number change to 101. i.e. name is 101 number is 101 on both phones. this happens every time, is there a way to stop this behavior as well ? Thanks Bob McCarthy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120229/c005643c/attachment.html From bob.mccarthy at experient.com Thu Mar 1 11:25:11 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Thu, 01 Mar 2012 03:25:11 -0500 Subject: [Freeswitch-users] send_dtmf fails on sla barge in Message-ID: <1330590312.30063.11.camel@CO-999-8> dialplan default.xml has prior to the bridge internal.xml has no dtmf statements uncommented when sending dtmf from the Polycom dialpad after a barge in using SLA, (also after the barging phone hangs up) I get the following error: 2012-03-01 03:05:53.841090 [ERR] switch_ivr_async.c:547 dmachine overflow error! FreeSWITCH Version 1.0.head (git-23645b6 2012-02-27 16-49-12 -0600) Bob McCarthy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/3098d6ba/attachment-0001.html From asilva at wirelessmundi.com Thu Mar 1 13:44:54 2012 From: asilva at wirelessmundi.com (Antonio) Date: Thu, 01 Mar 2012 11:44:54 +0100 Subject: [Freeswitch-users] fs_cli -x 'cdr_csv rotate' does not always rotate cdr_csv-files. Message-ID: <1330598694.11514.29.camel@vm.vm> Using the latest git version, on executing cdr rotation, using "cdr_csv rotate" doesn't work as expected... -- NO ROTATION IS PERFORMED: After reload the mod_cdr_csv: freeswitch at internal> reload mod_cdr_csv +OK module unloaded +OK Reloading XML +OK module loaded NOTE: (this output happens also happens after an entire restart of fresswitch) freeswitch at internal> cdr_csv rotate +OK After some call: freeswitch at internal> cdr_csv rotate +OK 2012-03-01 11:29:14.137741 [NOTICE] mod_cdr_csv.c:122 Re-opened CDR logfile /opt/freeswitch/log/cdr-csv/Master.csv -- I COULD GET ROTATION ONCE... but i don't know what i did... i just start work until i reload cdr_csv again... freeswitch at internal> cdr_csv rotate +OK 2012-03-01 11:23:44.397733 [NOTICE] mod_cdr_csv.c:122 Rotated CDR logfile /opt/freeswitch/log/cdr-csv/Master.csv CDR_CSV.XML: http://pastebin.freeswitch.org/18553 If you need more information to track the error, please let me know. If you think is appropriated i could open a jira issue. Thanks, Ant?nio From miha at softnet.si Thu Mar 1 14:07:27 2012 From: miha at softnet.si (Miha Zoubek) Date: Thu, 01 Mar 2012 12:07:27 +0100 Subject: [Freeswitch-users] destination_number a&b leg, please help Message-ID: <4F4F586F.3090203@softnet.si> Hi, how can I set in dialplan, that destination_number variable will be the same for a and b leg? thank you! Miha From potxoka at gmail.com Thu Mar 1 15:17:01 2012 From: potxoka at gmail.com (Anto) Date: Thu, 1 Mar 2012 13:17:01 +0100 Subject: [Freeswitch-users] Route to the profile and not the gateway Message-ID: Hello Once I download the XML from the server and see it with an editor, I have resolved many questions of how FreeSWITCH, but even I have several questions I have not found information (I spend all day reading the wiki). I am setting up multiple providers and use a format like this (do not know if it is the best): ......... Is there any way to set the extension when not having to configure all the gateway? (only use the profile). Sample dialplan: ........... ???? When set incoming calls, not very well how, not whether to route directly to dialplan or create a profile for this. I have to be routed to multiple internal servers (load balancer, failover, etc), so the above scenario I could serve to cover this. Thanks !! Best regards Anto From avi at avimarcus.net Thu Mar 1 15:18:46 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 1 Mar 2012 14:18:46 +0200 Subject: [Freeswitch-users] destination_number a&b leg, please help In-Reply-To: <4F4F586F.3090203@softnet.si> References: <4F4F586F.3090203@softnet.si> Message-ID: What are you trying to achieve? Matching up A and B legs of calls? -Avi On Thu, Mar 1, 2012 at 1:07 PM, Miha Zoubek wrote: > Hi, > > how can I set in dialplan, that destination_number variable will be the > same for a and b leg? > > > > thank you! > > Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/e010324c/attachment.html From potxoka at gmail.com Thu Mar 1 15:20:55 2012 From: potxoka at gmail.com (Anto) Date: Thu, 1 Mar 2012 13:20:55 +0100 Subject: [Freeswitch-users] Send PAI and RPID Message-ID: Hello To my FreeSWITCH servers, they come PAI and RPID headers, sent to the carrier but only one (the one I have configured with ). Is there any way to send both headers?. Thanks Best regards Anto From sharad at coraltele.com Thu Mar 1 15:06:04 2012 From: sharad at coraltele.com (Sharad Garg) Date: Thu, 1 Mar 2012 17:36:04 +0530 Subject: [Freeswitch-users] Update on Brian West's condition References: <33155901.11598.1330110388284.JavaMail.root@zmail.rockbochs.com> <278D737E-5A6B-4A85-86E0-0E7DD5C14D3C@freeswitch.org> Message-ID: <16133D17A49A4EAAB16B299BC56C94B6@sharad> Let us hope for this best. We are sure you will be in action very soon. God Bless you Man... Regards Sharad From fdelawarde at wirelessmundi.com Thu Mar 1 15:31:40 2012 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Thu, 01 Mar 2012 13:31:40 +0100 Subject: [Freeswitch-users] removing caller_id_name In-Reply-To: References: <1330527551.12944.47.camel@luna.madrid.commsmundi.com> <2AB1E3C03738764094BDD615FEE595AD037468A5@AMSPRD0402MB111.eurprd04.prod.outlook.com> <1330529520.12944.64.camel@luna.madrid.commsmundi.com> <1330531312.12944.65.camel@luna.madrid.commsmundi.com> <1330532922.12944.74.camel@luna.madrid.commsmundi.com> Message-ID: <1330605100.12944.132.camel@luna.madrid.commsmundi.com> Thanks for the help, did you find anything? If there is no way, I'll try to find time to do some patch that would take "_undef_" into account for PAI and RPID headers (right now it only works for the From header). Fran?ois. On Wed, 2012-02-29 at 11:32 -0500, Brian Foster wrote: > Hmm... let me do some research. > > > -BDF > > On Wed, Feb 29, 2012 at 11:28 AM, Fran?ois Delawarde > wrote: > Can't do that, "caller_id_name" is read-only (part of caller > profile). > > http://wiki.freeswitch.org/wiki/Channel_Variables#caller_id_name > > Please keep firing ideas, I'm sure it must be possible > somehow! > > Fran?ois. > > > On Wed, 2012-02-29 at 11:10 -0500, Brian Foster wrote: > > There's probably a couple of different ways to do it. It's > whatever > > works. I don't like setting things to an empty string just > as a > > personal preference (even though it accomplishes the same > result). > > > > > > > > > > > > -BDF > > > > On Wed, Feb 29, 2012 at 11:01 AM, Fran?ois Delawarde > > wrote: > > Nice try but no, it would unset > "effective_caller_id_name", > > and not > > "caller_id_name". > > > > I would need to be able to set it to an empty > string, but for > > FS, empty > > string is like an "unset". > > > > Fran?ois. > > > > > > On Wed, 2012-02-29 at 10:42 -0500, Brian Foster > wrote: > > > What about doing: > > > > > > > data="effective_caller_id_name"/> > > > > > > ..? > > > > > > -BDF > > > > > > On Feb 29, 2012 10:32 AM, "Fran?ois Delawarde" > > > wrote: > > > Yeah, my analog phone has that function to > be able > > to add the > > > last CID > > > to its directory, but it uses the > caller_id_name as > > contact > > > name and > > > caller_id_number as contact number. > > > > > > Each time I want to add someone that > called me to > > the phone's > > > directory, > > > I have to delete the preset contact name > and add > > what I want. > > > > > > It's a pain for lazy people like me. > > > > > > Fran?ois. > > > > > > > > > On Wed, 2012-02-29 at 07:23 -0800, Mitch > Capper > > wrote: > > > > Any reason not to set it to the phone > number? > > > > > > > > ~Mitch > > > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting > Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > FreeSWITCH-powered IP PBX: The CudaTel > > Communication Server > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting > Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > FreeSWITCH-powered IP PBX: The CudaTel > Communication > > Server > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > FreeSWITCH-powered IP PBX: The CudaTel > Communication Server > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Brian D. Foster > > Endigo Computer LLC > > Email: bdfoster at endigotech.com > > Phone: 317-429-1069 > > Indianapolis, Indiana, USA > > > > This message contains confidential information and is > intended for > > those listed in the "To:", "CC:", and/or "BCC:" fields of > the message > > header.If you are not the intended recipient you are > notified that > > disclosing, copying, distributing or taking any action in > reliance on > > the contents of this information is strictly prohibited. > E-mail > > transmission cannot be guaranteed to be secure or error-free > as > > information could be intercepted, corrupted, lost, > destroyed, arrive > > late or incomplete, or contain viruses. The sender therefore > does not > > accept liability for any errors or omissions in the contents > of this > > message, which arise as a result of e-mail transmission. If > > verification is required please request a hard-copy version. > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-429-1069 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for > those listed in the "To:", "CC:", and/or "BCC:" fields of the message > header.If you are not the intended recipient you are notified that > disclosing, copying, distributing or taking any action in reliance on > the contents of this information is strictly prohibited. E-mail > transmission cannot be guaranteed to be secure or error-free as > information could be intercepted, corrupted, lost, destroyed, arrive > late or incomplete, or contain viruses. The sender therefore does not > accept liability for any errors or omissions in the contents of this > message, which arise as a result of e-mail transmission. If > verification is required please request a hard-copy version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From miha at softnet.si Thu Mar 1 16:16:44 2012 From: miha at softnet.si (Miha Zoubek) Date: Thu, 01 Mar 2012 14:16:44 +0100 Subject: [Freeswitch-users] destination_number a&b leg, please help In-Reply-To: References: <4F4F586F.3090203@softnet.si> Message-ID: <4F4F76BC.9090402@softnet.si> On 03/01/2012 01:18 PM, Avi Marcus wrote: > What are you trying to achieve? Matching up A and B legs of calls? > > -Avi > > > On Thu, Mar 1, 2012 at 1:07 PM, Miha Zoubek > wrote: > > Hi, > > how can I set in dialplan, that destination_number variable will > be the > same for a and b leg? > > > > thank you! > > Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Hi @Avi, thank you for your quick replay. In cvs cdr table I have like this when I make a xfer from phone (cwfd): idDescending caller_id_name caller_id_number destination_number context 380 018108500 018108500.fs_kabelvoip1 051357952 fs_kabelvoip1.fs1.softnet.si 381 Outbound Call 38651357952 38651357952 default I would like that to have in row caller_id_name 018105800 instead of Outbound call. Or in row destination_number 38651357952 instead of 051357952. Thanks! Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/ce55c333/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: s_desc.png Type: image/png Size: 201 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/ce55c333/attachment-0001.png From miha at softnet.si Thu Mar 1 16:50:17 2012 From: miha at softnet.si (Miha Zoubek) Date: Thu, 01 Mar 2012 14:50:17 +0100 Subject: [Freeswitch-users] destination_number a&b leg, please help In-Reply-To: <4F4F76BC.9090402@softnet.si> References: <4F4F586F.3090203@softnet.si> <4F4F76BC.9090402@softnet.si> Message-ID: <4F4F7E99.6090100@softnet.si> On 03/01/2012 02:16 PM, Miha Zoubek wrote: > On 03/01/2012 01:18 PM, Avi Marcus wrote: >> What are you trying to achieve? Matching up A and B legs of calls? >> >> -Avi >> >> >> On Thu, Mar 1, 2012 at 1:07 PM, Miha Zoubek > > wrote: >> >> Hi, >> >> how can I set in dialplan, that destination_number variable will >> be the >> same for a and b leg? >> >> >> >> thank you! >> >> Miha >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > Hi @Avi, > > thank you for your quick replay. > > In cvs cdr table I have like this when I make a xfer from phone (cwfd): > > > idDescending > > caller_id_name > > caller_id_number > > destination_number > > context > > > > > 380 018108500 018108500.fs_kabelvoip1 051357952 > fs_kabelvoip1.fs1.softnet.si > > 381 Outbound Call 38651357952 38651357952 default > > I would like that to have in row caller_id_name 018105800 instead of > Outbound call. Or in row destination_number 38651357952 instead of > 051357952. > > Thanks! > > Miha > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org This happens because of: Flipping CID from "018108500" <018108500> to "Outbound Call" <38651357952> Is it possible to change CID, so that would stay the same (018108500)? thanks! miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/c1788050/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 201 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/c1788050/attachment-0001.png From benkokakao at gmail.com Thu Mar 1 17:45:37 2012 From: benkokakao at gmail.com (Christian Benke) Date: Thu, 1 Mar 2012 15:45:37 +0100 Subject: [Freeswitch-users] Choopy one-way noise Message-ID: Hello! On a box with a Sangoma A500 BRI card i have a serious audio quality problem. The "outside"-leg of a call hears a choppy robotic noise, it's difficult to understand the actual voice. I hope someone recognizes this issue, as i have tried several approaches and so far i neither know what causes the problem nor where it is caused exactly. Here's a sample file(The noise is not very noticeable, it's the oscillating hum you hear in the background, the short voice-sample you hear is mine, but it's inbound so it's not affected): http://poab.org/test1.wav If the call is set on-hold or if there's a playback/moh, the quality for the outside-leg is fine while there's no two-way-audio, so my assumption was that it's some kind of problem with codecs or transcoding. Capturing the RTP-Stream from the phone to the server proved though that the quality is fine before it arrives at FS(There are no audio problems on internal calls as well). Here's what else i've tried without success: - set disable-monotonic-timing to true - started FS with the -rt Flag(As well as -waste and -heavy-timer) - jitter "time_test" 100 100 responds with avg 130(Thats fine right?) - rearranged the codec-settings of the phone(Sangoma SPIP 450) to give A-law the highest priority - high-precision-timer is available in the kernel according to dmesg and /proc/timer_list (kernel is 2.6.32-5-686 on debian 6.0) Sangoma's Support appears to be clueless as well, they didn't have effective suggestions so far(Opened a ticket 2 weeks). Since i don't have this problem with identical hardware without Sangoma-Cards(SIP-Trunk instead), my first assumption was the card. Probably an interrupt issue? Any suggestions would be highly appreciated, i'm a bit at loss now and only know more drastic next steps(Replacing ISDN-Modules or the whole hardware). Best regards, Christian From steveu at coppice.org Thu Mar 1 17:57:55 2012 From: steveu at coppice.org (Steve Underwood) Date: Thu, 01 Mar 2012 22:57:55 +0800 Subject: [Freeswitch-users] Problems with t38_gateway In-Reply-To: <2AB1E3C03738764094BDD615FEE595AD03746708@AMSPRD0402MB111.eurprd04.prod.outlook.com> References: <2AB1E3C03738764094BDD615FEE595AD03746708@AMSPRD0402MB111.eurprd04.prod.outlook.com> Message-ID: <4F4F8E73.8030006@coppice.org> On 02/29/2012 09:21 PM, Steven Lam, KeenSystems B.V. wrote: > Hi all, > > I'm experimenting with the t38_gateway in mod_spandsp to build a gateway that fully supports "passive" T_38. My test config looks like this: Interesting term. What do you consider to be "passive" T.38? Steve From abaci64 at gmail.com Thu Mar 1 18:21:22 2012 From: abaci64 at gmail.com (a b) Date: Thu, 1 Mar 2012 10:21:22 -0500 Subject: [Freeswitch-users] Segmentation fault In-Reply-To: References: Message-ID: Jira created http://jira.freeswitch.org/browse/FS-3962 On Wed, Feb 29, 2012 at 6:13 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Getting_a_Backtrace > > On Wed, Feb 29, 2012 at 4:30 PM, a b wrote: > > Hi, > > I just noticed that if I bridge a call to a user that exists in the > > directory but doesn't have the dial-string param set, FreeSWITCH will > crash. > > this is happening on git from yesterday. can someone confirm if this is a > > bug or something in my setup? > > Thanks > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/2a69c560/attachment.html From s.lam at keensystems.eu Thu Mar 1 18:55:55 2012 From: s.lam at keensystems.eu (Steven Lam, KeenSystems B.V.) Date: Thu, 1 Mar 2012 15:55:55 +0000 Subject: [Freeswitch-users] Problems with t38_gateway In-Reply-To: <4F4F8E73.8030006@coppice.org> References: <2AB1E3C03738764094BDD615FEE595AD03746708@AMSPRD0402MB111.eurprd04.prod.outlook.com> <4F4F8E73.8030006@coppice.org> Message-ID: <2AB1E3C03738764094BDD615FEE595AD03749D35@AMSPRD0402MB111.eurprd04.prod.outlook.com> Hi, I consider a gateway only reacting to T38 re-INVITES and not initiating T38 re-INVITES "passive". My setup works great for this, the only problem is when both the A- and B-leg send a T38 re-INVITES something goes wrong. Steven > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Steve Underwood > Sent: donderdag 1 maart 2012 15:58 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Problems with t38_gateway > > On 02/29/2012 09:21 PM, Steven Lam, KeenSystems B.V. wrote: > > Hi all, > > > > I'm experimenting with the t38_gateway in mod_spandsp to build a > gateway that fully supports "passive" T_38. My test config looks like this: > Interesting term. What do you consider to be "passive" T.38? > > Steve > > __________________________________________________________ > _______________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Mar 1 19:21:23 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 1 Mar 2012 10:21:23 -0600 Subject: [Freeswitch-users] Choopy one-way noise In-Reply-To: References: Message-ID: You should choose your subject more carefully mentioning Sangoma or FreeTDM in the subject to attract the attention of Moy On Thu, Mar 1, 2012 at 8:45 AM, Christian Benke wrote: > Hello! > > On a box with a Sangoma A500 BRI card i have a serious audio quality > problem. The "outside"-leg of a call hears a choppy robotic noise, > it's difficult to understand the actual voice. I hope someone > recognizes this issue, as i have tried several approaches and so far i > neither know what causes the problem nor where it is caused exactly. > > Here's a sample file(The noise is not very noticeable, it's the > oscillating hum you hear in the background, the short voice-sample you > hear is mine, but it's inbound so it's not affected): > > http://poab.org/test1.wav > > If the call is set on-hold or if there's a playback/moh, the quality > for the outside-leg is fine while there's no two-way-audio, so my > assumption was that it's some kind of problem with codecs or > transcoding. Capturing the RTP-Stream from the phone to the server > proved though that the quality is fine before it arrives at FS(There > are no audio problems on internal calls as well). > > Here's what else i've tried without success: > - set disable-monotonic-timing to true > - started FS with the -rt Flag(As well as -waste and -heavy-timer) > - jitter "time_test" 100 100 responds with avg 130(Thats fine right?) > - rearranged the codec-settings of the phone(Sangoma SPIP 450) to give > A-law the highest priority > - high-precision-timer is available in the kernel according to dmesg > and /proc/timer_list (kernel is 2.6.32-5-686 on debian 6.0) > > Sangoma's Support appears to be clueless as well, they didn't have > effective suggestions so far(Opened a ticket 2 weeks). Since i don't > have this problem with identical hardware without > Sangoma-Cards(SIP-Trunk instead), my first assumption was the card. > Probably an interrupt issue? > > Any suggestions would be highly appreciated, i'm a bit at loss now and > only know more drastic next steps(Replacing ISDN-Modules or the whole > hardware). > > Best regards, > Christian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From moises.silva at gmail.com Thu Mar 1 19:45:33 2012 From: moises.silva at gmail.com (Moises Silva) Date: Thu, 1 Mar 2012 11:45:33 -0500 Subject: [Freeswitch-users] Choopy one-way noise In-Reply-To: References: Message-ID: On Thu, Mar 1, 2012 at 9:45 AM, Christian Benke wrote: > > Sangoma's Support appears to be clueless as well, they didn't have > effective suggestions so far(Opened a ticket 2 weeks). Since i don't > have this problem with identical hardware without > Sangoma-Cards(SIP-Trunk instead), my first assumption was the card. > Probably an interrupt issue? > Another test you can try to pinpoint the issue and determine once for all if the problem is below FreeSWITCH is to use the "ftdm trace" command. This command will create an input an output file (in the directory you specify) that contains the raw audio as it is read and written from/to the wanpipe device. Type "ftdm trace" to see the command syntax. Run the command whenever you reproduce the problem (can be run in the middle of the call) then stop the trace with "ftdm notrace" when you're done capturing. Note that the tracing will not stop automatically when the call is hung up, the trace is a raw I/O trace and has no knowledge of calls starting or being hung up. The audio format will be alaw (because is a BRI line), you can convert that to a wav (with sox for example) and see if the audio issue is there. If it is, we can start thinking on lower-level troubleshooting. *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/c0afd12e/attachment-0001.html From godson.g at gmail.com Thu Mar 1 20:57:22 2012 From: godson.g at gmail.com (Godson Gera) Date: Thu, 1 Mar 2012 23:27:22 +0530 Subject: [Freeswitch-users] a question about freeswitch conference caller-controls In-Reply-To: References: Message-ID: Freeswitch only sees the SIP side of the call according to your setup. So, make sure that your SIP provider is properly forwarding the DTMF in any one of the supported formats (inband, rfc2833, SIP INFO ). On Thu, Mar 1, 2012 at 7:31 AM, Erjian Li wrote: > Hi All, > > I have a querstion about freeswitch conference caller controls. If anyone > can give some hint/help, I'm very appreciated. > > -- Thanks & Regards, Godson Gera IVR India -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/e1e251f7/attachment.html From potxoka at gmail.com Thu Mar 1 21:35:29 2012 From: potxoka at gmail.com (Anto) Date: Thu, 1 Mar 2012 19:35:29 +0100 Subject: [Freeswitch-users] Route to the profile and not the gateway In-Reply-To: References: Message-ID: Hello, Setting up profiles and the other options, I can not make outgoing calls. With the standard configuration and adding the extension in the public folder, it worked specifying the gateway. Now removing it from that folder and setting the context "internal" (in addition to setting a context for provider), I can not make outgoing calls. I have reviewed documentation, the configuration and have tried various forms of make the bridge, but I can not make the call. Does anyone can guide me that could look at? Thank you very much. Best regards Anto sbc# cat outbound.xml freeswitch at sbc > sofia status Name Type Data State ================================================================================================= internal profile sip:mod_sofia at 192.168.20.5:5060 RUNNING (0) internal::server-1 gateway sip:user at 192.168.20.40 NOREG provider1 profile sip:mod_sofia at EXTERNAL_IP:5060 RUNNING (0) provider1::server-1 gateway sip:user at DOMAIN REGED ================================================================================================= 2 profiles 0 aliases freeswitch at sbc > 2012-03-01 17:56:32.401204 [NOTICE] switch_channel.c:915 New Channel sofia/internal/anto at 192.168.20.40 [3f191be1-c763-e111-bc8d-f46d041ca78e] 2012-03-01 17:56:32.401204 [DEBUG] sofia.c:5283 Channel sofia/internal/anto at 192.168.20.40 entering state [received][100] 2012-03-01 17:56:32.401204 [DEBUG] sofia.c:5294 Remote SDP: v=0 o=anto 0 0 IN IP4 192.168.20.40 s=- c=IN IP4 192.168.20.40 t=0 0 m=audio 59700 RTP/AVP 9 96 97 98 100 0 8 102 3 103 5 6 101 a=rtpmap:9 G722/8000 a=rtpmap:96 SILK/24000 a=rtpmap:97 SILK/16000 a=rtpmap:98 speex/32000 a=rtpmap:100 speex/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:102 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:103 speex/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:6 DVI4/16000 a=rtpmap:101 telephone-event/8000 a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level a=zrtp-hash:1.10 366f09eb8571b63fc76ad7df2417b1b145774d1076a99042721e6f5f3f3401ca m=video 60184 RTP/AVP 104 99 a=rtpmap:104 H264/90000 a=fmtp:104 profile-level-id=4DE01f;packetization-mode=1 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=4DE01f a=recvonly a=imageattr:104 send * recv [x=[0-1920],y=[0-1200]] a=imageattr:99 send * recv [x=[0-1920],y=[0-1200]] a=zrtp-hash:1.10 23f81f1a335dc819be01e9a794cb8e7a509019066dac01b8fbd89038aec44edb 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/anto at 192.168.20.40) Running State Change CS_NEW 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/anto at 192.168.20.40) State NEW 2012-03-01 17:56:32.401204 [DEBUG] sofia.c:5480 (sofia/internal/anto at 192.168.20.40) State Change CS_NEW -> CS_INIT 2012-03-01 17:56:32.401204 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/anto at 192.168.20.40 [BREAK] 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/anto at 192.168.20.40) Running State Change CS_INIT 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/anto at 192.168.20.40) State INIT 2012-03-01 17:56:32.401204 [DEBUG] mod_sofia.c:85 sofia/internal/anto at 192.168.20.40 SOFIA INIT 2012-03-01 17:56:32.401204 [DEBUG] mod_sofia.c:125 (sofia/internal/anto at 192.168.20.40) State Change CS_INIT -> CS_ROUTING 2012-03-01 17:56:32.401204 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/anto at 192.168.20.40 [BREAK] 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/anto at 192.168.20.40) State INIT going to sleep 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/anto at 192.168.20.40) Running State Change CS_ROUTING 2012-03-01 17:56:32.401204 [DEBUG] switch_channel.c:1844 (sofia/internal/anto at 192.168.20.40) Callstate Change DOWN -> RINGING 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/anto at 192.168.20.40) State ROUTING 2012-03-01 17:56:32.401204 [DEBUG] mod_sofia.c:148 sofia/internal/anto at 192.168.20.40 SOFIA ROUTING 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:104 sofia/internal/anto at 192.168.20.40 Standard ROUTING 2012-03-01 17:56:32.401204 [INFO] mod_dialplan_xml.c:336 Processing +57xxxxxxxxx <+57xxxxxxxxx>->57xxxxxxxxx in context internal Dialplan: sofia/internal/anto at 192.168.20.40 parsing [internal->provider1] continue=false Dialplan: sofia/internal/anto at 192.168.20.40 Regex (PASS) [provider1] network_addr(192.168.20.40) =~ /^192.168.20.40$/ break=on-false Dialplan: sofia/internal/anto at 192.168.20.40 Regex (PASS) [provider1] destination_number(57xxxxxxxxx) =~ /^(\d+)$/ break=on-false Dialplan: sofia/internal/anto at 192.168.20.40 Action bridge({sip_cid_type=pid,ignore_display_updates=true}sofia/gateway/provider1::server-1/57xxxxxxxxx) 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/anto at 192.168.20.40) State Change CS_ROUTING -> CS_EXECUTE 2012-03-01 17:56:32.401204 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/anto at 192.168.20.40 [BREAK] 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/anto at 192.168.20.40) State ROUTING going to sleep 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/anto at 192.168.20.40) Running State Change CS_EXECUTE 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/anto at 192.168.20.40) State EXECUTE 2012-03-01 17:56:32.401204 [DEBUG] mod_sofia.c:241 sofia/internal/anto at 192.168.20.40 SOFIA EXECUTE 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:192 sofia/internal/anto at 192.168.20.40 Standard EXECUTE 2012-03-01 17:56:32.401204 [ERR] switch_core_session.c:2096 Invalid Application bridge 2012-03-01 17:56:32.401204 [DEBUG] switch_channel.c:2804 (sofia/internal/anto at 192.168.20.40) Callstate Change RINGING -> HANGUP 2012-03-01 17:56:32.401204 [NOTICE] switch_core_session.c:2097 Hangup sofia/internal/anto at 192.168.20.40 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2012-03-01 17:56:32.401204 [DEBUG] switch_channel.c:2820 Send signal sofia/internal/anto at 192.168.20.40 [KILL] 2012-03-01 17:56:32.401204 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/anto at 192.168.20.40 [BREAK] 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/anto at 192.168.20.40) State EXECUTE going to sleep 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/anto at 192.168.20.40) Running State Change CS_HANGUP 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/anto at 192.168.20.40) State HANGUP 2012-03-01 17:56:32.401204 [DEBUG] mod_sofia.c:465 Channel sofia/internal/anto at 192.168.20.40 hanging up, cause: DESTINATION_OUT_OF_ORDER 2012-03-01 17:56:32.401204 [DEBUG] mod_sofia.c:530 Responding to INVITE with: 502 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:47 sofia/internal/anto at 192.168.20.40 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/anto at 192.168.20.40) State HANGUP going to sleep 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/anto at 192.168.20.40) State Change CS_HANGUP -> CS_REPORTING 2012-03-01 17:56:32.401204 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/anto at 192.168.20.40 [BREAK] 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/anto at 192.168.20.40) Running State Change CS_REPORTING 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/anto at 192.168.20.40) State REPORTING 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:79 sofia/internal/anto at 192.168.20.40 Standard REPORTING, cause: DESTINATION_OUT_OF_ORDER 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/anto at 192.168.20.40) State REPORTING going to sleep 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/anto at 192.168.20.40) State Change CS_REPORTING -> CS_DESTROY 2012-03-01 17:56:32.401204 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/anto at 192.168.20.40 [BREAK] 2012-03-01 17:56:32.401204 [DEBUG] switch_core_session.c:1377 Session 2 (sofia/internal/anto at 192.168.20.40) Locked, Waiting on external entities 2012-03-01 17:56:32.401204 [NOTICE] switch_core_session.c:1395 Session 2 (sofia/internal/anto at 192.168.20.40) Ended 2012-03-01 17:56:32.401204 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/anto at 192.168.20.40 [CS_DESTROY] 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/anto at 192.168.20.40) Callstate Change HANGUP -> DOWN 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/anto at 192.168.20.40) Running State Change CS_DESTROY 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/anto at 192.168.20.40) State DESTROY 2012-03-01 17:56:32.401204 [DEBUG] mod_sofia.c:370 sofia/internal/anto at 192.168.20.40 SOFIA DESTROY 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:86 sofia/internal/anto at 192.168.20.40 Standard DESTROY 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/anto at 192.168.20.40) State DESTROY going to sleep 2012/3/1 Anto : > Hello > > Once I download the XML from the server and see it with an editor, I > have resolved many questions of how FreeSWITCH, but even I have > several questions I have not found information (I spend all day > reading the wiki). > > I am setting up multiple providers and use a format like this (do not > know if it is the best): > > > ? > ? ? ? > ? > ......... > > Is there any way to set the extension when not having to configure all > the gateway? (only use the profile). Sample dialplan: > > data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/> > ? ? ? ? > ? ? ? ? > ? ? ? ? > ........... > ???? > > When set incoming calls, not very well how, not whether to route > directly to dialplan or create a profile for this. I have to be routed > to multiple internal servers (load balancer, failover, etc), so the > above scenario I could serve to cover this. Thanks !! > > Best regards > Anto From freeswitch at earthspike.net Thu Mar 1 22:14:52 2012 From: freeswitch at earthspike.net (John) Date: Thu, 01 Mar 2012 19:14:52 +0000 Subject: [Freeswitch-users] Choopy one-way noise (FreeTDM) In-Reply-To: References: Message-ID: <4F4FCAAC.5020307@earthspike.net> Christian, I have a similar problem with a Sangoma B700 card (I think the ISDN part is similar, though) and have a Sangoma ticket open (#1310). It's intermittent and relies on the outside party reporting the quality being poor, so I am waiting for another occasion to arise when I can grab some more detailed traces. Your sample is quiet, which makes it hard to see what is going on. I have generated a test tones file which I use as a voicemail outgoing message on a test extension which allows me to clearly see what is happening to the voice. It uses sox and goes like this: #!/bin/bash sox -r 8000 -n out300_0_2.wav synth 2 sine 300 sox -r 8000 -n out500_0_2.wav synth 2 sine 500 sox -r 8000 -n out1000_0_2.wav synth 2 sine 1000 sox -r 8000 -n out2000_0_2.wav synth 2 sine 2000 sox -r 8000 -n out3000_0_2.wav synth 2 sine 3000 sox -r 8000 -n out300-3400_0_3.wav synth 3 sine 300-3400 sox -r 8000 -n out1-4000_0_5.wav synth 5 sine 1-4000 sox -r 8000 -n out1000_0_5.wav synth 5 sine 1000 sox -r 8000 -n out1000_-6_5.wav synth 5 sine 1000 gain -6 sox -r 8000 -n out1000_-12_5.wav synth 5 sine 1000 gain -12 sox -r 8000 -n outwn_0_5.wav synth 5 noise sox -r 8000 -n outwn_-6_5.wav synth 5 noise gain -6 sox -r 8000 -n outwn_-12_5.wav synth 5 noise gain -12 sox -r 8000 -n outsilence_2.wav synth 2 sine 300 vol 0 amplitude sox out300_0_2.wav outsilence_2.wav out500_0_2.wav outsilence_2.wav out1000_0_2.wav outsilence_2.wav out2000_0_2.wav outsilence_2.wav out3000_0_2.wav outsilence_2.wav out300-3400_0_3.wav outsilence_2.wav out1-4000_0_5.wav outsilence_2.wav out1000_0_5.wav outsilence_2.wav out1000_-6_5.wav outsilence_2.wav out1000_-12_5.wav outsilence_2.wav outwn_0_5.wav outsilence_2.wav outwn_-6_5.wav outsilence_2.wav outwn_-12_5.wav testtones.wav You may find something similar useful, or you can download this from http://www.earthspike.net/FreeSWITCH/testtones.wav (2.2MB on my domestic broadband). This is what I have recorded so far (using another FreeSWITCH server on the PSTN side): http://www.earthspike.net/FreeSWITCH/choppy_voice.wav (3.3MB on domestic broadband). You will notice that the outgoing IVR message is perfectly clear, and the voicemail outgoing message (test tones) gets messed with. Both are 8kHz WAV, so there is no transcoding or anything like that going on. Sometimes hold music goes choppy, sometimes not, sometimes the IVR message goes choppy as well. Sometimes the choppiness persists until I restart FreeSWITCH; other times it is for just one call. I have also written a script to capture 15s of the voice legs, using capture commands outlined by Sangoma support to include stuff around the echo canceller. You may also find this useful: #!/bin/bash echo "Starting diagnostics" DIR_PREFIX=/usr/local/freeswitch/recordings/diagnostics FILE_PREFIX=${DIR_PREFIX}/`date +%Y%m%d-%H%M%S` FS_CLI_X="/usr/local/freeswitch/bin/fs_cli -x" WAN_EC_C=/usr/sbin/wan_ec_client SPAN=`${FS_CLI_X} 'show channels' | sed -e '/FreeTDM/!d;s|.*FreeTDM/\([0-9]\):\([0-9]\)/.*|\1|;'` CHNL=`${FS_CLI_X} 'show channels' | sed -e '/FreeTDM/!d;s|.*FreeTDM/\([0-9]\):\([0-9]\)/.*|\2|;'` echo Diagnostic capture utility echo ========================== echo Detected call on ${SPAN}:${CHNL} echo Setting up trace on span wp${SPAN} ${FS_CLI_X} "ftdm trace ${FILE_PREFIX} ${SPAN}" pushd ${DIR_PREFIX} echo Recording 15s EC sample on ${SPAN}:${CHNL} ${WAN_EC_C} wanpipe${SPAN} monitor ${CHNL} popd echo Stopping trace ${FS_CLI_X} "ftdm notrace ${SPAN}" echo Done. The script relies on there only being one channel active when you want to record the call, so you may have to tune this for your requirements. I have created a folder recordings/diagnostics for the captured output. The non-profit I do this for has low enough traffic to make the single call assumption usually valid. If I am near an ssh session, though, I would work out the SPAN/CHNL variables manually in case more than one was active. If you get anywhere with this, let me or the list know. Note for devs: I haven't opened a FreeSWITCH JIRA on this because it is only calls involving my Sangoma card that are affected. If the deeper tracing that Sangoma have requested show that it is not in their card, then it will go to the FreeSWITCH JIRA. If you wish, I can open a JIRA but I'm not sure it's a good thing to have 2 separate fixes going on. John On 01/03/12 16:45, Moises Silva wrote: > On Thu, Mar 1, 2012 at 9:45 AM, Christian Benke > wrote: > > > Sangoma's Support appears to be clueless as well, they didn't have > effective suggestions so far(Opened a ticket 2 weeks). Since i don't > have this problem with identical hardware without > Sangoma-Cards(SIP-Trunk instead), my first assumption was the card. > Probably an interrupt issue? > > > Another test you can try to pinpoint the issue and determine once for > all if the problem is below FreeSWITCH is to use the "ftdm trace" > command. This command > will create an input an output file (in the directory you specify) > that contains the raw audio as it is read and written from/to the > wanpipe device. > > Type "ftdm trace" to see the command syntax. Run the command whenever > you reproduce the problem (can be run in the middle of the call) then > stop the trace with "ftdm notrace" when you're > done capturing. Note that the tracing will not stop automatically when > the call is hung up, the trace is a raw I/O trace and has no knowledge > of calls starting or being hung up. > > The audio format will be alaw (because is a BRI line), you can convert > that to a wav (with sox for example) and see if the audio issue is > there. If it is, we can start thinking on lower-level troubleshooting. > > > *Moises Silva > **/Manager, Software Engineering/*** > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > ** > > > Products > | > Solutions > | > Events > | > Contact > | > Wiki > | > Facebook > | > Twitter > `| > | YouTube > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/49e79c41/attachment-0001.html From benkokakao at gmail.com Thu Mar 1 23:06:34 2012 From: benkokakao at gmail.com (Christian Benke) Date: Thu, 1 Mar 2012 21:06:34 +0100 Subject: [Freeswitch-users] Choopy one-way noise In-Reply-To: References: Message-ID: On 1 March 2012 17:45, Moises Silva wrote: > Another test you can try to pinpoint the issue and determine once for all > if the problem is below FreeSWITCH is to use the "ftdm trace" command. Thanks for the suggestion. I have taken a more drastic approach today and temporarily replaced the A500 with a OpenVOX-card. Will have to wait for results till there is more traffic tomorrow. For some reason it seems the broken audio does appear more often from morning to noon when there's more traffic, but does not depend on the number of concurrent calls(It also appears around noon when there are no other calls, but not in the evening when there's been no traffic for a while). However, this observation is not empiric. Cheers, Christian From benkokakao at gmail.com Thu Mar 1 23:38:50 2012 From: benkokakao at gmail.com (Christian Benke) Date: Thu, 1 Mar 2012 21:38:50 +0100 Subject: [Freeswitch-users] Choopy one-way noise (FreeTDM) In-Reply-To: <4F4FCAAC.5020307@earthspike.net> References: <4F4FCAAC.5020307@earthspike.net> Message-ID: Hi John! Thanks for your long mail, it's always good to hear that you're not alone with a problem :-) > It's > intermittent and relies on the outside party reporting the quality being > poor Exactly that! In my case though it's callers who annoyedly complain about the poor quality ;-) > Your sample is quiet, which makes it hard to see what is going on. Yeah well, i recorded it with a call to a phone set to auto-answer as the problem is already audible enough on a "quiet" call. I couldn't reproduce it so far with hold music or a playback. What do you mean by "voicemail outgoing message"? Thanks for the soundfile and script-examples too, will have to intensify my monitoring as well... Best regards, Christian From wagnerspi at gmail.com Fri Mar 2 01:05:15 2012 From: wagnerspi at gmail.com (Wagner) Date: Thu, 1 Mar 2012 19:05:15 -0300 Subject: [Freeswitch-users] web application to call to fs In-Reply-To: References: Message-ID: Thanks a lot Michael, it was just what I needed. Thanks 2012/2/29 Michael Collins > Yep, if you don't mind using the flex client and mod_rtmp. Check it out: > http://wiki.freeswitch.org/wiki/Mod_rtmp > > Be sure to see the note about where to get the flex client... > > -MC > > > On Wed, Feb 29, 2012 at 5:31 PM, Wagner wrote: > >> hello , >> >> is there any Web application that i could use to let a user call to my >> ivr through my website? >> >> thanks >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/4f04e487/attachment.html From spapas82 at gmail.com Fri Mar 2 01:16:19 2012 From: spapas82 at gmail.com (spyros papadopoulos) Date: Fri, 2 Mar 2012 00:16:19 +0200 Subject: [Freeswitch-users] get digits during entire session Message-ID: Hi, I am trying to figure out a way to save the digits pressed during the entire duration of a call. I am particularly interested in doing this after a call has been bridged to an external provider. I currently use a perl script to bridge connections accordingly, based on various database look ups. Are there any functions that could be used for this? Posibbly through ESL? I would like to save the result string in an db. thanks in advance, spyros -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/e57bd5c8/attachment.html From avi at avimarcus.net Fri Mar 2 01:39:37 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 2 Mar 2012 00:39:37 +0200 Subject: [Freeswitch-users] get digits during entire session In-Reply-To: References: Message-ID: it's in the xml_cdr as digits_dialed -Avi On Fri, Mar 2, 2012 at 12:16 AM, spyros papadopoulos wrote: > Hi, > > I am trying to figure out a way to save the digits pressed during the > entire duration of a call. I am particularly interested in doing this after > a call has been bridged to an external provider. > > I currently use a perl script to bridge connections accordingly, based on > various database look ups. Are there any functions that could be used for > this? Posibbly through ESL? > I would like to save the result string in an db. > > > thanks in advance, > spyros > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/84016649/attachment.html From mario_fs at mgtech.com Fri Mar 2 02:12:29 2012 From: mario_fs at mgtech.com (Mario G) Date: Thu, 1 Mar 2012 15:12:29 -0800 Subject: [Freeswitch-users] FreeSWITCH Cookbook Is Published! In-Reply-To: References: Message-ID: <89632AE0-65E9-4739-8C56-0C6EF3174DD4@mgtech.com> I noticed the FS 1.0.6 is on the iTunes store. Will the Cookbook be there too eventually? On Feb 27, 2012, at 11:23 AM, Michael Collins wrote: > Well, Amazon will let you see it: > http://www.amazon.com/FreeSWITCH-Cookbook-Anthony-Minessale/dp/1849515409/ref=sr_1_2?s=books&ie=UTF8&qid=1330370556&sr=1-2#reader_1849515409 > > Enjoy! > -MC > > On Sat, Feb 25, 2012 at 8:29 AM, Marcin Gozdalik wrote: > 2012/2/23 Michael Collins > > Just wanted to mention a little story I wrote up on the main FreeSWITCH website: > http://www.freeswitch.org/node/381 > > Packt has officially published the the cookbook, so get your copy today! > > Congrats! Is there a TOC somewhere to have a look at before purchase? The link at Packt Publishing shows only "Sorry, the table of contents for this book is not yet available". > > Best regards > > -- > Marcin Gozdalik > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/b12f3ad4/attachment-0001.html From freeswitch at earthspike.net Fri Mar 2 02:15:36 2012 From: freeswitch at earthspike.net (John) Date: Thu, 01 Mar 2012 23:15:36 +0000 Subject: [Freeswitch-users] Choopy one-way noise (FreeTDM) In-Reply-To: References: <4F4FCAAC.5020307@earthspike.net> Message-ID: <4F500318.2030501@earthspike.net> On 01/03/12 20:38, Christian Benke wrote: >> Your sample is quiet, which makes it hard to see what is going on. > Yeah well, i recorded it with a call to a phone set to auto-answer as > the problem is already audible enough on a "quiet" call. I couldn't > reproduce it so far with hold music or a playback. What do you mean by > "voicemail outgoing message"? I set up a test extension that has no registered handset/soft client, so calls go straight to voicemail. The sound file replaces greeting_1.wav in storage/voicemail/.../1999/ (or whatever number your test extension is). John From Hector.Geraldino at ipsoft.com Fri Mar 2 02:32:25 2012 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Thu, 1 Mar 2012 18:32:25 -0500 Subject: [Freeswitch-users] get digits during entire session In-Reply-To: References: Message-ID: <6A6B4C284AD15042B429EB9D904544AD0225EAF431@NY1-EXMB-01.ip-soft.net> And in case you decide to use ESL, you surely can register your client to receive DTMF events and store the digit values accordingly. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Thursday, March 01, 2012 5:40 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] get digits during entire session it's in the xml_cdr as digits_dialed -Avi On Fri, Mar 2, 2012 at 12:16 AM, spyros papadopoulos > wrote: Hi, I am trying to figure out a way to save the digits pressed during the entire duration of a call. I am particularly interested in doing this after a call has been bridged to an external provider. I currently use a perl script to bridge connections accordingly, based on various database look ups. Are there any functions that could be used for this? Posibbly through ESL? I would like to save the result string in an db. thanks in advance, spyros _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/59987061/attachment.html From dean at dyversesolutions.com.au Thu Mar 1 09:59:35 2012 From: dean at dyversesolutions.com.au (Dean Coulter) Date: Thu, 1 Mar 2012 16:59:35 +1000 Subject: [Freeswitch-users] Retrieve active call from voicemail Message-ID: <99F37B26-F412-49A6-9F6D-FCF8E788DF70@dyversesolutions.com.au> Hi, Does anyone know how to retrieve an active call from the voicemail system? I have googled and checked the lists but couldn't find anything. At a previous employer we had this feature so if you were heading back to your desk to answer the phone and it had gone to VM, you could retrieve the call if the calling party was in the process of leaving a message. This was a very useful feature as it saved time in listening to messages and returning calls. Dean From development.milos at gmail.com Thu Mar 1 20:23:19 2012 From: development.milos at gmail.com (Milos Jovanovic) Date: Thu, 1 Mar 2012 18:23:19 +0100 Subject: [Freeswitch-users] Simple dialplan extension Message-ID: Hello. I need a simple dialplan extension that will do the following: 1) answer the call 2) play a sound file in loop (unlimited loops) 3) hangup the call after X seconds (number of seconds should be defined in extension) What's the simplest way to do that? Thanks in advance, Milos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120301/a3704bb1/attachment-0001.html From shahzad.bhatti at g-r-v.com Thu Mar 1 22:10:00 2012 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Fri, 2 Mar 2012 00:10:00 +0500 Subject: [Freeswitch-users] Use of ESL and PHP for Call Message-ID: Hi, I am a new user and want to create an application to play a message using mod_tts (cepstral) like we have a meeting on Monday at 9 are you available, if yes press 1 and no then press 2 then record the key pressed in a variable or store it in the database. but when i originate call the call hangup after speak, but i need to make call continue so the caller press the key and then call hangup. now what i do, can i make it simple or make an IVR for it. i searched alot but not manage to get the solution. so far i get the digit pressed using $digit = getHeader("DTMF-Digit"); but i need to control the call to park it after playing the message till called press the digit. my code is as: sendRecv("api create_uuid"); $uuid = $e->getBody(); $user = '1500'; $cmd1 = "bgapi originate {origination_uuid=$uuid}user/$user &speak('cepstral|william|Hi, we have a meeting on Monday at nine, are you joining us, if yes then press one and if not press two')"; $e = $esl->sendRecv($cmd1); $e = $esl->sendRecv("events plain all"); $e = $esl->filter("Unique-ID",$uuid); while ($esl->connected()) { $e = $esl->recvEvent(); $result = $e->getType(); if($result=='CHANNEL_EXECUTE_COMPLETE') { // what i do here...! $esl->execute("&sleep",10000); } $digit = $e->getHeader("DTMF-Digit"); $state_no = $e->getHeader("Channel-State-Number"); $curr_uuid = $e->getHeader("Channel-Call-UUID"); /* $ans_state = $e->getHeader("Answer-State"); $state_no = $e->getHeader("Channel-State-Number"); $core_id = $e->getHeader("Core-UUID"); */ if($curr_uuid==$uuid && $digit!=NULL) { print "input number is " . $digit . "\n"; } $state = $e->getHeader("Channel-Call-State"); if ($state == 'HANGUP') { print "\n\nHangup Cause Number: " . $state_no; $esl->disconnect(); } } ?> please help me to solve the problem Regards Shahzad Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/074d9797/attachment-0001.html From bloodyiron at shaw.ca Fri Mar 2 02:34:58 2012 From: bloodyiron at shaw.ca (BloodyIron) Date: Thu, 1 Mar 2012 15:34:58 -0800 (PST) Subject: [Freeswitch-users] Calls drop after 1:48 (kinda) Message-ID: <1330644898790-7334603.post@n2.nabble.com> Hi Folks, Okay so this one is a bit tricky to reproduce. One of our extensions will have their calls dropped preicsely after 1 minutes 48 seconds ( 1:48 ) of call time. Another extension on the same segment of the network does not do the same thing. This extension does this for every single call type, be it external or internal calls. Now you may think, well it may be a busted phone. We reset the phone to factory defaults and saw no improvement. Furthermore we are seeing it elsewhere in our freeswitch installation, as in other extensions on other sites are seeing the same issue. These extensions are behind a NAT, however the freeswitch server is publically facing (as in public IP), with a passive firewall between it and the world (no routing, no NAT, etc). Just to be clear, we are also using fusionpbx to control the installation, as typing everything into an xml is not very efficient (but we're not scared of working with xml files either). Right now, we're gonna try adding " " to our sofia.conf.xml file to address it, beyond this we are unsure what to do. Can anyone speak on this matter? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Calls-drop-after-1-48-kinda-tp7334603p7334603.html Sent from the freeswitch-users mailing list archive at Nabble.com. From avi at avimarcus.net Fri Mar 2 03:40:31 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 2 Mar 2012 02:40:31 +0200 Subject: [Freeswitch-users] Retrieve active call from voicemail In-Reply-To: <99F37B26-F412-49A6-9F6D-FCF8E788DF70@dyversesolutions.com.au> References: <99F37B26-F412-49A6-9F6D-FCF8E788DF70@dyversesolutions.com.au> Message-ID: Via ESL or api you could do uuid_transfer but I'm not sure a) how you would know they were still in the VM system and b) what button you'd decide they should push. But you'd just use hash to store the UUID right before they went to the VM system. Like the redial or intercept code in the default config. -Avi On Thu, Mar 1, 2012 at 8:59 AM, Dean Coulter wrote: > Hi, > > > Does anyone know how to retrieve an active call from the voicemail system? > I have googled and checked the lists but couldn't find anything. > > At a previous employer we had this feature so if you were heading back to > your desk to answer the phone and it had gone to VM, you could retrieve the > call if the calling party was in the process of leaving a message. This > was a very useful feature as it saved time in listening to messages and > returning calls. > > Dean > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/3353acfb/attachment.html From avi at avimarcus.net Fri Mar 2 03:42:45 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 2 Mar 2012 02:42:45 +0200 Subject: [Freeswitch-users] Simple dialplan extension In-Reply-To: References: Message-ID: It would use these functions: 1) answer 2) sched_hangup-> +400 (e.g. 400 seconds, or set a specific time) 3) endless_playback ->file.wav -Avi On Thu, Mar 1, 2012 at 7:23 PM, Milos Jovanovic wrote: > Hello. > > I need a simple dialplan extension that will do the following: > > 1) answer the call > 2) play a sound file in loop (unlimited loops) > 3) hangup the call after X seconds (number of seconds should be defined in > extension) > > What's the simplest way to do that? > > Thanks in advance, > Milos > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/abdb9393/attachment.html From eli at netspectrum.com Fri Mar 2 04:30:22 2012 From: eli at netspectrum.com (Erjian Li) Date: Fri, 2 Mar 2012 09:30:22 +0800 Subject: [Freeswitch-users] a question about freeswitch conference caller-controls In-Reply-To: References: Message-ID: When I press cellphone's key, although Freeswitch can't see this DTMF, but the the other participant of the conference call can hear my DTMF tone. Does this situation indicates that the DTMF tone has been forwarded by SIP provider's server? On Fri, Mar 2, 2012 at 1:57 AM, Godson Gera wrote: > Freeswitch only sees the SIP side of the call according to your setup. So, > make sure that your SIP provider is properly forwarding the DTMF in any one > of the supported formats (inband, rfc2833, SIP INFO ). > > > > On Thu, Mar 1, 2012 at 7:31 AM, Erjian Li wrote: > >> Hi All, >> >> I have a querstion about freeswitch conference caller controls. If anyone >> can give some hint/help, I'm very appreciated. >> >> > > -- > Thanks & Regards, > Godson Gera > IVR India > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Best Regards Erjian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/87164f14/attachment.html From peter.olsson at visionutveckling.se Fri Mar 2 08:31:35 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 2 Mar 2012 05:31:35 +0000 Subject: [Freeswitch-users] Use of ESL and PHP for Call In-Reply-To: References: Message-ID: <1FFF97C269757C458224B7C895F35F1504BE62@cantor.std.visionutv.se> You should pass the originate command to &park() instead. Right now you originate the call and then execute the application speak. After this (when speak is finished) FS has no more instructions so it will hangup the call. If you execute &park() instead, the call will be "parked" and wait for furter instructions. Then you will also need to execute the speak from ESL instead - and afther that you continue to do whatever you want to do with the call. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Shahzad Bhatti [shahzad.bhatti at g-r-v.com] Skickat: den 1 mars 2012 20:10 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] Use of ESL and PHP for Call Hi, I am a new user and want to create an application to play a message using mod_tts (cepstral) like we have a meeting on Monday at 9 are you available, if yes press 1 and no then press 2 then record the key pressed in a variable or store it in the database. but when i originate call the call hangup after speak, but i need to make call continue so the caller press the key and then call hangup. now what i do, can i make it simple or make an IVR for it. i searched alot but not manage to get the solution. so far i get the digit pressed using $digit = getHeader("DTMF-Digit"); but i need to control the call to park it after playing the message till called press the digit. my code is as: sendRecv("api create_uuid"); $uuid = $e->getBody(); $user = '1500'; $cmd1 = "bgapi originate {origination_uuid=$uuid}user/$user &speak('cepstral|william|Hi, we have a meeting on Monday at nine, are you joining us, if yes then press one and if not press two')"; $e = $esl->sendRecv($cmd1); $e = $esl->sendRecv("events plain all"); $e = $esl->filter("Unique-ID",$uuid); while ($esl->connected()) { $e = $esl->recvEvent(); $result = $e->getType(); if($result=='CHANNEL_EXECUTE_COMPLETE') { // what i do here...! $esl->execute("&sleep",10000); } $digit = $e->getHeader("DTMF-Digit"); $state_no = $e->getHeader("Channel-State-Number"); $curr_uuid = $e->getHeader("Channel-Call-UUID"); /* $ans_state = $e->getHeader("Answer-State"); $state_no = $e->getHeader("Channel-State-Number"); $core_id = $e->getHeader("Core-UUID"); */ if($curr_uuid==$uuid && $digit!=NULL) { print "input number is " . $digit . "\n"; } $state = $e->getHeader("Channel-Call-State"); if ($state == 'HANGUP') { print "\n\nHangup Cause Number: " . $state_no; $esl->disconnect(); } } ?> please help me to solve the problem Regards Shahzad Bhatti !DSPAM:4f50163132762123817623! From anita.hall at simmortel.com Fri Mar 2 09:57:05 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Fri, 2 Mar 2012 12:27:05 +0530 Subject: [Freeswitch-users] ph_tor3_e1.c Message-ID: We are running FreeTDM on a very cheap Atcom card, which used another module ph_tor3_e1 on top of Dahdi. I believe this is derived from Torrenta. http://www.atcom.cn/downloads/TelephonyCard/drivers/AX-4ET/E1/ph_tor3_e1.c On Ubuntu 10.04 this gives problem as the module ph_tor3_e1 (and hence dahdi) does not unload. Sometimes the machine hangs and needs to be rebooted. This module has not been updated for the last 2 years during which the linux kernel has changed (I am told). Is there any other manufacturer of torrent card who would be using the same architecture and keeping his drivers updated ? If not, what steps do I need to take to update this driver to kernel version 2.6.32-37-server ? These are new waters and I feel so helpless :) regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/daf84634/attachment-0001.html From miha at softnet.si Fri Mar 2 10:29:40 2012 From: miha at softnet.si (Miha Zoubek) Date: Fri, 02 Mar 2012 08:29:40 +0100 Subject: [Freeswitch-users] Caller-Callee-ID-Name(please help) Message-ID: <4F5076E4.5080904@softnet.si> Hi, How can I set this variable on a leg(Caller-Callee-ID-Name)? I have try to put in user/dir but with no luck. Usr dir: log from cdr on b leg: Caller-Direction: [outbound] Caller-Username: [018108500.fs_kabelvoip1] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Outbound Call] Caller-Caller-ID-Number: [38651357952] a leg: Caller-Caller-ID-Name: [018108500] Caller-Caller-ID-Number: [018108500.fs_kabelvoip1] Caller-Callee-ID-Name: [Outbound Call] Caller-Callee-ID-Number: [38651357952] Thanks! From miha at softnet.si Fri Mar 2 13:04:57 2012 From: miha at softnet.si (Miha Zoubek) Date: Fri, 02 Mar 2012 11:04:57 +0100 Subject: [Freeswitch-users] Caller-Callee-ID-Name(please help) In-Reply-To: <4F5076E4.5080904@softnet.si> References: <4F5076E4.5080904@softnet.si> Message-ID: <4F509B49.7060703@softnet.si> On 03/02/2012 08:29 AM, Miha Zoubek wrote: > Hi, > > How can I set this variable on a leg(Caller-Callee-ID-Name)? I have try > to put in user/dir but with no luck. > > Usr dir: > > > > > > > > > > log from cdr on b leg: > > > > Caller-Direction: [outbound] > Caller-Username: [018108500.fs_kabelvoip1] > Caller-Dialplan: [XML] > Caller-Caller-ID-Name: [Outbound Call] > Caller-Caller-ID-Number: [38651357952] > > > > a leg: > Caller-Caller-ID-Name: [018108500] > Caller-Caller-ID-Number: [018108500.fs_kabelvoip1] > Caller-Callee-ID-Name: [Outbound Call] > Caller-Callee-ID-Number: [38651357952] > > Thanks! > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > I figure it out! Thanks guys! miha From miha at softnet.si Fri Mar 2 13:09:27 2012 From: miha at softnet.si (Miha Zoubek) Date: Fri, 02 Mar 2012 11:09:27 +0100 Subject: [Freeswitch-users] 302 redirect variable Message-ID: <4F509C57.20205@softnet.si> Hi, in my directory I set variable password () for every user. After I am doing 302 redirect in my public dialplan and transfer call to extension, I can not use varible pasword. How can I get varible password, so that I can authenticate call. public dialplan: default dialplan: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/a05def75/attachment.html From bob.mccarthy at experient.com Fri Mar 2 14:50:23 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Fri, 02 Mar 2012 06:50:23 -0500 Subject: [Freeswitch-users] Hang Up Cause is Blank in Dial Plan Message-ID: <1330689023.690.7.camel@CO-999-8> I am trying to play messages for failures on outbound Calls. When I try to use ${hangup_cause} after the failed bridge command it comes back as blank. what am I doing wrong ??? "/> 2012-03-02 06:42:38.005915 [DEBUG] switch_channel.c:2850 (sofia/external/2001 at 192.168.1.195) Callstate Change RINGING -> HANGUP 2012-03-02 06:42:38.005915 [NOTICE] switch_ivr_originate.c:3183 Hangup sofia/external/2001 at 192.168.1.195 [CS_CONSUME_MEDIA] [NO_ANSWER] 2012-03-02 06:42:38.005915 [DEBUG] switch_channel.c:2873 Send signal sofia/external/2001 at 192.168.1.195 [KILL] 2012-03-02 06:42:38.005915 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2001 at 192.168.1.195 [BREAK] 2012-03-02 06:42:38.005915 [INFO] mod_dptools.c:2922 Originate Failed. Cause: NO_ANSWER EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 set(test=) 2012-03-02 06:42:38.005915 [DEBUG] mod_dptools.c:1281 sofia/internal/CO999x1001.1 at 192.168.57.211 SET [test]=[UNDEF] EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 log(1 A-leg hangup cause: ) 2012-03-02 06:42:38.005915 [ALERT] mod_dptools.c:1420 A-leg hangup cause: EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 execute_extension(hangup_reason- XML features) 2012-03-02 06:42:38.005915 [INFO] mod_dialplan_xml.c:485 Processing Dispatch 2 ->hangup_reason- in context features Dialplan: sofia/internal/CO999x1001.1 at 192.168.57.211 parsing [features->dx] continue=false -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/8e044be1/attachment.html From bob.mccarthy at experient.com Fri Mar 2 15:31:26 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Fri, 2 Mar 2012 05:31:26 -0700 Subject: [Freeswitch-users] Hang Up Cause is Blank in Dial Plan In-Reply-To: <1330689023.690.7.camel@CO-999-8> References: <1330689023.690.7.camel@CO-999-8> Message-ID: <009201ccf870$642602a0$2c7207e0$@mccarthy@experient.com> Just Answering my Own Question here, I did the command and found that I can use: last_bridge_hangup_cause originate_disposition DIALSTATUS and not hangup_cause in the dial plan From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob McCarthy Sent: Friday, March 02, 2012 4:50 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Hang Up Cause is Blank in Dial Plan I am trying to play messages for failures on outbound Calls. When I try to use ${hangup_cause} after the failed bridge command it comes back as blank. what am I doing wrong ??? "/> 2012-03-02 06:42:38.005915 [DEBUG] switch_channel.c:2850 (sofia/external/2001 at 192.168.1.195) Callstate Change RINGING -> HANGUP 2012-03-02 06:42:38.005915 [NOTICE] switch_ivr_originate.c:3183 Hangup sofia/external/2001 at 192.168.1.195 [CS_CONSUME_MEDIA] [NO_ANSWER] 2012-03-02 06:42:38.005915 [DEBUG] switch_channel.c:2873 Send signal sofia/external/2001 at 192.168.1.195 [KILL] 2012-03-02 06:42:38.005915 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2001 at 192.168.1.195 [BREAK] 2012-03-02 06:42:38.005915 [INFO] mod_dptools.c:2922 Originate Failed. Cause: NO_ANSWER EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 set(test=) 2012-03-02 06:42:38.005915 [DEBUG] mod_dptools.c:1281 sofia/internal/CO999x1001.1 at 192.168.57.211 SET [test]=[UNDEF] EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 log(1 A-leg hangup cause: ) 2012-03-02 06:42:38.005915 [ALERT] mod_dptools.c:1420 A-leg hangup cause: EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 execute_extension(hangup_reason- XML features) 2012-03-02 06:42:38.005915 [INFO] mod_dialplan_xml.c:485 Processing Dispatch 2 ->hangup_reason- in context features Dialplan: sofia/internal/CO999x1001.1 at 192.168.57.211 parsing [features->dx] continue=false -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/b3cac9ba/attachment-0001.html From peder at networkoblivion.com Fri Mar 2 15:32:39 2012 From: peder at networkoblivion.com (peder at networkoblivion.com) Date: Fri, 2 Mar 2012 06:32:39 -0600 Subject: [Freeswitch-users] Calls drop after 1:48 (kinda) In-Reply-To: <1330644898790-7334603.post@n2.nabble.com> References: <1330644898790-7334603.post@n2.nabble.com> Message-ID: <071601ccf870$8e788370$ab698a50$@networkoblivion.com> We would need some console debugging to even begin to figure out what is wrong. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of BloodyIron Sent: Thursday, March 01, 2012 5:35 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Calls drop after 1:48 (kinda) Hi Folks, Okay so this one is a bit tricky to reproduce. One of our extensions will have their calls dropped preicsely after 1 minutes 48 seconds ( 1:48 ) of call time. Another extension on the same segment of the network does not do the same thing. This extension does this for every single call type, be it external or internal calls. Now you may think, well it may be a busted phone. We reset the phone to factory defaults and saw no improvement. Furthermore we are seeing it elsewhere in our freeswitch installation, as in other extensions on other sites are seeing the same issue. These extensions are behind a NAT, however the freeswitch server is publically facing (as in public IP), with a passive firewall between it and the world (no routing, no NAT, etc). Just to be clear, we are also using fusionpbx to control the installation, as typing everything into an xml is not very efficient (but we're not scared of working with xml files either). Right now, we're gonna try adding " " to our sofia.conf.xml file to address it, beyond this we are unsure what to do. Can anyone speak on this matter? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Calls-drop-after-1-48-kinda-tp 7334603p7334603.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From dgarcia at anew.com.ve Fri Mar 2 16:13:53 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Fri, 02 Mar 2012 08:43:53 -0430 Subject: [Freeswitch-users] Choopy one-way noise In-Reply-To: References: Message-ID: <4F50C791.1020109@anew.com.ve> Hi Chirstian, Have you checked your physical installation? The problem could be originated by an electrical peek or perhaps due to your equipment are not well isolated electrically (ground, peeks, etc). Perhaps an equipment or devise in your office introduce some noise when it is activated. You need do more monitoring. After change tdm board have you notice difference, the issue persist? On 3/1/2012 3:36 PM, Christian Benke wrote: > On 1 March 2012 17:45, Moises Silva wrote: >> Another test you can try to pinpoint the issue and determine once for all >> if the problem is below FreeSWITCH is to use the "ftdm trace" command. > Thanks for the suggestion. I have taken a more drastic approach today > and temporarily replaced the A500 with a OpenVOX-card. Will have to > wait for results till there is more traffic tomorrow. > For some reason it seems the broken audio does appear more often from > morning to noon when there's more traffic, but does not depend on the > number of concurrent calls(It also appears around noon when there are > no other calls, but not in the evening when there's been no traffic > for a while). However, this observation is not empiric. > > Cheers, > Christian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1913 / Virus Database: 2114/4844 - Release Date: 03/01/12 > > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/7f11bd87/attachment.html From dgarcia at anew.com.ve Fri Mar 2 16:24:47 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Fri, 02 Mar 2012 08:54:47 -0430 Subject: [Freeswitch-users] get digits during entire session In-Reply-To: References: Message-ID: <4F50CA1F.5050204@anew.com.ve> Hi, Spyros I don't know what are you trying to exactly but you have consider one point: "user privacy". Depending, in your local laws and policies, you have to consider in "how to protect sensible data" like account/credit card numbers, PIN, etc. I mention this because as you explained, you want to record/capture all dtmf session, specilly when a call has bridged to an external provider. What is the remote end is bank? How you could distinct what to/not to save? I think you are heading to dark waters. On 3/1/2012 5:46 PM, spyros papadopoulos wrote: > Hi, > I am trying to figure out a way to save the digits pressed during the > entire duration of a call. I am particularly interested in doing this > after a call has been bridged to an external provider. > I currently use a perl script to bridge connections accordingly, based > on various database look ups. Are there any functions that could be > used for this? Posibbly through ESL? > I would like to save the result string in an db. > thanks in advance, > spyros > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1913 / Virus Database: 2114/4844 - Release Date: 03/01/12 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/5500df5f/attachment.html From admharris at gmx.com Fri Mar 2 16:31:53 2012 From: admharris at gmx.com (adam harris) Date: Fri, 02 Mar 2012 14:31:53 +0100 Subject: [Freeswitch-users] execution ends after lcr command Message-ID: <20120302133153.142490@gmx.com> >From my esl application after executing lcr command, the freeswitch log says the last command has exectuted and the call is ended by hanging up The poision point where the lcr command is run in my application is not the last command, why does freeswitch think it has and hang up the call? From sip at inbox.com Fri Mar 2 16:43:34 2012 From: sip at inbox.com (Jimmy Godbout) Date: Fri, 2 Mar 2012 05:43:34 -0800 Subject: [Freeswitch-users] French sounds Message-ID: <64E8AB7758E.00000D96sip@inbox.com> Hi, When will the new french sounds be available ? Thanks, Sipster ____________________________________________________________ FREE 3D MARINE AQUARIUM SCREENSAVER - Watch dolphins, sharks & orcas on your desktop! Check it out at http://www.inbox.com/marineaquarium From krice at freeswitch.org Fri Mar 2 17:40:04 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 02 Mar 2012 08:40:04 -0600 Subject: [Freeswitch-users] Hang Up Cause is Blank in Dial Plan In-Reply-To: <1330689023.690.7.camel@CO-999-8> Message-ID: Variables are not expanded during the actual execution phase of dialplan processing ... What you are trying to do here will most likely require a transfer... You should read up on the wiki how the dialplan is processed and how variables are expanded... K On 3/2/12 5:50 AM, "Bob McCarthy" wrote: > I am trying to play messages for failures on outbound Calls. When I try to > use ${hangup_cause} after the failed bridge command it comes back as blank. > > what am I doing wrong ??? > > > > > > > > > data="sofia/external/${destination_number}@$${Switchvox}"/> > > > data="hangup_reason-${hangup_cause} XML features"/>"/> > > > > > > 2012-03-02 06:42:38.005915 [DEBUG] switch_channel.c:2850 > (sofia/external/2001 at 192.168.1.195) Callstate Change RINGING -> HANGUP > 2012-03-02 06:42:38.005915 [NOTICE] switch_ivr_originate.c:3183 Hangup > sofia/external/2001 at 192.168.1.195 [CS_CONSUME_MEDIA] [NO_ANSWER] > 2012-03-02 06:42:38.005915 [DEBUG] switch_channel.c:2873 Send signal > sofia/external/2001 at 192.168.1.195 [KILL] > 2012-03-02 06:42:38.005915 [DEBUG] switch_core_session.c:1180 Send signal > sofia/external/2001 at 192.168.1.195 [BREAK] > 2012-03-02 06:42:38.005915 [INFO] mod_dptools.c:2922 Originate Failed. Cause: > NO_ANSWER > EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 set(test=) > 2012-03-02 06:42:38.005915 [DEBUG] mod_dptools.c:1281 > sofia/internal/CO999x1001.1 at 192.168.57.211 SET [test]=[UNDEF] > EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 log(1 A-leg hangup cause: ) > 2012-03-02 06:42:38.005915 [ALERT] mod_dptools.c:1420 A-leg hangup cause: > EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 > execute_extension(hangup_reason- XML features) > 2012-03-02 06:42:38.005915 [INFO] mod_dialplan_xml.c:485 Processing Dispatch 2 > ->hangup_reason- in context features > Dialplan: sofia/internal/CO999x1001.1 at 192.168.57.211 parsing [features->dx] > continue=false > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/8851365d/attachment-0001.html From avi at avimarcus.net Fri Mar 2 18:27:53 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 2 Mar 2012 17:27:53 +0200 Subject: [Freeswitch-users] Hang Up Cause is Blank in Dial Plan In-Reply-To: References: <1330689023.690.7.camel@CO-999-8> Message-ID: Nah, it's executed after the bridge so that's fine Ken. The issue, as he found, is you can't use the hangup_cause within the A leg.. because it's not hung up yet. He found other variable that ARE set in the A leg, though. -Avi On Fri, Mar 2, 2012 at 4:40 PM, Ken Rice wrote: > Variables are not expanded during the actual execution phase of dialplan > processing ... > > What you are trying to do here will most likely require a transfer... You > should read up on the wiki how the dialplan is processed and how variables > are expanded... > > > K > > > On 3/2/12 5:50 AM, "Bob McCarthy" wrote: > > I am trying to play messages for failures on outbound Calls. When I try > to use ${hangup_cause} after the failed bridge command it comes back as > blank. > > what am I doing wrong ??? > > > > > > > > > data="sofia/external/${destination_number}@$${Switchvox}"/> > > > data="hangup_reason-${hangup_cause} XML features"/>"/> > > > > > > 2012-03-02 06:42:38.005915 [DEBUG] switch_channel.c:2850 ( > sofia/external/2001 at 192.168.1.195) Callstate Change RINGING -> HANGUP > 2012-03-02 06:42:38.005915 [NOTICE] switch_ivr_originate.c:3183 Hangup > sofia/external/2001 at 192.168.1.195 [CS_CONSUME_MEDIA] [NO_ANSWER] > 2012-03-02 06:42:38.005915 [DEBUG] switch_channel.c:2873 Send signal > sofia/external/2001 at 192.168.1.195 [KILL] > 2012-03-02 06:42:38.005915 [DEBUG] switch_core_session.c:1180 Send signal > sofia/external/2001 at 192.168.1.195 [BREAK] > 2012-03-02 06:42:38.005915 [INFO] mod_dptools.c:2922 *Originate Failed. > Cause: NO_ANSWER > *EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 *set(test=) > *2012-03-02 06:42:38.005915 [DEBUG] mod_dptools.c:1281 > sofia/internal/CO999x1001.1 at 192.168.57.211 *SET [test]=[UNDEF] > *EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 log(1 A-leg hangup > cause: ) > 2012-03-02 06:42:38.005915 [ALERT] mod_dptools.c:1420 A-leg hangup cause: > EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211execute_extension(hangup_reason- XML features) > 2012-03-02 06:42:38.005915 [INFO] mod_dialplan_xml.c:485 Processing > Dispatch 2 ->hangup_reason- in context features > Dialplan: sofia/internal/CO999x1001.1 at 192.168.57.211 parsing > [features->dx] continue=false > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/e4ede9f3/attachment.html From anthony.minessale at gmail.com Fri Mar 2 20:06:40 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 2 Mar 2012 11:06:40 -0600 Subject: [Freeswitch-users] ph_tor3_e1.c In-Reply-To: References: Message-ID: Try using the old zaptel drivers circa asterisk 1.2 Nothing has really changed in that card since then and it would work with FreeTDM an the zt io module. On Fri, Mar 2, 2012 at 12:57 AM, Anita Hall wrote: > We are? running FreeTDM on a very cheap Atcom card, which used another > module ph_tor3_e1 on top of Dahdi. I believe this is derived from Torrenta. > http://www.atcom.cn/downloads/TelephonyCard/drivers/AX-4ET/E1/ph_tor3_e1.c > > On Ubuntu 10.04 this gives problem as the module ph_tor3_e1 (and hence > dahdi) does not unload. Sometimes the machine hangs and needs to be > rebooted. > > This module has not been updated for the last 2 years during which the linux > kernel has changed (I am told). > > Is there any other manufacturer of torrent card who would be using the same > architecture and keeping his drivers updated ? > > If not, what steps do I need to take to update this driver to kernel version > 2.6.32-37-server ? > > These are new waters and I feel so helpless :) > > regards, > Anita > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From dean at dyversesolutions.com.au Fri Mar 2 07:46:49 2012 From: dean at dyversesolutions.com.au (Dean Coulter) Date: Fri, 02 Mar 2012 14:46:49 +1000 Subject: [Freeswitch-users] Retrieve active call from voicemail In-Reply-To: References: <99F37B26-F412-49A6-9F6D-FCF8E788DF70@dyversesolutions.com.au> Message-ID: <4F5050B9.6060400@dyversesolutions.com.au> Thanks Avi, I have it working using the UUID. Just before the call diverts to VM it inserts a hash. The user can then dial # to retrieve(intercept) the call from the VM system. The VM loopback hangsup at the time the call is intercepted. Note that in this instance, any extension can retrieve the call by dialling # (this is the main number which rings on all phones). Regards, Dean Coulter Dyverse Solutions Pty Ltd PO Box 10888 Adelaide Street BRISBANE QLD 4000 Mob +61 (0) 448 859 977 ACN 112 999 452 http://www.dyversesolutions.com.au Caution This message may contain privileged and confidential information intended only for the use of the addressee(s) named above. If you are not the intended recipient of this message you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify the sender immediately. On 02/03/12 10:40, Avi Marcus wrote: > Via ESL or api you could do uuid_transfer > but I'm > not sure a) how you would know they were still in the VM system and b) > what button you'd decide they should push. > But you'd just use hash to store the UUID right before they went to > the VM system. Like the redial or intercept code in the default config. > > -Avi > > > On Thu, Mar 1, 2012 at 8:59 AM, Dean Coulter > > > wrote: > > Hi, > > > Does anyone know how to retrieve an active call from the voicemail > system? I have googled and checked the lists but couldn't find > anything. > > At a previous employer we had this feature so if you were heading > back to your desk to answer the phone and it had gone to VM, you > could retrieve the call if the calling party was in the process of > leaving a message. This was a very useful feature as it saved > time in listening to messages and returning calls. > > Dean > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/e0a71f3f/attachment-0001.html From brent.paddon at gmail.com Fri Mar 2 11:18:27 2012 From: brent.paddon at gmail.com (Brent Paddon) Date: Fri, 2 Mar 2012 18:18:27 +1000 Subject: [Freeswitch-users] Calls drop after 1:48 (kinda) In-Reply-To: <1330644898790-7334603.post@n2.nabble.com> References: <1330644898790-7334603.post@n2.nabble.com> Message-ID: Have you captured the network traffic between the phone and FS to see whats there? That will probably determine your next step. Brent On Fri, Mar 2, 2012 at 9:34 AM, BloodyIron wrote: > Hi Folks, > > Okay so this one is a bit tricky to reproduce. One of our extensions will > have their calls dropped preicsely after 1 minutes 48 seconds ( 1:48 ) of > call time. Another extension on the same segment of the network does not do > the same thing. This extension does this for every single call type, be it > external or internal calls. > > Now you may think, well it may be a busted phone. We reset the phone to > factory defaults and saw no improvement. Furthermore we are seeing it > elsewhere in our freeswitch installation, as in other extensions on other > sites are seeing the same issue. > > These extensions are behind a NAT, however the freeswitch server is > publically facing (as in public IP), with a passive firewall between it and > the world (no routing, no NAT, etc). > > Just to be clear, we are also using fusionpbx to control the installation, > as typing everything into an xml is not very efficient (but we're not > scared > of working with xml files either). > > Right now, we're gonna try adding " " to our sofia.conf.xml file to > address > it, beyond this we are unsure what to do. > > Can anyone speak on this matter? > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Calls-drop-after-1-48-kinda-tp7334603p7334603.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/866031ec/attachment.html From shahzad.bhatti at g-r-v.com Fri Mar 2 15:14:02 2012 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Fri, 2 Mar 2012 17:14:02 +0500 Subject: [Freeswitch-users] Use of ESL and PHP for Call Message-ID: Hi, i park the call and then execute the speak command as but execute is not working any idea ? $cmd = "api originate {origination_uuid=$uuid}user/$user &park()"; $text = "Hi, we have a meeting on Monday at 9, are you joining us, if yes then press, 1. and if not press, 2."; $e = $esl->sendRecv($cmd); $esl->execute('speak',"(cepstral|callie|$text)",$uuid); my complete code is sendRecv("api create_uuid"); $uuid = $e->getBody(); $user = '1500'; $cmd = "bgapi originate {origination_uuid=$uuid}user/$user &park()"; $text = "Hi, we have a meeting on Monday at 9, are you joining us, if yes then press, 1. and if not press, 2."; $e = $esl->sendRecv($cmd); $esl->execute('speak',"(cepstral|callie|$text)",$uuid); $e = $esl->sendRecv("events plain all"); $e = $esl->filter("Unique-ID",$uuid); while ($esl->connected()) { $e = $esl->recvEvent(); $result = $e->getType(); if($result=='CHANNEL_EXECUTE_COMPLETE') { // what i do here...! $esl->execute("&sleep",10000); } $digit = $e->getHeader("DTMF-Digit"); $state_no = $e->getHeader("Channel-State-Number"); $curr_uuid = $e->getHeader("Channel-Call-UUID"); /* $ans_state = $e->getHeader("Answer-State"); $state_no = $e->getHeader("Channel-State-Number"); $core_id = $e->getHeader("Core-UUID"); */ if($curr_uuid==$uuid && $digit!=NULL) { print "input number is " . $digit . "\n"; } $state = $e->getHeader("Channel-Call-State"); if ($state == 'HANGUP') { print "\n\nHangup Cause Number: " . $state_no; $esl->disconnect(); } } ?> Regards Shahzad Bhatti ---------- Forwarded message ---------- From: Peter Olsson To: FreeSWITCH Users Help Cc: Date: Fri, 2 Mar 2012 05:31:35 +0000 Subject: Re: [Freeswitch-users] Use of ESL and PHP for Call You should pass the originate command to &park() instead. Right now you originate the call and then execute the application speak. After this (when speak is finished) FS has no more instructions so it will hangup the call. If you execute &park() instead, the call will be "parked" and wait for furter instructions. Then you will also need to execute the speak from ESL instead - and afther that you continue to do whatever you want to do with the call. /Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/0e9e6de2/attachment.html From phazlett at gmail.com Fri Mar 2 17:35:13 2012 From: phazlett at gmail.com (Paul Hazlett) Date: Fri, 2 Mar 2012 09:35:13 -0500 Subject: [Freeswitch-users] bypass_media_after_bridge codec Negotiation Problem Message-ID: I need some advice on how to handle a codec issue during a bridge setup with media bypass. Here is an example of what is going on. I use freeswitch to allow potential customers to try our service before signing up. They can call the PSTN for a limited number of minutes - then freeswitch cuts them off. When they dial a PSTN number, freeswitch first tells them how many remaining minutes they have, then it bidges to the PSTN and then jumps out of the media stream with bypass_media_after_bridge - this all works and the call is setup, but without audio because the codecs are not negotiated correctly and I cannot figure out how to make freeswitch cooperate. A concrete example; The caller supports codecs X and Y, with X being preferred (SILK, PCMU). Freeswitch supports codecs X and Y with X being preferred (SILK, PCMU). The PSTN gateway supports codecs Y and Z with Z being preferred (G729, PCMU). When freeswitch answers the inbound call using ESL pre_answer(), the codec in the 183 response is X (SILK). Freeswitch plays audio to the caller to tell them their remaining trial usage minutes and that works as expected. Next, freeswitch does a bridge to the PSTN, offering codecs X and Y (SILK, PCMU). The PSTN gateway replies 200OK with codec Y (PCMU) as it does not support codec X (SILK). Freeswitch then does media bypass, first sending a re-INVITE to the PSTN gateway and getting it pointed to the caller's IP/port. It again offers codecs X and Y and the PSTN again replies 200OK with codec Y (PCMU). Now for the problem (line 749 in the pastbin log); Freeswitch then re-INVITES the caller, but with codecs X and Y instead of just Y and since the caller prefers X it replies 200OK with codec X (SILK). The the call is set up. The caller and the PSTN gateway are bridged and talking to each other on the correct IP and ports - but with different codecs. So how can I get freeswitch to re-INVITE the A-leg with the same codes that the B-leg accepted? See the highlighted area at line 749: http://pastebin.freeswitch.org/18561 Also not that this was captured at our SIP load balancer, so you'll see all the messages passing on internal and external interfaces). Configuration info: FreeSWITCH Version 1.0.head git-b290936 2012-03-01 10-21-17 -0800 inbound-late-negotiation == false inbound-codec-negotiation == generous global_codec_prefs ==SILK,PCMU,H263,H264 outbound_codec_prefs == SILK,PCMU I am implementing this with ESL. Regards, Paul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/c7bc06fe/attachment.html From benkokakao at gmail.com Fri Mar 2 21:10:50 2012 From: benkokakao at gmail.com (Christian Benke) Date: Fri, 2 Mar 2012 19:10:50 +0100 Subject: [Freeswitch-users] Choopy one-way noise (FreeTDM) In-Reply-To: <4F500318.2030501@earthspike.net> References: <4F4FCAAC.5020307@earthspike.net> <4F500318.2030501@earthspike.net> Message-ID: > I set up a test extension that has no registered handset/soft client, so > calls go straight to voicemail. The sound file replaces greeting_1.wav > in storage/voicemail/.../1999/ (or whatever number your test extension is). Ah, ok - but wouldn't playback do the same? From benkokakao at gmail.com Fri Mar 2 21:12:40 2012 From: benkokakao at gmail.com (Christian Benke) Date: Fri, 2 Mar 2012 19:12:40 +0100 Subject: [Freeswitch-users] Choopy one-way noise In-Reply-To: <4F50C791.1020109@anew.com.ve> References: <4F50C791.1020109@anew.com.ve> Message-ID: On 2 March 2012 14:13, Saugort Dario Garcia Tovar wrote: > Have you checked your physical installation? The problem could be originated > by an electrical peek or perhaps due to your equipment are not well isolated > electrically (ground, peeks, etc).? Perhaps an equipment or devise in your > office introduce some noise when it is activated. I also have this suspicion, as it appears to happen more often at a certain time of the day. > You need do more monitoring. > > After change tdm board have you notice difference, the issue persist? It was better, but today there was not much traffic - will have to wait till Monday, when business starts again... Regards, Christian From bob.mccarthy at experient.com Fri Mar 2 21:18:40 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Fri, 2 Mar 2012 11:18:40 -0700 Subject: [Freeswitch-users] Hang Up Cause is Blank in Dial Plan In-Reply-To: References: <1330689023.690.7.camel@CO-999-8> Message-ID: <00d001ccf8a0$e64aa850$b2dff8f0$@mccarthy@experient.com> I did notice that during a CALL_REJECTED that the execute_extension statement did not get executed .. But it does work for NO_ANSWER. Would using a Transfer change that behavior ??? 2012-03-02 13:04:32.569026 [DEBUG] mod_dptools.c:1281 sofia/internal/CO999x1001.1 at 192.168.57.211 SET [hangup_after_bridge]=[false] EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 bridge(sofia/external/2010 at 192.168.1.195) 2012-03-02 13:04:32.589032 [DEBUG] switch_channel.c:1047 sofia/internal/CO999x1001.1 at 192.168.57.211 EXPORTING[export_vars] [RFC2822_DATE]=[Fri, 02 Mar 2012 13:04:32 -0500] to event 2012-03-02 13:04:32.589032 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-02 13:04:32.589032 [NOTICE] switch_channel.c:926 New Channel sofia/external/2010 at 192.168.1.195 [29bcf518-6492-11e1-b71d-ddc06b002bf3] 2012-03-02 13:04:32.589032 [DEBUG] mod_sofia.c:4679 (sofia/external/2010 at 192.168.1.195) State Change CS_NEW -> CS_INIT 2012-03-02 13:04:32.589032 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.589032 [DEBUG] switch_core_state_machine.c:362 (sofia/external/2010 at 192.168.1.195) Running State Change CS_INIT 2012-03-02 13:04:32.589032 [DEBUG] switch_core_state_machine.c:401 (sofia/external/2010 at 192.168.1.195) State INIT 2012-03-02 13:04:32.589032 [DEBUG] mod_sofia.c:85 sofia/external/2010 at 192.168.1.195 SOFIA INIT 2012-03-02 13:04:32.609021 [DEBUG] mod_sofia.c:125 (sofia/external/2010 at 192.168.1.195) State Change CS_INIT -> CS_ROUTING 2012-03-02 13:04:32.609021 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.609021 [DEBUG] switch_core_state_machine.c:401 (sofia/external/2010 at 192.168.1.195) State INIT going to sleep 2012-03-02 13:04:32.609021 [DEBUG] switch_core_state_machine.c:362 (sofia/external/2010 at 192.168.1.195) Running State Change CS_ROUTING 2012-03-02 13:04:32.609021 [DEBUG] switch_channel.c:1886 (sofia/external/2010 at 192.168.1.195) Callstate Change DOWN -> RINGING 2012-03-02 13:04:32.609021 [DEBUG] switch_core_state_machine.c:410 (sofia/external/2010 at 192.168.1.195) State ROUTING 2012-03-02 13:04:32.609021 [DEBUG] mod_sofia.c:148 sofia/external/2010 at 192.168.1.195 SOFIA ROUTING 2012-03-02 13:04:32.609021 [DEBUG] switch_ivr_originate.c:66 (sofia/external/2010 at 192.168.1.195) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-03-02 13:04:32.609021 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.609021 [DEBUG] switch_core_state_machine.c:410 (sofia/external/2010 at 192.168.1.195) State ROUTING going to sleep 2012-03-02 13:04:32.609021 [DEBUG] switch_core_state_machine.c:362 (sofia/external/2010 at 192.168.1.195) Running State Change CS_CONSUME_MEDIA 2012-03-02 13:04:32.609021 [DEBUG] switch_core_session.c:875 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.609021 [DEBUG] switch_core_state_machine.c:429 (sofia/external/2010 at 192.168.1.195) State CONSUME_MEDIA 2012-03-02 13:04:32.609021 [DEBUG] switch_core_state_machine.c:429 (sofia/external/2010 at 192.168.1.195) State CONSUME_MEDIA going to sleep 2012-03-02 13:04:32.609021 [DEBUG] sofia.c:5526 Channel sofia/external/2010 at 192.168.1.195 entering state [calling][0] 2012-03-02 13:04:32.692026 [DEBUG] switch_core_session.c:875 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.692026 [DEBUG] switch_core_session.c:875 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.692026 [DEBUG] switch_core_session.c:875 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.692026 [DEBUG] sofia.c:5526 Channel sofia/external/2010 at 192.168.1.195 entering state [terminated][603] 2012-03-02 13:04:32.692026 [DEBUG] switch_channel.c:2850 (sofia/external/2010 at 192.168.1.195) Callstate Change RINGING -> HANGUP 2012-03-02 13:04:32.692026 [NOTICE] sofia.c:6293 Hangup sofia/external/2010 at 192.168.1.195 [CS_CONSUME_MEDIA] [CALL_REJECTED] 2012-03-02 13:04:32.692026 [DEBUG] switch_channel.c:2873 Send signal sofia/external/2010 at 192.168.1.195 [KILL] 2012-03-02 13:04:32.692026 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:362 (sofia/external/2010 at 192.168.1.195) Running State Change CS_HANGUP 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:602 (sofia/external/2010 at 192.168.1.195) State HANGUP 2012-03-02 13:04:32.692026 [DEBUG] mod_sofia.c:469 Channel sofia/external/2010 at 192.168.1.195 hanging up, cause: CALL_REJECTED 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:47 sofia/external/2010 at 192.168.1.195 Standard HANGUP, cause: CALL_REJECTED 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:602 (sofia/external/2010 at 192.168.1.195) State HANGUP going to sleep 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:393 (sofia/external/2010 at 192.168.1.195) State Change CS_HANGUP -> CS_REPORTING 2012-03-02 13:04:32.692026 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:362 (sofia/external/2010 at 192.168.1.195) Running State Change CS_REPORTING 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:662 (sofia/external/2010 at 192.168.1.195) State REPORTING 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:79 sofia/external/2010 at 192.168.1.195 Standard REPORTING, cause: CALL_REJECTED 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:662 (sofia/external/2010 at 192.168.1.195) State REPORTING going to sleep 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:387 (sofia/external/2010 at 192.168.1.195) State Change CS_REPORTING -> CS_DESTROY 2012-03-02 13:04:32.692026 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.692026 [DEBUG] switch_core_session.c:1380 Session 2 (sofia/external/2010 at 192.168.1.195) Locked, Waiting on external entities 2012-03-02 13:04:32.712027 [DEBUG] switch_ivr_originate.c:3365 Originate Resulted in Error Cause: 21 [CALL_REJECTED] 2012-03-02 13:04:32.712027 [INFO] mod_dptools.c:2922 Originate Failed. Cause: CALL_REJECTED 2012-03-02 13:04:32.712027 [DEBUG] switch_channel.c:2850 (sofia/internal/CO999x1001.1 at 192.168.57.211) Callstate Change RINGING -> HANGUP 2012-03-02 13:04:32.712027 [NOTICE] mod_dptools.c:3041 Hangup sofia/internal/CO999x1001.1 at 192.168.57.211 [CS_EXECUTE] [CALL_REJECTED] 2012-03-02 13:04:32.712027 [DEBUG] switch_channel.c:2873 Send signal sofia/internal/CO999x1001.1 at 192.168.57.211 [KILL] 2012-03-02 13:04:32.712027 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/CO999x1001.1 at 192.168.57.211 [BREAK] 2012-03-02 13:04:32.712027 [DEBUG] switch_core_session.c:2285 sofia/internal/CO999x1001.1 at 192.168.57.211 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Friday, March 02, 2012 8:28 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Hang Up Cause is Blank in Dial Plan Nah, it's executed after the bridge so that's fine Ken. The issue, as he found, is you can't use the hangup_cause within the A leg.. because it's not hung up yet. He found other variable that ARE set in the A leg, though. -Avi On Fri, Mar 2, 2012 at 4:40 PM, Ken Rice wrote: Variables are not expanded during the actual execution phase of dialplan processing ... What you are trying to do here will most likely require a transfer... You should read up on the wiki how the dialplan is processed and how variables are expanded... K On 3/2/12 5:50 AM, "Bob McCarthy" wrote: I am trying to play messages for failures on outbound Calls. When I try to use ${hangup_cause} after the failed bridge command it comes back as blank. what am I doing wrong ??? "/> 2012-03-02 06:42:38.005915 [DEBUG] switch_channel.c:2850 (sofia/external/2001 at 192.168.1.195) Callstate Change RINGING -> HANGUP 2012-03-02 06:42:38.005915 [NOTICE] switch_ivr_originate.c:3183 Hangup sofia/external/2001 at 192.168.1.195 [CS_CONSUME_MEDIA] [NO_ANSWER] 2012-03-02 06:42:38.005915 [DEBUG] switch_channel.c:2873 Send signal sofia/external/2001 at 192.168.1.195 [KILL] 2012-03-02 06:42:38.005915 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2001 at 192.168.1.195 [BREAK] 2012-03-02 06:42:38.005915 [INFO] mod_dptools.c:2922 Originate Failed. Cause: NO_ANSWER EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 set(test=) 2012-03-02 06:42:38.005915 [DEBUG] mod_dptools.c:1281 sofia/internal/CO999x1001.1 at 192.168.57.211 SET [test]=[UNDEF] EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 log(1 A-leg hangup cause: ) 2012-03-02 06:42:38.005915 [ALERT] mod_dptools.c:1420 A-leg hangup cause: EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 execute_extension(hangup_reason- XML features) 2012-03-02 06:42:38.005915 [INFO] mod_dialplan_xml.c:485 Processing Dispatch 2 ->hangup_reason- in context features Dialplan: sofia/internal/CO999x1001.1 at 192.168.57.211 parsing [features->dx] continue=false _____ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/c01cb1cc/attachment-0001.html From johnrose at google.hm Fri Mar 2 21:27:15 2012 From: johnrose at google.hm (John Rose) Date: Fri, 2 Mar 2012 13:27:15 -0500 Subject: [Freeswitch-users] ManagedESL Run Issues Message-ID: <000401ccf8a2$18c42580$4a4c7080$@google.hm> Has anyone tried to run the latest ManagedEslTest.2010 and run into esl.dll issues such as: "An attempt was made to load a program with an incorrect format. (Exception from HRESULT: 0x8007000B)" at ESLPINVOKE.SWIGExceptionHelper.SWIGRegisterExceptionCallbacks_ESL(ExceptionD elegate applicationDelegate, ExceptionDelegate arithmeticDelegate, ExceptionDelegate divideByZeroDelegate, ExceptionDelegate indexOutOfRangeDelegate, ExceptionDelegate invalidCastDelegate, ExceptionDelegate invalidOperationDelegate, ExceptionDelegate ioDelegate, ExceptionDelegate nullReferenceDelegate, ExceptionDelegate outOfMemoryDelegate, ExceptionDelegate overflowDelegate, ExceptionDelegate systemExceptionDelegate) at ESLPINVOKE.SWIGExceptionHelper..cctor() in D:\FreeSWITCH\freeswitch\libs\esl\managed\ESLPINVOKE.cs:line 105 John -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/cdeb0022/attachment.html From sat at calgaryit.com Sat Mar 3 00:03:50 2012 From: sat at calgaryit.com (George Sapak) Date: Fri, 2 Mar 2012 14:03:50 -0700 (MST) Subject: [Freeswitch-users] Server crashed In-Reply-To: <479987783.21589.1330722115278.JavaMail.root@server3> Message-ID: <1744445147.21598.1330722230718.JavaMail.root@server3> this is what was in my syslog: Mar 2 11:50:36 pbx kernel: [7508355.081072] freeswitch[16804]: segfault at 31c0 ip 00007fa012ec8f04 sp 00007fa0001c9a80 error 4 in libfreeswitch.so.1.0.0[7fa012e70000+1c2000] any ideas, running ubuntu 10 64bit and switch version FreeSWITCH Version 1.0.head (git-2e651c8 2011-07-03 22-35-44 -0500) Thank You, George. From krice at freeswitch.org Sat Mar 3 00:14:55 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 02 Mar 2012 15:14:55 -0600 Subject: [Freeswitch-users] Server crashed In-Reply-To: <1744445147.21598.1330722230718.JavaMail.root@server3> Message-ID: That doesn't really tell us anything... Do you have a core file? See http://wiki.freeswitch.org/wiki/Debugging_Freeswitch for more debugging in formation K On 3/2/12 3:03 PM, "George Sapak" wrote: > this is what was in my syslog: > > > Mar 2 11:50:36 pbx kernel: [7508355.081072] freeswitch[16804]: segfault at > 31c0 ip 00007fa012ec8f04 sp 00007fa0001c9a80 error 4 in > libfreeswitch.so.1.0.0[7fa012e70000+1c2000] > > any ideas, running ubuntu 10 64bit and switch version FreeSWITCH Version > 1.0.head (git-2e651c8 2011-07-03 22-35-44 -0500) > > Thank You, George. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch at earthspike.net Sat Mar 3 00:16:28 2012 From: freeswitch at earthspike.net (John) Date: Fri, 02 Mar 2012 21:16:28 +0000 Subject: [Freeswitch-users] Server crashed In-Reply-To: <1744445147.21598.1330722230718.JavaMail.root@server3> References: <1744445147.21598.1330722230718.JavaMail.root@server3> Message-ID: <4F5138AC.4010903@earthspike.net> Update to the latest git; in 8 months there have been many patches. If it does crash again, get a backtrace and file a JIRA ; FreeSWITCH crashing is always a bug. http://wiki.freeswitch.org/wiki/Reporting_Bugs#Reporting_A_Bug_With_JIRA http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Getting_a_Backtrace John On 02/03/12 21:03, George Sapak wrote: > this is what was in my syslog: > > > Mar 2 11:50:36 pbx kernel: [7508355.081072] freeswitch[16804]: segfault at 31c0 ip 00007fa012ec8f04 sp 00007fa0001c9a80 error 4 in libfreeswitch.so.1.0.0[7fa012e70000+1c2000] > > any ideas, running ubuntu 10 64bit and switch version FreeSWITCH Version 1.0.head (git-2e651c8 2011-07-03 22-35-44 -0500) > > Thank You, George. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/db579c4c/attachment.html From sat at calgaryit.com Sat Mar 3 00:21:28 2012 From: sat at calgaryit.com (George Sapak) Date: Fri, 2 Mar 2012 14:21:28 -0700 (MST) Subject: [Freeswitch-users] Server crashed In-Reply-To: <4F5138AC.4010903@earthspike.net> Message-ID: <2134369044.21726.1330723288905.JavaMail.root@server3> Ok I'll do that, this is the first crash since that build. Thanks, George. ----- Original Message ----- From: "John" To: freeswitch-users at lists.freeswitch.org Sent: Friday, March 2, 2012 2:16:28 PM Subject: Re: [Freeswitch-users] Server crashed Update to the latest git; in 8 months there have been many patches. If it does crash again, get a backtrace and file a JIRA ; FreeSWITCH crashing is always a bug. http://wiki.freeswitch.org/wiki/Reporting_Bugs#Reporting_A_Bug_With_JIRA http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Getting_a_Backtrace John On 02/03/12 21:03, George Sapak wrote: this is what was in my syslog: Mar 2 11:50:36 pbx kernel: [7508355.081072] freeswitch[16804]: segfault at 31c0 ip 00007fa012ec8f04 sp 00007fa0001c9a80 error 4 in libfreeswitch.so.1.0.0[7fa012e70000+1c2000] any ideas, running ubuntu 10 64bit and switch version FreeSWITCH Version 1.0.head (git-2e651c8 2011-07-03 22-35-44 -0500) Thank You, George. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From dgarcia at anew.com.ve Sat Mar 3 00:50:38 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Fri, 02 Mar 2012 17:20:38 -0430 Subject: [Freeswitch-users] web application to call to fs In-Reply-To: References: Message-ID: <4F5140AE.6000708@anew.com.ve> Hi, You have more info about mod_rtmp? How to put it to work? I got an error about could not autenticate when I login in flex client On 2/29/2012 9:01 PM, Wagner wrote: > > hello , > > is there any Web application that i could use to let a user call to my > ivr through my website? > > thanks > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1913 / Virus Database: 2114/4844 - Release Date: 03/01/12 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/62f615aa/attachment.html From jack at livecall.com Sat Mar 3 01:07:05 2012 From: jack at livecall.com (Jack) Date: Fri, 02 Mar 2012 14:07:05 -0800 Subject: [Freeswitch-users] web application to call to fs In-Reply-To: <4F5140AE.6000708@anew.com.ve> References: <4F5140AE.6000708@anew.com.ve> Message-ID: <4F514489.40500@livecall.com> Make sure you use a fully qualified username not just the extension 1001 at xxx.xxx.xxx.xxx the XXX would be the IP of your FreeSwitch Server. did you compile with mod_rtmp did you make sure you have in your modules.conf.xml? did you configure your rtmp.conf.xml and have a matching context in your dial plan? jack On 3/2/2012 1:50 PM, Saugort Dario Garcia Tovar wrote: > Hi, > > You have more info about mod_rtmp? How to put it to work? I got an > error about could not autenticate when I login in flex client > > On 2/29/2012 9:01 PM, Wagner wrote: >> >> hello , >> >> is there any Web application that i could use to let a user call to >> my ivr through my website? >> >> thanks >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.1913 / Virus Database: 2114/4844 - Release Date: 03/01/12 >> > > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/d35c577b/attachment-0001.html From brian at freeswitch.org Sat Mar 3 01:10:08 2012 From: brian at freeswitch.org (Brian West) Date: Fri, 2 Mar 2012 16:10:08 -0600 Subject: [Freeswitch-users] BKW-1 Resolved Message-ID: I have resolved my jira! :P I'm home resting. I want to thank everyone that has sent in donations, letters and gifts while I was in the hospital. I still have weeks of recovery time to endure. I'm very fortunate to not have that require me to lift heavy objects. Thanks, Brian From amelton at gmail.com Fri Mar 2 22:01:46 2012 From: amelton at gmail.com (Andrew Melton) Date: Fri, 2 Mar 2012 11:01:46 -0800 Subject: [Freeswitch-users] TCAPI_User in Contact Message-ID: I know this has been asked before, but I can't seem to put my finger on the answer, any suggestions of where to look are appreciated. Freeswitch (Sofia) is rewriting the Conact header on calls originating on a trunk ('inbound' context), routing to another trunk ('outbound' context). None of the calls are from registered users or extensions, this is more of an SBC configuration (LCR). For example, incoming to Freeswitch: Contact: Outbound from Freeswitch: Contact: I really don't want to modify the from userpary, nor do I want to include gw=trunk_2. I just can't figure out the right syntax or parameter to change it. Is it a parameter in the dialplan, or should I be using a regex on the outbound interface? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/a1bb8af2/attachment.html From stephane at lux.io Sat Mar 3 00:25:27 2012 From: stephane at lux.io (=?iso-8859-1?Q?St=E9phane_Lux?=) Date: Fri, 2 Mar 2012 22:25:27 +0100 Subject: [Freeswitch-users] *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Message-ID: <1388D9CF-ACC4-4DDF-B512-5FB9A68A5506@lux.io> Hi, when compiling FS on Mac OS 10.7.3 I get this error: make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. How can I solve it? Kind Regards, Stephane From jack at livecall.com Sat Mar 3 01:25:55 2012 From: jack at livecall.com (Jack) Date: Fri, 02 Mar 2012 14:25:55 -0800 Subject: [Freeswitch-users] web application to call to fs In-Reply-To: <4F5140AE.6000708@anew.com.ve> References: <4F5140AE.6000708@anew.com.ve> Message-ID: <4F5148F3.6030006@livecall.com> Make sure You directory .xml file for your user has the same user_context as your rtmp.config.xml On 3/2/2012 1:50 PM, Saugort Dario Garcia Tovar wrote: > Hi, > > You have more info about mod_rtmp? How to put it to work? I got an > error about could not autenticate when I login in flex client > > On 2/29/2012 9:01 PM, Wagner wrote: >> >> hello , >> >> is there any Web application that i could use to let a user call to >> my ivr through my website? >> >> thanks >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.1913 / Virus Database: 2114/4844 - Release Date: 03/01/12 >> > > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/422ee902/attachment.html From msc at freeswitch.org Sat Mar 3 01:38:56 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 2 Mar 2012 14:38:56 -0800 Subject: [Freeswitch-users] BKW-1 Resolved In-Reply-To: References: Message-ID: Yay! Glad to hear you're back home. Keep resting - as much as we've missed you we still want you to come back 100%. Thanks again to all those who stepped up to take care of one of our own! It is much appreciated. -MC On Fri, Mar 2, 2012 at 2:10 PM, Brian West wrote: > I have resolved my jira! :P I'm home resting. I want to thank everyone > that has sent in donations, letters and gifts while I was in the hospital. > I still have weeks of recovery time to endure. I'm very fortunate to not > have that require me to lift heavy objects. > > Thanks, > Brian > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/fe0997e1/attachment.html From msc at freeswitch.org Sat Mar 3 01:42:35 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 2 Mar 2012 14:42:35 -0800 Subject: [Freeswitch-users] *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. In-Reply-To: <1388D9CF-ACC4-4DDF-B512-5FB9A68A5506@lux.io> References: <1388D9CF-ACC4-4DDF-B512-5FB9A68A5506@lux.io> Message-ID: We need to see the output that comes prior to this error. Grab the previous 20+ lines of output prior to this error and let us take a look. Hopefully the true cause will be apparent. -MC 2012/3/2 St?phane Lux > Hi, > > when compiling FS on Mac OS 10.7.3 I get this error: > > make[8]: *** No rule to make target `tport/libtport.la', needed by ` > libsofia-sip-ua.la'. Stop. > > How can I solve it? > > Kind Regards, > Stephane > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/d3b796ce/attachment.html From gabe at gundy.org Sat Mar 3 01:44:59 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Fri, 2 Mar 2012 15:44:59 -0700 Subject: [Freeswitch-users] BKW-1 Resolved In-Reply-To: References: Message-ID: On Fri, Mar 2, 2012 at 3:10 PM, Brian West wrote: > I have resolved my jira! :P Do we have the proper regression testing in place? > I'm very fortunate to not have that require me to lift heavy objects. A 32 oz Dr. Pepper shouldn't be too heavy ;) Take it easy and heal up. Glad to hear the worst is behind you. Best, Gabe From admharris at gmx.com Sat Mar 3 01:51:28 2012 From: admharris at gmx.com (adam harris) Date: Fri, 02 Mar 2012 23:51:28 +0100 Subject: [Freeswitch-users] lcr problem Message-ID: <20120302225129.142450@gmx.com> lcr api command seems to have executed, debug log pasted below however I am not seeing any lcr channels variables Why are lcr channel variables missing? [DEBUG] esl.c:1123 esl_recv_event() RECV HEADER [Content-Type] = [api/response] [DEBUG] esl.c:1123 esl_recv_event() RECV HEADER [Content-Length] = [1] [DEBUG] esl.c:1321 esl_send() SEND sendmsg call-command: execute execute-app-name: lcr execute-app-arg: 8801753733050 event-lock: true [DEBUG] esl.c:1123 esl_recv_event() RECV HEADER [Content-Length] = [1949] [DEBUG] esl.c:1123 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] esl.c:1123 esl_recv_event() RECV HEADER [Content-Length] = [6259] [DEBUG] esl.c:1123 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] esl.c:1123 esl_recv_event() RECV HEADER [Content-Length] = [1968] [DEBUG] esl.c:1123 esl_recv_event() RECV HEADER [Content-Type] = [text/event-plain] [DEBUG] esl.c:1123 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] esl.c:1123 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] esl.c:1321 esl_send() SEND sendmsg From stephane at lux.io Sat Mar 3 01:54:47 2012 From: stephane at lux.io (=?iso-8859-1?Q?St=E9phane_Lux?=) Date: Fri, 2 Mar 2012 23:54:47 +0100 Subject: [Freeswitch-users] *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. In-Reply-To: References: <1388D9CF-ACC4-4DDF-B512-5FB9A68A5506@lux.io> Message-ID: <77953D5C-1EA1-4FA5-AC23-85ED95853A75@lux.io> Here is the output: Making all in nua LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo LTCOMPILE nua_notifier.lo LTCOMPILE nua_event_server.lo LTCOMPILE nua_tag.lo LTCOMPILE nua_tag_ref.lo LINK libnua.la make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. On 02.03.2012, at 23:42, Michael Collins wrote: > We need to see the output that comes prior to this error. Grab the previous 20+ lines of output prior to this error and let us take a look. Hopefully the true cause will be apparent. > > -MC > > 2012/3/2 St?phane Lux > Hi, > > when compiling FS on Mac OS 10.7.3 I get this error: > > make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. > > How can I solve it? > > Kind Regards, > Stephane > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/60b55cb3/attachment.html From stephane at lux.io Sat Mar 3 01:57:22 2012 From: stephane at lux.io (=?iso-8859-1?Q?St=E9phane_Lux?=) Date: Fri, 2 Mar 2012 23:57:22 +0100 Subject: [Freeswitch-users] *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. In-Reply-To: References: <1388D9CF-ACC4-4DDF-B512-5FB9A68A5506@lux.io> Message-ID: <384CFAFC-7D1A-48FA-8A4A-9F0A9CD04FB1@lux.io> I have tried to compile it again: tport_tls.c: In function 'tls_error': tport_tls.c:665: warning: 'SSL_get_error' is deprecated (declared at /usr/include/openssl/ssl.h:1501) tport_tls.c:680: warning: 'SSL_get_shutdown' is deprecated (declared at /usr/include/openssl/ssl.h:1568) tport_tls.c: In function 'tls_read': tport_tls.c:723: warning: 'SSL_read' is deprecated (declared at /usr/include/openssl/ssl.h:1493) tport_tls.c: In function 'tls_pending': tport_tls.c:739: warning: 'SSL_pending' is deprecated (declared at /usr/include/openssl/ssl.h:1368) tport_tls.c: In function 'tls_write': tport_tls.c:807: warning: 'SSL_write' is deprecated (declared at /usr/include/openssl/ssl.h:1495) tport_tls.c: In function 'tls_connect': tport_tls.c:899: warning: 'SSL_accept' is deprecated (declared at /usr/include/openssl/ssl.h:1491) tport_tls.c:899: warning: 'SSL_connect' is deprecated (declared at /usr/include/openssl/ssl.h:1492) tport_tls.c:900: warning: 'SSL_get_error' is deprecated (declared at /usr/include/openssl/ssl.h:1501) tport_tls.c:954: warning: 'ERR_error_string_n' is deprecated (declared at /usr/include/openssl/err.h:280) make[9]: *** [tport_tls.lo] Error 1 make[8]: *** [all] Error 2 Making all in nta Making all in nth Making all in nea Making all in iptsec Making all in nua make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. make[7]: *** [all-recursive] Error 1 Making all in packages make[6]: *** [all-recursive] Error 1 make[5]: *** [all] Error 2 make[4]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 On 02.03.2012, at 23:42, Michael Collins wrote: > We need to see the output that comes prior to this error. Grab the previous 20+ lines of output prior to this error and let us take a look. Hopefully the true cause will be apparent. > > -MC > > 2012/3/2 St?phane Lux > Hi, > > when compiling FS on Mac OS 10.7.3 I get this error: > > make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. > > How can I solve it? > > Kind Regards, > Stephane > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/1bba0f3a/attachment.html From msc at freeswitch.org Sat Mar 3 02:02:33 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 2 Mar 2012 15:02:33 -0800 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi Message-ID: Hello all! I just saw an article in Engadget that said that some "hobbyists" were trying to put FS on Raspberry Pi. I recall Richard Neese talking this subject a few months ago but I hadn't heard anything recently. If you are working to put FS on RP then please contact me off list. I would like to keep track of these projects. Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/342e7764/attachment.html From itamar at ispbrasil.com.br Sat Mar 3 02:33:21 2012 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Fri, 2 Mar 2012 20:33:21 -0300 Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH on Raspberry Pi In-Reply-To: References: Message-ID: On Fri, Mar 2, 2012 at 8:02 PM, Michael Collins wrote: > Hello all! > > I just saw an article in Engadget that said that some "hobbyists" were > trying to put FS on Raspberry Pi. I recall Richard Neese talking this > subject a few months ago but I hadn't heard anything recently. If you are > working to put FS on RP then please contact me off list. I would like to > keep track of these projects. > > Thanks, > Michael > Raspberry Pi, runs fedora. adding FS to fedora will make it available to Raspberry Pi ------------ Itamar Reis Peixoto msn, google talk: itamar at ispbrasil.com.br +55 11 4063 5033 (FIXO SP) +55 34 9158 9329 (TIM) +55 34 8806 3989 (OI) +55 34 3221 8599 (FIXO MG) From msc at freeswitch.org Sat Mar 3 02:49:10 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 2 Mar 2012 15:49:10 -0800 Subject: [Freeswitch-users] Use of ESL and PHP for Call In-Reply-To: References: Message-ID: Shahzad, I think you may have taken an unnecessarily complicated approach to this. If I were you I would simply use the play_and_get_digits application which does all the work for you. I would also handle that in the XML dialplan and then use the api_hangup_hook to call your script to process the result. (I.e. did the user press 1 or 2 or something else...) I recommend you start with something simple like a basic originate command that calls the user and then drops that call to a dialplan extension: originate user/1500 meeting_request_1500 Then create that extension: Then you just need to write your handler.php script (or whatever you want to call it) and have it parse the command line args - the first one is the user and the second one is the digit pressed. PHP scripts will work for hangup hooks but they aren't the best choice since there isn't a "mod_php" like there is a mod_lua or mod_perl. Those modules let you do cool things like "session_in_hangup_hook." However, what you're doing doesn't look like it's very difficult so the example I've given should get you going. Please note that I just did this off the top of my head w/o testing it so be sure double-check everything if you run into any unusual or unexpected behavior. Keep in mind that in this example you will need two different PHP scripts - one that actually initiates the ESL connection and does the originate and one that handles the call results. Instead of one semi-complicated script you will have two very easy scripts. Hope this helps... -MC On Fri, Mar 2, 2012 at 4:14 AM, Shahzad Bhatti wrote: > > Hi, > > i park the call and then execute the speak command as but execute is not > working any idea ? > > $cmd = "api originate {origination_uuid=$uuid}user/$user &park()"; > $text = "Hi, we have a meeting on Monday at 9, are you joining us, if yes > then press, 1. and if not press, 2."; > > $e = $esl->sendRecv($cmd); > $esl->execute('speak',"(cepstral|callie|$text)",$uuid); > > > > my complete code is > > > require_once('ESL.php'); > > $host_name = 'localhost'; > $port = '8021'; > $password = 'ClueCon'; > > $esl = new ESLconnection($host_name,$port,$password); > > $e = $esl->sendRecv("api create_uuid"); > $uuid = $e->getBody(); > $user = '1500'; > > $cmd = "bgapi originate {origination_uuid=$uuid}user/$user &park()"; > $text = "Hi, we have a meeting on Monday at 9, are you joining us, if yes > then press, 1. and if not press, 2."; > > $e = $esl->sendRecv($cmd); > $esl->execute('speak',"(cepstral|callie|$text)",$uuid); > > > $e = $esl->sendRecv("events plain all"); > $e = $esl->filter("Unique-ID",$uuid); > > while ($esl->connected()) > { > $e = $esl->recvEvent(); > $result = $e->getType(); > if($result=='CHANNEL_EXECUTE_COMPLETE') > { > // what i do here...! > $esl->execute("&sleep",10000); > } > $digit = $e->getHeader("DTMF-Digit"); > $state_no = $e->getHeader("Channel-State-Number"); > $curr_uuid = $e->getHeader("Channel-Call-UUID"); > > /* > $ans_state = $e->getHeader("Answer-State"); > $state_no = $e->getHeader("Channel-State-Number"); > $core_id = $e->getHeader("Core-UUID"); > */ > if($curr_uuid==$uuid && $digit!=NULL) > { > print "input number is " . $digit . "\n"; > } > $state = $e->getHeader("Channel-Call-State"); > if ($state == 'HANGUP') { > print "\n\nHangup Cause Number: " . $state_no; > $esl->disconnect(); > } > } > > ?> > > > Regards > > Shahzad Bhatti > > > ---------- Forwarded message ---------- > From: Peter Olsson > To: FreeSWITCH Users Help > Cc: > Date: Fri, 2 Mar 2012 05:31:35 +0000 > Subject: Re: [Freeswitch-users] Use of ESL and PHP for Call > You should pass the originate command to &park() instead. > > Right now you originate the call and then execute the application speak. > After this (when speak is finished) FS has no more instructions so it will > hangup the call. > > If you execute &park() instead, the call will be "parked" and wait for > furter instructions. Then you will also need to execute the speak from ESL > instead - and afther that you continue to do whatever you want to do with > the call. > > /Peter > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/85af9003/attachment.html From msc at freeswitch.org Sat Mar 3 03:01:44 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 2 Mar 2012 16:01:44 -0800 Subject: [Freeswitch-users] a question about freeswitch conference caller-controls In-Reply-To: References: Message-ID: Try this in your dialplan before sending the call to the conference: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf -MC On Thu, Mar 1, 2012 at 5:30 PM, Erjian Li wrote: > When I press cellphone's key, although Freeswitch can't see this DTMF, but > the the other participant of the conference call can hear my DTMF tone. > Does this situation indicates that the DTMF tone has been forwarded by SIP > provider's server? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/14bc2462/attachment.html From krice at freeswitch.org Sat Mar 3 03:02:02 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 02 Mar 2012 18:02:02 -0600 Subject: [Freeswitch-users] *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. In-Reply-To: <384CFAFC-7D1A-48FA-8A4A-9F0A9CD04FB1@lux.io> Message-ID: This is because Apple Deprecated SSL on Lion... They don?t want you using it... Configure ?--without-openssl? and the problem goes away... There is an open jira on this issue I just cant recall the bug number off the top of my head On 3/2/12 4:57 PM, "St?phane Lux" wrote: > I have tried to compile it again: > > > tport_tls.c: In function 'tls_error': > tport_tls.c:665: warning: 'SSL_get_error' is deprecated (declared at > /usr/include/openssl/ssl.h:1501) > tport_tls.c:680: warning: 'SSL_get_shutdown' is deprecated (declared at > /usr/include/openssl/ssl.h:1568) > tport_tls.c: In function 'tls_read': > tport_tls.c:723: warning: 'SSL_read' is deprecated (declared at > /usr/include/openssl/ssl.h:1493) > tport_tls.c: In function 'tls_pending': > tport_tls.c:739: warning: 'SSL_pending' is deprecated (declared at > /usr/include/openssl/ssl.h:1368) > tport_tls.c: In function 'tls_write': > tport_tls.c:807: warning: 'SSL_write' is deprecated (declared at > /usr/include/openssl/ssl.h:1495) > tport_tls.c: In function 'tls_connect': > tport_tls.c:899: warning: 'SSL_accept' is deprecated (declared at > /usr/include/openssl/ssl.h:1491) > tport_tls.c:899: warning: 'SSL_connect' is deprecated (declared at > /usr/include/openssl/ssl.h:1492) > tport_tls.c:900: warning: 'SSL_get_error' is deprecated (declared at > /usr/include/openssl/ssl.h:1501) > tport_tls.c:954: warning: 'ERR_error_string_n' is deprecated (declared at > /usr/include/openssl/err.h:280) > make[9]: *** [tport_tls.lo] Error 1 > make[8]: *** [all] Error 2 > Making all in nta > Making all in nth > Making all in nea > Making all in iptsec > Making all in nua > make[8]: *** No rule to make target `tport/libtport.la', needed by > `libsofia-sip-ua.la'. Stop. > make[7]: *** [all-recursive] Error 1 > Making all in packages > make[6]: *** [all-recursive] Error 1 > make[5]: *** [all] Error 2 > make[4]: *** > [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] > Error 2 > make[3]: *** [mod_sofia-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > > On 02.03.2012, at 23:42, Michael Collins wrote: > >> We need to see the output that comes prior to this error. Grab the previous >> 20+ lines of output prior to this error and let us take a look. Hopefully the >> true cause will be apparent. >> >> -MC >> >> 2012/3/2 St?phane Lux >>> Hi, >>> >>> when compiling FS on Mac OS 10.7.3 I get this error: >>> >>> make[8]: *** No rule to make target `tport/libtport.la >>> ', needed by `libsofia-sip-ua.la '. Stop. >>> >>> How can I solve it? >>> >>> Kind Regards, >>> Stephane >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/a033aa07/attachment-0001.html From curriegrad2004 at gmail.com Sat Mar 3 03:03:10 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 2 Mar 2012 16:03:10 -0800 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: Message-ID: On Fri, Mar 2, 2012 at 3:02 PM, Michael Collins wrote: > > Hello all! > > I just saw an article in Engadget that said that some "hobbyists" were > trying to put FS on Raspberry Pi. I recall Richard Neese talking this > subject a few months ago but I hadn't heard anything recently. If you are > working to put FS on RP then please contact me off list. I would like to > keep track of these projects. Can you post a link to the said article? And from my early experimentation, yes it is highly possible to put FreeSWITCH on a raspberry pi board as evidenced with the QEMU ARM emulator for which the R Pi community was using. Heck, if FreeSWITCH can run properly on uClibc (albeit some linker issues with mod_spidermonkey), it should run just fine on an embedded platform like the R Pi. However, I wouldn't expect much out of it performance wise. > Thanks, > Michael > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Sat Mar 3 03:21:45 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 2 Mar 2012 16:21:45 -0800 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: Message-ID: Oops, it was gizmodo, not engadget... http://gizmodo.com/5889245/five-things-you-can-do-with-the-new-raspberry-pi -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/de919471/attachment.html From krice at freeswitch.org Sat Mar 3 03:24:13 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 02 Mar 2012 18:24:13 -0600 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: Message-ID: Who cares about spidermonkey... I would just turn that off and pretty much anything else with too many problems... All you really are going to do on a RaspPi is a small config anyway K On 3/2/12 6:03 PM, "curriegrad2004" wrote: > On Fri, Mar 2, 2012 at 3:02 PM, Michael Collins wrote: >> >> Hello all! >> >> I just saw an article in Engadget that said that some "hobbyists" were >> trying to put FS on Raspberry Pi. I recall Richard Neese talking this >> subject a few months ago but I hadn't heard anything recently. If you are >> working to put FS on RP then please contact me off list. I would like to >> keep track of these projects. > > Can you post a link to the said article? > > And from my early experimentation, yes it is highly possible to put > FreeSWITCH on a raspberry pi board as evidenced with the QEMU ARM > emulator for which the R Pi community was using. Heck, if FreeSWITCH > can run properly on uClibc (albeit some linker issues with > mod_spidermonkey), it should run just fine on an embedded platform > like the R Pi. > > However, I wouldn't expect much out of it performance wise. > >> Thanks, >> Michael >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From curriegrad2004 at gmail.com Sat Mar 3 05:16:16 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 2 Mar 2012 18:16:16 -0800 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: Message-ID: Agreed. Transcoding media formats would be out of the question. somebody running the current voicemail sound files against mod_native_file would be very helpful for users who use that device. On Fri, Mar 2, 2012 at 4:24 PM, Ken Rice wrote: > Who cares about spidermonkey... I would just turn that off and pretty much > anything else with too many problems... > > All you really are going to do on a RaspPi is a small config anyway > > K > > > On 3/2/12 6:03 PM, "curriegrad2004" wrote: > > > On Fri, Mar 2, 2012 at 3:02 PM, Michael Collins > wrote: > >> > >> Hello all! > >> > >> I just saw an article in Engadget that said that some "hobbyists" were > >> trying to put FS on Raspberry Pi. I recall Richard Neese talking this > >> subject a few months ago but I hadn't heard anything recently. If you > are > >> working to put FS on RP then please contact me off list. I would like to > >> keep track of these projects. > > > > Can you post a link to the said article? > > > > And from my early experimentation, yes it is highly possible to put > > FreeSWITCH on a raspberry pi board as evidenced with the QEMU ARM > > emulator for which the R Pi community was using. Heck, if FreeSWITCH > > can run properly on uClibc (albeit some linker issues with > > mod_spidermonkey), it should run just fine on an embedded platform > > like the R Pi. > > > > However, I wouldn't expect much out of it performance wise. > > > >> Thanks, > >> Michael > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/24a8e436/attachment.html From bob.mccarthy at experient.com Sat Mar 3 09:54:06 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Fri, 2 Mar 2012 23:54:06 -0700 Subject: [Freeswitch-users] Hang Up Cause is Blank in Dial Plan In-Reply-To: <00d001ccf8a0$e64aa850$b2dff8f0$@mccarthy@experient.com> References: <1330689023.690.7.camel@CO-999-8> <00d001ccf8a0$e64aa850$b2dff8f0$@mccarthy@experient.com> Message-ID: <014401ccf90a$6f133260$4d399720$@mccarthy@experient.com> Answering my last observation again, I needed to add Bob From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob McCarthy Sent: Friday, March 02, 2012 11:19 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] Hang Up Cause is Blank in Dial Plan I did notice that during a CALL_REJECTED that the execute_extension statement did not get executed .. But it does work for NO_ANSWER. Would using a Transfer change that behavior ??? 2012-03-02 13:04:32.569026 [DEBUG] mod_dptools.c:1281 sofia/internal/CO999x1001.1 at 192.168.57.211 SET [hangup_after_bridge]=[false] EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 bridge(sofia/external/2010 at 192.168.1.195) 2012-03-02 13:04:32.589032 [DEBUG] switch_channel.c:1047 sofia/internal/CO999x1001.1 at 192.168.57.211 EXPORTING[export_vars] [RFC2822_DATE]=[Fri, 02 Mar 2012 13:04:32 -0500] to event 2012-03-02 13:04:32.589032 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-02 13:04:32.589032 [NOTICE] switch_channel.c:926 New Channel sofia/external/2010 at 192.168.1.195 [29bcf518-6492-11e1-b71d-ddc06b002bf3] 2012-03-02 13:04:32.589032 [DEBUG] mod_sofia.c:4679 (sofia/external/2010 at 192.168.1.195) State Change CS_NEW -> CS_INIT 2012-03-02 13:04:32.589032 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.589032 [DEBUG] switch_core_state_machine.c:362 (sofia/external/2010 at 192.168.1.195) Running State Change CS_INIT 2012-03-02 13:04:32.589032 [DEBUG] switch_core_state_machine.c:401 (sofia/external/2010 at 192.168.1.195) State INIT 2012-03-02 13:04:32.589032 [DEBUG] mod_sofia.c:85 sofia/external/2010 at 192.168.1.195 SOFIA INIT 2012-03-02 13:04:32.609021 [DEBUG] mod_sofia.c:125 (sofia/external/2010 at 192.168.1.195) State Change CS_INIT -> CS_ROUTING 2012-03-02 13:04:32.609021 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.609021 [DEBUG] switch_core_state_machine.c:401 (sofia/external/2010 at 192.168.1.195) State INIT going to sleep 2012-03-02 13:04:32.609021 [DEBUG] switch_core_state_machine.c:362 (sofia/external/2010 at 192.168.1.195) Running State Change CS_ROUTING 2012-03-02 13:04:32.609021 [DEBUG] switch_channel.c:1886 (sofia/external/2010 at 192.168.1.195) Callstate Change DOWN -> RINGING 2012-03-02 13:04:32.609021 [DEBUG] switch_core_state_machine.c:410 (sofia/external/2010 at 192.168.1.195) State ROUTING 2012-03-02 13:04:32.609021 [DEBUG] mod_sofia.c:148 sofia/external/2010 at 192.168.1.195 SOFIA ROUTING 2012-03-02 13:04:32.609021 [DEBUG] switch_ivr_originate.c:66 (sofia/external/2010 at 192.168.1.195) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-03-02 13:04:32.609021 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.609021 [DEBUG] switch_core_state_machine.c:410 (sofia/external/2010 at 192.168.1.195) State ROUTING going to sleep 2012-03-02 13:04:32.609021 [DEBUG] switch_core_state_machine.c:362 (sofia/external/2010 at 192.168.1.195) Running State Change CS_CONSUME_MEDIA 2012-03-02 13:04:32.609021 [DEBUG] switch_core_session.c:875 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.609021 [DEBUG] switch_core_state_machine.c:429 (sofia/external/2010 at 192.168.1.195) State CONSUME_MEDIA 2012-03-02 13:04:32.609021 [DEBUG] switch_core_state_machine.c:429 (sofia/external/2010 at 192.168.1.195) State CONSUME_MEDIA going to sleep 2012-03-02 13:04:32.609021 [DEBUG] sofia.c:5526 Channel sofia/external/2010 at 192.168.1.195 entering state [calling][0] 2012-03-02 13:04:32.692026 [DEBUG] switch_core_session.c:875 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.692026 [DEBUG] switch_core_session.c:875 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.692026 [DEBUG] switch_core_session.c:875 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.692026 [DEBUG] sofia.c:5526 Channel sofia/external/2010 at 192.168.1.195 entering state [terminated][603] 2012-03-02 13:04:32.692026 [DEBUG] switch_channel.c:2850 (sofia/external/2010 at 192.168.1.195) Callstate Change RINGING -> HANGUP 2012-03-02 13:04:32.692026 [NOTICE] sofia.c:6293 Hangup sofia/external/2010 at 192.168.1.195 [CS_CONSUME_MEDIA] [CALL_REJECTED] 2012-03-02 13:04:32.692026 [DEBUG] switch_channel.c:2873 Send signal sofia/external/2010 at 192.168.1.195 [KILL] 2012-03-02 13:04:32.692026 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:362 (sofia/external/2010 at 192.168.1.195) Running State Change CS_HANGUP 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:602 (sofia/external/2010 at 192.168.1.195) State HANGUP 2012-03-02 13:04:32.692026 [DEBUG] mod_sofia.c:469 Channel sofia/external/2010 at 192.168.1.195 hanging up, cause: CALL_REJECTED 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:47 sofia/external/2010 at 192.168.1.195 Standard HANGUP, cause: CALL_REJECTED 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:602 (sofia/external/2010 at 192.168.1.195) State HANGUP going to sleep 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:393 (sofia/external/2010 at 192.168.1.195) State Change CS_HANGUP -> CS_REPORTING 2012-03-02 13:04:32.692026 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:362 (sofia/external/2010 at 192.168.1.195) Running State Change CS_REPORTING 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:662 (sofia/external/2010 at 192.168.1.195) State REPORTING 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:79 sofia/external/2010 at 192.168.1.195 Standard REPORTING, cause: CALL_REJECTED 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:662 (sofia/external/2010 at 192.168.1.195) State REPORTING going to sleep 2012-03-02 13:04:32.692026 [DEBUG] switch_core_state_machine.c:387 (sofia/external/2010 at 192.168.1.195) State Change CS_REPORTING -> CS_DESTROY 2012-03-02 13:04:32.692026 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2010 at 192.168.1.195 [BREAK] 2012-03-02 13:04:32.692026 [DEBUG] switch_core_session.c:1380 Session 2 (sofia/external/2010 at 192.168.1.195) Locked, Waiting on external entities 2012-03-02 13:04:32.712027 [DEBUG] switch_ivr_originate.c:3365 Originate Resulted in Error Cause: 21 [CALL_REJECTED] 2012-03-02 13:04:32.712027 [INFO] mod_dptools.c:2922 Originate Failed. Cause: CALL_REJECTED 2012-03-02 13:04:32.712027 [DEBUG] switch_channel.c:2850 (sofia/internal/CO999x1001.1 at 192.168.57.211) Callstate Change RINGING -> HANGUP 2012-03-02 13:04:32.712027 [NOTICE] mod_dptools.c:3041 Hangup sofia/internal/CO999x1001.1 at 192.168.57.211 [CS_EXECUTE] [CALL_REJECTED] 2012-03-02 13:04:32.712027 [DEBUG] switch_channel.c:2873 Send signal sofia/internal/CO999x1001.1 at 192.168.57.211 [KILL] 2012-03-02 13:04:32.712027 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/CO999x1001.1 at 192.168.57.211 [BREAK] 2012-03-02 13:04:32.712027 [DEBUG] switch_core_session.c:2285 sofia/internal/CO999x1001.1 at 192.168.57.211 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Friday, March 02, 2012 8:28 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Hang Up Cause is Blank in Dial Plan Nah, it's executed after the bridge so that's fine Ken. The issue, as he found, is you can't use the hangup_cause within the A leg.. because it's not hung up yet. He found other variable that ARE set in the A leg, though. -Avi On Fri, Mar 2, 2012 at 4:40 PM, Ken Rice wrote: Variables are not expanded during the actual execution phase of dialplan processing ... What you are trying to do here will most likely require a transfer... You should read up on the wiki how the dialplan is processed and how variables are expanded... K On 3/2/12 5:50 AM, "Bob McCarthy" wrote: I am trying to play messages for failures on outbound Calls. When I try to use ${hangup_cause} after the failed bridge command it comes back as blank. what am I doing wrong ??? "/> 2012-03-02 06:42:38.005915 [DEBUG] switch_channel.c:2850 (sofia/external/2001 at 192.168.1.195) Callstate Change RINGING -> HANGUP 2012-03-02 06:42:38.005915 [NOTICE] switch_ivr_originate.c:3183 Hangup sofia/external/2001 at 192.168.1.195 [CS_CONSUME_MEDIA] [NO_ANSWER] 2012-03-02 06:42:38.005915 [DEBUG] switch_channel.c:2873 Send signal sofia/external/2001 at 192.168.1.195 [KILL] 2012-03-02 06:42:38.005915 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2001 at 192.168.1.195 [BREAK] 2012-03-02 06:42:38.005915 [INFO] mod_dptools.c:2922 Originate Failed. Cause: NO_ANSWER EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 set(test=) 2012-03-02 06:42:38.005915 [DEBUG] mod_dptools.c:1281 sofia/internal/CO999x1001.1 at 192.168.57.211 SET [test]=[UNDEF] EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 log(1 A-leg hangup cause: ) 2012-03-02 06:42:38.005915 [ALERT] mod_dptools.c:1420 A-leg hangup cause: EXECUTE sofia/internal/CO999x1001.1 at 192.168.57.211 execute_extension(hangup_reason- XML features) 2012-03-02 06:42:38.005915 [INFO] mod_dialplan_xml.c:485 Processing Dispatch 2 ->hangup_reason- in context features Dialplan: sofia/internal/CO999x1001.1 at 192.168.57.211 parsing [features->dx] continue=false _____ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120302/6bd15e1e/attachment-0001.html From stephane at lux.io Sat Mar 3 15:21:34 2012 From: stephane at lux.io (=?iso-8859-1?Q?St=E9phane_Lux?=) Date: Sat, 3 Mar 2012 13:21:34 +0100 Subject: [Freeswitch-users] *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. In-Reply-To: References: Message-ID: Hi Ken, thanks a lot. It worked! On 03.03.2012, at 01:02, Ken Rice wrote: > This is because Apple Deprecated SSL on Lion... They don?t want you using it... > > Configure ?--without-openssl? and the problem goes away... > > There is an open jira on this issue I just cant recall the bug number off the top of my head > > > On 3/2/12 4:57 PM, "St?phane Lux" wrote: > >> I have tried to compile it again: >> >> >> tport_tls.c: In function 'tls_error': >> tport_tls.c:665: warning: 'SSL_get_error' is deprecated (declared at /usr/include/openssl/ssl.h:1501) >> tport_tls.c:680: warning: 'SSL_get_shutdown' is deprecated (declared at /usr/include/openssl/ssl.h:1568) >> tport_tls.c: In function 'tls_read': >> tport_tls.c:723: warning: 'SSL_read' is deprecated (declared at /usr/include/openssl/ssl.h:1493) >> tport_tls.c: In function 'tls_pending': >> tport_tls.c:739: warning: 'SSL_pending' is deprecated (declared at /usr/include/openssl/ssl.h:1368) >> tport_tls.c: In function 'tls_write': >> tport_tls.c:807: warning: 'SSL_write' is deprecated (declared at /usr/include/openssl/ssl.h:1495) >> tport_tls.c: In function 'tls_connect': >> tport_tls.c:899: warning: 'SSL_accept' is deprecated (declared at /usr/include/openssl/ssl.h:1491) >> tport_tls.c:899: warning: 'SSL_connect' is deprecated (declared at /usr/include/openssl/ssl.h:1492) >> tport_tls.c:900: warning: 'SSL_get_error' is deprecated (declared at /usr/include/openssl/ssl.h:1501) >> tport_tls.c:954: warning: 'ERR_error_string_n' is deprecated (declared at /usr/include/openssl/err.h:280) >> make[9]: *** [tport_tls.lo] Error 1 >> make[8]: *** [all] Error 2 >> Making all in nta >> Making all in nth >> Making all in nea >> Making all in iptsec >> Making all in nua >> make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. >> make[7]: *** [all-recursive] Error 1 >> Making all in packages >> make[6]: *** [all-recursive] Error 1 >> make[5]: *** [all] Error 2 >> make[4]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 >> make[3]: *** [mod_sofia-all] Error 1 >> make[2]: *** [all-recursive] Error 1 >> make[1]: *** [all-recursive] Error 1 >> make: *** [all] Error 2 >> >> >> On 02.03.2012, at 23:42, Michael Collins wrote: >> >>> We need to see the output that comes prior to this error. Grab the previous 20+ lines of output prior to this error and let us take a look. Hopefully the true cause will be apparent. >>> >>> -MC >>> >>> 2012/3/2 St?phane Lux >>>> Hi, >>>> >>>> when compiling FS on Mac OS 10.7.3 I get this error: >>>> >>>> make[8]: *** No rule to make target `tport/libtport.la ', needed by `libsofia-sip-ua.la '. Stop. >>>> >>>> How can I solve it? >>>> >>>> Kind Regards, >>>> Stephane >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120303/1cb7050c/attachment.html From andrew at cassidywebservices.co.uk Sat Mar 3 16:13:34 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Sat, 3 Mar 2012 13:13:34 +0000 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: Message-ID: Even on the Model B you've still got to cram it into 256MB RAM, which is not going to be fun. However I am also interested in the idea of using the Raspberry Pi as the base for a low-cost PBX that I could add to my product offerings. It may even be the case that I just end up supplying them as a proxy using milkfish to a hosted service. On 3 March 2012 02:16, curriegrad2004 wrote: > Agreed. Transcoding media formats would be out of the question. somebody > running the current voicemail sound files against mod_native_file would be > very helpful for users who use that device. > > > On Fri, Mar 2, 2012 at 4:24 PM, Ken Rice wrote: > >> Who cares about spidermonkey... I would just turn that off and pretty much >> anything else with too many problems... >> >> All you really are going to do on a RaspPi is a small config anyway >> >> K >> >> >> On 3/2/12 6:03 PM, "curriegrad2004" wrote: >> >> > On Fri, Mar 2, 2012 at 3:02 PM, Michael Collins >> wrote: >> >> >> >> Hello all! >> >> >> >> I just saw an article in Engadget that said that some "hobbyists" were >> >> trying to put FS on Raspberry Pi. I recall Richard Neese talking this >> >> subject a few months ago but I hadn't heard anything recently. If you >> are >> >> working to put FS on RP then please contact me off list. I would like >> to >> >> keep track of these projects. >> > >> > Can you post a link to the said article? >> > >> > And from my early experimentation, yes it is highly possible to put >> > FreeSWITCH on a raspberry pi board as evidenced with the QEMU ARM >> > emulator for which the R Pi community was using. Heck, if FreeSWITCH >> > can run properly on uClibc (albeit some linker issues with >> > mod_spidermonkey), it should run just fine on an embedded platform >> > like the R Pi. >> > >> > However, I wouldn't expect much out of it performance wise. >> > >> >> Thanks, >> >> Michael >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120303/348f8da9/attachment-0001.html From curriegrad2004 at gmail.com Sat Mar 3 20:43:43 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sat, 3 Mar 2012 09:43:43 -0800 Subject: [Freeswitch-users] FreeSWITCH on Raspberry Pi In-Reply-To: References: Message-ID: Model B is the only viable choice unless you want to stick a USB to Ethernet bridge over there. There were last minute changes that Model A now also features 256MB of RAM On Sat, Mar 3, 2012 at 5:13 AM, Andrew Cassidy wrote: > Even on the Model B you've still got to cram it into 256MB RAM, which is not > going to be fun. > > However I am also interested in the idea of using the Raspberry Pi as the > base for a low-cost PBX that I could add to my product offerings. It may > even be the case that I just end up supplying them as a proxy using milkfish > to a hosted service. > > > On 3 March 2012 02:16, curriegrad2004 wrote: >> >> Agreed. Transcoding media formats would be out of the question. somebody >> running the current voicemail sound files against mod_native_file would be >> very helpful for users who use that device. >> >> >> On Fri, Mar 2, 2012 at 4:24 PM, Ken Rice wrote: >>> >>> Who cares about spidermonkey... I would just turn that off and pretty >>> much >>> anything else with too many problems... >>> >>> All you really are going to do on a RaspPi is a small config anyway >>> >>> K >>> >>> >>> On 3/2/12 6:03 PM, "curriegrad2004" wrote: >>> >>> > On Fri, Mar 2, 2012 at 3:02 PM, Michael Collins >>> > wrote: >>> >> >>> >> Hello all! >>> >> >>> >> I just saw an article in Engadget that said that some "hobbyists" were >>> >> trying to put FS on Raspberry Pi. I recall Richard Neese talking this >>> >> subject a few months ago but I hadn't heard anything recently. If you >>> >> are >>> >> working to put FS on RP then please contact me off list. I would like >>> >> to >>> >> keep track of these projects. >>> > >>> > Can you post a link to the said article? >>> > >>> > And from my early experimentation, yes it is highly possible to put >>> > FreeSWITCH on a raspberry pi board as evidenced with the QEMU ARM >>> > emulator for which the R Pi community was using. Heck, if FreeSWITCH >>> > can run properly on uClibc (albeit some linker issues with >>> > mod_spidermonkey), it should run just fine on an embedded platform >>> > like the R Pi. >>> > >>> > However, I wouldn't expect much out of it performance wise. >>> > >>> >> Thanks, >>> >> Michael >>> >> >>> >> >>> >> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://wiki.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Andrew Cassidy BSc (Hons) MBCS > Managing Director; Cassidy Web Services Ltd > T: 03300 100 960 F: 03300 100 961 > E: andrew at cassidywebservices.co.uk > W:?www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mstockton at harqen.com Sat Mar 3 20:53:39 2012 From: mstockton at harqen.com (Matt Stockton) Date: Sat, 3 Mar 2012 11:53:39 -0600 Subject: [Freeswitch-users] Using mod_shout over ssl, curl issue with curl-ca-bundle.crt file location Message-ID: Hi all, I just upgraded to the latest git, and I'm trying to dive into an issue I'm having. I am using mod_shout and in some instances am playing files that are hosted on web servers protected by https. This seemed to be working fine before I upgraded, but now I am getting the following issues, which is preventing the streaming of the files: 12-03-02 19:06:57.926919 [WARNING] mod_shout.c:468 CURL returned error:[77] problem with the SSL CA cert (path? access rights?) : error setting certificate verify locations: CAfile: /usr/local/freeswitch/share/curl/curl-ca-bundle.crt CApath: none I looked at the code and the git history in mod_shout.c where it is setting all the curl options, nothing seems to have changed there since I last updated FS (12/07), however, the curl-ca-bundle file is certainly not located at /usr/local/freeswitch/share/curl/curl-ca-bundle.crt and never has been as far as I know. I also looked at other mods that are using curl and where they are calling switch_curl_easy_setopt (mod_xml_curl, mod_httapi), and noticed that those mods are setting options that might be related to what I need? CURLOPT_SSLCERT I am confused as to what is causing the breakage, since mod_shout hasn't changed since I last updated, yet none of the ssl curl options are set in mod_shout..and I never had any problems with the mod_shout curl usage finding the certificate verify locations by default. Is there some other default that used to be set in the freeswitch configuration that I need to set manually? Any help is appreciated!!! Thanks! Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120303/59e743c0/attachment.html From avi at avimarcus.net Sat Mar 3 20:54:58 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 3 Mar 2012 19:54:58 +0200 Subject: [Freeswitch-users] Retrieve active call from voicemail In-Reply-To: <4F5050B9.6060400@dyversesolutions.com.au> References: <99F37B26-F412-49A6-9F6D-FCF8E788DF70@dyversesolutions.com.au> <4F5050B9.6060400@dyversesolutions.com.au> Message-ID: So you are sharing your working extension for retrieving a call from VM? Thanks for sharing! -Avi On Fri, Mar 2, 2012 at 6:46 AM, Dean Coulter wrote: > Thanks Avi, > > I have it working using the UUID. Just before the call diverts to VM it > inserts a hash. The user can then dial # to retrieve(intercept) the call > from the VM system. The VM loopback hangsup at the time the call is > intercepted. Note that in this instance, any extension can retrieve the > call by dialling # (this is the main number which rings on all phones). > > > > > > > > > > > data="{ignore_early_media=true}sofia/internal/302%${domain_name},sofia/internal/301%${domain_name}"/> > > > data="insert/${domain_name}-retrieve_vm/global/${uuid}"/> > data="loopback/app=voicemail:default ${domain_name} 301"/> > > > > > > > > > > data="${hash(select/${domain_name}-retrieve_vm/global/${uuid}"/> > > > > > > > Regards, > > Dean Coulter > > Dyverse Solutions Pty Ltd > PO Box 10888 > Adelaide Street > BRISBANE QLD 4000 > > Mob +61 (0) 448 859 977 > > ACN 112 999 452 > http://www.dyversesolutions.com.au > > Caution > > This message may contain privileged and confidential information intended only for the use of the addressee(s) named above. If you are not the intended recipient of this message you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify the sender immediately. > > > On 02/03/12 10:40, Avi Marcus wrote: > > Via ESL or api you could do uuid_transfer but > I'm not sure a) how you would know they were still in the VM system and b) > what button you'd decide they should push. > But you'd just use hash to store the UUID right before they went to the VM > system. Like the redial or intercept code in the default config. > > -Avi > > > On Thu, Mar 1, 2012 at 8:59 AM, Dean Coulter > wrote: > >> Hi, >> >> >> Does anyone know how to retrieve an active call from the voicemail >> system? I have googled and checked the lists but couldn't find anything.. >> >> At a previous employer we had this feature so if you were heading back to >> your desk to answer the phone and it had gone to VM, you could retrieve the >> call if the calling party was in the process of leaving a message. This >> was a very useful feature as it saved time in listening to messages and >> returning calls. >> >> Dean >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120303/b18c9810/attachment-0001.html From miha at softnet.si Sat Mar 3 22:24:13 2012 From: miha at softnet.si (Miha) Date: Sat, 03 Mar 2012 20:24:13 +0100 Subject: [Freeswitch-users] 302 Message-ID: Hi, scenario. A calls B. On B is set 302 redirect to C. How can I authenticate call with mod_radius, as FS sets all variables for call A(CID), so I can not authenticate outgoing call from B to C. I can not use params variable password, which is set in user/dir for B. I can only use varibles set for A. Thanks! Miha From fluixab at bellsouth.net Sat Mar 3 17:17:08 2012 From: fluixab at bellsouth.net (Bernard Fluixa) Date: Sat, 3 Mar 2012 09:17:08 -0500 Subject: [Freeswitch-users] mod shell stream Message-ID: Hi there, I'm trying to make a playback of streamed sound data coming from a database. I could make the example included in documentation work (cat file.wav | sox -t wav - $@ -t raw -). However, I can't make it work with same sox options after having uploaded that same file into my PGSQL database and select it instead of a cat command. My shell script is #!/bin/bash psql -h -U postgres -E -t -q -d -c "select from " | sox -t wav - $@ -t raw - Freeswitch gives me a "sox FAIL formats: can't open input `-': WAVE: RIFF header not found" error message. Has anybody streamed data from database into Freeswitch? Thank you. Bernard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120303/ea9d9d20/attachment.html From gregor at infomedia.si Sun Mar 4 00:00:41 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Sat, 3 Mar 2012 22:00:41 +0100 Subject: [Freeswitch-users] Freeswitch + MSSQL Message-ID: Hi! I am using Freeswitch with Windows. I hope I am not the only one. I also want to make mod_sofia to talk via ODBC with MSSQL. I can connect, but on call I get error, when sofia wants to write to mssql: [STATE: 42000 CODE 102 ERROR: [Microsoft][ODBC SQL Server Driver][SQL Server]Incorrect syntax near '|'. ] Is this know issue? Any suggestions? On MySQL it is working ok.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120303/18b6435d/attachment.html From bdfoster at endigotech.com Sun Mar 4 02:23:17 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 3 Mar 2012 18:23:17 -0500 Subject: [Freeswitch-users] New batch of sound files Message-ID: Hola, I briefly heard in channel and on conference about a new batch of sound files for freeswitch. Is there a download link out there for this? Also, is there any notable prompts for conferences? I'm specifically looking for some prompts that would take care of telling the incoming participant how many other participants are in the conference. Thanks! -BDF -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120303/a5deee90/attachment.html From bdfoster at endigotech.com Sun Mar 4 02:26:32 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 3 Mar 2012 18:26:32 -0500 Subject: [Freeswitch-users] Freeswitch + MSSQL In-Reply-To: References: Message-ID: I think that has to do with ymthe specific ODBC driver you are using. I'd check your configs and see of there's anything amiss. I could be wrong though, as I haven't dealt with MSSQL. I have experience with MYSQL and postgres with ODBC though. -BDF On Mar 3, 2012 6:19 PM, "Gregor Nanger" wrote: > Hi! > > I am using Freeswitch with Windows. I hope I am not the only one. > > I also want to make mod_sofia to talk via ODBC with MSSQL. I can connect, > but on call I get error, when sofia wants to write to mssql: > > [STATE: 42000 CODE 102 ERROR: [Microsoft][ODBC SQL Server Driver][SQL > Server]Incorrect syntax near '|'. > ] > > Is this know issue? Any suggestions? > > On MySQL it is working ok.. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120303/ea5fd8ed/attachment.html From dujinfang at gmail.com Sun Mar 4 06:20:13 2012 From: dujinfang at gmail.com (Seven Du) Date: Sun, 4 Mar 2012 11:20:13 +0800 Subject: [Freeswitch-users] send message via gateway from mod_sms Message-ID: <05B1435B9174403E87DF5844F4DF27DA@gmail.com> Hi, I'd like to know is there a way to send message to a gateway like bridge for sound? Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/4f67000a/attachment.html From jmesquita at freeswitch.org Sun Mar 4 07:14:13 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Sun, 4 Mar 2012 01:14:13 -0300 Subject: [Freeswitch-users] 302 redirect variable In-Reply-To: <4F509C57.20205@softnet.si> References: <4F509C57.20205@softnet.si> Message-ID: <9BCA494E297D4430B191119091E31C1A@freeswitch.org> You won't get the user's password in plain text like that EVER. If we did that, we would be considered to be insanely insecure. I am guessing you are using SIP only so you can take a look at the dialplan/public.xml file of the default configs. On the end of that file you will see that there is a verification to do dial plan based authentication. Look for this extension in particular: Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday, March 2, 2012 at 7:09 AM, Miha Zoubek wrote: > Hi, > > in my directory I set variable password () for every user. > > After I am doing 302 redirect in my public dialplan and transfer call to extension, I can not use varible pasword. > How can I get varible password, so that I can authenticate call. > > public dialplan: > > > > > > > > > default dialplan: > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/6603fd7d/attachment-0001.html From jmesquita at freeswitch.org Sun Mar 4 07:23:12 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Sun, 4 Mar 2012 01:23:12 -0300 Subject: [Freeswitch-users] TCAPI_User in Contact In-Reply-To: References: Message-ID: <43070B72CB024DDC91BF10A94F0A6400@freeswitch.org> The answer to this one was not all that easy because it is hidden on the source code and undocumented as far as I can tell. sofia_glue.c has a function called sofia_glue_do_invite. Right there is where all this is generated and you can pretty much change any header you want using channel variables. The contact relevant ones you are looking for are: sip_invite_contact_params sip_contact_user contact_params is what goes after the URI (after the ;) and the contact_user is 5122046107 (mailto:5122046107 at 74.37.201.198), which in your case you want to change to TCAPI_User. What I would really ask you to do is to go ahead and document those on the wiki, please. A lot of these variables are not documented anywhere and few ppl know of their existence. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday, March 2, 2012 at 4:01 PM, Andrew Melton wrote: > I know this has been asked before, but I can't seem to put my finger on the answer, any suggestions of where to look are appreciated. > > Freeswitch (Sofia) is rewriting the Conact header on calls originating on a trunk ('inbound' context), routing to another trunk ('outbound' context). None of the calls are from registered users or extensions, this is more of an SBC configuration (LCR). > > For example, > > incoming to Freeswitch: > Contact: > > > Outbound from Freeswitch: > Contact: > > I really don't want to modify the from userpary, nor do I want to include gw=trunk_2. I just can't figure out the right syntax or parameter to change it. Is it a parameter in the dialplan, or should I be using a regex on the outbound interface? > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/8ceffc5d/attachment.html From jean.marc.hyppolite at gmail.com Sun Mar 4 08:09:39 2012 From: jean.marc.hyppolite at gmail.com (Jean-Marc Hyppolite) Date: Sun, 4 Mar 2012 00:09:39 -0500 Subject: [Freeswitch-users] Freeswitch-users] spandsp error reading frame Message-ID: <4f52f917.f21c340a.7f41.ffff9060@mx.google.com> Hello, I am trying to use mod_spandsp to detect call progress. I haven t been successful so far. I am getting the following error messages. ================================================ 2012-03-03 23:55:08.398205 [DEBUG] mod_spandsp_dsp.c:379 (sofia/outbound/. at ...) Starting tone detection for '1' 2012-03-03 23:55:08.398205 [INFO] mod_spandsp_dsp.c:411 (sofia/outbound/. at ...) initializing tone detector 2012-03-03 23:55:08.398205 [DEBUG] switch_core_media_bug.c:462 Attaching BUG to sofia/outbound/. at ... 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:422 (sofia/outbound/. at ...) error reading frame 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:447 (sofia/outbound/. at ...) destroying tone detector ================================================ Any help would be appreciated. Thanks Jean-Marc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/14952ded/attachment.html From dujinfang at gmail.com Sun Mar 4 09:03:36 2012 From: dujinfang at gmail.com (Seven Du) Date: Sun, 4 Mar 2012 14:03:36 +0800 Subject: [Freeswitch-users] For all In-Reply-To: References: Message-ID: <35A7A8F4E59E4306B4AA6E9E9BF52D4E@gmail.com> I think you'd better ask the freeswitch-dev list. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) On Wednesday, February 29, 2012 at 9:18 PM, ???? wrote: > > Hi, > > > I would like to use my own equipment for the endpoints transcoding not by freeswitch. > > > So I want to modify the sdp. The media stream to point to the transcoding device. > > > Is there anyway of modifying o= and c= in sdp protocol? > > > (modify ip and port of rtp server in sdp) > > > How to modify the sdp with in C code. > > > > > > Other methods to achieve it? Any idea? > > > > > > > > > > Thanks ! > > > > > > > > > > best regards! > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/199c781c/attachment-0001.html From dujinfang at gmail.com Sun Mar 4 09:12:47 2012 From: dujinfang at gmail.com (Seven Du) Date: Sun, 4 Mar 2012 14:12:47 +0800 Subject: [Freeswitch-users] mod shell stream In-Reply-To: References: Message-ID: <5BA3C64DA53B4EAC9B1DA367F8FFD398@gmail.com> I think you might need a script to strip the column header ? psql -c "select 1" |cat ?column? ---------- 1 (1 row) -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) On Saturday, March 3, 2012 at 10:17 PM, Bernard Fluixa wrote: > Hi there, > > I'm trying to make a playback of streamed sound data coming from a database. I could make the example included in documentation work (cat file.wav | sox -t wav - $@ -t raw -). However, I can't make it work with same sox options after having uploaded that same file into my PGSQL database and select it instead of a cat command. > > My shell script is > > #!/bin/bash > > psql -h -U postgres -E -t -q -d -c "select from
" | sox -t wav - $@ -t raw - > > Freeswitch gives me a "sox FAIL formats: can't open input `-': WAVE: RIFF header not found" error message. > > > Has anybody streamed data from database into Freeswitch? > > Thank you. > > Bernard > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/3a950430/attachment.html From curriegrad2004 at gmail.com Sun Mar 4 10:16:04 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sat, 3 Mar 2012 23:16:04 -0800 Subject: [Freeswitch-users] For all In-Reply-To: <35A7A8F4E59E4306B4AA6E9E9BF52D4E@gmail.com> References: <35A7A8F4E59E4306B4AA6E9E9BF52D4E@gmail.com> Message-ID: Try bypass_media. If I were you I'd get FS to pass that media stream to another machine that does transcoding instead of messing with the SDP headers On Sat, Mar 3, 2012 at 10:03 PM, Seven Du wrote: > I think you'd better ask the freeswitch-dev list. > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: ?http://www.freeswitch.org.cn > > Sent with Sparrow > > On Wednesday, February 29, 2012 at 9:18 PM, ???? wrote: > > Hi, > > I?would?like?to?use?my?own?equipment?for?the?endpoints?transcoding?not?by??freeswitch. > > So?I?want?to?modify?the?sdp.?The?media?stream?to?point?to?the?transcoding?device. > > Is?there?anyway?of?modifying?o=?and?c=?in?sdp?protocol? > > (modify?ip?and?port?of?rtp?server?in?sdp) > > How?to?modify?the?sdp?with?in?C?code. > > Other?methods?to?achieve?it??Any?idea? > > > Thanks?! > > > best?regards! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From hjz51 at qq.com Sun Mar 4 11:18:21 2012 From: hjz51 at qq.com (=?gbk?B?xObLrtDQ1ts=?=) Date: Sun, 4 Mar 2012 16:18:21 +0800 Subject: [Freeswitch-users] how to alter the SDP information in freeswitch Message-ID: Hi, I'd like to know how to alter the SDP information in freeswitch ? Any help is appreciated! Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/dc08d139/attachment.html From gregor at infomedia.si Sun Mar 4 03:22:01 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 4 Mar 2012 01:22:01 +0100 Subject: [Freeswitch-users] Freeswitch + MSSQL In-Reply-To: References: Message-ID: This is sql that is generated: select sip_registrations.sip_user, sip_registrations.sub_host, sip_registrations.status, sip_registrations.rpid, '', sip_dialogs.uuid, sip_dialogs.state, sip_dialogs.direction, sip_dialogs.sip_to_user, sip_dialogs.sip_to_host,sip_presence.status,sip_presence.rpid,sip_dialogs.presence_id, sip_pre sence.open_closed,'','','' from sip_registrations left join sip_dialogs on sip_dialogs.hostname = sip_registrations.host name and sip_dialogs.profile_name = sip_registrations.profile_name and (sip_dialogs.presence_id = sip_registrations.sip_ user || '@' || sip_registrations.sub_host or (sip_dialogs.sip_from_user = sip_registrations.sip_user and sip_dialogs.sip _from_host = sip_registrations.sip_host)) left join sip_presence on sip_presence.hostname=sip_registrations.hostname and (sip_registrations.sip_user=sip_presence.sip_user and sip_registrations.orig_server_host=sip_presence.sip_host and sip_ registrations.profile_name=sip_presence.profile_name) where sip_registrations.hostname='WIN-SERVER2008' and sip_registra tions.profile_name='internal' and sip_dialogs.call_info_state != 'seized' and sip_dialogs.presence_id='1002 at 192.168.1. 150' or (sip_registrations.sip_user='1002' and (sip_registrations.orig_server_host='192.168.1.150' or sip_registration s.sub_host='192.168.1.150' ))] [STATE: 42000 CODE 102 ERROR: [Microsoft][ODBC SQL Server Driver][SQL Server]Incorrect syntax near '|'. But in source code it is: "sip_dialogs.presence_id = sip_registrations.sip_user %q '@' %q sip_registrations.sub_host " It looks like parameters are not inserted at runtime... What do you think? 2012/3/4 Brian Foster > I think that has to do with ymthe specific ODBC driver you are using. I'd > check your configs and see of there's anything amiss. I could be wrong > though, as I haven't dealt with MSSQL. I have experience with MYSQL and > postgres with ODBC though. > > -BDF > On Mar 3, 2012 6:19 PM, "Gregor Nanger" wrote: > >> Hi! >> >> I am using Freeswitch with Windows. I hope I am not the only one. >> >> I also want to make mod_sofia to talk via ODBC with MSSQL. I can connect, >> but on call I get error, when sofia wants to write to mssql: >> >> [STATE: 42000 CODE 102 ERROR: [Microsoft][ODBC SQL Server Driver][SQL >> Server]Incorrect syntax near '|'. >> ] >> >> Is this know issue? Any suggestions? >> >> On MySQL it is working ok.. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/16e3815a/attachment.html From peter.olsson at visionutveckling.se Sun Mar 4 13:02:16 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 4 Mar 2012 10:02:16 +0000 Subject: [Freeswitch-users] Freeswitch + MSSQL In-Reply-To: References: , Message-ID: <1FFF97C269757C458224B7C895F35F1504CC69@cantor.std.visionutv.se> Did you try to set this parameter in switch.conf.xml? According to the code this should enable different method for concat() for MS SQL. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Gregor Nanger [gregor at infomedia.si] Skickat: den 4 mars 2012 01:22 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Freeswitch + MSSQL This is sql that is generated: select sip_registrations.sip_user, sip_registrations.sub_host, sip_registrations.status, sip_registrations.rpid, '', sip_dialogs.uuid, sip_dialogs.state, sip_dialogs.direction, sip_dialogs.sip_to_user, sip_dialogs.sip_to_host,sip_presence.status,sip_presence.rpid,sip_dialogs.presence_id, sip_pre sence.open_closed,'','','' from sip_registrations left join sip_dialogs on sip_dialogs.hostname = sip_registrations.host name and sip_dialogs.profile_name = sip_registrations.profile_name and (sip_dialogs.presence_id = sip_registrations.sip_ user || '@' || sip_registrations.sub_host or (sip_dialogs.sip_from_user = sip_registrations.sip_user and sip_dialogs.sip _from_host = sip_registrations.sip_host)) left join sip_presence on sip_presence.hostname=sip_registrations.hostname and (sip_registrations.sip_user=sip_presence.sip_user and sip_registrations.orig_server_host=sip_presence.sip_host and sip_ registrations.profile_name=sip_presence.profile_name) where sip_registrations.hostname='WIN-SERVER2008' and sip_registra tions.profile_name='internal' and sip_dialogs.call_info_state != 'seized' and sip_dialogs.presence_id='1002 at 192.168.1. 150' or (sip_registrations.sip_user='1002' and (sip_registrations.orig_server_host='192.168.1.150' or sip_registration s.sub_host='192.168.1.150' ))] [STATE: 42000 CODE 102 ERROR: [Microsoft][ODBC SQL Server Driver][SQL Server]Incorrect syntax near '|'. But in source code it is: "sip_dialogs.presence_id = sip_registrations.sip_user %q '@' %q sip_registrations.sub_host " It looks like parameters are not inserted at runtime... What do you think? 2012/3/4 Brian Foster > I think that has to do with ymthe specific ODBC driver you are using. I'd check your configs and see of there's anything amiss. I could be wrong though, as I haven't dealt with MSSQL. I have experience with MYSQL and postgres with ODBC though. -BDF On Mar 3, 2012 6:19 PM, "Gregor Nanger" > wrote: Hi! I am using Freeswitch with Windows. I hope I am not the only one. I also want to make mod_sofia to talk via ODBC with MSSQL. I can connect, but on call I get error, when sofia wants to write to mssql: [STATE: 42000 CODE 102 ERROR: [Microsoft][ODBC SQL Server Driver][SQL Server]Incorrect syntax near '|'. ] Is this know issue? Any suggestions? On MySQL it is working ok.. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f53366632766912397640! From gregor at infomedia.si Sun Mar 4 15:05:05 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 4 Mar 2012 13:05:05 +0100 Subject: [Freeswitch-users] Freeswitch + MSSQL In-Reply-To: <1FFF97C269757C458224B7C895F35F1504CC69@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1504CC69@cantor.std.visionutv.se> Message-ID: Thanks Peter! It's working now. I also had to set core-db-dsn in switch.conf to works... 2012/3/4 Peter Olsson > Did you try to set this parameter in switch.conf.xml? > > > > According to the code this should enable different method for concat() for > MS SQL. > > /Peter > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Gregor Nanger [ > gregor at infomedia.si] > Skickat: den 4 mars 2012 01:22 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Freeswitch + MSSQL > > This is sql that is generated: > select sip_registrations.sip_user, sip_registrations.sub_host, > sip_registrations.status, sip_registrations.rpid, '', sip_dialogs.uuid, > sip_dialogs.state, sip_dialogs.direction, sip_dialogs.sip_to_user, > sip_dialogs.sip_to_host,sip_presence.status,sip_presence.rpid,sip_dialogs.presence_id, > sip_pre > sence.open_closed,'','','' from sip_registrations left join sip_dialogs on > sip_dialogs.hostname = sip_registrations.host > name and sip_dialogs.profile_name = sip_registrations.profile_name and > (sip_dialogs.presence_id = sip_registrations.sip_ > user || '@' || sip_registrations.sub_host or (sip_dialogs.sip_from_user = > sip_registrations.sip_user and sip_dialogs.sip > _from_host = sip_registrations.sip_host)) left join sip_presence on > sip_presence.hostname=sip_registrations.hostname and > (sip_registrations.sip_user=sip_presence.sip_user and > sip_registrations.orig_server_host=sip_presence.sip_host and sip_ > registrations.profile_name=sip_presence.profile_name) where > sip_registrations.hostname='WIN-SERVER2008' and sip_registra > tions.profile_name='internal' and sip_dialogs.call_info_state != 'seized' > and sip_dialogs.presence_id='1002 at 192.168.1. > 150' or (sip_registrations.sip_user='1002' and > (sip_registrations.orig_server_host='192.168.1.150' or sip_registration > s.sub_host='192.168.1.150' ))] > [STATE: 42000 CODE 102 ERROR: [Microsoft][ODBC SQL Server Driver][SQL > Server]Incorrect syntax near '|'. > > > But in source code it is: > "sip_dialogs.presence_id = sip_registrations.sip_user %q '@' %q > sip_registrations.sub_host " > > It looks like parameters are not inserted at runtime... > > What do you think? > > > 2012/3/4 Brian Foster bdfoster at endigotech.com>> > > I think that has to do with ymthe specific ODBC driver you are using. I'd > check your configs and see of there's anything amiss. I could be wrong > though, as I haven't dealt with MSSQL. I have experience with MYSQL and > postgres with ODBC though. > > -BDF > > On Mar 3, 2012 6:19 PM, "Gregor Nanger" gregor at infomedia.si>> wrote: > Hi! > > I am using Freeswitch with Windows. I hope I am not the only one. > > I also want to make mod_sofia to talk via ODBC with MSSQL. I can connect, > but on call I get error, when sofia wants to write to mssql: > > [STATE: 42000 CODE 102 ERROR: [Microsoft][ODBC SQL Server Driver][SQL > Server]Incorrect syntax near '|'. > ] > > Is this know issue? Any suggestions? > > On MySQL it is working ok.. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > !DSPAM:4f53366632766912397640! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/a439457d/attachment-0001.html From gregor at infomedia.si Sun Mar 4 15:10:34 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 4 Mar 2012 13:10:34 +0100 Subject: [Freeswitch-users] mod_managed Message-ID: I'm so close to anounce Freeswitch as the best voip software ever created :-) I am using mod_managed with c# and trying Test Visual studio project that is included. I can connect inbound and outbound and I can control call. But I cannot read header. If I use method getHeader("EVENT-NAME") , response is null. Whichever header I read, I get null. Serialize method prints all headers.. Any suggestion? I am using Freeswitch with Windows OS. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/9ded0b34/attachment-0001.html From fluixab at bellsouth.net Sun Mar 4 15:30:05 2012 From: fluixab at bellsouth.net (Bernard Fluixa) Date: Sun, 4 Mar 2012 07:30:05 -0500 Subject: [Freeswitch-users] mod shell stream In-Reply-To: <5BA3C64DA53B4EAC9B1DA367F8FFD398@gmail.com> References: <5BA3C64DA53B4EAC9B1DA367F8FFD398@gmail.com> Message-ID: That is already done with the -t option (tuples only) Thanks anyway Bernard On Mar 4, 2012, at 1:12 AM, Seven Du wrote: > I think you might need a script to strip the column header ? > > psql -c "select 1" |cat > ?column? > ---------- > 1 > (1 row) > > > > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow > > On Saturday, March 3, 2012 at 10:17 PM, Bernard Fluixa wrote: > >> Hi there, >> >> I'm trying to make a playback of streamed sound data coming from a database. I could make the example included in documentation work (cat file.wav | sox -t wav - $@ -t raw -). However, I can't make it work with same sox options after having uploaded that same file into my PGSQL database and select it instead of a cat command. >> >> My shell script is >> >> #!/bin/bash >> >> psql -h -U postgres -E -t -q -d -c "select from
" | sox -t wav - $@ -t raw - >> >> Freeswitch gives me a "sox FAIL formats: can't open input `-': WAVE: RIFF header not found" error message. >> >> >> Has anybody streamed data from database into Freeswitch? >> >> Thank you. >> >> Bernard >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/bfba8766/attachment-0001.html From peter.olsson at visionutveckling.se Sun Mar 4 16:02:51 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 4 Mar 2012 13:02:51 +0000 Subject: [Freeswitch-users] mod_managed In-Reply-To: References: Message-ID: <1FFF97C269757C458224B7C895F35F1504CCF5@cantor.std.visionutv.se> I'm not really sure about this, but the getHeader is maybe case sensitive? Have you tried "Event-Name" as the header name instead? /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Gregor Nanger [gregor at infomedia.si] Skickat: den 4 mars 2012 13:10 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] mod_managed I'm so close to anounce Freeswitch as the best voip software ever created :-) I am using mod_managed with c# and trying Test Visual studio project that is included. I can connect inbound and outbound and I can control call. But I cannot read header. If I use method getHeader("EVENT-NAME") , response is null. Whichever header I read, I get null. Serialize method prints all headers.. Any suggestion? I am using Freeswitch with Windows OS. !DSPAM:4f53615d32762189818904! From lists at telefaks.de Sun Mar 4 16:57:23 2012 From: lists at telefaks.de (Peter Steinbach) Date: Sun, 04 Mar 2012 14:57:23 +0100 Subject: [Freeswitch-users] Audio problems with unimrcp and Vestec ASR Message-ID: <4F5374C3.800@telefaks.de> I have an Audio problem with unimrcp and Vestec ASR with mrcp v1. This seems to be a similar problem to thread: "ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem" What does work: * MRCP command are passed successsfull in both ways on port 1554 (vestec mrcp server) * Grammar is accepted and recognition is started * I can see RTP stream (wireshark) from FS port 400x to the mrcp server * Result (002 no-input-timeout) is sent back to Freeeswicth and is processed by FS successfully. What does NOT work: * RTP stream is always empty (silence), i grepped this on the network with ngrep, RTP data is empty, there is no change in the audio data when I speak. * so voice recognition will always time out (002 no-input-timeout) Here are my configs: unimrcp.conf.xml unimrcpserver-mrcp-v1.xml See logs here http://pastebin.freeswitch.org/18570 Any help is appreciated. Best egards Peter -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/8da082f9/attachment.html From gregor at infomedia.si Sun Mar 4 16:59:11 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 4 Mar 2012 14:59:11 +0100 Subject: [Freeswitch-users] mod_managed In-Reply-To: <1FFF97C269757C458224B7C895F35F1504CCF5@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1504CCF5@cantor.std.visionutv.se> Message-ID: I did tried all permutations... Even getBody() returns null.... Do you think that this is because of Windows installation? Otherwise library works ok. 2012/3/4 Peter Olsson > I'm not really sure about this, but the getHeader is maybe case sensitive? > Have you tried > > "Event-Name" as the header name instead? > > /Peter > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Gregor Nanger [ > gregor at infomedia.si] > Skickat: den 4 mars 2012 13:10 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] mod_managed > > > I'm so close to anounce Freeswitch as the best voip software ever created > :-) > > I am using mod_managed with c# and trying Test Visual studio project that > is included. I can connect inbound and outbound and I can control call. But > I cannot read header. If I use method getHeader("EVENT-NAME") , response is > null. Whichever header I read, I get null. Serialize method prints all > headers.. > > Any suggestion? I am using Freeswitch with Windows OS. > > !DSPAM:4f53615d32762189818904! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/ddab5884/attachment.html From peter.olsson at visionutveckling.se Sun Mar 4 17:12:23 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 4 Mar 2012 14:12:23 +0000 Subject: [Freeswitch-users] mod_managed In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1504CCF5@cantor.std.visionutv.se>, Message-ID: <1FFF97C269757C458224B7C895F35F1504CD24@cantor.std.visionutv.se> Please post your sample code here, and I will have a look. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Gregor Nanger [gregor at infomedia.si] Skickat: den 4 mars 2012 14:59 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] mod_managed I did tried all permutations... Even getBody() returns null.... Do you think that this is because of Windows installation? Otherwise library works ok. 2012/3/4 Peter Olsson > I'm not really sure about this, but the getHeader is maybe case sensitive? Have you tried "Event-Name" as the header name instead? /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Gregor Nanger [gregor at infomedia.si] Skickat: den 4 mars 2012 13:10 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] mod_managed I'm so close to anounce Freeswitch as the best voip software ever created :-) I am using mod_managed with c# and trying Test Visual studio project that is included. I can connect inbound and outbound and I can control call. But I cannot read header. If I use method getHeader("EVENT-NAME") , response is null. Whichever header I read, I get null. Serialize method prints all headers.. Any suggestion? I am using Freeswitch with Windows OS. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f5373e932764490361988! From peter.olsson at visionutveckling.se Sun Mar 4 17:17:46 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 4 Mar 2012 14:17:46 +0000 Subject: [Freeswitch-users] Audio problems with unimrcp and Vestec ASR In-Reply-To: <4F5374C3.800@telefaks.de> References: <4F5374C3.800@telefaks.de> Message-ID: <1FFF97C269757C458224B7C895F35F1504CD2F@cantor.std.visionutv.se> What codec is used on the call leg? I tried some ASR (UniMRCP to Nuance) once, and I noticed that the audio passed to Nuance seemed to fail when using HD audio, when I did the same test with standard PCMA it worked. I never had the time to investigate any further though, since this was onle for a test, and nothing I needed to implement yet. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Peter Steinbach [lists at telefaks.de] Skickat: den 4 mars 2012 14:57 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Audio problems with unimrcp and Vestec ASR I have an Audio problem with unimrcp and Vestec ASR with mrcp v1. This seems to be a similar problem to thread: "ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem" What does work: * MRCP command are passed successsfull in both ways on port 1554 (vestec mrcp server) * Grammar is accepted and recognition is started * I can see RTP stream (wireshark) from FS port 400x to the mrcp server * Result (002 no-input-timeout) is sent back to Freeeswicth and is processed by FS successfully. What does NOT work: * RTP stream is always empty (silence), i grepped this on the network with ngrep, RTP data is empty, there is no change in the audio data when I speak. * so voice recognition will always time out (002 no-input-timeout) Here are my configs: unimrcp.conf.xml unimrcpserver-mrcp-v1.xml See logs here http://pastebin.freeswitch.org/18570 Any help is appreciated. Best egards Peter -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de !DSPAM:4f5373f932761937219840! From gregor at infomedia.si Sun Mar 4 17:38:24 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 4 Mar 2012 15:38:24 +0100 Subject: [Freeswitch-users] mod_managed In-Reply-To: <1FFF97C269757C458224B7C895F35F1504CD24@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1504CCF5@cantor.std.visionutv.se> <1FFF97C269757C458224B7C895F35F1504CD24@cantor.std.visionutv.se> Message-ID: Well, it is code from ManagedEslTest static void OutboundModeAsync(Object stateInfo) { /* add next line to a dialplan */ TcpListener tcpListener = new TcpListener(IPAddress.Parse("127.0.0.1"), 8022); try { tcpListener.Start(); Console.WriteLine("OutboundModeAsync, waiting for connections..."); while (true) { tcpListener.BeginAcceptSocket((asyncCallback) => { TcpListener tcpListened = (TcpListener)asyncCallback.AsyncState; Socket sckClient = tcpListened.EndAcceptSocket(asyncCallback); //Initializes a new instance of ESLconnection, and connects to the host $host on the port $port, and supplies $password to freeswitch ESLconnection eslConnection = new ESLconnection(sckClient.Handle.ToInt32()); ESLevent eslEvent = eslConnection.GetInfo(); string strUuid =* eslEvent.GetHeader("UNIQUE-ID", 0); //THERE I GET NULL* eslConnection.SendRecv("myevents"); eslConnection.SendRecv("divert_events on"); eslConnection.Execute("answer", String.Empty, String.Empty); eslConnection.Execute("playback", "../../../music/8000/suite-espanola-op-47-leyenda.wav", String.Empty); while (eslConnection.Connected() == ESL_SUCCESS) { eslEvent = eslConnection.RecvEvent(); Console.WriteLine(eslEvent.Serialize(String.Empty)); } sckClient.Close(); Console.WriteLine("Connection closed uuid:{0}", strUuid); }, tcpListener); Thread.Sleep(50); } } catch (Exception ex) { Console.WriteLine(ex); } finally { tcpListener.Stop(); } } 2012/3/4 Peter Olsson > Please post your sample code here, and I will have a look. > > /Peter > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Gregor Nanger [ > gregor at infomedia.si] > Skickat: den 4 mars 2012 14:59 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] mod_managed > > I did tried all permutations... > > Even getBody() returns null.... Do you think that this is because of > Windows installation? > > Otherwise library works ok. > > > > > 2012/3/4 Peter Olsson peter.olsson at visionutveckling.se>> > I'm not really sure about this, but the getHeader is maybe case sensitive? > Have you tried > > "Event-Name" as the header name instead? > > /Peter > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> [ > freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>] f?r Gregor Nanger [ > gregor at infomedia.si] > Skickat: den 4 mars 2012 13:10 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] mod_managed > > > I'm so close to anounce Freeswitch as the best voip software ever created > :-) > > I am using mod_managed with c# and trying Test Visual studio project that > is included. I can connect inbound and outbound and I can control call. But > I cannot read header. If I use method getHeader("EVENT-NAME") , response is > null. Whichever header I read, I get null. Serialize method prints all > headers.. > > Any suggestion? I am using Freeswitch with Windows OS. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f5373e932764490361988! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/ce780075/attachment-0001.html From peter.olsson at visionutveckling.se Sun Mar 4 18:03:36 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 4 Mar 2012 15:03:36 +0000 Subject: [Freeswitch-users] mod_managed In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1504CCF5@cantor.std.visionutv.se> <1FFF97C269757C458224B7C895F35F1504CD24@cantor.std.visionutv.se>, Message-ID: <1FFF97C269757C458224B7C895F35F1504CD5F@cantor.std.visionutv.se> What happens if you use index -1 in getHeader() call? /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Gregor Nanger [gregor at infomedia.si] Skickat: den 4 mars 2012 15:38 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] mod_managed Well, it is code from ManagedEslTest static void OutboundModeAsync(Object stateInfo) { /* add next line to a dialplan */ TcpListener tcpListener = new TcpListener(IPAddress.Parse("127.0.0.1"), 8022); try { tcpListener.Start(); Console.WriteLine("OutboundModeAsync, waiting for connections..."); while (true) { tcpListener.BeginAcceptSocket((asyncCallback) => { TcpListener tcpListened = (TcpListener)asyncCallback.AsyncState; Socket sckClient = tcpListened.EndAcceptSocket(asyncCallback); //Initializes a new instance of ESLconnection, and connects to the host $host on the port $port, and supplies $password to freeswitch ESLconnection eslConnection = new ESLconnection(sckClient.Handle.ToInt32()); ESLevent eslEvent = eslConnection.GetInfo(); string strUuid = eslEvent.GetHeader("UNIQUE-ID", 0); //THERE I GET NULL eslConnection.SendRecv("myevents"); eslConnection.SendRecv("divert_events on"); eslConnection.Execute("answer", String.Empty, String.Empty); eslConnection.Execute("playback", "../../../music/8000/suite-espanola-op-47-leyenda.wav", String.Empty); while (eslConnection.Connected() == ESL_SUCCESS) { eslEvent = eslConnection.RecvEvent(); Console.WriteLine(eslEvent.Serialize(String.Empty)); } sckClient.Close(); Console.WriteLine("Connection closed uuid:{0}", strUuid); }, tcpListener); Thread.Sleep(50); } } catch (Exception ex) { Console.WriteLine(ex); } finally { tcpListener.Stop(); } } 2012/3/4 Peter Olsson > Please post your sample code here, and I will have a look. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Gregor Nanger [gregor at infomedia.si] Skickat: den 4 mars 2012 14:59 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] mod_managed I did tried all permutations... Even getBody() returns null.... Do you think that this is because of Windows installation? Otherwise library works ok. 2012/3/4 Peter Olsson >> I'm not really sure about this, but the getHeader is maybe case sensitive? Have you tried "Event-Name" as the header name instead? /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org>] f?r Gregor Nanger [gregor at infomedia.si>] Skickat: den 4 mars 2012 13:10 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] mod_managed I'm so close to anounce Freeswitch as the best voip software ever created :-) I am using mod_managed with c# and trying Test Visual studio project that is included. I can connect inbound and outbound and I can control call. But I cannot read header. If I use method getHeader("EVENT-NAME") , response is null. Whichever header I read, I get null. Serialize method prints all headers.. Any suggestion? I am using Freeswitch with Windows OS. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f537db532764595775111! From godson.g at gmail.com Sun Mar 4 19:33:50 2012 From: godson.g at gmail.com (Godson Gera) Date: Sun, 4 Mar 2012 22:03:50 +0530 Subject: [Freeswitch-users] how to alter the SDP information in freeswitch In-Reply-To: References: Message-ID: Here is some info related to altering SDP http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation -- Thanks & Regards, Godson Gera IVR India 2012/3/4 ???? > Hi, > > I'd like to know how to alter the SDP information in freeswitch ? > > Any help is appreciated! > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/1c834024/attachment.html From miha at softnet.si Sun Mar 4 20:17:52 2012 From: miha at softnet.si (Miha) Date: Sun, 04 Mar 2012 18:17:52 +0100 Subject: [Freeswitch-users] 302 redirect variable In-Reply-To: <9BCA494E297D4430B191119091E31C1A@freeswitch.org> References: <4F509C57.20205@softnet.si> <9BCA494E297D4430B191119091E31C1A@freeswitch.org> Message-ID: Thank you for that! After that I will be able to use variables from user/dir? Regards, Miha On Sun, 4 Mar 2012 01:14:13 -0300 Jo?o Mesquita wrote: > You won't get the user's password in plain text like that > EVER. If we did that, we would be considered to be > insanely insecure. > > I am guessing you are using SIP only so you can take a > look at the dialplan/public.xml file of the default > configs. On the end of that file you will see that there > is a verification to do dial plan based authentication. > > Look for this extension in particular: > > > expression="^true$" break="never"> > > > > > > data="${destination_number} XML default"/> > > > > > Regards, > > -- > Jo?o Mesquita > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > On Friday, March 2, 2012 at 7:09 AM, Miha Zoubek wrote: > > > Hi, > > > > in my directory I set variable password ( name="password" value="52166"/>) for every user. > > > > After I am doing 302 redirect in my public dialplan and > transfer call to extension, I can not use varible > pasword. > > How can I get varible password, so that I can > authenticate call. > > > > public dialplan: > > > > > > > > > > > > data="process_cdr=false"/> > > data="IZVEDI_PREUSMERITEV XML default"/> > > > > default dialplan: > > > > expression="IZVEDI_PREUSMERITEV" /> > > expression="^0(\d+)$" > > > > > data="CALLINGNUMBER=${sip_redirect_contact_user_0}"/> > > data="USERNAME=${sip_req_user}"/> > > > > > > > > > > > > > > data="effective_caller_id_name=${sip_req_user}"/> > > data="origination_caller_id_name=${sip_req_user} "/> > > > > > > data="PASSWD=${password}"/> > > data="RADIUS_ANI_AUTH XML default"/> > > data="386${sip_redirect_contact_user_0:1} > enumsbc.softnet.si (http://enumsbc.softnet.si)"/> > > > > data="{origination_callee_id_name='${effective_caller_id_name}'}${enum_auto_route}"/> > > > > > > > > data="sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx" > (mailto:sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx) > /> > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > From jean.marc.hyppolite at gmail.com Sun Mar 4 20:22:27 2012 From: jean.marc.hyppolite at gmail.com (Jean-Marc Hyppolite) Date: Sun, 4 Mar 2012 12:22:27 -0500 Subject: [Freeswitch-users] spandsp error reading frame Message-ID: <4f53a4d6.a526340a.306b.ffff9ea1@mx.google.com> Hello, I am trying to use mod_spandsp to detect call progress. I haven t been successful so far. I am getting the following error messages. ================================================ 2012-03-03 23:55:08.398205 [DEBUG] mod_spandsp_dsp.c:379 (sofia/outbound/. at ...) Starting tone detection for '1' 2012-03-03 23:55:08.398205 [INFO] mod_spandsp_dsp.c:411 (sofia/outbound/. at ...) initializing tone detector 2012-03-03 23:55:08.398205 [DEBUG] switch_core_media_bug.c:462 Attaching BUG to sofia/outbound/. at ... 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:422 (sofia/outbound/. at ...) error reading frame 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:447 (sofia/outbound/. at ...) destroying tone detector ================================================ Any help would be appreciated. Thanks Jean-Marc. N.B. Sorry if some people received this e-mail twice. I am not sure this email went through the first time. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/c5246c3d/attachment.html From lists at telefaks.de Sun Mar 4 20:25:50 2012 From: lists at telefaks.de (Peter Steinbach) Date: Sun, 04 Mar 2012 18:25:50 +0100 Subject: [Freeswitch-users] Audio problems with unimrcp and Vestec ASR In-Reply-To: <1FFF97C269757C458224B7C895F35F1504CD2F@cantor.std.visionutv.se> References: <4F5374C3.800@telefaks.de> <1FFF97C269757C458224B7C895F35F1504CD2F@cantor.std.visionutv.se> Message-ID: <4F53A59E.9070001@telefaks.de> Hello Peter, thanks for the hint. It was using L16 codec, so I changed the avaliable codecs for the mrcp profile, so that it only accepts PCMA. Now it negociates PCMA, see FS log: 2012-03-04 18:12:40.266537 [INFO] rtsp_client.c:929 () Receive RTSP Stream 192.168.178.221:54797 <-> 192.168.178.180:1554 [324 bytes] RTSP/1.0 200 OK CSeq: 1 Transport: RTP/AVP;unicast;client_port=4008-4009;server_port=5022-5023 Session: dafbbc3e7efb4474 Content-Type: application/sdp Content-Length: 145 v=0 o=UniMRCPServer 0 0 IN IP4 192.168.178.180 s=- c=IN IP4 192.168.178.180 t=0 0 m=audio 5022 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=recvonly But the behaviour is the same. The RTP stream I grepped on the network seems always to be empty (silence). All UDP RTP Packets sent from FS look like this (hex values): 80:08:00:08:00:00:05:00:f9:53:b5:e6:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5 Best regards Peter Am 04.03.2012 15:17, schrieb Peter Olsson: > What codec is used on the call leg? I tried some ASR (UniMRCP to Nuance) once, and I noticed that the audio passed to Nuance seemed to fail when using HD audio, when I did the same test with standard PCMA it worked. I never had the time to investigate any further though, since this was onle for a test, and nothing I needed to implement yet. > > /Peter > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Peter Steinbach [lists at telefaks.de] > Skickat: den 4 mars 2012 14:57 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] Audio problems with unimrcp and Vestec ASR > > I have an Audio problem with unimrcp and Vestec ASR with mrcp v1. > > This seems to be a similar problem to thread: "ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem" > > What does work: > > * MRCP command are passed successsfull in both ways on port 1554 (vestec mrcp server) > * Grammar is accepted and recognition is started > * I can see RTP stream (wireshark) from FS port 400x to the mrcp server > * Result (002 no-input-timeout) is sent back to Freeeswicth and is processed by FS successfully. > > What does NOT work: > > * RTP stream is always empty (silence), i grepped this on the network with ngrep, RTP data is empty, there is no change in the audio data when I speak. > * so voice recognition will always time out (002 no-input-timeout) > > Here are my configs: > unimrcp.conf.xml > > > > > > > > > > > > > > > > unimrcpserver-mrcp-v1.xml > > > > > > > > > > > > > > > > > > > > > > See logs here > http://pastebin.freeswitch.org/18570 > > Any help is appreciated. > > Best egards > Peter > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > > !DSPAM:4f5373f932761937219840! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From jmesquita at freeswitch.org Sun Mar 4 23:31:34 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Sun, 4 Mar 2012 17:31:34 -0300 Subject: [Freeswitch-users] 302 redirect variable In-Reply-To: References: <4F509C57.20205@softnet.si> <9BCA494E297D4430B191119091E31C1A@freeswitch.org> Message-ID: <1F1CE9939D004E86A0679267391ABA92@freeswitch.org> Yes -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Sunday, March 4, 2012 at 2:17 PM, Miha wrote: > Thank you for that! > > After that I will be able to use variables from user/dir? > > Regards, > Miha > > On Sun, 4 Mar 2012 01:14:13 -0300 > Jo?o Mesquita wrote: > > You won't get the user's password in plain text like that > > EVER. If we did that, we would be considered to be > > insanely insecure. > > > > I am guessing you are using SIP only so you can take a > > look at the dialplan/public.xml file of the default > > configs. On the end of that file you will see that there > > is a verification to do dial plan based authentication. > > > > Look for this extension in particular: > > > > > > > expression="^true$" break="never"> > > > > > > > > > > > > > data="${destination_number} XML default"/> > > > > > > > > > > Regards, > > > > -- > > Jo?o Mesquita > > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > > > > On Friday, March 2, 2012 at 7:09 AM, Miha Zoubek wrote: > > > > > Hi, > > > > > > in my directory I set variable password ( > > > > > > name="password" value="52166"/>) for every user. > > > > > > After I am doing 302 redirect in my public dialplan and > > > > > > > transfer call to extension, I can not use varible > > pasword. > > > How can I get varible password, so that I can > > > > authenticate call. > > > > > > public dialplan: > > > > > > > > > > > > > > > > > > > > > > > > data="process_cdr=false"/> > > > > > > data="IZVEDI_PREUSMERITEV XML default"/> > > > > > > default dialplan: > > > > > > > > > > > > expression="IZVEDI_PREUSMERITEV" /> > > > > > > expression="^0(\d+)$" > > > > > > > > > > > > > data="CALLINGNUMBER=${sip_redirect_contact_user_0}"/> > > > > > > data="USERNAME=${sip_req_user}"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > data="effective_caller_id_name=${sip_req_user}"/> > > > > > > data="origination_caller_id_name=${sip_req_user} "/> > > > > > > > > > > > > > > > data="PASSWD=${password}"/> > > > > > > data="RADIUS_ANI_AUTH XML default"/> > > > > > > data="386${sip_redirect_contact_user_0:1} > > enumsbc.softnet.si (http://enumsbc.softnet.si)"/> > > > > > > > > > > > > > > data="{origination_callee_id_name='${effective_caller_id_name}'}${enum_auto_route}"/> > > > > > > > > > > > > > > > > > > data="sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx (http://xxx.xxx.xxx.xxx)" > > > > (mailto:sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx) > > /> > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > > > > > (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > > > > > > > Server > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > > > (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/df818340/attachment.html From jeff at jefflenk.com Sun Mar 4 23:32:18 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Sun, 4 Mar 2012 12:32:18 -0800 (PST) Subject: [Freeswitch-users] ManagedESL Run Issues In-Reply-To: <000401ccf8a2$18c42580$4a4c7080$@google.hm> References: <000401ccf8a2$18c42580$4a4c7080$@google.hm> Message-ID: <1330893138108-7342951.post@n2.nabble.com> Make sure you are not mixing 32/64 bit versions of the dlls/exe. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ManagedESL-Run-Issues-tp7337511p7342951.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gregor at infomedia.si Mon Mar 5 02:04:10 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 5 Mar 2012 00:04:10 +0100 Subject: [Freeswitch-users] ManagedESL Run Issues In-Reply-To: <1330893138108-7342951.post@n2.nabble.com> References: <000401ccf8a2$18c42580$4a4c7080$@google.hm> <1330893138108-7342951.post@n2.nabble.com> Message-ID: Jeff, you are right. I had same error and it was because i installed x64 freeswitch, but managedesl compiled as win32... 2012/3/4 Jeff Lenk > Make sure you are not mixing 32/64 bit versions of the dlls/exe. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/ManagedESL-Run-Issues-tp7337511p7342951.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/757bb4c7/attachment.html From gregor at infomedia.si Mon Mar 5 02:07:46 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 5 Mar 2012 00:07:46 +0100 Subject: [Freeswitch-users] mod_managed In-Reply-To: <1FFF97C269757C458224B7C895F35F1504CD5F@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1504CCF5@cantor.std.visionutv.se> <1FFF97C269757C458224B7C895F35F1504CD24@cantor.std.visionutv.se> <1FFF97C269757C458224B7C895F35F1504CD5F@cantor.std.visionutv.se> Message-ID: Peter, that was it, thank you. It's funny that in test code is 0 in getHeader() Now I can anounce that FS is best ever... :-) 2012/3/4 Peter Olsson > What happens if you use index -1 in getHeader() call? > > /Peter > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Gregor Nanger [ > gregor at infomedia.si] > Skickat: den 4 mars 2012 15:38 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] mod_managed > > Well, it is code from ManagedEslTest > > static void OutboundModeAsync(Object stateInfo) > { > /* add next line to a dialplan > > */ > TcpListener tcpListener = new > TcpListener(IPAddress.Parse("127.0.0.1"), 8022); > > try > { > tcpListener.Start(); > > Console.WriteLine("OutboundModeAsync, waiting for connections..."); > > while (true) > { > tcpListener.BeginAcceptSocket((asyncCallback) => > { > TcpListener tcpListened = (TcpListener)asyncCallback.AsyncState; > > Socket sckClient = tcpListened.EndAcceptSocket(asyncCallback); > > //Initializes a new instance of ESLconnection, and connects to > the host $host on the port $port, and supplies $password to freeswitch > ESLconnection eslConnection = new > ESLconnection(sckClient.Handle.ToInt32()); > > ESLevent eslEvent = eslConnection.GetInfo(); > string strUuid = eslEvent.GetHeader("UNIQUE-ID", 0); //THERE I > GET NULL > > eslConnection.SendRecv("myevents"); > eslConnection.SendRecv("divert_events on"); > > eslConnection.Execute("answer", String.Empty, String.Empty); > eslConnection.Execute("playback", > "../../../music/8000/suite-espanola-op-47-leyenda.wav", String.Empty); > > while (eslConnection.Connected() == ESL_SUCCESS) > { > eslEvent = eslConnection.RecvEvent(); > Console.WriteLine(eslEvent.Serialize(String.Empty)); > } > > sckClient.Close(); > Console.WriteLine("Connection closed uuid:{0}", strUuid); > > }, tcpListener); > > Thread.Sleep(50); > } > } > catch (Exception ex) > { > Console.WriteLine(ex); > } > finally > { > tcpListener.Stop(); > } > } > > > > 2012/3/4 Peter Olsson peter.olsson at visionutveckling.se>> > Please post your sample code here, and I will have a look. > > /Peter > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> [ > freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>] f?r Gregor Nanger [ > gregor at infomedia.si] > Skickat: den 4 mars 2012 14:59 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] mod_managed > > I did tried all permutations... > > Even getBody() returns null.... Do you think that this is because of > Windows installation? > > Otherwise library works ok. > > > > > 2012/3/4 Peter Olsson peter.olsson at visionutveckling.se> >> > I'm not really sure about this, but the getHeader is maybe case sensitive? > Have you tried > > "Event-Name" as the header name instead? > > /Peter > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>> [ > freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>>] f?r Gregor Nanger [ > gregor at infomedia.si >] > Skickat: den 4 mars 2012 13:10 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] mod_managed > > > I'm so close to anounce Freeswitch as the best voip software ever created > :-) > > I am using mod_managed with c# and trying Test Visual studio project that > is included. I can connect inbound and outbound and I can control call. But > I cannot read header. If I use method getHeader("EVENT-NAME") , response is > null. Whichever header I read, I get null. Serialize method prints all > headers.. > > Any suggestion? I am using Freeswitch with Windows OS. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org consulting at freeswitch.org> > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f537db532764595775111! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/51031d9c/attachment-0001.html From jeff at jefflenk.com Mon Mar 5 02:50:01 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Sun, 4 Mar 2012 15:50:01 -0800 (PST) Subject: [Freeswitch-users] mod_managed In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1504CCF5@cantor.std.visionutv.se> <1FFF97C269757C458224B7C895F35F1504CD24@cantor.std.visionutv.se> <1FFF97C269757C458224B7C895F35F1504CD5F@cantor.std.visionutv.se> Message-ID: <1330905001738-7343192.post@n2.nabble.com> Thanks for verifying the sample and the correction. Fix was applied to Git Head. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-managed-tp7342274p7343192.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mstockton at harqen.com Mon Mar 5 03:41:08 2012 From: mstockton at harqen.com (Matt Stockton) Date: Sun, 4 Mar 2012 18:41:08 -0600 Subject: [Freeswitch-users] Using mod_shout over ssl, curl issue with curl-ca-bundle.crt file location In-Reply-To: References: Message-ID: I just rolled back to the 12/07 FS version and confirmed that mod_shout with SSL is working for me in that version...no complaints about the cert file I'm speculating that the curl call in the 12/07 version is somehow referencing the CA file at /etc/ssl/certs/ca-certificates.crt , but is no longer referencing that file in the latest, and is trying to reference: /usr/local/freeswitch/share/curl/curl-ca-bundle.crt instead, which doesn't exist. I guess I could put a sym link in there during my deployment process, but my question is: is this the appropriate way to handle the situation? Or should I be doing something different during the make and install? Or is there something I need to add to the FS configuration? Thanks in advance!! Matt On Sat, Mar 3, 2012 at 11:53 AM, Matt Stockton wrote: > Hi all, > > I just upgraded to the latest git, and I'm trying to dive into an issue > I'm having. I am using mod_shout and in some instances am playing files > that are hosted on web servers protected by https. This seemed to be > working fine before I upgraded, but now I am getting the following issues, > which is preventing the streaming of the files: > > 12-03-02 19:06:57.926919 [WARNING] mod_shout.c:468 CURL returned > error:[77] problem with the SSL CA cert (path? access rights?) : error > setting certificate verify locations: > CAfile: /usr/local/freeswitch/share/curl/curl-ca-bundle.crt > CApath: none > > I looked at the code and the git history in mod_shout.c where it is > setting all the curl options, nothing seems to have changed there since I > last updated FS (12/07), however, the curl-ca-bundle file is certainly not > located at /usr/local/freeswitch/share/curl/curl-ca-bundle.crt and never > has been as far as I know. > > I also looked at other mods that are using curl and where they are > calling switch_curl_easy_setopt (mod_xml_curl, mod_httapi), and noticed > that those mods are setting options that might be related to what I > need? CURLOPT_SSLCERT > > I am confused as to what is causing the breakage, since mod_shout hasn't > changed since I last updated, yet none of the ssl curl options are set in > mod_shout..and I never had any problems with the mod_shout curl usage > finding the certificate verify locations by default. Is there some other > default that used to be set in the freeswitch configuration that I need to > set manually? > > Any help is appreciated!!! Thanks! > Matt > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/398eb39d/attachment.html From krice at freeswitch.org Mon Mar 5 04:39:11 2012 From: krice at freeswitch.org (Ken Rice) Date: Sun, 04 Mar 2012 19:39:11 -0600 Subject: [Freeswitch-users] Using mod_shout over ssl, curl issue with curl-ca-bundle.crt file location In-Reply-To: Message-ID: Are you using System Libcurl or in tree libcurl? We did recently fix a problem with linkink syste, libcurl.... On 3/4/12 6:41 PM, "Matt Stockton" wrote: > I just rolled back to the 12/07 FS version and confirmed that mod_shout with > SSL is working for me in that version...no complaints about the cert file > > I'm speculating that the curl call in the 12/07 version is somehow referencing > the CA file at?/etc/ssl/certs/ca-certificates.crt , but is no longer > referencing that file in the latest, and is trying to > reference:?/usr/local/freeswitch/share/curl/curl-ca-bundle.crt instead, which > doesn't exist.? > > I guess I could put a sym link in there during my deployment process, but my > question is: is this the appropriate way to handle the situation? Or should I > be doing something different during the make and install? Or is there > something I need to add to the FS configuration? > > Thanks in advance!! > Matt > > On Sat, Mar 3, 2012 at 11:53 AM, Matt Stockton wrote: >> Hi all, >> >> I just upgraded to the latest git, and I'm trying to dive into an issue I'm >> having. I am using mod_shout and in some instances am playing files that are >> hosted on web servers protected by https. This seemed to be working fine >> before I upgraded, but now I am getting the following issues, which is >> preventing the streaming of the files: >> >> 12-03-02 19:06:57.926919 [WARNING] mod_shout.c:468 CURL returned error:[77] >> problem with the SSL CA cert (path? access rights?) : error setting >> certificate verify locations: >> ? CAfile: /usr/local/freeswitch/share/curl/curl-ca-bundle.crt >> ? CApath: none >> >> I looked at the code and the git history in mod_shout.c?where it is setting >> all the curl options, nothing seems to have changed there since I last >> updated FS (12/07), however, the curl-ca-bundle file is certainly not located >> at /usr/local/freeswitch/share/curl/curl-ca-bundle.crt and never has been as >> far as I know. >> >> I also looked at other mods that are using curl and where they are >> calling?switch_curl_easy_setopt (mod_xml_curl, mod_httapi), and noticed that >> those mods are setting options that might be related to what I >> need??CURLOPT_SSLCERT >> >> I am confused as to what is causing the breakage, since mod_shout hasn't >> changed since I last updated, yet none of the ssl curl options are set in >> mod_shout..and I never had any problems with the mod_shout curl usage finding >> the certificate verify locations by default. Is there some other default that >> used to be set in the freeswitch configuration that I need to set manually? >> >> Any help is appreciated!!! Thanks! >> Matt > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/9dc4cf48/attachment.html From mstockton at harqen.com Mon Mar 5 05:52:00 2012 From: mstockton at harqen.com (Matt Stockton) Date: Sun, 4 Mar 2012 20:52:00 -0600 Subject: [Freeswitch-users] Using mod_shout over ssl, curl issue with curl-ca-bundle.crt file location In-Reply-To: References: Message-ID: I'm not doing anything special during the installation process, (e.g. just doing bootstrap, configure, make, and make install), so I'm assuming I'm using the in tree libcurl? Is there an easy way to tell? What was the recent fix? Would it affect where it looks for the cert file? On Sun, Mar 4, 2012 at 7:39 PM, Ken Rice wrote: > Are you using System Libcurl or in tree libcurl? We did recently fix a > problem with linkink syste, libcurl.... > > > > > > On 3/4/12 6:41 PM, "Matt Stockton" wrote: > > I just rolled back to the 12/07 FS version and confirmed that mod_shout > with SSL is working for me in that version...no complaints about the cert > file > > I'm speculating that the curl call in the 12/07 version is somehow > referencing the CA file at /etc/ssl/certs/ca-certificates.crt , but is no > longer referencing that file in the latest, and is trying to > reference: /usr/local/freeswitch/share/curl/curl-ca-bundle.crt instead, > which doesn't exist. > > I guess I could put a sym link in there during my deployment process, but > my question is: is this the appropriate way to handle the situation? Or > should I be doing something different during the make and install? Or is > there something I need to add to the FS configuration? > > Thanks in advance!! > Matt > > On Sat, Mar 3, 2012 at 11:53 AM, Matt Stockton > wrote: > > Hi all, > > I just upgraded to the latest git, and I'm trying to dive into an issue > I'm having. I am using mod_shout and in some instances am playing files > that are hosted on web servers protected by https. This seemed to be > working fine before I upgraded, but now I am getting the following issues, > which is preventing the streaming of the files: > > 12-03-02 19:06:57.926919 [WARNING] mod_shout.c:468 CURL returned > error:[77] problem with the SSL CA cert (path? access rights?) : error > setting certificate verify locations: > CAfile: /usr/local/freeswitch/share/curl/curl-ca-bundle.crt > CApath: none > > I looked at the code and the git history in mod_shout.c where it is > setting all the curl options, nothing seems to have changed there since I > last updated FS (12/07), however, the curl-ca-bundle file is certainly not > located at /usr/local/freeswitch/share/curl/curl-ca-bundle.crt and never > has been as far as I know. > > I also looked at other mods that are using curl and where they are > calling switch_curl_easy_setopt (mod_xml_curl, mod_httapi), and noticed > that those mods are setting options that might be related to what I > need? CURLOPT_SSLCERT > > I am confused as to what is causing the breakage, since mod_shout hasn't > changed since I last updated, yet none of the ssl curl options are set in > mod_shout..and I never had any problems with the mod_shout curl usage > finding the certificate verify locations by default. Is there some other > default that used to be set in the freeswitch configuration that I need to > set manually? > > Any help is appreciated!!! Thanks! > Matt > > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/e77935c0/attachment-0001.html From krice at freeswitch.org Mon Mar 5 05:56:46 2012 From: krice at freeswitch.org (Ken Rice) Date: Sun, 04 Mar 2012 20:56:46 -0600 Subject: [Freeswitch-users] Using mod_shout over ssl, curl issue with curl-ca-bundle.crt file location In-Reply-To: Message-ID: What Platform are you using and do you have libcurl installed on the system? K On 3/4/12 8:52 PM, "Matt Stockton" wrote: > I'm not doing anything special during the installation process, (e.g. just > doing bootstrap, configure, make, and make install), so I'm assuming I'm using > the in tree libcurl? Is there an easy way to tell? > > What was the recent fix? Would it affect where it looks for the cert file? > > On Sun, Mar 4, 2012 at 7:39 PM, Ken Rice wrote: >> Are you using System Libcurl or in tree libcurl? We did recently fix a >> problem with linkink syste, libcurl.... >> >> >> >> >> >> On 3/4/12 6:41 PM, "Matt Stockton" > > wrote: >> >>> I just rolled back to the 12/07 FS version and confirmed that mod_shout with >>> SSL is working for me in that version...no complaints about the cert file >>> >>> I'm speculating that the curl call in the 12/07 version is somehow >>> referencing the CA file at?/etc/ssl/certs/ca-certificates.crt , but is no >>> longer referencing that file in the latest, and is trying to >>> reference:?/usr/local/freeswitch/share/curl/curl-ca-bundle.crt instead, >>> which doesn't exist.? >>> >>> I guess I could put a sym link in there during my deployment process, but my >>> question is: is this the appropriate way to handle the situation? Or should >>> I be doing something different during the make and install? Or is there >>> something I need to add to the FS configuration? >>> >>> Thanks in advance!! >>> Matt >>> >>> On Sat, Mar 3, 2012 at 11:53 AM, Matt Stockton >> > wrote: >>>> Hi all, >>>> >>>> I just upgraded to the latest git, and I'm trying to dive into an issue I'm >>>> having. I am using mod_shout and in some instances am playing files that >>>> are hosted on web servers protected by https. This seemed to be working >>>> fine before I upgraded, but now I am getting the following issues, which is >>>> preventing the streaming of the files: >>>> >>>> 12-03-02 19:06:57.926919 [WARNING] mod_shout.c:468 CURL returned error:[77] >>>> problem with the SSL CA cert (path? access rights?) : error setting >>>> certificate verify locations: >>>> ? CAfile: /usr/local/freeswitch/share/curl/curl-ca-bundle.crt >>>> ? CApath: none >>>> >>>> I looked at the code and the git history in mod_shout.c?where it is setting >>>> all the curl options, nothing seems to have changed there since I last >>>> updated FS (12/07), however, the curl-ca-bundle file is certainly not >>>> located at /usr/local/freeswitch/share/curl/curl-ca-bundle.crt and never >>>> has been as far as I know. >>>> >>>> I also looked at other mods that are using curl and where they are >>>> calling?switch_curl_easy_setopt (mod_xml_curl, mod_httapi), and noticed >>>> that those mods are setting options that might be related to what I >>>> need??CURLOPT_SSLCERT >>>> >>>> I am confused as to what is causing the breakage, since mod_shout hasn't >>>> changed since I last updated, yet none of the ssl curl options are set in >>>> mod_shout..and I never had any problems with the mod_shout curl usage >>>> finding the certificate verify locations by default. Is there some other >>>> default that used to be set in the freeswitch configuration that I need to >>>> set manually? >>>> >>>> Any help is appreciated!!! Thanks! >>>> Matt >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/a15a7a5c/attachment.html From mstockton at harqen.com Mon Mar 5 06:03:39 2012 From: mstockton at harqen.com (Matt Stockton) Date: Sun, 4 Mar 2012 21:03:39 -0600 Subject: [Freeswitch-users] Using mod_shout over ssl, curl issue with curl-ca-bundle.crt file location In-Reply-To: References: Message-ID: I am using Ubtuntu 11.10 64bit and I have libcurl3 installed On Sun, Mar 4, 2012 at 8:56 PM, Ken Rice wrote: > What Platform are you using and do you have libcurl installed on the > system? > > K > > > > On 3/4/12 8:52 PM, "Matt Stockton" wrote: > > I'm not doing anything special during the installation process, (e.g. just > doing bootstrap, configure, make, and make install), so I'm assuming I'm > using the in tree libcurl? Is there an easy way to tell? > > What was the recent fix? Would it affect where it looks for the cert file? > > On Sun, Mar 4, 2012 at 7:39 PM, Ken Rice wrote: > > Are you using System Libcurl or in tree libcurl? We did recently fix a > problem with linkink syste, libcurl.... > > > > > > On 3/4/12 6:41 PM, "Matt Stockton" http://mstockton at harqen.com> > wrote: > > I just rolled back to the 12/07 FS version and confirmed that mod_shout > with SSL is working for me in that version...no complaints about the cert > file > > I'm speculating that the curl call in the 12/07 version is somehow > referencing the CA file at /etc/ssl/certs/ca-certificates.crt , but is no > longer referencing that file in the latest, and is trying to > reference: /usr/local/freeswitch/share/curl/curl-ca-bundle.crt instead, > which doesn't exist. > > I guess I could put a sym link in there during my deployment process, but > my question is: is this the appropriate way to handle the situation? Or > should I be doing something different during the make and install? Or is > there something I need to add to the FS configuration? > > Thanks in advance!! > Matt > > On Sat, Mar 3, 2012 at 11:53 AM, Matt Stockton http://mstockton at harqen.com> > wrote: > > Hi all, > > I just upgraded to the latest git, and I'm trying to dive into an issue > I'm having. I am using mod_shout and in some instances am playing files > that are hosted on web servers protected by https. This seemed to be > working fine before I upgraded, but now I am getting the following issues, > which is preventing the streaming of the files: > > 12-03-02 19:06:57.926919 [WARNING] mod_shout.c:468 CURL returned > error:[77] problem with the SSL CA cert (path? access rights?) : error > setting certificate verify locations: > CAfile: /usr/local/freeswitch/share/curl/curl-ca-bundle.crt > CApath: none > > I looked at the code and the git history in mod_shout.c where it is > setting all the curl options, nothing seems to have changed there since I > last updated FS (12/07), however, the curl-ca-bundle file is certainly not > located at /usr/local/freeswitch/share/curl/curl-ca-bundle.crt and never > has been as far as I know. > > I also looked at other mods that are using curl and where they are > calling switch_curl_easy_setopt (mod_xml_curl, mod_httapi), and noticed > that those mods are setting options that might be related to what I > need? CURLOPT_SSLCERT > > I am confused as to what is causing the breakage, since mod_shout hasn't > changed since I last updated, yet none of the ssl curl options are set in > mod_shout..and I never had any problems with the mod_shout curl usage > finding the certificate verify locations by default. Is there some other > default that used to be set in the freeswitch configuration that I need to > set manually? > > Any help is appreciated!!! Thanks! > Matt > > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org < > http://FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/71d5a92f/attachment-0001.html From hjz51 at qq.com Mon Mar 5 09:56:19 2012 From: hjz51 at qq.com (hjz51) Date: Mon, 5 Mar 2012 14:56:19 +0800 Subject: [Freeswitch-users] how to use " remote_media_ip, remote_media_port" in dialplan Message-ID: <00d401ccfa9d$132ff020$398fd060$@com> Hi, I tried to use the variable remote_media_ip, remote_media_port within dialplan, but it is not returning anything. Does anyone know when this variable gets set and how to have this variable to be set as soon as an INVITE hit freeswitch? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/86204e31/attachment.html From asteriskcoding at gmail.com Mon Mar 5 07:39:50 2012 From: asteriskcoding at gmail.com (Ast Coder) Date: Sun, 4 Mar 2012 23:39:50 -0500 Subject: [Freeswitch-users] What is the best way to predictive dial with FreeSwitch? What is the best way to pull real-time data from FreeSwitch? In-Reply-To: References: Message-ID: Hello everyone, New to FreeSwitch here. Trying to make the change from Asterisk to FreeSwitch but need to plan ahead for our GUI implementation. As I read more of the documentation, I would appreciate some guidance on the following: - What is the best way to pull and display real-time call status from FreeSwitch to a web GUI. E.g. extension status, call connection status, etc... - What is best way to do predictive dialling? I guess I am looking for something like Asterisk spool files or the AMI equivalent. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120304/6f6758e1/attachment.html From peter.olsson at visionutveckling.se Mon Mar 5 10:41:24 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 5 Mar 2012 07:41:24 +0000 Subject: [Freeswitch-users] how to use " remote_media_ip, remote_media_port" in dialplan Message-ID: <1FFF97C269757C458224B7C895F35F1504CFE1@cantor.std.visionutv.se> My guess is that it's set as soon as media is up, so either during early media or when the call has been answered. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r hjz51 Skickat: den 5 mars 2012 07:56 Till: 'freeswitch-users' ?mne: [Freeswitch-users] how to use " remote_media_ip, remote_media_port" in dialplan Hi, I tried to use the variable remote_media_ip, remote_media_port within dialplan, but it is not returning anything. Does anyone know when this variable gets set and how to have this variable to be set as soon as an INVITE hit freeswitch? Thanks, !DSPAM:4f54629c32761872512829! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/7114dae4/attachment.html From miha at softnet.si Mon Mar 5 11:01:12 2012 From: miha at softnet.si (Miha Zoubek) Date: Mon, 05 Mar 2012 09:01:12 +0100 Subject: [Freeswitch-users] 302 redirect variable In-Reply-To: <1F1CE9939D004E86A0679267391ABA92@freeswitch.org> References: <4F509C57.20205@softnet.si> <9BCA494E297D4430B191119091E31C1A@freeswitch.org> <1F1CE9939D004E86A0679267391ABA92@freeswitch.org> Message-ID: <4F5472C8.2050004@softnet.si> Hi @Jo?o, first thank you for your quick response. I have add ${sip_authorized} condition, but still no luck, as when 302 happens FS indicate user as inbound call who is calling on redirected number due to this I can not use password which is set for phone who is having redirect set and is connected on FS. SO, I can only use variables which are set for incoming call. This is my public dialplan: I have also add log to pastebin: http://pastebin.freeswitch.org/18575 Thanks! Miha On 03/04/2012 09:31 PM, Jo?o Mesquita wrote: > Yes > > -- > Jo?o Mesquita > Sent with Sparrow > > On Sunday, March 4, 2012 at 2:17 PM, Miha wrote: > >> Thank you for that! >> >> After that I will be able to use variables from user/dir? >> >> Regards, >> Miha >> >> On Sun, 4 Mar 2012 01:14:13 -0300 >> Jo?o Mesquita > > wrote: >>> You won't get the user's password in plain text like that >>> EVER. If we did that, we would be considered to be >>> insanely insecure. >>> >>> I am guessing you are using SIP only so you can take a >>> look at the dialplan/public.xml file of the default >>> configs. On the end of that file you will see that there >>> is a verification to do dial plan based authentication. >>> >>> Look for this extension in particular: >>> >>> >>> >> expression="^true$" break="never"> >>> >>> >>> >>> >>> >>> >> data="${destination_number} XML default"/> >>> >>> >>> >>> >>> Regards, >>> >>> -- >>> Jo?o Mesquita >>> Sent with Sparrow (http://www.sparrowmailapp.com/?sig) >>> >>> >>> On Friday, March 2, 2012 at 7:09 AM, Miha Zoubek wrote: >>> >>>> Hi, >>>> in my directory I set variable password (>> name="password" value="52166"/>) for every user. >>>> After I am doing 302 redirect in my public dialplan and >>> transfer call to extension, I can not use varible >>> pasword. >>>> How can I get varible password, so that I can >>> authenticate call. >>>> public dialplan: >>>> >>>> >>>> >>>> >>>> >> data="process_cdr=false"/> >>>> >> data="IZVEDI_PREUSMERITEV XML default"/> >>>> default dialplan: >>>> >> expression="IZVEDI_PREUSMERITEV" /> >>>> >> expression="^0(\d+)$" > >>>> >>>> >> data="CALLINGNUMBER=${sip_redirect_contact_user_0}"/> >>>> >> data="USERNAME=${sip_req_user}"/> >>>> >>>> >> data="effective_caller_id_name=${sip_req_user}"/> >>>> >> data="origination_caller_id_name=${sip_req_user} "/> >>>> >> data="PASSWD=${password}"/> >>>> >> data="RADIUS_ANI_AUTH XML default"/> >>>> >> data="386${sip_redirect_contact_user_0:1} >>> enumsbc.softnet.si (http://enumsbc.softnet.si)"/> >>>> > data="{origination_callee_id_name='${effective_caller_id_name}'}${enum_auto_route}"/> >>>> >>>> > data="sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx >> " >> (mailto:sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx >> ) >>> /> >> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>> (mailto:consulting at freeswitch.org) >>>> http://www.freeswitchsolutions.com >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>> Server >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>> (mailto:FreeSWITCH-users at lists.freeswitch.org) >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/273f7a87/attachment-0001.html From gregor at infomedia.si Mon Mar 5 11:28:08 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 5 Mar 2012 09:28:08 +0100 Subject: [Freeswitch-users] What is the best way to predictive dial with FreeSwitch? What is the best way to pull real-time data from FreeSwitch? In-Reply-To: References: Message-ID: If you want web GUI, then your web application needs to talk with FS. You can talk via mod_event_socket: http://wiki.freeswitch.org/wiki/Mod_event_socket And you can set FS to use MySql as DB backend and you can make query's about active calls... 2012/3/5 Ast Coder > Hello everyone, > > New to FreeSwitch here. Trying to make the change from Asterisk to > FreeSwitch but need to plan ahead for our GUI implementation. As I read > more of the documentation, I would appreciate some guidance on the > following: > > - What is the best way to pull and display real-time call status from > FreeSwitch to a web GUI. E.g. extension status, call connection status, > etc... > - What is best way to do predictive dialling? I guess I am looking for > something like Asterisk spool files or the AMI equivalent. > > > Thanks, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/4b3dd2db/attachment.html From gregor at infomedia.si Mon Mar 5 11:33:05 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 5 Mar 2012 09:33:05 +0100 Subject: [Freeswitch-users] 302 redirect variable In-Reply-To: <4F5472C8.2050004@softnet.si> References: <4F509C57.20205@softnet.si> <9BCA494E297D4430B191119091E31C1A@freeswitch.org> <1F1CE9939D004E86A0679267391ABA92@freeswitch.org> <4F5472C8.2050004@softnet.si> Message-ID: Yes Miha! I think that is impossible to get password in raw format via variables. If you have registration data somewhere else in database, you can make sql query. You need to make, that on redirection your sbc is using IP authentication, not username/pass. That way, you wouldn't need password. BR Grega 2012/3/5 Miha Zoubek > Hi @Jo?o, > > first thank you for your quick response. > > I have add ${sip_authorized} condition, but still no luck, as when 302 > happens FS indicate user as inbound call who is calling on redirected > number due to this I can not use password which is set for phone who is > having redirect set and is connected on FS. SO, I can only use variables > which are set for incoming call. > > This is my public dialplan: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="sofia/internal/${destination_number}.fs_kabelvoip1% > fs_kabelvoip1.fs1.softnet.si"/> > > > > > > > > > > data="{origination_callee_id_name='${sip_req_user}'}IZVEDI_PREUSMERITEV XML > default"/> > > > > > > > > > I have also add log to pastebin: http://pastebin.freeswitch.org/18575 > > Thanks! > Miha > > > On 03/04/2012 09:31 PM, Jo?o Mesquita wrote: > > Yes > > -- > Jo?o Mesquita > Sent with Sparrow > > On Sunday, March 4, 2012 at 2:17 PM, Miha wrote: > > Thank you for that! > > After that I will be able to use variables from user/dir? > > Regards, > Miha > > On Sun, 4 Mar 2012 01:14:13 -0300 > Jo?o Mesquita wrote: > > You won't get the user's password in plain text like that > EVER. If we did that, we would be considered to be > insanely insecure. > > I am guessing you are using SIP only so you can take a > look at the dialplan/public.xml file of the default > configs. On the end of that file you will see that there > is a verification to do dial plan based authentication. > > Look for this extension in particular: > > > expression="^true$" break="never"> > > > > > > data="${destination_number} XML default"/> > > > > > Regards, > > -- > Jo?o Mesquita > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > On Friday, March 2, 2012 at 7:09 AM, Miha Zoubek wrote: > > Hi, > in my directory I set variable password ( > name="password" value="52166"/>) for every user. > > After I am doing 302 redirect in my public dialplan and > > transfer call to extension, I can not use varible > pasword. > > How can I get varible password, so that I can > > authenticate call. > > public dialplan: > > > > > > > > > data="process_cdr=false"/> > > > data="IZVEDI_PREUSMERITEV XML default"/> > > default dialplan: > > expression="IZVEDI_PREUSMERITEV" /> > > > expression="^0(\d+)$" > > > > > data="CALLINGNUMBER=${sip_redirect_contact_user_0}"/> > > > data="USERNAME=${sip_req_user}"/> > > > > data="effective_caller_id_name=${sip_req_user}"/> > > > data="origination_caller_id_name=${sip_req_user} "/> > > > data="PASSWD=${password}"/> > > > data="RADIUS_ANI_AUTH XML default"/> > > > data="386${sip_redirect_contact_user_0:1} > enumsbc.softnet.si (http://enumsbc.softnet.si)"/> > > > > data="{origination_callee_id_name='${effective_caller_id_name}'}${enum_auto_route}"/> > > > > data="sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx" > (mailto:sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx > ) > > /> > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > (mailto:consulting at freeswitch.org ) > > http://www.freeswitchsolutions.com > FreeSWITCH-powered IP PBX: The CudaTel Communication > > Server > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > (mailto:FreeSWITCH-users at lists.freeswitch.org > ) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/a80f2263/attachment-0001.html From anita.hall at simmortel.com Mon Mar 5 12:41:23 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Mon, 5 Mar 2012 15:11:23 +0530 Subject: [Freeswitch-users] uuid in spandsp log Message-ID: Hi I have verbose enabled in fax which gives me all the lower level fax related message transfers by spandsp. Also, uuid is enabled so all log lines are preceded by uuid. But spandsp related logs do not have a uuid. How can I enable uuid in spandsp log ? Thanks. regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/587641c9/attachment.html From anita.hall at simmortel.com Mon Mar 5 12:43:10 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Mon, 5 Mar 2012 15:13:10 +0530 Subject: [Freeswitch-users] What is the best way to predictive dial with FreeSwitch? What is the best way to pull real-time data from FreeSwitch? In-Reply-To: References: Message-ID: You can use mod_event_socket for things that you used AMI and AGI for. It can be used for both sending and receiving events, call control, status, CDR etc. regards, Anita On Mon, Mar 5, 2012 at 1:58 PM, Gregor Nanger wrote: > If you want web GUI, then your web application needs to talk with FS. You > can talk via mod_event_socket: > http://wiki.freeswitch.org/wiki/Mod_event_socket > > And you can set FS to use MySql as DB backend and you can make query's > about active calls... > > > > > 2012/3/5 Ast Coder > >> Hello everyone, >> >> New to FreeSwitch here. Trying to make the change from Asterisk to >> FreeSwitch but need to plan ahead for our GUI implementation. As I read >> more of the documentation, I would appreciate some guidance on the >> following: >> >> - What is the best way to pull and display real-time call status from >> FreeSwitch to a web GUI. E.g. extension status, call connection status, >> etc... >> - What is best way to do predictive dialling? I guess I am looking for >> something like Asterisk spool files or the AMI equivalent. >> >> >> Thanks, >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/6ee7b4fe/attachment.html From hjz51 at qq.com Mon Mar 5 14:09:59 2012 From: hjz51 at qq.com (hjz51) Date: Mon, 5 Mar 2012 19:09:59 +0800 Subject: [Freeswitch-users] =?utf-8?b?562U5aSNOiAgaG93IHRvIHVzZSAiIHJlbW90?= =?utf-8?q?e=5Fmedia=5Fip=2C=09remote=5Fmedia=5Fport=22_in_dialplan?= In-Reply-To: <1FFF97C269757C458224B7C895F35F1504CFE1@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1504CFE1@cantor.std.visionutv.se> Message-ID: <00f901ccfac0$814a8070$83df8150$@com> Hello peter, Thanks very much for your reply. I agree with you . That you have any good way to make them work. Thanks! ???: Peter Olsson [mailto:peter.olsson at visionutveckling.se] ????: 2012?3?5???? 15:41 ???: 'FreeSWITCH Users Help' ??: Re: [Freeswitch-users] how to use " remote_media_ip, remote_media_port" in dialplan My guess is that it's set as soon as media is up, so either during early media or when the call has been answered. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r hjz51 Skickat: den 5 mars 2012 07:56 Till: 'freeswitch-users' ?mne: [Freeswitch-users] how to use " remote_media_ip, remote_media_port" in dialplan Hi, I tried to use the variable remote_media_ip, remote_media_port within dialplan, but it is not returning anything. Does anyone know when this variable gets set and how to have this variable to be set as soon as an INVITE hit freeswitch? Thanks, !DSPAM:4f54629c32761872512829! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/f9eeed44/attachment.html From avi at avimarcus.net Mon Mar 5 15:22:06 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 5 Mar 2012 14:22:06 +0200 Subject: [Freeswitch-users] What is the best way to predictive dial with FreeSwitch? What is the best way to pull real-time data from FreeSwitch? In-Reply-To: References: Message-ID: Or if you want something pre-written, take a look at: http://www.newfies-dialer.org/ -Avi On Mon, Mar 5, 2012 at 11:43 AM, Anita Hall wrote: > You can use mod_event_socket for things that you used AMI and AGI for. It > can be used for both sending and receiving events, call control, status, > CDR etc. > > regards, > Anita > > > > > On Mon, Mar 5, 2012 at 1:58 PM, Gregor Nanger wrote: > >> If you want web GUI, then your web application needs to talk with FS. You >> can talk via mod_event_socket: >> http://wiki.freeswitch.org/wiki/Mod_event_socket >> >> And you can set FS to use MySql as DB backend and you can make query's >> about active calls... >> >> >> >> >> 2012/3/5 Ast Coder >> >>> Hello everyone, >>> >>> New to FreeSwitch here. Trying to make the change from Asterisk to >>> FreeSwitch but need to plan ahead for our GUI implementation. As I read >>> more of the documentation, I would appreciate some guidance on the >>> following: >>> >>> - What is the best way to pull and display real-time call status from >>> FreeSwitch to a web GUI. E.g. extension status, call connection status, >>> etc... >>> - What is best way to do predictive dialling? I guess I am looking for >>> something like Asterisk spool files or the AMI equivalent. >>> >>> >>> Thanks, >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/52d56b19/attachment-0001.html From shahzad.bhatti at g-r-v.com Mon Mar 5 15:47:45 2012 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Mon, 5 Mar 2012 17:47:45 +0500 Subject: [Freeswitch-users] Use of ESL and PHP for Call Message-ID: Hi, Thanks for the tip but i want to know that what hsndler.php code is and where to put that file i means path of file. second i have an error on this example and log show like that in fs_cli 2012-03-05 22:36:53.293238 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/&speak:cepstral|callie|Hi,.PCMU 2012-03-05 22:36:53.293238 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/we.PCMU 2012-03-05 22:36:53.293238 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/&speak:cepstral|callie|Hi,.PCMU 2012-03-05 22:36:53.293238 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/we.PCMU 2012-03-05 22:36:53.293238 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/&speak:cepstral|callie|Hi,.PCMU 2012-03-05 22:36:53.293238 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/we.PCMU 2012-03-05 22:36:53.293238 [WARNING] switch_ivr_play_say.c:2072 PAGD failure! Transfer to: on / Monday / at Please Guide me Regards Shahzad Bhatti On Sat, Mar 3, 2012 at 5:02 AM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Use of ESL and PHP for Call (Michael Collins) > 2. Re: a question about freeswitch conference caller-controls > (Michael Collins) > 3. Re: *** No rule to make target `tport/libtport.la', needed by > `libsofia-sip-ua.la'. (Ken Rice) > > > ---------- Forwarded message ---------- > From: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Fri, 2 Mar 2012 15:49:10 -0800 > Subject: Re: [Freeswitch-users] Use of ESL and PHP for Call > Shahzad, > > I think you may have taken an unnecessarily complicated approach to this. > If I were you I would simply use the play_and_get_digits application which > does all the work for you. I would also handle that in the XML dialplan and > then use the api_hangup_hook to call your script to process the result. > (I.e. did the user press 1 or 2 or something else...) > > I recommend you start with something simple like a basic originate command > that calls the user and then drops that call to a dialplan extension: > > originate user/1500 meeting_request_1500 > > Then create that extension: > > > > > > > > data="1 1 3 5000 # speak:${text} > ivr/ivr-that_was_an_invalid_entry.wav answer \d"/> > > > > > > > > Then you just need to write your handler.php script (or whatever you want > to call it) and have it parse the command line args - the first one is the > user and the second one is the digit pressed. PHP scripts will work for > hangup hooks but they aren't the best choice since there isn't a "mod_php" > like there is a mod_lua or mod_perl. Those modules let you do cool things > like "session_in_hangup_hook." However, what you're doing doesn't look like > it's very difficult so the example I've given should get you going. Please > note that I just did this off the top of my head w/o testing it so be sure > double-check everything if you run into any unusual or unexpected behavior. > > Keep in mind that in this example you will need two different PHP scripts > - one that actually initiates the ESL connection and does the originate and > one that handles the call results. Instead of one semi-complicated script > you will have two very easy scripts. > > Hope this helps... > -MC > > On Fri, Mar 2, 2012 at 4:14 AM, Shahzad Bhatti wrote: > >> >> Hi, >> >> i park the call and then execute the speak command as but execute is not >> working any idea ? >> >> $cmd = "api originate {origination_uuid=$uuid}user/$user &park()"; >> $text = "Hi, we have a meeting on Monday at 9, are you joining us, if >> yes then press, 1. and if not press, 2."; >> >> $e = $esl->sendRecv($cmd); >> $esl->execute('speak',"(cepstral|callie|$text)",$uuid); >> >> >> >> my complete code is >> >> > >> require_once('ESL.php'); >> >> $host_name = 'localhost'; >> $port = '8021'; >> $password = 'ClueCon'; >> >> $esl = new ESLconnection($host_name,$port,$password); >> >> $e = $esl->sendRecv("api create_uuid"); >> $uuid = $e->getBody(); >> $user = '1500'; >> >> $cmd = "bgapi originate {origination_uuid=$uuid}user/$user &park()"; >> $text = "Hi, we have a meeting on Monday at 9, are you joining us, if >> yes then press, 1. and if not press, 2."; >> >> $e = $esl->sendRecv($cmd); >> $esl->execute('speak',"(cepstral|callie|$text)",$uuid); >> >> >> $e = $esl->sendRecv("events plain all"); >> $e = $esl->filter("Unique-ID",$uuid); >> >> while ($esl->connected()) >> { >> $e = $esl->recvEvent(); >> $result = $e->getType(); >> if($result=='CHANNEL_EXECUTE_COMPLETE') >> { >> // what i do here...! >> $esl->execute("&sleep",10000); >> } >> $digit = $e->getHeader("DTMF-Digit"); >> $state_no = $e->getHeader("Channel-State-Number"); >> $curr_uuid = $e->getHeader("Channel-Call-UUID"); >> >> /* >> $ans_state = $e->getHeader("Answer-State"); >> $state_no = $e->getHeader("Channel-State-Number"); >> $core_id = $e->getHeader("Core-UUID"); >> */ >> if($curr_uuid==$uuid && $digit!=NULL) >> { >> print "input number is " . $digit . "\n"; >> } >> $state = $e->getHeader("Channel-Call-State"); >> if ($state == 'HANGUP') { >> print "\n\nHangup Cause Number: " . $state_no; >> $esl->disconnect(); >> } >> } >> >> ?> >> >> >> Regards >> >> Shahzad Bhatti >> >> >> ---------- Forwarded message ---------- >> From: Peter Olsson >> To: FreeSWITCH Users Help >> Cc: >> Date: Fri, 2 Mar 2012 05:31:35 +0000 >> Subject: Re: [Freeswitch-users] Use of ESL and PHP for Call >> You should pass the originate command to &park() instead. >> >> Right now you originate the call and then execute the application speak. >> After this (when speak is finished) FS has no more instructions so it will >> hangup the call. >> >> If you execute &park() instead, the call will be "parked" and wait for >> furter instructions. Then you will also need to execute the speak from ESL >> instead - and afther that you continue to do whatever you want to do with >> the call. >> >> /Peter >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > ---------- Forwarded message ---------- > From: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Fri, 2 Mar 2012 16:01:44 -0800 > Subject: Re: [Freeswitch-users] a question about freeswitch conference > caller-controls > Try this in your dialplan before sending the call to the conference: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf > -MC > > On Thu, Mar 1, 2012 at 5:30 PM, Erjian Li wrote: > >> When I press cellphone's key, although Freeswitch can't see this DTMF, >> but the the other participant of the conference call can hear my DTMF tone. >> Does this situation indicates that the DTMF tone has been forwarded by SIP >> provider's server? >> >> > > ---------- Forwarded message ---------- > From: Ken Rice > To: FreeSWITCH Users Help > Cc: > Date: Fri, 02 Mar 2012 18:02:02 -0600 > Subject: Re: [Freeswitch-users] *** No rule to make target `tport/ > libtport.la', needed by `libsofia-sip-ua.la'. > This is because Apple Deprecated SSL on Lion... They don?t want you > using it... > > Configure ?--without-openssl? and the problem goes away... > > There is an open jira on this issue I just cant recall the bug number off > the top of my head > > > On 3/2/12 4:57 PM, "St?phane Lux" wrote: > > I have tried to compile it again: > > > tport_tls.c: In function 'tls_error': > tport_tls.c:665: warning: 'SSL_get_error' is deprecated (declared at > /usr/include/openssl/ssl.h:1501) > tport_tls.c:680: warning: 'SSL_get_shutdown' is deprecated (declared at > /usr/include/openssl/ssl.h:1568) > tport_tls.c: In function 'tls_read': > tport_tls.c:723: warning: 'SSL_read' is deprecated (declared at > /usr/include/openssl/ssl.h:1493) > tport_tls.c: In function 'tls_pending': > tport_tls.c:739: warning: 'SSL_pending' is deprecated (declared at > /usr/include/openssl/ssl.h:1368) > tport_tls.c: In function 'tls_write': > tport_tls.c:807: warning: 'SSL_write' is deprecated (declared at > /usr/include/openssl/ssl.h:1495) > tport_tls.c: In function 'tls_connect': > tport_tls.c:899: warning: 'SSL_accept' is deprecated (declared at > /usr/include/openssl/ssl.h:1491) > tport_tls.c:899: warning: 'SSL_connect' is deprecated (declared at > /usr/include/openssl/ssl.h:1492) > tport_tls.c:900: warning: 'SSL_get_error' is deprecated (declared at > /usr/include/openssl/ssl.h:1501) > tport_tls.c:954: warning: 'ERR_error_string_n' is deprecated (declared at > /usr/include/openssl/err.h:280) > make[9]: *** [tport_tls.lo] Error 1 > make[8]: *** [all] Error 2 > Making all in nta > Making all in nth > Making all in nea > Making all in iptsec > Making all in nua > make[8]: *** No rule to make target `tport/libtport.la', needed by ` > libsofia-sip-ua.la'. Stop. > make[7]: *** [all-recursive] Error 1 > Making all in packages > make[6]: *** [all-recursive] Error 1 > make[5]: *** [all] Error 2 > make[4]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/ > libsofia-sip-ua.la] Error 2 > make[3]: *** [mod_sofia-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > > On 02.03.2012, at 23:42, Michael Collins wrote: > > We need to see the output that comes prior to this error. Grab the > previous 20+ lines of output prior to this error and let us take a look. > Hopefully the true cause will be apparent. > > -MC > > 2012/3/2 St?phane Lux > > Hi, > > when compiling FS on Mac OS 10.7.3 I get this error: > > make[8]: *** No rule to make target `tport/libtport.la < > http://libtport.la/> ', needed by `libsofia-sip-ua.la < > http://libsofia-sip-ua.la/> '. Stop. > > How can I solve it? > > Kind Regards, > Stephane > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/c667afd2/attachment-0001.html From miha at softnet.si Mon Mar 5 17:51:04 2012 From: miha at softnet.si (Miha Zoubek) Date: Mon, 05 Mar 2012 15:51:04 +0100 Subject: [Freeswitch-users] 302 redirect variable In-Reply-To: References: <4F509C57.20205@softnet.si> <9BCA494E297D4430B191119091E31C1A@freeswitch.org> <1F1CE9939D004E86A0679267391ABA92@freeswitch.org> <4F5472C8.2050004@softnet.si> Message-ID: <4F54D2D8.7070900@softnet.si> On 03/05/2012 09:33 AM, Gregor Nanger wrote: > Yes Miha! > > I think that is impossible to get password in raw format via > variables. If you have registration data somewhere else in database, > you can make sql query. > > You need to make, that on redirection your sbc is using IP > authentication, not username/pass. That way, you wouldn't need password. > > BR Grega > > > 2012/3/5 Miha Zoubek > > > Hi @Jo?o, > > first thank you for your quick response. > > I have add ${sip_authorized} condition, but still no luck, as when > 302 happens FS indicate user as inbound call who is calling on > redirected number due to this I can not use password which is set > for phone who is having redirect set and is connected on FS. SO, I > can only use variables which are set for incoming call. > > This is my public dialplan: > > > > > > > break="never"> > > > > > > > > > > > > > > > data="domain_name=fs_kabelvoip1.fs1.softnet.si > "/> > data="domain=fs_kabelvoip1.fs1.softnet.si > "/> > > > > > > > data="sofia/internal/${destination_number}.fs_kabelvoip1%fs_kabelvoip1.fs1.softnet.si > "/> > > > > > > > > > > data="{origination_callee_id_name='${sip_req_user}'}IZVEDI_PREUSMERITEV > XML default"/> > > > > > > > > > I have also add log to pastebin: http://pastebin.freeswitch.org/18575 > > Thanks! > Miha > > > On 03/04/2012 09:31 PM, Jo?o Mesquita wrote: >> Yes >> >> -- >> Jo?o Mesquita >> Sent with Sparrow >> >> On Sunday, March 4, 2012 at 2:17 PM, Miha wrote: >> >>> Thank you for that! >>> >>> After that I will be able to use variables from user/dir? >>> >>> Regards, >>> Miha >>> >>> On Sun, 4 Mar 2012 01:14:13 -0300 >>> Jo?o Mesquita >> > wrote: >>>> You won't get the user's password in plain text like that >>>> EVER. If we did that, we would be considered to be >>>> insanely insecure. >>>> >>>> I am guessing you are using SIP only so you can take a >>>> look at the dialplan/public.xml file of the default >>>> configs. On the end of that file you will see that there >>>> is a verification to do dial plan based authentication. >>>> >>>> Look for this extension in particular: >>>> >>>> >>>> >>> expression="^true$" break="never"> >>>> >>>> >>>> >>>> >>>> >>>> >>> data="${destination_number} XML default"/> >>>> >>>> >>>> >>>> >>>> Regards, >>>> >>>> -- >>>> Jo?o Mesquita >>>> Sent with Sparrow (http://www.sparrowmailapp.com/?sig) >>>> >>>> >>>> On Friday, March 2, 2012 at 7:09 AM, Miha Zoubek wrote: >>>> >>>>> Hi, >>>>> in my directory I set variable password (>>> name="password" value="52166"/>) for every user. >>>>> After I am doing 302 redirect in my public dialplan and >>>> transfer call to extension, I can not use varible >>>> pasword. >>>>> How can I get varible password, so that I can >>>> authenticate call. >>>>> public dialplan: >>>>> >>>>> >>>>> >>>>> >>>>> >>> data="process_cdr=false"/> >>>>> >>> data="IZVEDI_PREUSMERITEV XML default"/> >>>>> default dialplan: >>>>> >>> expression="IZVEDI_PREUSMERITEV" /> >>>>> >>> expression="^0(\d+)$" > >>>>> >>>>> >>> data="CALLINGNUMBER=${sip_redirect_contact_user_0}"/> >>>>> >>> data="USERNAME=${sip_req_user}"/> >>>>> >>>>> >>> data="effective_caller_id_name=${sip_req_user}"/> >>>>> >>> data="origination_caller_id_name=${sip_req_user} "/> >>>>> >>> data="PASSWD=${password}"/> >>>>> >>> data="RADIUS_ANI_AUTH XML default"/> >>>>> >>> data="386${sip_redirect_contact_user_0:1} >>>> enumsbc.softnet.si >>>> (http://enumsbc.softnet.si)"/> >>>>> >> data="{origination_callee_id_name='${effective_caller_id_name}'}${enum_auto_route}"/> >>>>> >>>>> >> data="sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx >>> " >>> (mailto:sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx >>> ) >>>> /> >>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>> (mailto:consulting at freeswitch.org) >>>>> http://www.freeswitchsolutions.com >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>> Server >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>>> (mailto:FreeSWITCH-users at lists.freeswitch.org) >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Hi, so you say that I can not authenticate call via radius when is goes for 302 redirect? Regards, Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/e52ec49d/attachment-0001.html From jmesquita at freeswitch.org Mon Mar 5 18:17:43 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Mon, 5 Mar 2012 12:17:43 -0300 Subject: [Freeswitch-users] 302 redirect variable In-Reply-To: <4F54D2D8.7070900@softnet.si> References: <4F509C57.20205@softnet.si> <9BCA494E297D4430B191119091E31C1A@freeswitch.org> <1F1CE9939D004E86A0679267391ABA92@freeswitch.org> <4F5472C8.2050004@softnet.si> <4F54D2D8.7070900@softnet.si> Message-ID: <7546743BA27045A198FFEFA99DB7C821@freeswitch.org> Miha, I think that there is some misunderstanding on your part. Let me try to make that clear. You will NEVER see plain text passwords in FreeSWITCH. No matter what state the channel is on, you just won't. On the other hand, you client is being able to authenticate correctly. If you look at a SIP dialog, you will see that it responds to the 401 correctly somewhere, meaning that it has been authenticated. Re-reading your emails, now it has crossed my mind that you are looking to do per-call authentication on RADIUS and that authentication has nothing to do with SIP. In that case, you won't be able to use the SIP password on the directory either. Just set a new variable for that on the user directory and then you can use that. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Monday, March 5, 2012 at 11:51 AM, Miha Zoubek wrote: > On 03/05/2012 09:33 AM, Gregor Nanger wrote: > > Yes Miha! > > > > I think that is impossible to get password in raw format via variables. If you have registration data somewhere else in database, you can make sql query. > > > > You need to make, that on redirection your sbc is using IP authentication, not username/pass. That way, you wouldn't need password. > > > > BR Grega > > > > > > > > > > 2012/3/5 Miha Zoubek > > > Hi @Jo?o, > > > > > > first thank you for your quick response. > > > > > > I have add ${sip_authorized} condition, but still no luck, as when 302 happens FS indicate user as inbound call who is calling on redirected number due to this I can not use password which is set for phone who is having redirect set and is connected on FS. SO, I can only use variables which are set for incoming call. > > > > > > This is my public dialplan: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > I have also add log to pastebin: http://pastebin.freeswitch.org/18575 > > > > > > Thanks! > > > Miha > > > > > > > > > On 03/04/2012 09:31 PM, Jo?o Mesquita wrote: > > > > Yes > > > > > > > > -- > > > > Jo?o Mesquita > > > > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > > > > > > > > > > On Sunday, March 4, 2012 at 2:17 PM, Miha wrote: > > > > > > > > > Thank you for that! > > > > > > > > > > After that I will be able to use variables from user/dir? > > > > > > > > > > Regards, > > > > > Miha > > > > > > > > > > On Sun, 4 Mar 2012 01:14:13 -0300 > > > > > Jo?o Mesquita wrote: > > > > > > You won't get the user's password in plain text like that > > > > > > EVER. If we did that, we would be considered to be > > > > > > insanely insecure. > > > > > > > > > > > > I am guessing you are using SIP only so you can take a > > > > > > look at the dialplan/public.xml file of the default > > > > > > configs. On the end of that file you will see that there > > > > > > is a verification to do dial plan based authentication. > > > > > > > > > > > > Look for this extension in particular: > > > > > > > > > > > > > > > > > > > > > > > expression="^true$" break="never"> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="${destination_number} XML default"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Regards, > > > > > > > > > > > > -- > > > > > > Jo?o Mesquita > > > > > > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > > > > > > > > > > > > > > > > On Friday, March 2, 2012 at 7:09 AM, Miha Zoubek wrote: > > > > > > > > > > > > > Hi, > > > > > > > > > > > > > > in my directory I set variable password ( > > > > > > > > > > > > > > > > > > > > > > > > > name="password" value="52166"/>) for every user. > > > > > > > > > > > > > > After I am doing 302 redirect in my public dialplan and > > > > > > > > > > > > > > > > > > > > > > > > > > transfer call to extension, I can not use varible > > > > > > pasword. > > > > > > > How can I get varible password, so that I can > > > > > > > > > > > > > > > > > > > authenticate call. > > > > > > > > > > > > > > public dialplan: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="process_cdr=false"/> > > > > > > > > > > > > > > > > > > > > > > > > > data="IZVEDI_PREUSMERITEV XML default"/> > > > > > > > > > > > > > > default dialplan: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > expression="IZVEDI_PREUSMERITEV" /> > > > > > > > > > > > > > > > > > > > > > > > > > expression="^0(\d+)$" > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="CALLINGNUMBER=${sip_redirect_contact_user_0}"/> > > > > > > > > > > > > > > > > > > > > > > > > > data="USERNAME=${sip_req_user}"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="effective_caller_id_name=${sip_req_user}"/> > > > > > > > > > > > > > > > > > > > > > > > > > data="origination_caller_id_name=${sip_req_user} "/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="PASSWD=${password}"/> > > > > > > > > > > > > > > > > > > > > > > > > > data="RADIUS_ANI_AUTH XML default"/> > > > > > > > > > > > > > > > > > > > > > > > > > data="386${sip_redirect_contact_user_0:1} > > > > > > enumsbc.softnet.si (http://enumsbc.softnet.si) (http://enumsbc.softnet.si)"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="{origination_callee_id_name='${effective_caller_id_name}'}${enum_auto_route}"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx (http://xxx.xxx.xxx.xxx)" > > > > > (mailto:sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx (mailto:sofia/external/386$%7Bsip_redirect_contact_user_0:1%7D at xxx.xxx.xxx.xxx)) > > > > > > /> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > > > > > > > > > > > > > > > > > > > > > > > > (mailto:consulting at freeswitch.org) > > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > > > > > > > > > > > > > > > > > > > > > > > > > > Server > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > > http://www.freeswitch.org > > > > > > > http://wiki.freeswitch.org > > > > > > > http://www.cluecon.com > > > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > > > > > > > > > > > > > > > > > > > > > > (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > > > > > > > > > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > Professional FreeSWITCH Consulting Services: > > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > http://www.freeswitch.org > > > > > http://wiki.freeswitch.org > > > > > http://www.cluecon.com > > > > > > > > > > FreeSWITCH-users mailing list > > > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org (mailto:consulting at freeswitch.org) http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org (mailto:consulting at freeswitch.org) http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Hi, > > so you say that I can not authenticate call via radius when is goes for 302 redirect? > > Regards, > Miha > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/5bcaaad9/attachment-0001.html From bob.mccarthy at experient.com Mon Mar 5 19:00:00 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Mon, 5 Mar 2012 09:00:00 -0700 Subject: [Freeswitch-users] send_dtmf fails on sla barge in In-Reply-To: <1330590312.30063.11.camel@CO-999-8> References: <1330590312.30063.11.camel@CO-999-8> Message-ID: <002301ccfae9$06657c60$13307520$@mccarthy@experient.com> I fixed it by putting into SLA profile in conference.conf.xml (I had it on in the default) ? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob McCarthy Sent: Thursday, March 01, 2012 1:25 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] send_dtmf fails on sla barge in dialplan default.xml has prior to the bridge internal.xml has no dtmf statements uncommented when sending dtmf from the Polycom dialpad after a barge in using SLA, (also after the barging phone hangs up) I get the following error: 2012-03-01 03:05:53.841090 [ERR] switch_ivr_async.c:547 dmachine overflow error! FreeSWITCH Version 1.0.head (git-23645b6 2012-02-27 16-49-12 -0600) Bob McCarthy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/5405863e/attachment.html From b2m at a-cti.com Mon Mar 5 19:14:28 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Mon, 5 Mar 2012 21:44:28 +0530 Subject: [Freeswitch-users] Voicemail Transcription Message-ID: Hi all, We are looking for a tool to transcribe voicemail, I was trying with Sphinx but the quality is not okay to use. Please suggest me the right tool for Voicemail Transcription. Thanks, Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/da21fe1d/attachment.html From anita.hall at simmortel.com Mon Mar 5 19:15:08 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Mon, 5 Mar 2012 21:45:08 +0530 Subject: [Freeswitch-users] spandsp error reading frame In-Reply-To: <4f53a4d6.a526340a.306b.ffff9ea1@mx.google.com> References: <4f53a4d6.a526340a.306b.ffff9ea1@mx.google.com> Message-ID: How does your spandsp.conf.xml look like? regards, Anita On Sun, Mar 4, 2012 at 10:52 PM, Jean-Marc Hyppolite < jean.marc.hyppolite at gmail.com> wrote: > Hello,**** > > ** ** > > I am trying to use mod_spandsp to detect call progress. I haven t been > successful so far. I am getting the following error messages.**** > > ** ** > > ================================================**** > > 2012-03-03 23:55:08.398205 [DEBUG] mod_spandsp_dsp.c:379 > (sofia/outbound/?@...) Starting tone detection for '1'**** > > 2012-03-03 23:55:08.398205 [INFO] mod_spandsp_dsp.c:411 > (sofia/outbound/?@...) initializing tone detector**** > > 2012-03-03 23:55:08.398205 [DEBUG] switch_core_media_bug.c:462 Attaching > BUG to sofia/outbound/?@...**** > > 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:422 > (sofia/outbound/?@...) error reading frame**** > > 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:447 > (sofia/outbound/?@...) destroying tone detector**** > > ================================================**** > > ** ** > > Any help would be appreciated. **** > > ** ** > > Thanks**** > > ** ** > > Jean-Marc.**** > > ** ** > > N.B. Sorry if some people received this e-mail twice. I am not sure this > email went through the first time.**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/1ff24d35/attachment.html From avi at avimarcus.net Mon Mar 5 19:18:50 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 5 Mar 2012 18:18:50 +0200 Subject: [Freeswitch-users] Voicemail Transcription In-Reply-To: References: Message-ID: Here are some service APIs - one free and several paid options, of varying qualities: http://wiki.freeswitch.org/wiki/Transcribing_Voicemail There's other options for on-system transcription but if you want FOSS.. there nothing ready to install that I found. -Avi On Mon, Mar 5, 2012 at 6:14 PM, Balamurugan Mahendran wrote: > Hi all, > > We are looking for a tool to transcribe voicemail, I was trying with > Sphinx but the quality is not okay to use. Please suggest me the right tool > for Voicemail Transcription. > > Thanks, > Bala > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/341befb0/attachment.html From anita.hall at simmortel.com Mon Mar 5 19:20:46 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Mon, 5 Mar 2012 21:50:46 +0530 Subject: [Freeswitch-users] Voicemail Transcription In-Reply-To: References: Message-ID: Sphinx is the right tool provided that it has the right acoustic (AM) and language models (LM) for your task. Chances are high that you are using the default AM and LM which are not the right fit for you. Differences like frequency, telephony quality speech will play an important role in AM. While the words and grammar will affect the LM. You could look at HTK if your application is not commercial. regards, Anita On Mon, Mar 5, 2012 at 9:44 PM, Balamurugan Mahendran wrote: > Hi all, > > We are looking for a tool to transcribe voicemail, I was trying with > Sphinx but the quality is not okay to use. Please suggest me the right tool > for Voicemail Transcription. > > Thanks, > Bala > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/5d87c25b/attachment-0001.html From peter.olsson at visionutveckling.se Mon Mar 5 19:22:20 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 5 Mar 2012 16:22:20 +0000 Subject: [Freeswitch-users] Voicemail Transcription Message-ID: <1FFF97C269757C458224B7C895F35F1504D8C5@cantor.std.visionutv.se> I think Nuance (www.nuance.com) is the only serious option for this - it's quite expensive though. Even though I don't think that any solution really work that good... /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Balamurugan Mahendran Skickat: den 5 mars 2012 17:14 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Voicemail Transcription Hi all, We are looking for a tool to transcribe voicemail, I was trying with Sphinx but the quality is not okay to use. Please suggest me the right tool for Voicemail Transcription. Thanks, Bala !DSPAM:4f54e52332761375616741! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/ea5ac5b0/attachment.html From anita.hall at simmortel.com Mon Mar 5 19:31:31 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Mon, 5 Mar 2012 22:01:31 +0530 Subject: [Freeswitch-users] Sangoma Card Originate Not Working on Forwarded Call Numbers Message-ID: Hi There are some numbers on which I am not able to originate calls from a Sangoma Card, while I am able to do the same from a Digium like card. I am using FreeTDM. >originate freetdm/4/a/07305880672 &park() Log is here http://pastebin.freeswitch.org/18578 Note that I am able to call other numbers from this system and I have a working PRI line and FreeTDM. Also the number 07305880672 is working fine. It unconditionally forwards the call to another number. I am problem with calling all such numbers. I am able to do q931 traces using sangoma_isdn but how to view them ? regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/7facf1dd/attachment.html From miha at softnet.si Mon Mar 5 19:43:56 2012 From: miha at softnet.si (Miha) Date: Mon, 05 Mar 2012 17:43:56 +0100 Subject: [Freeswitch-users] 302 redirect variable In-Reply-To: <7546743BA27045A198FFEFA99DB7C821@freeswitch.org> References: <4F509C57.20205@softnet.si> <9BCA494E297D4430B191119091E31C1A@freeswitch.org> <1F1CE9939D004E86A0679267391ABA92@freeswitch.org> <4F5472C8.2050004@softnet.si> <4F54D2D8.7070900@softnet.si> <7546743BA27045A198FFEFA99DB7C821@freeswitch.org> Message-ID: Hi @Jo?o, I think that we are having few problems to understand each other due to my english:) OK, I hope I will make it clear this time:) In user dir, I am having set like this: for each user. I can use this variable in my dialplan like this: Problem is when 302 redirect from phone appear (manual redirect on all calls). A calls B, B has set redirect to C. FS interpretate that A in calling B, so I can not use variables which are set for B in FS. I hope I make in clear:) Thanks! MIha On Mon, 5 Mar 2012 12:17:43 -0300 Jo?o Mesquita wrote: > Miha, I think that there is some misunderstanding on your > part. Let me try to make that clear. > > You will NEVER see plain text passwords in FreeSWITCH. No > matter what state the channel is on, you just won't. > > On the other hand, you client is being able to > authenticate correctly. If you look at a SIP dialog, you > will see that it responds to the 401 correctly somewhere, > meaning that it has been authenticated. > > Re-reading your emails, now it has crossed my mind that > you are looking to do per-call authentication on RADIUS > and that authentication has nothing to do with SIP. In > that case, you won't be able to use the SIP password on > the directory either. Just set a new variable for that on > the user directory and then you can use that. > > Regards, > > -- > Jo?o Mesquita > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > On Monday, March 5, 2012 at 11:51 AM, Miha Zoubek wrote: > > > On 03/05/2012 09:33 AM, Gregor Nanger wrote: > > > Yes Miha! > > > > > > I think that is impossible to get password in raw > format via variables. If you have registration data > somewhere else in database, you can make sql query. > > > > > > You need to make, that on redirection your sbc is > using IP authentication, not username/pass. That way, you > wouldn't need password. > > > > > > BR Grega > > > > > > > > > > > > > > > 2012/3/5 Miha Zoubek (mailto:miha at softnet.si)> > > > > Hi @Jo?o, > > > > > > > > first thank you for your quick response. > > > > > > > > I have add ${sip_authorized} condition, but still > no luck, as when 302 happens FS indicate user as inbound > call who is calling on redirected number due to this I > can not use password which is set for phone who is having > redirect set and is connected on FS. SO, I can only use > variables which are set for incoming call. > > > > > > > > This is my public dialplan: > > > > > > > > > > > > > > > > > > > > > > > > > > > > expression="^true$" break="never"> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > expression="true"> > > > > > > > > > > > > > > > > data="domain_name=fs_kabelvoip1.fs1.softnet.si > (http://fs_kabelvoip1.fs1.softnet.si)"/> > > > > data="domain=fs_kabelvoip1.fs1.softnet.si > (http://fs_kabelvoip1.fs1.softnet.si)"/> > > > > data="destination_number"=$1"/> > > > > > > > > > > > > data="process_cdr=false"/> > > > > data="domain_name=$${domain}"/> > > > > > > > > data="sofia/internal/${destination_number}.fs_kabelvoip1%fs_kabelvoip1.fs1.softnet.si > (http://fs_kabelvoip1.fs1.softnet.si)"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="process_cdr=false"/> > > > > data="{origination_callee_id_name='${sip_req_user}'}IZVEDI_PREUSMERITEV > XML default"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > I have also add log to pastebin: > http://pastebin.freeswitch.org/18575 > > > > > > > > Thanks! > > > > Miha > > > > > > > > > > > > On 03/04/2012 09:31 PM, Jo?o Mesquita wrote: > > > > > Yes > > > > > > > > > > -- > > > > > Jo?o Mesquita > > > > > Sent with Sparrow > (http://www.sparrowmailapp.com/?sig) > > > > > > > > > > > > > > > On Sunday, March 4, 2012 at 2:17 PM, Miha wrote: > > > > > > > > > > > Thank you for that! > > > > > > > > > > > > After that I will be able to use variables from > user/dir? > > > > > > > > > > > > Regards, > > > > > > Miha > > > > > > > > > > > > On Sun, 4 Mar 2012 01:14:13 -0300 > > > > > > Jo?o Mesquita (mailto:jmesquita at freeswitch.org)> wrote: > > > > > > > You won't get the user's password in plain > text like that > > > > > > > EVER. If we did that, we would be considered > to be > > > > > > > insanely insecure. > > > > > > > > > > > > > > I am guessing you are using SIP only so you > can take a > > > > > > > look at the dialplan/public.xml file of the > default > > > > > > > configs. On the end of that file you will see > that there > > > > > > > is a verification to do dial plan based > authentication. > > > > > > > > > > > > > > Look for this extension in particular: > > > > > > > > > > > > > > > > > > > > > > > > > > > > expression="^true$" break="never"> > > > > > > > data="407"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="${destination_number} XML default"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Regards, > > > > > > > > > > > > > > -- > > > > > > > Jo?o Mesquita > > > > > > > Sent with Sparrow > (http://www.sparrowmailapp.com/?sig) > > > > > > > > > > > > > > > > > > > > > On Friday, March 2, 2012 at 7:09 AM, Miha > Zoubek wrote: > > > > > > > > > > > > > > > Hi, > > > > > > > > > > > > > > > > in my directory I set variable password > ( > > > > > > > > > > > > > > > > > > > > > > > > > > > > > name="password" value="52166"/>) for every > user. > > > > > > > > > > > > > > > > After I am doing 302 redirect in my public > dialplan and > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > transfer call to extension, I can not use > varible > > > > > > > pasword. > > > > > > > > How can I get varible password, so that I > can > > > > > > > > > > > > > > > > > > > > > > authenticate call. > > > > > > > > > > > > > > > > public dialplan: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="process_cdr=false"/> > > > > > > > > application="execute_extension" > > > > > > > > > > > > > > > > > > > > > > data="IZVEDI_PREUSMERITEV XML default"/> > > > > > > > > > > > > > > > > default dialplan: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > expression="IZVEDI_PREUSMERITEV" /> > > > > > > > > field="${sip_redirect_contact_user_0}" > > > > > > > > > > > > > > > > > > > > > > expression="^0(\d+)$" > > > > > > > > > data="process_cdr=true"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="CALLINGNUMBER=${sip_redirect_contact_user_0}"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="USERNAME=${sip_req_user}"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="effective_caller_id_name=${sip_req_user}"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="origination_caller_id_name=${sip_req_user} "/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="PASSWD=${password}"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="RADIUS_ANI_AUTH XML default"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="386${sip_redirect_contact_user_0:1} > > > > > > > enumsbc.softnet.si > (http://enumsbc.softnet.si) > (http://enumsbc.softnet.si)"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="{origination_callee_id_name='${effective_caller_id_name}'}${enum_auto_route}"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx > (http://xxx.xxx.xxx.xxx)" > > > > > > > (mailto:sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx > (mailto:sofia/external/386$%7Bsip_redirect_contact_user_0:1%7D at xxx.xxx.xxx.xxx)) > > > > > > > /> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > > > Professional FreeSWITCH Consulting > Services: > > > > > > > > consulting at freeswitch.org > (mailto:consulting at freeswitch.org) > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > (mailto:consulting at freeswitch.org) > > > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > FreeSWITCH-powered IP PBX: The CudaTel > Communication > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Server > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > > > http://www.freeswitch.org > > > > > > > > http://wiki.freeswitch.org > > > > > > > > http://www.cluecon.com > > > > > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > > > FreeSWITCH-users at lists.freeswitch.org > (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > > > > > > > > > > > > > > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > consulting at freeswitch.org > (mailto:consulting at freeswitch.org) > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > FreeSWITCH-powered IP PBX: The CudaTel > Communication Server > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > http://www.freeswitch.org > > > > > > http://wiki.freeswitch.org > > > > > > http://www.cluecon.com > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > FreeSWITCH-users at lists.freeswitch.org > (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com FreeSWITCH-powered IP > PBX: The CudaTel Communication Server > Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > (mailto:consulting at freeswitch.org) > > > > http://www.freeswitchsolutions.com > > > > > > > > FreeSWITCH-powered IP PBX: The CudaTel > Communication Server > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com FreeSWITCH-powered IP > PBX: The CudaTel Communication Server > Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Hi, > > > > so you say that I can not authenticate call via radius > when is goes for 302 redirect? > > > > Regards, > > Miha > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > From basit.engg at gmail.com Mon Mar 5 20:26:51 2012 From: basit.engg at gmail.com (Abdul Basit) Date: Mon, 5 Mar 2012 22:26:51 +0500 Subject: [Freeswitch-users] ph_tor3_e1.c In-Reply-To: References: Message-ID: You can use sangoma/digium/OpenVOX cards. These are good, tested and has up-to-date drivers. Also try updating you atom card firmware. This may help. Ask Atom. -- regards, Abdul Basit On Fri, Mar 2, 2012 at 10:06 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Try using the old zaptel drivers circa asterisk 1.2 > Nothing has really changed in that card since then and it would work > with FreeTDM an the zt io module. > > > On Fri, Mar 2, 2012 at 12:57 AM, Anita Hall > wrote: > > We are running FreeTDM on a very cheap Atcom card, which used another > > module ph_tor3_e1 on top of Dahdi. I believe this is derived from > Torrenta. > > > http://www.atcom.cn/downloads/TelephonyCard/drivers/AX-4ET/E1/ph_tor3_e1.c > > > > On Ubuntu 10.04 this gives problem as the module ph_tor3_e1 (and hence > > dahdi) does not unload. Sometimes the machine hangs and needs to be > > rebooted. > > > > This module has not been updated for the last 2 years during which the > linux > > kernel has changed (I am told). > > > > Is there any other manufacturer of torrent card who would be using the > same > > architecture and keeping his drivers updated ? > > > > If not, what steps do I need to take to update this driver to kernel > version > > 2.6.32-37-server ? > > > > These are new waters and I feel so helpless :) > > > > regards, > > Anita > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/7a8349a2/attachment-0001.html From freeswitch at earthspike.net Mon Mar 5 21:27:15 2012 From: freeswitch at earthspike.net (John) Date: Mon, 05 Mar 2012 18:27:15 +0000 Subject: [Freeswitch-users] Sangoma Card Originate Not Working on Forwarded Call Numbers In-Reply-To: References: Message-ID: <4F550583.4060105@earthspike.net> On 05/03/12 16:31, Anita Hall wrote: > Hi > > There are some numbers on which I am not able to originate calls from > a Sangoma Card, while I am able to do the same from a Digium like > card. I am using FreeTDM. > > >originate freetdm/4/a/07305880672 &park() > > Log is here http://pastebin.freeswitch.org/18578 > > Note that I am able to call other numbers from this system and I have > a working PRI line and FreeTDM. > > Also the number 07305880672 is working fine. It unconditionally > forwards the call to another number. I am problem with calling all > such numbers. > > I am able to do q931 traces using sangoma_isdn but how to view them ? > > regards, > Anita > Anita, You can use wireshark to decode the q.931 traces; the output files are .pcap format. John From anita.hall at simmortel.com Mon Mar 5 21:55:09 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Tue, 6 Mar 2012 00:25:09 +0530 Subject: [Freeswitch-users] Sangoma Card Originate Not Working on Forwarded Call Numbers In-Reply-To: <4F550583.4060105@earthspike.net> References: <4F550583.4060105@earthspike.net> Message-ID: Thanks John. I could take a pcap trace using wanpipemon -i w4g1 -pcap -pcap_file sng_isdn_fwd.pc -prot ISDN -full -systime -c trd The q931 for a call which is not maturing is as follows. This is the case when the called number is set to unconditional forwarding. 3 6.057463 Local User Remote Network Q.931 SETUP 5 6.142822 Remote Network Local User Q.931 CALL PROCEEDING 7 Remote Network Local User Q.931 DISCONNECT 9 Local User Remote Network Q.931 RELEASE 11 Remote Network Local User Q.931 RELEASE COMPLETE This happens only with Sangoma Card. I am using the sangoma_isdn stack and FreeTDM. Somehow, it is not able to understand that the call is getting forwarded. The q931 for a call which is maturing is as follows 11 Local User Remote Network Q.931 SETUP 13 Remote Network Local User Q.931 CALL PROCEEDING 15 Remote Network Local User Q.931 ALERTING 17 Remote Network Local User Q.931 CONNECT 23 Remote Network Local User Q.931 DISCONNECT 25 Local User Remote Network Q.931 RELEASE 27 Remote Network Local User Q.931 RELEASE COMPLETE regards, Anita On Mon, Mar 5, 2012 at 11:57 PM, John wrote: > On 05/03/12 16:31, Anita Hall wrote: > > Hi > > > > There are some numbers on which I am not able to originate calls from > > a Sangoma Card, while I am able to do the same from a Digium like > > card. I am using FreeTDM. > > > > >originate freetdm/4/a/07305880672 &park() > > > > Log is here http://pastebin.freeswitch.org/18578 > > > > Note that I am able to call other numbers from this system and I have > > a working PRI line and FreeTDM. > > > > Also the number 07305880672 is working fine. It unconditionally > > forwards the call to another number. I am problem with calling all > > such numbers. > > > > I am able to do q931 traces using sangoma_isdn but how to view them ? > > > > regards, > > Anita > > > Anita, > > You can use wireshark to decode the q.931 traces; the output files are > .pcap format. > > John > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/fc64b549/attachment.html From msc at freeswitch.org Mon Mar 5 22:42:16 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 5 Mar 2012 11:42:16 -0800 Subject: [Freeswitch-users] Send PAI and RPID In-Reply-To: References: Message-ID: FYI, A quick glance through the source suggests that PAI and RPID are mutually exclusive - i.e. you can set one or the other but not both. I'll defer to the experts on whether or not the SIP spec says you SHOULD or SHOULD NOT have both headers in a single message. -MC On Thu, Mar 1, 2012 at 4:20 AM, Anto wrote: > Hello > > To my FreeSWITCH servers, they come PAI and RPID headers, sent to the > carrier but only one (the one I have configured with name="caller-id-type" value="pid"/>). Is there any way to send both > headers?. Thanks > > Best regards > Anto > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/34394d2d/attachment.html From msc at freeswitch.org Mon Mar 5 22:50:47 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 5 Mar 2012 11:50:47 -0800 Subject: [Freeswitch-users] New batch of sound files In-Reply-To: References: Message-ID: Ask and ye shall receive! The files have been out on files.freeswitch.org for the past week or so. Look for version 1.0.18. Also, in my previous email on the subject I mentioned this trick for forcing your system to get the latest sounds: edit ${fs_src}/build/sounds_version.txt change callie to "1.0.18" save & exit "make cd-sounds-install" New sounds will be downloaded. As far as the prompts you mentioned, I added several prompts that we can use to piece together some phrase files or whatnot. I've got two different wordings depending on your preference: There is one other person in this conference There is one other member in this conference There are... ...other members in this conference ...other persons in this conference Now you have the pieces you need to build a nice little phrase. -MC On Sat, Mar 3, 2012 at 3:23 PM, Brian Foster wrote: > Hola, > > I briefly heard in channel and on conference about a new batch of sound > files for freeswitch. Is there a download link out there for this? Also, is > there any notable prompts for conferences? I'm specifically looking for > some prompts that would take care of telling the incoming participant how > many other participants are in the conference. > > Thanks! > -BDF > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/dbbb043c/attachment.html From krice at freeswitch.org Mon Mar 5 22:59:46 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 05 Mar 2012 13:59:46 -0600 Subject: [Freeswitch-users] Send PAI and RPID In-Reply-To: Message-ID: You can send both... They arent really mutually exclusive... Just some providers whine about it... Its really redundant... Also RPID was never a RFC, it was a draft that has stuck around and refuses to die... PAI is the only ratified RFC standard On 3/5/12 1:42 PM, "Michael Collins" wrote: > FYI, > > A quick glance through the source suggests that PAI and RPID are mutually > exclusive - i.e. you can set one or the other but not both. I'll defer to the > experts on whether or not the SIP spec says you SHOULD or SHOULD NOT have both > headers in a single message. > > -MC > > On Thu, Mar 1, 2012 at 4:20 AM, Anto wrote: >> Hello >> >> To my FreeSWITCH servers, they come PAI and RPID headers, sent to the >> carrier but only one (the one I have configured with > name="caller-id-type" value="pid"/>). Is there any way to send both >> headers?. Thanks >> >> Best regards >> Anto >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/d2bad0a5/attachment-0001.html From wstephen80 at gmail.com Mon Mar 5 23:28:03 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Mon, 5 Mar 2012 21:28:03 +0100 Subject: [Freeswitch-users] Sangoma Card Originate Not Working on Forwarded Call Numbers In-Reply-To: References: <4F550583.4060105@earthspike.net> Message-ID: In your log I see an "UNALLOCATED_NUMBER" release cause. If the number is ok, this can be due to the ISDN numbering plan/number type of the called party. What are required by your isdn provider? You can try with different parameters (for your called number I think that you have to specify numbering plan=ISDN and number type=National). These parameter can be specified in profile or dialplan. http://wiki.sangoma.com/wanpipe-freeswitch-config-appendix#ton-npi-per-call Stephen On Mon, Mar 5, 2012 at 7:55 PM, Anita Hall wrote: > Thanks John. > > I could take a pcap trace using > wanpipemon -i w4g1 -pcap -pcap_file sng_isdn_fwd.pc -prot ISDN -full > -systime -c trd > > The q931 for a call which is not maturing is as follows. This is the case > when the called number is set to unconditional forwarding. > 3 6.057463 Local User Remote Network Q.931 SETUP > 5 6.142822 Remote Network Local User Q.931 CALL PROCEEDING > 7 Remote Network Local User Q.931 DISCONNECT > 9 Local User Remote Network Q.931 RELEASE > 11 Remote Network Local User Q.931 RELEASE COMPLETE > > This happens only with Sangoma Card. I am using the sangoma_isdn stack and > FreeTDM. Somehow, it is not able to understand that the call is getting > forwarded. > > The q931 for a call which is maturing is as follows > 11 Local User Remote Network Q.931 SETUP > 13 Remote Network Local User Q.931 CALL PROCEEDING > 15 Remote Network Local User Q.931 ALERTING > 17 Remote Network Local User Q.931 CONNECT > 23 Remote Network Local User Q.931 DISCONNECT > 25 Local User Remote Network Q.931 RELEASE > 27 Remote Network Local User Q.931 RELEASE COMPLETE > > > > regards, > Anita > > > > > On Mon, Mar 5, 2012 at 11:57 PM, John wrote: > >> On 05/03/12 16:31, Anita Hall wrote: >> > Hi >> > >> > There are some numbers on which I am not able to originate calls from >> > a Sangoma Card, while I am able to do the same from a Digium like >> > card. I am using FreeTDM. >> > >> > >originate freetdm/4/a/07305880672 &park() >> > >> > Log is here http://pastebin.freeswitch.org/18578 >> > >> > Note that I am able to call other numbers from this system and I have >> > a working PRI line and FreeTDM. >> > >> > Also the number 07305880672 is working fine. It unconditionally >> > forwards the call to another number. I am problem with calling all >> > such numbers. >> > >> > I am able to do q931 traces using sangoma_isdn but how to view them ? >> > >> > regards, >> > Anita >> > >> Anita, >> >> You can use wireshark to decode the q.931 traces; the output files are >> .pcap format. >> >> John >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/09359b9c/attachment.html From rsaavedra at ecogizmos.com Tue Mar 6 00:01:39 2012 From: rsaavedra at ecogizmos.com (rsaavedra at ecogizmos.com) Date: Mon, 5 Mar 2012 16:01:39 -0500 Subject: [Freeswitch-users] RTMP Codec Message-ID: <60705295ef231830817ddf006f70e6cc.squirrel@emailmg.globat.com> Hello everybody, Using the Flex client of mod_rtmp can I specify the codec to use? Thank you, Ricardo Saavedra From msc at freeswitch.org Tue Mar 6 00:18:44 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 5 Mar 2012 13:18:44 -0800 Subject: [Freeswitch-users] mod shell stream In-Reply-To: References: <5BA3C64DA53B4EAC9B1DA367F8FFD398@gmail.com> Message-ID: On Sun, Mar 4, 2012 at 4:30 AM, Bernard Fluixa wrote: > That is already done with the -t option (tuples only) > > Thanks anyway > > Bernard > > Does your script work if you do it in two steps, i.e. read from the database and drop into a file, then cat the file into sox? I'm just curious if something unexpected is happening on the command line when piping from psql to sox. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/34f3083e/attachment.html From krice at freeswitch.org Tue Mar 6 00:24:40 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 05 Mar 2012 15:24:40 -0600 Subject: [Freeswitch-users] RTMP Codec In-Reply-To: <60705295ef231830817ddf006f70e6cc.squirrel@emailmg.globat.com> Message-ID: It uses Speex by default on mod_rtmp that's your only real choice... The other choice is Nellymoser but that's not really a good choice (altho it is better then ADPCM) K On 3/5/12 3:01 PM, "rsaavedra at ecogizmos.com" wrote: > Hello everybody, > > Using the Flex client of mod_rtmp can I specify the codec to use? > > Thank you, > > Ricardo Saavedra > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rsaavedra at ecogizmos.com Tue Mar 6 00:53:00 2012 From: rsaavedra at ecogizmos.com (rsaavedra at ecogizmos.com) Date: Mon, 5 Mar 2012 16:53:00 -0500 Subject: [Freeswitch-users] RTMP Codec In-Reply-To: References: Message-ID: Thank you, Is better to use Speex in the extension that will receive the call? Ricardo Saavedra > It uses Speex by default on mod_rtmp that's your only real choice... The > other choice is Nellymoser but that's not really a good choice (altho it > is > better then ADPCM) > > K > > > On 3/5/12 3:01 PM, "rsaavedra at ecogizmos.com" > wrote: > >> Hello everybody, >> >> Using the Flex client of mod_rtmp can I specify the codec to use? >> >> Thank you, >> >> Ricardo Saavedra >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From krice at freeswitch.org Tue Mar 6 00:57:10 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 05 Mar 2012 15:57:10 -0600 Subject: [Freeswitch-users] RTMP Codec In-Reply-To: Message-ID: If they support it, however freeswitch can transcode speex to other codecs without issue. The only penalty is CPU costs... You'll want to test in your actual configuration to see what the loading on the CPU does tho. K On 3/5/12 3:53 PM, "rsaavedra at ecogizmos.com" wrote: > Thank you, > > Is better to use Speex in the extension that will receive the call? > > Ricardo Saavedra > >> It uses Speex by default on mod_rtmp that's your only real choice... The >> other choice is Nellymoser but that's not really a good choice (altho it >> is >> better then ADPCM) >> >> K >> >> >> On 3/5/12 3:01 PM, "rsaavedra at ecogizmos.com" >> wrote: >> >>> Hello everybody, >>> >>> Using the Flex client of mod_rtmp can I specify the codec to use? >>> >>> Thank you, >>> >>> Ricardo Saavedra >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fs-list at communicatefreely.net Tue Mar 6 01:36:05 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 05 Mar 2012 17:36:05 -0500 Subject: [Freeswitch-users] MWI does not working anymore? In-Reply-To: References: Message-ID: <4F553FD5.4010208@communicatefreely.net> I'm having a similar problem. Can someone explain the mechanics behind MWI? I did an upgrade on Friday, and lost most of my MWI functionality (although it fixed all the BLF issues we had). The version I was using was several months old, so this may not be new. We have two profiles - nonat and internal. nonat is purely private IP space, internal is public IP, with most clients behind NAT. Our curl directory scripts will return all domains whenever asked for a generic domain section with no tags. In the profiles, I have this little bit at the top: When I do a sofia status, I see that all our domains are aliased to the internal profile. I also have In the extension's directory, I have Until I upgraded, a phone could register on either profile, and expect a notify on both registration, and any time the message count changed. I read some of the earlier threads about domains only being mapped to one profile, so I changed the directory script to only return domains to one profile - the one they have more phones on. Now, if the phone registers on the mapped profile, changes produce a notify, but no notify comes on a registration. On the non-mapped profile, I sometimes get a notify when it registers, but never on a change. What changed and how should I set this up. I tried using subscriptions, but this is a problem as the phones we use can't subscribe to something other than what they register as, and that is a problem in our case. How do you set a system up so that a domain is accessible from more than one profile. Yehavi, what did you do in the alias section that you talked about? I don't expect someone to fix this for me, but I can't find enough information to fix it on my own. -Tim Anthony Minessale wrote: > alias allows you to map domains to your profile so the domain can be > used to find that profile in place of the profile name. > so in the case of mwi it looks up the domain name and the alias allows > it to resolve the domain to the correct profile. > > similar to dns only for sofia profiles. > > On Fri, Feb 17, 2012 at 8:02 AM, Yehavi Bourvine > wrote: > >> I played around with the ALIAS definitions and it seems to solve the >> problem. >> >> I've tried searching the WIKI for a description of ALIAS use and did not >> find. Anyone can explain what does the alias do and what is the correct >> usage when you have more than one profile? >> >> Thanks, __Yehavi: >> >> >> 2012/2/17 Yehavi Bourvine >> >>> Hello Anthony, >>> >>> I know, and I've put a comment there that it still does not work. I am >>> downloading the recent GIT every two days to my test system (the last one >>> was yesterday) and it still behaves the same. >>> >>> >>> Thanks, __Yehavi: >>> >>> 2012/2/17 Anthony Minessale >>> >>>> you opened a jira on it and i fixed it and committed it >>>> >>>> >>>> [FS-3866] MWI is not updated >>>> Reporter: Yehavi Bourvine [yehavi] >>>> Assignee: Anthony Minessale II [anthm] >>>> Status: Resolved >>>> http://jira.freeswitch.org/browse/FS-3866 >>>> >>>> commit ff379a97e57a94f11ce75e8894b1ceb79d10a9ed >>>> Author: Anthony Minessale >>>> Date: Fri Feb 10 11:02:41 2012 -0600 >>>> >>>> >>>> >>>> On Thu, Feb 16, 2012 at 2:56 AM, Yehavi Bourvine >>>> wrote: >>>> >>>>> Hello, >>>>> >>>>> I've noticed a problem started in the last two weeks, but I don't see >>>>> anyone else reporting it... >>>>> >>>>> MWI status is not updated anymore. It is only updated at the initial >>>>> boot of >>>>> the phone. Do others that use GIT from the last two weeks notice it? >>>>> >>>>> Thanks, __Yehavi: >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > From potxoka at gmail.com Tue Mar 6 01:43:12 2012 From: potxoka at gmail.com (Anto) Date: Mon, 5 Mar 2012 23:43:12 +0100 Subject: [Freeswitch-users] Route to the profile and not the gateway In-Reply-To: References: Message-ID: Hi Is solved, if someone else happens, look if you have commented erroneously dptools module. Thanks ! Best regards Anto 2012/3/1 Anto : > Hello, > > Setting up profiles and the other options, I can not make outgoing > calls. With the standard configuration and adding the extension in the > public folder, it worked specifying the gateway. Now removing it from > that folder and setting the context "internal" (in addition to setting > a context for provider), I can not make outgoing calls. I have > reviewed documentation, the configuration and have tried various forms > of make the bridge, but I can not make the call. Does anyone can guide > me that could look at? Thank you very much. > > Best regards > Anto > > sbc# cat outbound.xml > > > ? ? > ? ? ? > ? ? ? > ? ? ? ? > ? ? ? ? data="{sip_cid_type=pid,ignore_display_updates=true}sofia/gateway/provider1::server-1/$1"/> > ? ? ? ? > ? ? ? > ? ? > > > > freeswitch at sbc > sofia status > > ? ? ? ? ? ? ? ? ? ? Name ? ? ? ? ?Type > ?Data ? State > ================================================================================================= > ? ? ? ? ? ? ? ? ? internal ? ? ? ?profile > sip:mod_sofia at 192.168.20.5:5060 RUNNING (0) > ? ?internal::server-1 ? ? gateway > sip:user at 192.168.20.40 ?NOREG > ? ? ? ? ? ? ? ?provider1 ? ?profile > sip:mod_sofia at EXTERNAL_IP:5060 ?RUNNING (0) > ?provider1::server-1 ? ? ? gateway > sip:user at DOMAIN REGED > ================================================================================================= > 2 profiles 0 aliases > > > freeswitch at sbc > > 2012-03-01 17:56:32.401204 [NOTICE] switch_channel.c:915 New Channel > sofia/internal/anto at 192.168.20.40 > [3f191be1-c763-e111-bc8d-f46d041ca78e] > 2012-03-01 17:56:32.401204 [DEBUG] sofia.c:5283 Channel > sofia/internal/anto at 192.168.20.40 entering state [received][100] > 2012-03-01 17:56:32.401204 [DEBUG] sofia.c:5294 Remote SDP: > v=0 > o=anto 0 0 IN IP4 192.168.20.40 > s=- > c=IN IP4 192.168.20.40 > t=0 0 > m=audio 59700 RTP/AVP 9 96 97 98 100 0 8 102 3 103 5 6 101 > a=rtpmap:9 G722/8000 > a=rtpmap:96 SILK/24000 > a=rtpmap:97 SILK/16000 > a=rtpmap:98 speex/32000 > a=rtpmap:100 speex/16000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:102 iLBC/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:103 speex/8000 > a=rtpmap:5 DVI4/8000 > a=rtpmap:6 DVI4/16000 > a=rtpmap:101 telephone-event/8000 > a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level > a=zrtp-hash:1.10 > 366f09eb8571b63fc76ad7df2417b1b145774d1076a99042721e6f5f3f3401ca > m=video 60184 RTP/AVP 104 99 > a=rtpmap:104 H264/90000 > a=fmtp:104 profile-level-id=4DE01f;packetization-mode=1 > a=rtpmap:99 H264/90000 > a=fmtp:99 profile-level-id=4DE01f > a=recvonly > a=imageattr:104 send * recv [x=[0-1920],y=[0-1200]] > a=imageattr:99 send * recv [x=[0-1920],y=[0-1200]] > a=zrtp-hash:1.10 > 23f81f1a335dc819be01e9a794cb8e7a509019066dac01b8fbd89038aec44edb > > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/anto at 192.168.20.40) Running State Change CS_NEW > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:380 > (sofia/internal/anto at 192.168.20.40) State NEW > 2012-03-01 17:56:32.401204 [DEBUG] sofia.c:5480 > (sofia/internal/anto at 192.168.20.40) State Change CS_NEW -> CS_INIT > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/anto at 192.168.20.40 [BREAK] > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/anto at 192.168.20.40) Running State Change CS_INIT > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/anto at 192.168.20.40) State INIT > 2012-03-01 17:56:32.401204 [DEBUG] mod_sofia.c:85 > sofia/internal/anto at 192.168.20.40 SOFIA INIT > 2012-03-01 17:56:32.401204 [DEBUG] mod_sofia.c:125 > (sofia/internal/anto at 192.168.20.40) State Change CS_INIT -> CS_ROUTING > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/anto at 192.168.20.40 [BREAK] > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/anto at 192.168.20.40) State INIT going to sleep > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/anto at 192.168.20.40) Running State Change CS_ROUTING > 2012-03-01 17:56:32.401204 [DEBUG] switch_channel.c:1844 > (sofia/internal/anto at 192.168.20.40) Callstate Change DOWN -> RINGING > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/anto at 192.168.20.40) State ROUTING > 2012-03-01 17:56:32.401204 [DEBUG] mod_sofia.c:148 > sofia/internal/anto at 192.168.20.40 SOFIA ROUTING > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/anto at 192.168.20.40 Standard ROUTING > 2012-03-01 17:56:32.401204 [INFO] mod_dialplan_xml.c:336 Processing > +57xxxxxxxxx <+57xxxxxxxxx>->57xxxxxxxxx in context internal > Dialplan: sofia/internal/anto at 192.168.20.40 parsing > [internal->provider1] continue=false > Dialplan: sofia/internal/anto at 192.168.20.40 Regex (PASS) [provider1] > network_addr(192.168.20.40) =~ /^192.168.20.40$/ break=on-false > Dialplan: sofia/internal/anto at 192.168.20.40 Regex (PASS) [provider1] > destination_number(57xxxxxxxxx) =~ /^(\d+)$/ break=on-false > Dialplan: sofia/internal/anto at 192.168.20.40 Action > bridge({sip_cid_type=pid,ignore_display_updates=true}sofia/gateway/provider1::server-1/57xxxxxxxxx) > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/anto at 192.168.20.40) State Change CS_ROUTING -> > CS_EXECUTE > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/anto at 192.168.20.40 [BREAK] > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/anto at 192.168.20.40) State ROUTING going to sleep > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/anto at 192.168.20.40) Running State Change CS_EXECUTE > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/anto at 192.168.20.40) State EXECUTE > 2012-03-01 17:56:32.401204 [DEBUG] mod_sofia.c:241 > sofia/internal/anto at 192.168.20.40 SOFIA EXECUTE > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/anto at 192.168.20.40 Standard EXECUTE > 2012-03-01 17:56:32.401204 [ERR] switch_core_session.c:2096 Invalid > Application bridge > 2012-03-01 17:56:32.401204 [DEBUG] switch_channel.c:2804 > (sofia/internal/anto at 192.168.20.40) Callstate Change RINGING -> HANGUP > 2012-03-01 17:56:32.401204 [NOTICE] switch_core_session.c:2097 Hangup > sofia/internal/anto at 192.168.20.40 [CS_EXECUTE] > [DESTINATION_OUT_OF_ORDER] > 2012-03-01 17:56:32.401204 [DEBUG] switch_channel.c:2820 Send signal > sofia/internal/anto at 192.168.20.40 [KILL] > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/anto at 192.168.20.40 [BREAK] > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/anto at 192.168.20.40) State EXECUTE going to sleep > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/anto at 192.168.20.40) Running State Change CS_HANGUP > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/anto at 192.168.20.40) State HANGUP > 2012-03-01 17:56:32.401204 [DEBUG] mod_sofia.c:465 Channel > sofia/internal/anto at 192.168.20.40 hanging up, cause: > DESTINATION_OUT_OF_ORDER > 2012-03-01 17:56:32.401204 [DEBUG] mod_sofia.c:530 Responding to > INVITE with: 502 > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/anto at 192.168.20.40 Standard HANGUP, cause: > DESTINATION_OUT_OF_ORDER > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/anto at 192.168.20.40) State HANGUP going to sleep > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/anto at 192.168.20.40) State Change CS_HANGUP -> > CS_REPORTING > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/anto at 192.168.20.40 [BREAK] > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/anto at 192.168.20.40) Running State Change CS_REPORTING > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/anto at 192.168.20.40) State REPORTING > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/anto at 192.168.20.40 Standard REPORTING, cause: > DESTINATION_OUT_OF_ORDER > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/anto at 192.168.20.40) State REPORTING going to sleep > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/anto at 192.168.20.40) State Change CS_REPORTING -> > CS_DESTROY > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/anto at 192.168.20.40 [BREAK] > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_session.c:1377 Session > 2 (sofia/internal/anto at 192.168.20.40) Locked, Waiting on external > entities > 2012-03-01 17:56:32.401204 [NOTICE] switch_core_session.c:1395 Session > 2 (sofia/internal/anto at 192.168.20.40) Ended > 2012-03-01 17:56:32.401204 [NOTICE] switch_core_session.c:1397 Close > Channel sofia/internal/anto at 192.168.20.40 [CS_DESTROY] > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/anto at 192.168.20.40) Callstate Change HANGUP -> DOWN > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/anto at 192.168.20.40) Running State Change CS_DESTROY > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/anto at 192.168.20.40) State DESTROY > 2012-03-01 17:56:32.401204 [DEBUG] mod_sofia.c:370 > sofia/internal/anto at 192.168.20.40 SOFIA DESTROY > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/anto at 192.168.20.40 Standard DESTROY > 2012-03-01 17:56:32.401204 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/anto at 192.168.20.40) State DESTROY going to sleep > > 2012/3/1 Anto : >> Hello >> >> Once I download the XML from the server and see it with an editor, I >> have resolved many questions of how FreeSWITCH, but even I have >> several questions I have not found information (I spend all day >> reading the wiki). >> >> I am setting up multiple providers and use a format like this (do not >> know if it is the best): >> >> >> ? >> ? ? ? >> ? >> ......... >> >> Is there any way to set the extension when not having to configure all >> the gateway? (only use the profile). Sample dialplan: >> >> > data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/> >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ........... >> ???? >> >> When set incoming calls, not very well how, not whether to route >> directly to dialplan or create a profile for this. I have to be routed >> to multiple internal servers (load balancer, failover, etc), so the >> above scenario I could serve to cover this. Thanks !! >> >> Best regards >> Anto From potxoka at gmail.com Tue Mar 6 02:09:23 2012 From: potxoka at gmail.com (Anto) Date: Tue, 6 Mar 2012 00:09:23 +0100 Subject: [Freeswitch-users] Send PAI and RPID In-Reply-To: References: Message-ID: Hi Thank you very much. So asked to send both, as some providers accept PAI and other RPID :-S. Best regards Anto 2012/3/5 Ken Rice : > You can send both... They arent really mutually exclusive... Just some > providers whine about it... Its really redundant... Also RPID was never a > RFC, it was a draft that has stuck around and refuses to die... PAI is the > only ratified RFC standard > > > > On 3/5/12 1:42 PM, "Michael Collins" wrote: > > FYI, > > A quick glance through the source suggests that PAI and RPID are mutually > exclusive - i.e. you can set one or the other but not both. I'll defer to > the experts on whether or not the SIP spec says you SHOULD or SHOULD NOT > have both headers in a single message. > > -MC > > On Thu, Mar 1, 2012 at 4:20 AM, Anto wrote: > > Hello > > To my FreeSWITCH servers, they come PAI and RPID headers, sent to the > carrier but only one (the one I have configured with name="caller-id-type" value="pid"/>). Is there any way to send both > headers?. Thanks > > Best regards > Anto > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ________________________________ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From wagnerspi at gmail.com Tue Mar 6 03:57:13 2012 From: wagnerspi at gmail.com (Wagner) Date: Mon, 5 Mar 2012 21:57:13 -0300 Subject: [Freeswitch-users] Flipping CID from .... In-Reply-To: <4F4E36E0.90101@softnet.si> References: <4F4E36E0.90101@softnet.si> Message-ID: Hello, have you tried changing the vars.xml file? there is one variable that you could set it: Thanks 2012/2/29 Miha Zoubek > Hi, > > Flipping CID from "018108500" <018108500> to "Outbound Call" > <38651357952>, FS does this after attended trafers happens. > > For b leg in my outbound call I can see there written Outbound Call not > number. > > Is is possible to change this? I was looking at variable but did not > noticed that caller-id is set to Outbound Call. > > Thanks! > > Miha > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/ab7493ef/attachment-0001.html From yehavi.bourvine at gmail.com Tue Mar 6 06:29:45 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 6 Mar 2012 05:29:45 +0200 Subject: [Freeswitch-users] MWI does not working anymore? In-Reply-To: <4F553FD5.4010208@communicatefreely.net> References: <4F553FD5.4010208@communicatefreely.net> Message-ID: In my case the problem was an erratic profile ALIAS. Do F5 and see whether you have aliases shown there. If you do not need them, simply remove them from your SIP profile (this solved my problem). I am now running without any alias. __Yehavi: 2012/3/6 Tim St. Pierre > I'm having a similar problem. Can someone explain the mechanics behind > MWI? > > I did an upgrade on Friday, and lost most of my MWI functionality > (although it fixed all the BLF issues we had). The version I was using > was several months old, so this may not be new. > > We have two profiles - nonat and internal. nonat is purely private IP > space, internal is public IP, with most clients behind NAT. > > Our curl directory scripts will return all domains whenever asked for a > generic domain section with no tags. > > In the profiles, I have this little bit at the top: > > > > > > > > > > > > > > > > When I do a sofia status, I see that all our domains are aliased to the > internal profile. > > I also have > > > > > In the extension's directory, I have > '"/> > > Until I upgraded, a phone could register on either profile, and expect a > notify on both registration, and any time the message count changed. > > > I read some of the earlier threads about domains only being mapped to > one profile, so I changed the directory script to only return domains to > one profile - the one they have more phones on. > > Now, if the phone registers on the mapped profile, changes produce a > notify, but no notify comes on a registration. On the non-mapped > profile, I sometimes get a notify when it registers, but never on a change. > > What changed and how should I set this up. I tried using subscriptions, > but this is a problem as the phones we use can't subscribe to something > other than what they register as, and that is a problem in our case. > > How do you set a system up so that a domain is accessible from more than > one profile. > > Yehavi, what did you do in the alias section that you talked about? > > I don't expect someone to fix this for me, but I can't find enough > information to fix it on my own. > > -Tim > > > > Anthony Minessale wrote: > > alias allows you to map domains to your profile so the domain can be > > used to find that profile in place of the profile name. > > so in the case of mwi it looks up the domain name and the alias allows > > it to resolve the domain to the correct profile. > > > > similar to dns only for sofia profiles. > > > > On Fri, Feb 17, 2012 at 8:02 AM, Yehavi Bourvine > > wrote: > > > >> I played around with the ALIAS definitions and it seems to solve the > >> problem. > >> > >> I've tried searching the WIKI for a description of ALIAS use and did not > >> find. Anyone can explain what does the alias do and what is the correct > >> usage when you have more than one profile? > >> > >> Thanks, __Yehavi: > >> > >> > >> 2012/2/17 Yehavi Bourvine > >> > >>> Hello Anthony, > >>> > >>> I know, and I've put a comment there that it still does not work. I > am > >>> downloading the recent GIT every two days to my test system (the last > one > >>> was yesterday) and it still behaves the same. > >>> > >>> > >>> Thanks, __Yehavi: > >>> > >>> 2012/2/17 Anthony Minessale > >>> > >>>> you opened a jira on it and i fixed it and committed it > >>>> > >>>> > >>>> [FS-3866] MWI is not updated > >>>> Reporter: Yehavi Bourvine [yehavi] > >>>> Assignee: Anthony Minessale II [anthm] > >>>> Status: Resolved > >>>> http://jira.freeswitch.org/browse/FS-3866 > >>>> > >>>> commit ff379a97e57a94f11ce75e8894b1ceb79d10a9ed > >>>> Author: Anthony Minessale > >>>> Date: Fri Feb 10 11:02:41 2012 -0600 > >>>> > >>>> > >>>> > >>>> On Thu, Feb 16, 2012 at 2:56 AM, Yehavi Bourvine > >>>> wrote: > >>>> > >>>>> Hello, > >>>>> > >>>>> I've noticed a problem started in the last two weeks, but I don't > see > >>>>> anyone else reporting it... > >>>>> > >>>>> MWI status is not updated anymore. It is only updated at the initial > >>>>> boot of > >>>>> the phone. Do others that use GIT from the last two weeks notice it? > >>>>> > >>>>> Thanks, __Yehavi: > >>>>> > >>>>> > >>>>> > _________________________________________________________________________ > >>>>> Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://wiki.freeswitch.org > >>>>> http://www.cluecon.com > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>> > >>>> -- > >>>> Anthony Minessale II > >>>> > >>>> FreeSWITCH http://www.freeswitch.org/ > >>>> ClueCon http://www.cluecon.com/ > >>>> Twitter: http://twitter.com/FreeSWITCH_wire > >>>> > >>>> AIM: anthm > >>>> MSN:anthony_minessale at hotmail.com > >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >>>> IRC: irc.freenode.net #freeswitch > >>>> > >>>> FreeSWITCH Developer Conference > >>>> sip:888 at conference.freeswitch.org > >>>> googletalk:conf+888 at conference.freeswitch.org > >>>> pstn:+19193869900 > >>>> > >>>> > _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> http://www.cluecon.com > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/6e66bb55/attachment.html From jean.marc.hyppolite at gmail.com Tue Mar 6 06:56:27 2012 From: jean.marc.hyppolite at gmail.com (Jean-Marc Hyppolite) Date: Mon, 5 Mar 2012 22:56:27 -0500 Subject: [Freeswitch-users] spandsp error reading frame In-Reply-To: References: <4f53a4d6.a526340a.306b.ffff9ea1@mx.google.com> Message-ID: <4f558af0.451b340a.6a62.119f@mx.google.com> Hello, Thanks for your email. This is my spandsp.conf.xml file: Thanks again. Jean-Marc. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anita Hall Sent: March-05-12 11:15 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] spandsp error reading frame How does your spandsp.conf.xml look like? regards, Anita On Sun, Mar 4, 2012 at 10:52 PM, Jean-Marc Hyppolite wrote: Hello, I am trying to use mod_spandsp to detect call progress. I haven t been successful so far. I am getting the following error messages. ================================================ 2012-03-03 23:55:08.398205 [DEBUG] mod_spandsp_dsp.c:379 (sofia/outbound/. at ...) Starting tone detection for '1' 2012-03-03 23:55:08.398205 [INFO] mod_spandsp_dsp.c:411 (sofia/outbound/. at ...) initializing tone detector 2012-03-03 23:55:08.398205 [DEBUG] switch_core_media_bug.c:462 Attaching BUG to sofia/outbound/. at ... 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:422 (sofia/outbound/. at ...) error reading frame 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:447 (sofia/outbound/. at ...) destroying tone detector ================================================ Any help would be appreciated. Thanks Jean-Marc. N.B. Sorry if some people received this e-mail twice. I am not sure this email went through the first time. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120305/92d1b639/attachment-0001.html From hjz51 at qq.com Tue Mar 6 08:15:52 2012 From: hjz51 at qq.com (hjz51) Date: Tue, 6 Mar 2012 13:15:52 +0800 Subject: [Freeswitch-users] switch_r_sdp or switch_l_sdp modify the media ip or port? Message-ID: <005801ccfb58$33a077b0$9ae16710$@com> Hi , Who can tell me how to use switch_r_sdp or switch_l_sdp modify the media ip or port? Now I am want to redirect media to a transcoding card. But I don't know how to rewrite the rtp ip and port point to the transcoding card. Please can you give me the way to do it. Thanks very much! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/33481ca1/attachment.html From sdame at 207me.com Thu Mar 1 23:08:04 2012 From: sdame at 207me.com (Stephen Dame) Date: Thu, 01 Mar 2012 15:08:04 -0500 Subject: [Freeswitch-users] 401 Unauthorized in iptel external gateway Message-ID: Sent from my U.S. Cellular? Android phone Vitalie Colosov wrote: >Yes I was not sure if you was able to call from registered endpoint thru FS >and also directly, or only directly. >So FS does not create correct #3 query...? >Probably as was suggested above, someone can investigate directly at your >server. >Or if anybody has better ideas... > > >2012/2/26 Anita Hall > >> Vitalie is right. I have seen similar logs. >> >> I am currently using localphone and rapidvox as SIP provider and both >> register Ok. >> >> This is my sip_profile >> http://wiki.freeswitch.org/wiki/Provider_Configuration:_Localphone.com >> >> I could look it up for you if you can give remote access to your machine >> and watch over screen or byobu. >> >> Alternatively, you could give me your account name and password. I will >> check if it works from my installation. I promise not to make any calls and >> you can change the password later :) >> >> regards, >> Anita >> >> >> >> >> On Sun, Feb 26, 2012 at 11:34 AM, Denis Gasparin < >> denis.gasparin at edistar.com> wrote: >> >>> Sorry, i did not exposed well the problem... >>> >>> I was able to call iptel account directly without the gateway (so without >>> registering to iptel). >>> >>> When i setup the gateway i see sofia trying continuosly to register to >>> iptel without success. >>> I see many 401 messages like this: >>> >>> 1. FS->IPTEL Register >>> 2. IPTEL->FS Unauthorized >>> 3. FS->IPTEL Register >>> 4. IPTEL->FS Unauthorized >>> 5. FS->IPTEL Register >>> ... >>> >>> After a while sofia stops trying to register and retries after 30 seconds >>> and so on. >>> >>> Thank you for your help, >>> Denid >>> >>> Il giorno 26/feb/2012, alle ore 02:20, Vitalie Colosov < >>> vetali100 at gmail.com> ha scritto: >>> >>> Usually SIP registration goes like this: >>> >>> 1. FS->IPEL Register >>> >>> 2. IPTEL->FS Unauthorized >>> >>> 3. FS->IPTEL Register (yes, again, with additional credentials) >>> 4. IPTEL->FS OK >>> >>> You say that you was able to call iptel successfully, so FS is registered. >>> You just did not observed and did not provide SIP messages #3 and #4. >>> >>> >>> >>> 2012/2/25 Denis Gasparin >>> >>>> Hi to all. >>>> >>>> Today I installed freeswitch following the guide in Freeswitch book. >>>> I made successfully all the tests in the book... well all but one. >>>> >>>> When I tried to setup an external gateway on my iptel account, the >>>> registration process failed with 401 sip message. >>>> >>>> This is the sip trace: >>>> >>>> freeswitch at dhcppc7> send 647 bytes to udp/[217.9.36.145]:5060 at >>>> 17:59:20.965613: >>>> >>>> ------------------------------------------------------------------------ >>>> REGISTER sip:sip.iptel.org;transport=udp SIP/2.0 >>>> Via: SIP/2.0/UDP MY_PUBLIC_IP:5080;rport;branch=z9hG4bKXNc7jeQB7BXrp >>>> Max-Forwards: 70 >>>> From: ;tag=XQ58BvHmj15eD >>>> To: >>>> Call-ID: 8c2823cd-573d-4cc2-8e01-cba340ae890f >>>> CSeq: 24760296 REGISTER >>>> Contact: >>>> Expires: 3600 >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 >>>> 14-15-32 -0600 >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY >>>> Supported: timer, precondition, path, replaces >>>> Content-Length: 0 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> recv 710 bytes from udp/[217.9.36.145]:5060 at 17:59:21.070238: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 401 Unauthorized >>>> Via: SIP/2.0/UDP 192.168.1.9:5080 >>>> :5080;rport=19198;branch=z9hG4bKXNc7jeQB7BXrp >>>> From: ;tag=XQ58BvHmj15eD >>>> To: >>> ;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-5904 >>>> Call-ID: 8c2823cd-573d-4cc2-8e01-cba340ae890f >>>> CSeq: 24760296 REGISTER >>>> Expires: 600 >>>> Min-Expires: 240 >>>> WWW-Authenticate: Digest realm="iptel.org", >>>> nonce="T0ki/U9JIsE+F7QdNo/qIktZaDvpS0sn" >>>> Server: ser (3.3.0-dev0 (i386/linux)) >>>> Content-Length: 0 >>>> Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=863 >>>> req_src_ip=MY_PUBLIC_IP req_src_port=19198 in_uri=sip:sip.iptel.org;transport=udp >>>> out_uri=sip:sip.iptel.org;transport=udp via_cnt==1" >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> I'm sure that user account and password are valid. >>>> I also tested (successfully) a call from my extension to my >>>> account at iptel.org. >>>> Registration to iptel.org using a sip phone with the same user works >>>> just fine. >>>> >>>> What's wrong? >>>> >>>> Thank you in advance for your help, >>>> >>>> Denis >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org From benkokakao at gmail.com Tue Mar 6 14:03:43 2012 From: benkokakao at gmail.com (Christian Benke) Date: Tue, 6 Mar 2012 12:03:43 +0100 Subject: [Freeswitch-users] Choopy one-way noise (FreeTDM) Message-ID: On 2 March 2012 14:13, Saugort Dario Garcia Tovar wrote: > After change tdm board have you notice difference, the issue persist? No more audio-quality issues after replacing the A500 with a OpenVOX B100p. I forgot to mention we where able to reproduce this issue in our lab, after upgrading wanpipe the problem appeared to be gone - but we didn't test throughly and when we did the same at the clients machine, it didn't have the same effect. Will try to reproduce the problem in the lab again. Regards, Christian From freeswitch at earthspike.net Tue Mar 6 15:53:14 2012 From: freeswitch at earthspike.net (John) Date: Tue, 06 Mar 2012 12:53:14 +0000 Subject: [Freeswitch-users] Choopy one-way noise (FreeTDM) In-Reply-To: References: Message-ID: <4F5608BA.4020904@earthspike.net> On 06/03/12 11:03, Christian Benke wrote: > On 2 March 2012 14:13, Saugort Dario Garcia Tovar wrote: >> After change tdm board have you notice difference, the issue persist? > No more audio-quality issues after replacing the A500 with a OpenVOX B100p. > > I forgot to mention we where able to reproduce this issue in our lab, > after upgrading wanpipe the problem appeared to be gone - but we > didn't test throughly and when we did the same at the clients machine, > it didn't have the same effect. Will try to reproduce the problem in > the lab again. Christian, Which version of wanpipe did you upgrade to? I believe a new version has just been released that I am going to try this week. John From benkokakao at gmail.com Tue Mar 6 16:34:02 2012 From: benkokakao at gmail.com (Christian Benke) Date: Tue, 6 Mar 2012 14:34:02 +0100 Subject: [Freeswitch-users] Choopy one-way noise (FreeTDM) In-Reply-To: <4F5608BA.4020904@earthspike.net> References: <4F5608BA.4020904@earthspike.net> Message-ID: On 6 March 2012 13:53, John wrote: > Which version of wanpipe did you upgrade to? ?I believe a new version > has just been released that I am going to try this week. We used to have 3.5.24(2011-11-15) and upgraded to 3.5.25(2012-02-21) The Changelog says: - Fixed BRI noise issue: Where a BRI channel could be started in a corrupted state. Didn't help with the noise we experienced though :-( From ga at steadfasttelecom.com Tue Mar 6 17:40:03 2012 From: ga at steadfasttelecom.com (Gilad Abada) Date: Tue, 6 Mar 2012 09:40:03 -0500 Subject: [Freeswitch-users] Calls drop after 1:48 (kinda) In-Reply-To: References: <1330644898790-7334603.post@n2.nabble.com> Message-ID: Hi I am having the same exact issue. I am using Grandstream HT 502's Has anyone come up with a fix? Thanks Gill On Fri, Mar 2, 2012 at 3:18 AM, Brent Paddon wrote: > Have you captured the network traffic between the phone and FS to see whats > there? > > That will probably determine your next step. > > Brent > > > On Fri, Mar 2, 2012 at 9:34 AM, BloodyIron wrote: >> >> Hi Folks, >> >> Okay so this one is a bit tricky to reproduce. One of our extensions will >> have their calls dropped preicsely after 1 minutes 48 seconds ( 1:48 ) of >> call time. Another extension on the same segment of the network does not >> do >> the same thing. This extension does this for every single call type, be it >> external or internal calls. >> >> Now you may think, well it may be a busted phone. We reset the phone to >> factory defaults and saw no improvement. Furthermore we are seeing it >> elsewhere in our freeswitch installation, as in other extensions on other >> sites are seeing the same issue. >> >> These extensions are behind a NAT, however the freeswitch server is >> publically facing (as in public IP), with a passive firewall between it >> and >> the world (no routing, no NAT, etc). >> >> Just to be clear, we are also using fusionpbx to control the installation, >> as typing everything into an xml is not very efficient (but we're not >> scared >> of working with xml files either). >> >> Right now, we're gonna try adding " ?" to our sofia.conf.xml file to >> address >> it, beyond this we are unsure what to do. >> >> Can anyone speak on this matter? >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Calls-drop-after-1-48-kinda-tp7334603p7334603.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. From fs-list at communicatefreely.net Tue Mar 6 17:41:57 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Tue, 06 Mar 2012 09:41:57 -0500 Subject: [Freeswitch-users] MWI does not working anymore? In-Reply-To: References: <4F553FD5.4010208@communicatefreely.net> Message-ID: <4F562235.9090608@communicatefreely.net> If I do sofia status, I see all my profiles, and an alias for each customer domain to the internal (nat'd) profile. I did a little experiment, changing my directory script so that it will return all domains for each profile as it starts. I started the internal profile first, so all the domains are aliased to it. Now, I get a notify on registration for any profile, which is what I want. If a phone is registered to the internal profile, it gets a notify on change. If I register to the nonat profile, I get a notify on register only if it didn't already have a registration, but no notify on change. How did you make yours work without any aliases? If I take my aliases out, I get all these errors about "can't find profile". -Tim Yehavi Bourvine wrote: > In my case the problem was an erratic profile ALIAS. Do F5 and see > whether you have aliases shown there. If you do not need them, simply > remove them from your SIP profile (this solved my problem). I am now > running without any alias. > > __Yehavi: > > 2012/3/6 Tim St. Pierre > > > I'm having a similar problem. Can someone explain the mechanics > behind MWI? > > I did an upgrade on Friday, and lost most of my MWI functionality > (although it fixed all the BLF issues we had). The version I was > using > was several months old, so this may not be new. > > We have two profiles - nonat and internal. nonat is purely private IP > space, internal is public IP, with most clients behind NAT. > > Our curl directory scripts will return all domains whenever asked > for a > generic domain section with no tags. > > In the profiles, I have this little bit at the top: > > > > > > > > > > > > > > > > When I do a sofia status, I see that all our domains are aliased > to the > internal profile. > > I also have > > > > > In the extension's directory, I have > data=mailbox at subdomain.communicatefreely.net > '"/> > > Until I upgraded, a phone could register on either profile, and > expect a > notify on both registration, and any time the message count changed. > > > I read some of the earlier threads about domains only being mapped to > one profile, so I changed the directory script to only return > domains to > one profile - the one they have more phones on. > > Now, if the phone registers on the mapped profile, changes produce a > notify, but no notify comes on a registration. On the non-mapped > profile, I sometimes get a notify when it registers, but never on > a change. > > What changed and how should I set this up. I tried using > subscriptions, > but this is a problem as the phones we use can't subscribe to > something > other than what they register as, and that is a problem in our case. > > How do you set a system up so that a domain is accessible from > more than > one profile. > > Yehavi, what did you do in the alias section that you talked about? > > I don't expect someone to fix this for me, but I can't find enough > information to fix it on my own. > > -Tim > > > > Anthony Minessale wrote: > > alias allows you to map domains to your profile so the domain can be > > used to find that profile in place of the profile name. > > so in the case of mwi it looks up the domain name and the alias > allows > > it to resolve the domain to the correct profile. > > > > similar to dns only for sofia profiles. > > > > On Fri, Feb 17, 2012 at 8:02 AM, Yehavi Bourvine > > > > wrote: > > > >> I played around with the ALIAS definitions and it seems to > solve the > >> problem. > >> > >> I've tried searching the WIKI for a description of ALIAS use > and did not > >> find. Anyone can explain what does the alias do and what is the > correct > >> usage when you have more than one profile? > >> > >> Thanks, __Yehavi: > >> > >> > >> 2012/2/17 Yehavi Bourvine > > >> > >>> Hello Anthony, > >>> > >>> I know, and I've put a comment there that it still does not > work. I am > >>> downloading the recent GIT every two days to my test system > (the last one > >>> was yesterday) and it still behaves the same. > >>> > >>> > >>> Thanks, __Yehavi: > >>> > >>> 2012/2/17 Anthony Minessale > > >>> > >>>> you opened a jira on it and i fixed it and committed it > >>>> > >>>> > >>>> [FS-3866] MWI is not updated > >>>> Reporter: Yehavi Bourvine [yehavi] > >>>> Assignee: Anthony Minessale II [anthm] > >>>> Status: Resolved > >>>> http://jira.freeswitch.org/browse/FS-3866 > >>>> > >>>> commit ff379a97e57a94f11ce75e8894b1ceb79d10a9ed > >>>> Author: Anthony Minessale > > >>>> Date: Fri Feb 10 11:02:41 2012 -0600 > >>>> > >>>> > >>>> > >>>> On Thu, Feb 16, 2012 at 2:56 AM, Yehavi Bourvine > >>>> > wrote: > >>>> > >>>>> Hello, > >>>>> > >>>>> I've noticed a problem started in the last two weeks, but > I don't see > >>>>> anyone else reporting it... > >>>>> > >>>>> MWI status is not updated anymore. It is only updated at the > initial > >>>>> boot of > >>>>> the phone. Do others that use GIT from the last two weeks > notice it? > >>>>> > >>>>> Thanks, __Yehavi: > >>>>> > >>>>> > >>>>> > _________________________________________________________________________ > >>>>> Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://wiki.freeswitch.org > >>>>> http://www.cluecon.com > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>> > >>>> -- > >>>> Anthony Minessale II > >>>> > >>>> FreeSWITCH http://www.freeswitch.org/ > >>>> ClueCon http://www.cluecon.com/ > >>>> Twitter: http://twitter.com/FreeSWITCH_wire > >>>> > >>>> AIM: anthm > >>>> MSN:anthony_minessale at hotmail.com > > >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > >>>> IRC: irc.freenode.net #freeswitch > >>>> > >>>> FreeSWITCH Developer Conference > >>>> sip:888 at conference.freeswitch.org > > >>>> googletalk:conf+888 at conference.freeswitch.org > > >>>> pstn:+19193869900 > >>>> > >>>> > _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> http://www.cluecon.com > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------------------------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ga at steadfasttelecom.com Tue Mar 6 17:42:56 2012 From: ga at steadfasttelecom.com (Gilad Abada) Date: Tue, 6 Mar 2012 09:42:56 -0500 Subject: [Freeswitch-users] Calls drop after 1:48 (kinda) In-Reply-To: References: <1330644898790-7334603.post@n2.nabble.com> Message-ID: Here is my CLI debug from when the call drops. The extension that is called out is 1003. I also have pcaps if they are needed. 2012-03-06 09:24:27.580387 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/1003 at example.domain [BREAK] 2012-03-06 09:24:27.580387 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/1003 at example.domain [BREAK] 2012-03-06 09:24:27.580387 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/1003 at example.domain [BREAK] 2012-03-06 09:24:27.590418 [DEBUG] sofia.c:5532 Channel sofia/internal/1003 at example.domain entering state [terminated][200] 2012-03-06 09:24:27.590418 [DEBUG] switch_channel.c:2848 (sofia/internal/1003 at example.domain) Callstate Change ACTIVE -> HANGUP 2012-03-06 09:24:27.590418 [NOTICE] sofia.c:6299 Hangup sofia/internal/1003 at example.domain [CS_EXECUTE] [NORMAL_CLEARING] 2012-03-06 09:24:27.590418 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/1003 at example.domain [KILL] 2012-03-06 09:24:27.590418 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1003 at example.domain [BREAK] 2012-03-06 09:24:27.590418 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/1003 at example.domain] 2012-03-06 09:24:27.590418 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/external/XXXXXXXXXXX [BREAK] 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:499 sofia/internal/1003 at example.domain ending bridge by request from write function 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/external/XXXXXXXXXXX] 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/1003 at example.domain [BREAK] 2012-03-06 09:24:27.610398 [DEBUG] switch_channel.c:2848 (sofia/external/XXXXXXXXXXX) Callstate Change ACTIVE -> HANGUP 2012-03-06 09:24:27.610398 [NOTICE] switch_ivr_bridge.c:669 Hangup sofia/external/XXXXXXXXXXX [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2012-03-06 09:24:27.610398 [DEBUG] switch_channel.c:2871 Send signal sofia/external/XXXXXXXXXXX [KILL] 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/XXXXXXXXXXX [BREAK] 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:420 (sofia/external/XXXXXXXXXXX) State EXCHANGE_MEDIA going to sleep 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 (sofia/external/XXXXXXXXXXX) Running State Change CS_HANGUP 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 (sofia/external/XXXXXXXXXXX) State HANGUP 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:463 sofia/external/XXXXXXXXXXX Overriding SIP cause 480 with 200 from the other leg 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:469 Channel sofia/external/XXXXXXXXXXX hanging up, cause: NORMAL_CLEARING 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:513 Sending BYE to sofia/external/XXXXXXXXXXX 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:47 sofia/external/XXXXXXXXXXX Standard HANGUP, cause: NORMAL_CLEARING 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 (sofia/external/XXXXXXXXXXX) State HANGUP going to sleep 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:393 (sofia/external/XXXXXXXXXXX) State Change CS_HANGUP -> CS_REPORTING 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/XXXXXXXXXXX [BREAK] 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 (sofia/external/XXXXXXXXXXX) Running State Change CS_REPORTING 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:662 (sofia/external/XXXXXXXXXXX) State REPORTING 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:1403 sofia/external/XXXXXXXXXXX skip receive message [UNBRIDGE] (channel is hungup already) 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:1406 sofia/internal/1003 at example.domain skip receive message [UNBRIDGE] (channel is hungup already) 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:2285 sofia/internal/1003 at example.domain skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1003 at example.domain) State EXECUTE going to sleep 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1003 at example.domain) Running State Change CS_HANGUP 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1003 at example.domain) State HANGUP 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1003 at example.domain hanging up, cause: NORMAL_CLEARING 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1003 at example.domain Standard HANGUP, cause: NORMAL_CLEARING 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1003 at example.domain) State HANGUP going to sleep 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1003 at example.domain) State Change CS_HANGUP -> CS_REPORTING 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1003 at example.domain [BREAK] 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1003 at example.domain) Running State Change CS_REPORTING 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1003 at example.domain) State REPORTING 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:79 sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:662 (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:387 (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY 2012-03-06 09:24:27.630498 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/XXXXXXXXXXX [BREAK] 2012-03-06 09:24:27.630498 [DEBUG] switch_core_session.c:1380 Session 4 (sofia/external/XXXXXXXXXXX) Locked, Waiting on external entities 2012-03-06 09:24:27.630498 [NOTICE] switch_core_session.c:1398 Session 4 (sofia/external/XXXXXXXXXXX) Ended 2012-03-06 09:24:27.630498 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external/XXXXXXXXXXX [CS_DESTROY] 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:491 (sofia/external/XXXXXXXXXXX) Callstate Change HANGUP -> DOWN 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:494 (sofia/external/XXXXXXXXXXX) Running State Change CS_DESTROY 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:504 (sofia/external/XXXXXXXXXXX) State DESTROY 2012-03-06 09:24:27.630498 [DEBUG] mod_sofia.c:374 sofia/external/XXXXXXXXXXX SOFIA DESTROY 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:86 sofia/external/XXXXXXXXXXX Standard DESTROY 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:504 (sofia/external/XXXXXXXXXXX) State DESTROY going to sleep 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1003 at example.domain Standard REPORTING, cause: NORMAL_CLEARING 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1003 at example.domain) State REPORTING going to sleep 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1003 at example.domain) State Change CS_REPORTING -> CS_DESTROY 2012-03-06 09:24:27.640384 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1003 at example.domain [BREAK] 2012-03-06 09:24:27.640384 [DEBUG] switch_core_session.c:1380 Session 3 (sofia/internal/1003 at example.domain) Locked, Waiting on external entities 2012-03-06 09:24:27.640384 [NOTICE] switch_core_session.c:1398 Session 3 (sofia/internal/1003 at example.domain) Ended 2012-03-06 09:24:27.640384 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/1003 at example.domain [CS_DESTROY] 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1003 at example.domain) Callstate Change HANGUP -> DOWN 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1003 at example.domain) Running State Change CS_DESTROY 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1003 at example.domain) State DESTROY 2012-03-06 09:24:27.640384 [DEBUG] mod_sofia.c:374 sofia/internal/1003 at example.domain SOFIA DESTROY 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1003 at example.domain Standard DESTROY 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1003 at example.domain) State DESTROY going to sleep 2012-03-06 09:24:28.190377 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:499 sofia/external/+XXXXXXXXXXX at flowroute.com ending bridge by request from write function 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/sip:1020 at xxx.xxx.xxx.xxx] 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2848 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Callstate Change ACTIVE -> HANGUP 2012-03-06 09:24:28.210377 [NOTICE] switch_ivr_bridge.c:669 Hangup sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [KILL] 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:420 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State EXCHANGE_MEDIA going to sleep 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Running State Change CS_HANGUP 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2848 (sofia/external/+XXXXXXXXXXX at flowroute.com) Callstate Change ACTIVE -> HANGUP 2012-03-06 09:24:28.210377 [NOTICE] sofia.c:628 Hangup sofia/external/+XXXXXXXXXXX at flowroute.com [CS_EXECUTE] [NORMAL_CLEARING] 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State HANGUP 2012-03-06 09:24:28.210377 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1020 at xxx.xxx.xxx.xxx hanging up, cause: NORMAL_CLEARING 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2871 Send signal sofia/external/+XXXXXXXXXXX at flowroute.com [KILL] 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/external/+XXXXXXXXXXX at flowroute.com] 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:1403 sofia/internal/sip:1020 at xxx.xxx.xxx.xxx skip receive message [UNBRIDGE] (channel is hungup already) 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:1406 sofia/external/+XXXXXXXXXXX at flowroute.com skip receive message [UNBRIDGE] (channel is hungup already) 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:2285 sofia/external/+XXXXXXXXXXX at flowroute.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:2100 sofia/external/+XXXXXXXXXXX at flowroute.com Channel is hungup and application 'bridge' does not have the zombie_exec flag. 2012-03-06 09:24:28.210377 [DEBUG] switch_cpp.cpp:1007 sofia/external/+XXXXXXXXXXX at flowroute.com destroy/unlink session from object 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:2285 sofia/external/+XXXXXXXXXXX at flowroute.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:417 (sofia/external/+XXXXXXXXXXX at flowroute.com) State EXECUTE going to sleep 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+XXXXXXXXXXX at flowroute.com) Running State Change CS_HANGUP 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 (sofia/external/+XXXXXXXXXXX at flowroute.com) State HANGUP 2012-03-06 09:24:28.210377 [DEBUG] mod_sofia.c:469 Channel sofia/external/+XXXXXXXXXXX at flowroute.com hanging up, cause: NORMAL_CLEARING 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:47 sofia/external/+XXXXXXXXXXX at flowroute.com Standard HANGUP, cause: NORMAL_CLEARING 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 (sofia/external/+XXXXXXXXXXX at flowroute.com) State HANGUP going to sleep 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:393 (sofia/external/+XXXXXXXXXXX at flowroute.com) State Change CS_HANGUP -> CS_REPORTING 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+XXXXXXXXXXX at flowroute.com) Running State Change CS_REPORTING 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:662 (sofia/external/+XXXXXXXXXXX at flowroute.com) State REPORTING 2012-03-06 09:24:28.210377 [DEBUG] mod_sofia.c:513 Sending BYE to sofia/internal/sip:1020 at xxx.xxx.xxx.xxx 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1020 at xxx.xxx.xxx.xxx Standard HANGUP, cause: NORMAL_CLEARING 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State HANGUP going to sleep 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State Change CS_HANGUP -> CS_REPORTING 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Running State Change CS_REPORTING 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State REPORTING 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:79 sofia/external/+XXXXXXXXXXX at flowroute.com Standard REPORTING, cause: NORMAL_CLEARING 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:662 (sofia/external/+XXXXXXXXXXX at flowroute.com) State REPORTING going to sleep 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:387 (sofia/external/+XXXXXXXXXXX at flowroute.com) State Change CS_REPORTING -> CS_DESTROY 2012-03-06 09:24:28.230382 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] 2012-03-06 09:24:28.230382 [DEBUG] switch_core_session.c:1380 Session 5 (sofia/external/+XXXXXXXXXXX at flowroute.com) Locked, Waiting on external entities 2012-03-06 09:24:28.230382 [NOTICE] switch_core_session.c:1398 Session 5 (sofia/external/+XXXXXXXXXXX at flowroute.com) Ended 2012-03-06 09:24:28.230382 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external/+XXXXXXXXXXX at flowroute.com [CS_DESTROY] 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:491 (sofia/external/+XXXXXXXXXXX at flowroute.com) Callstate Change HANGUP -> DOWN 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:494 (sofia/external/+XXXXXXXXXXX at flowroute.com) Running State Change CS_DESTROY 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:504 (sofia/external/+XXXXXXXXXXX at flowroute.com) State DESTROY 2012-03-06 09:24:28.230382 [DEBUG] mod_sofia.c:374 sofia/external/+XXXXXXXXXXX at flowroute.com SOFIA DESTROY 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:86 sofia/external/+XXXXXXXXXXX at flowroute.com Standard DESTROY 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:504 (sofia/external/+XXXXXXXXXXX at flowroute.com) State DESTROY going to sleep 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1020 at xxx.xxx.xxx.xxx Standard REPORTING, cause: NORMAL_CLEARING 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State REPORTING going to sleep 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State Change CS_REPORTING -> CS_DESTROY 2012-03-06 09:24:28.240382 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] 2012-03-06 09:24:28.240382 [DEBUG] switch_core_session.c:1380 Session 6 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Locked, Waiting on external entities 2012-03-06 09:24:28.240382 [NOTICE] switch_core_session.c:1398 Session 6 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Ended 2012-03-06 09:24:28.240382 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [CS_DESTROY] 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Callstate Change HANGUP -> DOWN 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Running State Change CS_DESTROY 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State DESTROY 2012-03-06 09:24:28.240382 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1020 at xxx.xxx.xxx.xxx SOFIA DESTROY 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1020 at xxx.xxx.xxx.xxx Standard DESTROY 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State DESTROY going to sleep On Tue, Mar 6, 2012 at 9:40 AM, Gilad Abada wrote: > Hi > > I am having the same exact issue. I am using Grandstream HT 502's > > Has anyone come up with a fix? > > Thanks > Gill > > On Fri, Mar 2, 2012 at 3:18 AM, Brent Paddon wrote: >> Have you captured the network traffic between the phone and FS to see whats >> there? >> >> That will probably determine your next step. >> >> Brent >> >> >> On Fri, Mar 2, 2012 at 9:34 AM, BloodyIron wrote: >>> >>> Hi Folks, >>> >>> Okay so this one is a bit tricky to reproduce. One of our extensions will >>> have their calls dropped preicsely after 1 minutes 48 seconds ( 1:48 ) of >>> call time. Another extension on the same segment of the network does not >>> do >>> the same thing. This extension does this for every single call type, be it >>> external or internal calls. >>> >>> Now you may think, well it may be a busted phone. We reset the phone to >>> factory defaults and saw no improvement. Furthermore we are seeing it >>> elsewhere in our freeswitch installation, as in other extensions on other >>> sites are seeing the same issue. >>> >>> These extensions are behind a NAT, however the freeswitch server is >>> publically facing (as in public IP), with a passive firewall between it >>> and >>> the world (no routing, no NAT, etc). >>> >>> Just to be clear, we are also using fusionpbx to control the installation, >>> as typing everything into an xml is not very efficient (but we're not >>> scared >>> of working with xml files either). >>> >>> Right now, we're gonna try adding " ?" to our sofia.conf.xml file to >>> address >>> it, beyond this we are unsure what to do. >>> >>> Can anyone speak on this matter? >>> >>> -- >>> View this message in context: >>> http://freeswitch-users.2379917.n2.nabble.com/Calls-drop-after-1-48-kinda-tp7334603p7334603.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Gilad Abada > > SteadFast Telecommunications, Inc. > > Call us to find out how much you can save with VoIP! > > V: 212.589.1001 > F: 212.589.1011 > > > For 35 years, Steadfast Telecommunications has been providing > state-of-the-art communications technology to businesses and > government agencies - large and small. Steadfast Telecommunications > tailors Unified Communications and Voice-Over IP Solutions to > single-site offices or multi-site and worldwide enterprises.?? Make > your virtual office a reality.? Enjoy the freedom to travel while > remaining connected to your office. -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. From yehavi.bourvine at gmail.com Tue Mar 6 18:52:57 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 6 Mar 2012 17:52:57 +0200 Subject: [Freeswitch-users] MWI does not working anymore? In-Reply-To: <4F562235.9090608@communicatefreely.net> References: <4F553FD5.4010208@communicatefreely.net> <4F562235.9090608@communicatefreely.net> Message-ID: Most of our phones are on the same profile. We have very few on a different one, but since they are less important I never tested this on them. Sorry, __Yehavi: 2012/3/6 Tim St. Pierre > If I do sofia status, I see all my profiles, and an alias for each > customer domain to the internal (nat'd) profile. > > I did a little experiment, changing my directory script so that it will > return all domains for each profile as it starts. I started the > internal profile first, so all the domains are aliased to it. > > Now, I get a notify on registration for any profile, which is what I want. > > If a phone is registered to the internal profile, it gets a notify on > change. > > If I register to the nonat profile, I get a notify on register only if > it didn't already have a registration, but no notify on change. > > How did you make yours work without any aliases? If I take my aliases > out, I get all these errors about "can't find profile". > > -Tim > > Yehavi Bourvine wrote: > > In my case the problem was an erratic profile ALIAS. Do F5 and see > > whether you have aliases shown there. If you do not need them, simply > > remove them from your SIP profile (this solved my problem). I am now > > running without any alias. > > > > __Yehavi: > > > > 2012/3/6 Tim St. Pierre > > > > > > I'm having a similar problem. Can someone explain the mechanics > > behind MWI? > > > > I did an upgrade on Friday, and lost most of my MWI functionality > > (although it fixed all the BLF issues we had). The version I was > > using > > was several months old, so this may not be new. > > > > We have two profiles - nonat and internal. nonat is purely private > IP > > space, internal is public IP, with most clients behind NAT. > > > > Our curl directory scripts will return all domains whenever asked > > for a > > generic domain section with no tags. > > > > In the profiles, I have this little bit at the top: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > When I do a sofia status, I see that all our domains are aliased > > to the > > internal profile. > > > > I also have > > > > > > > > > > In the extension's directory, I have > > > data=mailbox at subdomain.communicatefreely.net > > '"/> > > > > Until I upgraded, a phone could register on either profile, and > > expect a > > notify on both registration, and any time the message count changed. > > > > > > I read some of the earlier threads about domains only being mapped to > > one profile, so I changed the directory script to only return > > domains to > > one profile - the one they have more phones on. > > > > Now, if the phone registers on the mapped profile, changes produce a > > notify, but no notify comes on a registration. On the non-mapped > > profile, I sometimes get a notify when it registers, but never on > > a change. > > > > What changed and how should I set this up. I tried using > > subscriptions, > > but this is a problem as the phones we use can't subscribe to > > something > > other than what they register as, and that is a problem in our case. > > > > How do you set a system up so that a domain is accessible from > > more than > > one profile. > > > > Yehavi, what did you do in the alias section that you talked about? > > > > I don't expect someone to fix this for me, but I can't find enough > > information to fix it on my own. > > > > -Tim > > > > > > > > Anthony Minessale wrote: > > > alias allows you to map domains to your profile so the domain can > be > > > used to find that profile in place of the profile name. > > > so in the case of mwi it looks up the domain name and the alias > > allows > > > it to resolve the domain to the correct profile. > > > > > > similar to dns only for sofia profiles. > > > > > > On Fri, Feb 17, 2012 at 8:02 AM, Yehavi Bourvine > > > > > > wrote: > > > > > >> I played around with the ALIAS definitions and it seems to > > solve the > > >> problem. > > >> > > >> I've tried searching the WIKI for a description of ALIAS use > > and did not > > >> find. Anyone can explain what does the alias do and what is the > > correct > > >> usage when you have more than one profile? > > >> > > >> Thanks, __Yehavi: > > >> > > >> > > >> 2012/2/17 Yehavi Bourvine > > > > >> > > >>> Hello Anthony, > > >>> > > >>> I know, and I've put a comment there that it still does not > > work. I am > > >>> downloading the recent GIT every two days to my test system > > (the last one > > >>> was yesterday) and it still behaves the same. > > >>> > > >>> > > >>> Thanks, __Yehavi: > > >>> > > >>> 2012/2/17 Anthony Minessale > > > > >>> > > >>>> you opened a jira on it and i fixed it and committed it > > >>>> > > >>>> > > >>>> [FS-3866] MWI is not updated > > >>>> Reporter: Yehavi Bourvine [yehavi] > > >>>> Assignee: Anthony Minessale II [anthm] > > >>>> Status: Resolved > > >>>> http://jira.freeswitch.org/browse/FS-3866 > > >>>> > > >>>> commit ff379a97e57a94f11ce75e8894b1ceb79d10a9ed > > >>>> Author: Anthony Minessale > > > > >>>> Date: Fri Feb 10 11:02:41 2012 -0600 > > >>>> > > >>>> > > >>>> > > >>>> On Thu, Feb 16, 2012 at 2:56 AM, Yehavi Bourvine > > >>>> > > wrote: > > >>>> > > >>>>> Hello, > > >>>>> > > >>>>> I've noticed a problem started in the last two weeks, but > > I don't see > > >>>>> anyone else reporting it... > > >>>>> > > >>>>> MWI status is not updated anymore. It is only updated at the > > initial > > >>>>> boot of > > >>>>> the phone. Do others that use GIT from the last two weeks > > notice it? > > >>>>> > > >>>>> Thanks, __Yehavi: > > >>>>> > > >>>>> > > >>>>> > > > _________________________________________________________________________ > > >>>>> Professional FreeSWITCH Consulting Services: > > >>>>> consulting at freeswitch.org > > >>>>> http://www.freeswitchsolutions.com > > >>>>> > > >>>>> > > >>>>> > > >>>>> > > >>>>> Official FreeSWITCH Sites > > >>>>> http://www.freeswitch.org > > >>>>> http://wiki.freeswitch.org > > >>>>> http://www.cluecon.com > > >>>>> > > >>>>> FreeSWITCH-users mailing list > > >>>>> FreeSWITCH-users at lists.freeswitch.org > > > > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>>>> > > >>>>> > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>>>> http://www.freeswitch.org > > >>>>> > > >>>>> > > >>>> > > >>>> -- > > >>>> Anthony Minessale II > > >>>> > > >>>> FreeSWITCH http://www.freeswitch.org/ > > >>>> ClueCon http://www.cluecon.com/ > > >>>> Twitter: http://twitter.com/FreeSWITCH_wire > > >>>> > > >>>> AIM: anthm > > >>>> MSN:anthony_minessale at hotmail.com > > > > >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >>>> IRC: irc.freenode.net #freeswitch > > >>>> > > >>>> FreeSWITCH Developer Conference > > >>>> sip:888 at conference.freeswitch.org > > > > >>>> googletalk:conf+888 at conference.freeswitch.org > > > > >>>> pstn:+19193869900 > > >>>> > > >>>> > > > _________________________________________________________________________ > > >>>> Professional FreeSWITCH Consulting Services: > > >>>> consulting at freeswitch.org > > >>>> http://www.freeswitchsolutions.com > > >>>> > > >>>> > > >>>> > > >>>> > > >>>> Official FreeSWITCH Sites > > >>>> http://www.freeswitch.org > > >>>> http://wiki.freeswitch.org > > >>>> http://www.cluecon.com > > >>>> > > >>>> FreeSWITCH-users mailing list > > >>>> FreeSWITCH-users at lists.freeswitch.org > > > > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>>> > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>>> http://www.freeswitch.org > > >>>> > > >> > > > _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: > > >> consulting at freeswitch.org > > >> http://www.freeswitchsolutions.com > > >> > > >> > > >> > > >> > > >> Official FreeSWITCH Sites > > >> http://www.freeswitch.org > > >> http://wiki.freeswitch.org > > >> http://www.cluecon.com > > >> > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > >> > > >> > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > ------------------------------------------------------------------------ > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/12abed4c/attachment-0001.html From fs-list at communicatefreely.net Tue Mar 6 19:03:08 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Tue, 06 Mar 2012 11:03:08 -0500 Subject: [Freeswitch-users] MWI does not working anymore? In-Reply-To: References: <4F553FD5.4010208@communicatefreely.net> <4F562235.9090608@communicatefreely.net> Message-ID: <4F56353C.1010800@communicatefreely.net> No problem. It was worth a try. I'm willing to bet that they don't get a notify until they re-register. -Tim Yehavi Bourvine wrote: > Most of our phones are on the same profile. We have very few on a > different one, but since they are less important I never tested this > on them. > > Sorry, __Yehavi: > > 2012/3/6 Tim St. Pierre > > > If I do sofia status, I see all my profiles, and an alias for each > customer domain to the internal (nat'd) profile. > > I did a little experiment, changing my directory script so that it > will > return all domains for each profile as it starts. I started the > internal profile first, so all the domains are aliased to it. > > Now, I get a notify on registration for any profile, which is what > I want. > > If a phone is registered to the internal profile, it gets a notify on > change. > > If I register to the nonat profile, I get a notify on register only if > it didn't already have a registration, but no notify on change. > > How did you make yours work without any aliases? If I take my > aliases > out, I get all these errors about "can't find profile". > > -Tim > > Yehavi Bourvine wrote: > > In my case the problem was an erratic profile ALIAS. Do F5 and see > > whether you have aliases shown there. If you do not need them, > simply > > remove them from your SIP profile (this solved my problem). I am now > > running without any alias. > > > > __Yehavi: > > > > 2012/3/6 Tim St. Pierre > > >> > > > > I'm having a similar problem. Can someone explain the mechanics > > behind MWI? > > > > I did an upgrade on Friday, and lost most of my MWI > functionality > > (although it fixed all the BLF issues we had). The version > I was > > using > > was several months old, so this may not be new. > > > > We have two profiles - nonat and internal. nonat is purely > private IP > > space, internal is public IP, with most clients behind NAT. > > > > Our curl directory scripts will return all domains whenever > asked > > for a > > generic domain section with no tags. > > > > In the profiles, I have this little bit at the top: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > When I do a sofia status, I see that all our domains are aliased > > to the > > internal profile. > > > > I also have > > > > > > > > > > In the extension's directory, I have > > > data=mailbox at subdomain.communicatefreely.net > > > >'"/> > > > > Until I upgraded, a phone could register on either profile, and > > expect a > > notify on both registration, and any time the message count > changed. > > > > > > I read some of the earlier threads about domains only being > mapped to > > one profile, so I changed the directory script to only return > > domains to > > one profile - the one they have more phones on. > > > > Now, if the phone registers on the mapped profile, changes > produce a > > notify, but no notify comes on a registration. On the > non-mapped > > profile, I sometimes get a notify when it registers, but > never on > > a change. > > > > What changed and how should I set this up. I tried using > > subscriptions, > > but this is a problem as the phones we use can't subscribe to > > something > > other than what they register as, and that is a problem in > our case. > > > > How do you set a system up so that a domain is accessible from > > more than > > one profile. > > > > Yehavi, what did you do in the alias section that you talked > about? > > > > I don't expect someone to fix this for me, but I can't find > enough > > information to fix it on my own. > > > > -Tim > > > > > > > > Anthony Minessale wrote: > > > alias allows you to map domains to your profile so the > domain can be > > > used to find that profile in place of the profile name. > > > so in the case of mwi it looks up the domain name and the > alias > > allows > > > it to resolve the domain to the correct profile. > > > > > > similar to dns only for sofia profiles. > > > > > > On Fri, Feb 17, 2012 at 8:02 AM, Yehavi Bourvine > > > > >> > > wrote: > > > > > >> I played around with the ALIAS definitions and it seems to > > solve the > > >> problem. > > >> > > >> I've tried searching the WIKI for a description of ALIAS use > > and did not > > >> find. Anyone can explain what does the alias do and what > is the > > correct > > >> usage when you have more than one profile? > > >> > > >> Thanks, __Yehavi: > > >> > > >> > > >> 2012/2/17 Yehavi Bourvine > > >> > > >> > > >>> Hello Anthony, > > >>> > > >>> I know, and I've put a comment there that it still > does not > > work. I am > > >>> downloading the recent GIT every two days to my test system > > (the last one > > >>> was yesterday) and it still behaves the same. > > >>> > > >>> > > >>> Thanks, __Yehavi: > > >>> > > >>> 2012/2/17 Anthony Minessale > > >> > > >>> > > >>>> you opened a jira on it and i fixed it and committed it > > >>>> > > >>>> > > >>>> [FS-3866] MWI is not updated > > >>>> Reporter: Yehavi Bourvine [yehavi] > > >>>> Assignee: Anthony Minessale II [anthm] > > >>>> Status: Resolved > > >>>> http://jira.freeswitch.org/browse/FS-3866 > > >>>> > > >>>> commit ff379a97e57a94f11ce75e8894b1ceb79d10a9ed > > >>>> Author: Anthony Minessale > > >> > > >>>> Date: Fri Feb 10 11:02:41 2012 -0600 > > >>>> > > >>>> > > >>>> > > >>>> On Thu, Feb 16, 2012 at 2:56 AM, Yehavi Bourvine > > >>>> > > >> wrote: > > >>>> > > >>>>> Hello, > > >>>>> > > >>>>> I've noticed a problem started in the last two > weeks, but > > I don't see > > >>>>> anyone else reporting it... > > >>>>> > > >>>>> MWI status is not updated anymore. It is only updated > at the > > initial > > >>>>> boot of > > >>>>> the phone. Do others that use GIT from the last two weeks > > notice it? > > >>>>> > > >>>>> Thanks, __Yehavi: > > >>>>> > > >>>>> > > >>>>> > > > _________________________________________________________________________ > > >>>>> Professional FreeSWITCH Consulting Services: > > >>>>> consulting at freeswitch.org > > > > > >>>>> http://www.freeswitchsolutions.com > > >>>>> > > >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > >>>>> > > >>>>> > > >>>>> Official FreeSWITCH Sites > > >>>>> http://www.freeswitch.org > > >>>>> http://wiki.freeswitch.org > > >>>>> http://www.cluecon.com > > >>>>> > > >>>>> FreeSWITCH-users mailing list > > >>>>> FreeSWITCH-users at lists.freeswitch.org > > > > > > >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>>>> > > >>>>> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>>>> http://www.freeswitch.org > > >>>>> > > >>>>> > > >>>> > > >>>> -- > > >>>> Anthony Minessale II > > >>>> > > >>>> FreeSWITCH http://www.freeswitch.org/ > > >>>> ClueCon http://www.cluecon.com/ > > >>>> Twitter: http://twitter.com/FreeSWITCH_wire > > >>>> > > >>>> AIM: anthm > > >>>> MSN:anthony_minessale at hotmail.com > > > > > > >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > >>>> IRC: irc.freenode.net > #freeswitch > > >>>> > > >>>> FreeSWITCH Developer Conference > > >>>> sip:888 at conference.freeswitch.org > > > > > > >>>> googletalk:conf+888 at conference.freeswitch.org > > > > > > >>>> pstn:+19193869900 > > >>>> > > >>>> > > > _________________________________________________________________________ > > >>>> Professional FreeSWITCH Consulting Services: > > >>>> consulting at freeswitch.org > > > > > >>>> http://www.freeswitchsolutions.com > > >>>> > > >>>> > > >>>> > > >>>> > > >>>> Official FreeSWITCH Sites > > >>>> http://www.freeswitch.org > > >>>> http://wiki.freeswitch.org > > >>>> http://www.cluecon.com > > >>>> > > >>>> FreeSWITCH-users mailing list > > >>>> FreeSWITCH-users at lists.freeswitch.org > > > > > > >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>>> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>>> http://www.freeswitch.org > > >>>> > > >> > > > _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: > > >> consulting at freeswitch.org > > > > > >> http://www.freeswitchsolutions.com > > >> > > >> > > >> > > >> > > >> Official FreeSWITCH Sites > > >> http://www.freeswitch.org > > >> http://wiki.freeswitch.org > > >> http://www.cluecon.com > > >> > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > >> > > >> > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ------------------------------------------------------------------------ > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------------------------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ga at steadfasttelecom.com Tue Mar 6 19:16:05 2012 From: ga at steadfasttelecom.com (Gilad Abada) Date: Tue, 6 Mar 2012 11:16:05 -0500 Subject: [Freeswitch-users] Calls drop after 1:48 (kinda) In-Reply-To: References: <1330644898790-7334603.post@n2.nabble.com> Message-ID: Hey I just wanted to keep you posted on progress. I downgraded to this version hash ID: 77c01bc4b7e8b455cc4c73ec6263b395a1823cbb and its working looks like a bug in the latest. On Tue, Mar 6, 2012 at 9:42 AM, Gilad Abada wrote: > Here is my CLI debug from when the call drops. The extension that is > called out is 1003. I also have pcaps if they are needed. > > 2012-03-06 09:24:27.580387 [DEBUG] switch_core_session.c:875 Send > signal sofia/internal/1003 at example.domain [BREAK] > 2012-03-06 09:24:27.580387 [DEBUG] switch_core_session.c:875 Send > signal sofia/internal/1003 at example.domain [BREAK] > 2012-03-06 09:24:27.580387 [DEBUG] switch_core_session.c:875 Send > signal sofia/internal/1003 at example.domain [BREAK] > 2012-03-06 09:24:27.590418 [DEBUG] sofia.c:5532 Channel > sofia/internal/1003 at example.domain entering state [terminated][200] > 2012-03-06 09:24:27.590418 [DEBUG] switch_channel.c:2848 > (sofia/internal/1003 at example.domain) Callstate Change ACTIVE -> HANGUP > 2012-03-06 09:24:27.590418 [NOTICE] sofia.c:6299 Hangup > sofia/internal/1003 at example.domain [CS_EXECUTE] [NORMAL_CLEARING] > 2012-03-06 09:24:27.590418 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/1003 at example.domain [KILL] > 2012-03-06 09:24:27.590418 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1003 at example.domain [BREAK] > 2012-03-06 09:24:27.590418 [DEBUG] switch_ivr_bridge.c:586 BRIDGE > THREAD DONE [sofia/internal/1003 at example.domain] > 2012-03-06 09:24:27.590418 [DEBUG] switch_ivr_bridge.c:611 Send signal > sofia/external/XXXXXXXXXXX [BREAK] > 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:499 > sofia/internal/1003 at example.domain ending bridge by request from write > function > 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:586 BRIDGE > THREAD DONE [sofia/external/XXXXXXXXXXX] > 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:611 Send signal > sofia/internal/1003 at example.domain [BREAK] > 2012-03-06 09:24:27.610398 [DEBUG] switch_channel.c:2848 > (sofia/external/XXXXXXXXXXX) Callstate Change ACTIVE -> HANGUP > 2012-03-06 09:24:27.610398 [NOTICE] switch_ivr_bridge.c:669 Hangup > sofia/external/XXXXXXXXXXX [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2012-03-06 09:24:27.610398 [DEBUG] switch_channel.c:2871 Send signal > sofia/external/XXXXXXXXXXX [KILL] > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:1180 Send > signal sofia/external/XXXXXXXXXXX [BREAK] > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:420 > (sofia/external/XXXXXXXXXXX) State EXCHANGE_MEDIA going to sleep > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/XXXXXXXXXXX) Running State Change CS_HANGUP > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 > (sofia/external/XXXXXXXXXXX) State HANGUP > 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:463 > sofia/external/XXXXXXXXXXX Overriding SIP cause 480 with 200 from the > other leg > 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:469 Channel > sofia/external/XXXXXXXXXXX hanging up, cause: NORMAL_CLEARING > 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:513 Sending BYE to > sofia/external/XXXXXXXXXXX > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:47 > sofia/external/XXXXXXXXXXX Standard HANGUP, cause: NORMAL_CLEARING > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 > (sofia/external/XXXXXXXXXXX) State HANGUP going to sleep > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:393 > (sofia/external/XXXXXXXXXXX) State Change CS_HANGUP -> CS_REPORTING > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:1180 Send > signal sofia/external/XXXXXXXXXXX [BREAK] > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/XXXXXXXXXXX) Running State Change CS_REPORTING > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:662 > (sofia/external/XXXXXXXXXXX) State REPORTING > 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:1403 > sofia/external/XXXXXXXXXXX skip receive message [UNBRIDGE] (channel is > hungup already) > 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:1406 > sofia/internal/1003 at example.domain skip receive message [UNBRIDGE] > (channel is hungup already) > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:2285 > sofia/internal/1003 at example.domain skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/1003 at example.domain) State EXECUTE going to sleep > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1003 at example.domain) Running State Change CS_HANGUP > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1003 at example.domain) State HANGUP > 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/1003 at example.domain hanging up, cause: NORMAL_CLEARING > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/1003 at example.domain Standard HANGUP, cause: > NORMAL_CLEARING > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1003 at example.domain) State HANGUP going to sleep > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/1003 at example.domain) State Change CS_HANGUP -> > CS_REPORTING > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1003 at example.domain [BREAK] > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1003 at example.domain) Running State Change CS_REPORTING > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1003 at example.domain) State REPORTING > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:79 > sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:662 > (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:387 > (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_session.c:1180 Send > signal sofia/external/XXXXXXXXXXX [BREAK] > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_session.c:1380 Session > 4 (sofia/external/XXXXXXXXXXX) Locked, Waiting on external entities > 2012-03-06 09:24:27.630498 [NOTICE] switch_core_session.c:1398 Session > 4 (sofia/external/XXXXXXXXXXX) Ended > 2012-03-06 09:24:27.630498 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/external/XXXXXXXXXXX [CS_DESTROY] > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:491 > (sofia/external/XXXXXXXXXXX) Callstate Change HANGUP -> DOWN > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:494 > (sofia/external/XXXXXXXXXXX) Running State Change CS_DESTROY > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:504 > (sofia/external/XXXXXXXXXXX) State DESTROY > 2012-03-06 09:24:27.630498 [DEBUG] mod_sofia.c:374 > sofia/external/XXXXXXXXXXX SOFIA DESTROY > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:86 > sofia/external/XXXXXXXXXXX Standard DESTROY > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:504 > (sofia/external/XXXXXXXXXXX) State DESTROY going to sleep > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/1003 at example.domain Standard REPORTING, cause: > NORMAL_CLEARING > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1003 at example.domain) State REPORTING going to sleep > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/1003 at example.domain) State Change CS_REPORTING -> > CS_DESTROY > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1003 at example.domain [BREAK] > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_session.c:1380 Session > 3 (sofia/internal/1003 at example.domain) Locked, Waiting on external > entities > 2012-03-06 09:24:27.640384 [NOTICE] switch_core_session.c:1398 Session > 3 (sofia/internal/1003 at example.domain) Ended > 2012-03-06 09:24:27.640384 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/1003 at example.domain [CS_DESTROY] > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/1003 at example.domain) Callstate Change HANGUP -> DOWN > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/1003 at example.domain) Running State Change CS_DESTROY > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1003 at example.domain) State DESTROY > 2012-03-06 09:24:27.640384 [DEBUG] mod_sofia.c:374 > sofia/internal/1003 at example.domain SOFIA DESTROY > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/1003 at example.domain Standard DESTROY > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1003 at example.domain) State DESTROY going to sleep > 2012-03-06 09:24:28.190377 [DEBUG] switch_core_session.c:875 Send > signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] > 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:499 > sofia/external/+XXXXXXXXXXX at flowroute.com ending bridge by request > from write function > 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:586 BRIDGE > THREAD DONE [sofia/internal/sip:1020 at xxx.xxx.xxx.xxx] > 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:611 Send signal > sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] > 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2848 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Callstate Change ACTIVE -> > HANGUP > 2012-03-06 09:24:28.210377 [NOTICE] switch_ivr_bridge.c:669 Hangup > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [CS_EXCHANGE_MEDIA] > [NORMAL_CLEARING] > 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [KILL] > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:420 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State EXCHANGE_MEDIA going > to sleep > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Running State Change > CS_HANGUP > 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2848 > (sofia/external/+XXXXXXXXXXX at flowroute.com) Callstate Change ACTIVE -> > HANGUP > 2012-03-06 09:24:28.210377 [NOTICE] sofia.c:628 Hangup > sofia/external/+XXXXXXXXXXX at flowroute.com [CS_EXECUTE] > [NORMAL_CLEARING] > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State HANGUP > 2012-03-06 09:24:28.210377 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx hanging up, cause: > NORMAL_CLEARING > 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2871 Send signal > sofia/external/+XXXXXXXXXXX at flowroute.com [KILL] > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send > signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] > 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:586 BRIDGE > THREAD DONE [sofia/external/+XXXXXXXXXXX at flowroute.com] > 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:611 Send signal > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] > 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:1403 > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx skip receive message > [UNBRIDGE] (channel is hungup already) > 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:1406 > sofia/external/+XXXXXXXXXXX at flowroute.com skip receive message > [UNBRIDGE] (channel is hungup already) > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:2285 > sofia/external/+XXXXXXXXXXX at flowroute.com skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:2100 > sofia/external/+XXXXXXXXXXX at flowroute.com Channel is hungup and > application 'bridge' does not have the zombie_exec flag. > 2012-03-06 09:24:28.210377 [DEBUG] switch_cpp.cpp:1007 > sofia/external/+XXXXXXXXXXX at flowroute.com destroy/unlink session from > object > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:2285 > sofia/external/+XXXXXXXXXXX at flowroute.com skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:417 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State EXECUTE going to > sleep > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/+XXXXXXXXXXX at flowroute.com) Running State Change > CS_HANGUP > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State HANGUP > 2012-03-06 09:24:28.210377 [DEBUG] mod_sofia.c:469 Channel > sofia/external/+XXXXXXXXXXX at flowroute.com hanging up, cause: > NORMAL_CLEARING > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:47 > sofia/external/+XXXXXXXXXXX at flowroute.com Standard HANGUP, cause: > NORMAL_CLEARING > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State HANGUP going to > sleep > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:393 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State Change CS_HANGUP -> > CS_REPORTING > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send > signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/+XXXXXXXXXXX at flowroute.com) Running State Change > CS_REPORTING > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:662 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State REPORTING > 2012-03-06 09:24:28.210377 [DEBUG] mod_sofia.c:513 Sending BYE to > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx Standard HANGUP, cause: > NORMAL_CLEARING > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State HANGUP going to sleep > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State Change CS_HANGUP -> > CS_REPORTING > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Running State Change > CS_REPORTING > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State REPORTING > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:79 > sofia/external/+XXXXXXXXXXX at flowroute.com Standard REPORTING, cause: > NORMAL_CLEARING > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:662 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State REPORTING going to > sleep > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:387 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State Change CS_REPORTING > -> CS_DESTROY > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_session.c:1180 Send > signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_session.c:1380 Session > 5 (sofia/external/+XXXXXXXXXXX at flowroute.com) Locked, Waiting on > external entities > 2012-03-06 09:24:28.230382 [NOTICE] switch_core_session.c:1398 Session > 5 (sofia/external/+XXXXXXXXXXX at flowroute.com) Ended > 2012-03-06 09:24:28.230382 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/external/+XXXXXXXXXXX at flowroute.com [CS_DESTROY] > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:491 > (sofia/external/+XXXXXXXXXXX at flowroute.com) Callstate Change HANGUP -> > DOWN > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:494 > (sofia/external/+XXXXXXXXXXX at flowroute.com) Running State Change > CS_DESTROY > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:504 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State DESTROY > 2012-03-06 09:24:28.230382 [DEBUG] mod_sofia.c:374 > sofia/external/+XXXXXXXXXXX at flowroute.com SOFIA DESTROY > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:86 > sofia/external/+XXXXXXXXXXX at flowroute.com Standard DESTROY > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:504 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State DESTROY going to > sleep > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx Standard REPORTING, cause: > NORMAL_CLEARING > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State REPORTING going to > sleep > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State Change CS_REPORTING -> > CS_DESTROY > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_session.c:1380 Session > 6 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Locked, Waiting on > external entities > 2012-03-06 09:24:28.240382 [NOTICE] switch_core_session.c:1398 Session > 6 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Ended > 2012-03-06 09:24:28.240382 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [CS_DESTROY] > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Callstate Change HANGUP -> > DOWN > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Running State Change > CS_DESTROY > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State DESTROY > 2012-03-06 09:24:28.240382 [DEBUG] mod_sofia.c:374 > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx SOFIA DESTROY > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx Standard DESTROY > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State DESTROY going to sleep > > On Tue, Mar 6, 2012 at 9:40 AM, Gilad Abada wrote: >> Hi >> >> I am having the same exact issue. I am using Grandstream HT 502's >> >> Has anyone come up with a fix? >> >> Thanks >> Gill >> >> On Fri, Mar 2, 2012 at 3:18 AM, Brent Paddon wrote: >>> Have you captured the network traffic between the phone and FS to see whats >>> there? >>> >>> That will probably determine your next step. >>> >>> Brent >>> >>> >>> On Fri, Mar 2, 2012 at 9:34 AM, BloodyIron wrote: >>>> >>>> Hi Folks, >>>> >>>> Okay so this one is a bit tricky to reproduce. One of our extensions will >>>> have their calls dropped preicsely after 1 minutes 48 seconds ( 1:48 ) of >>>> call time. Another extension on the same segment of the network does not >>>> do >>>> the same thing. This extension does this for every single call type, be it >>>> external or internal calls. >>>> >>>> Now you may think, well it may be a busted phone. We reset the phone to >>>> factory defaults and saw no improvement. Furthermore we are seeing it >>>> elsewhere in our freeswitch installation, as in other extensions on other >>>> sites are seeing the same issue. >>>> >>>> These extensions are behind a NAT, however the freeswitch server is >>>> publically facing (as in public IP), with a passive firewall between it >>>> and >>>> the world (no routing, no NAT, etc). >>>> >>>> Just to be clear, we are also using fusionpbx to control the installation, >>>> as typing everything into an xml is not very efficient (but we're not >>>> scared >>>> of working with xml files either). >>>> >>>> Right now, we're gonna try adding " ?" to our sofia.conf.xml file to >>>> address >>>> it, beyond this we are unsure what to do. >>>> >>>> Can anyone speak on this matter? >>>> >>>> -- >>>> View this message in context: >>>> http://freeswitch-users.2379917.n2.nabble.com/Calls-drop-after-1-48-kinda-tp7334603p7334603.html >>>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Gilad Abada >> >> SteadFast Telecommunications, Inc. >> >> Call us to find out how much you can save with VoIP! >> >> V: 212.589.1001 >> F: 212.589.1011 >> >> >> For 35 years, Steadfast Telecommunications has been providing >> state-of-the-art communications technology to businesses and >> government agencies - large and small. Steadfast Telecommunications >> tailors Unified Communications and Voice-Over IP Solutions to >> single-site offices or multi-site and worldwide enterprises.?? Make >> your virtual office a reality.? Enjoy the freedom to travel while >> remaining connected to your office. > > > > -- > Gilad Abada > > SteadFast Telecommunications, Inc. > > Call us to find out how much you can save with VoIP! > > V: 212.589.1001 > F: 212.589.1011 > > > For 35 years, Steadfast Telecommunications has been providing > state-of-the-art communications technology to businesses and > government agencies - large and small. Steadfast Telecommunications > tailors Unified Communications and Voice-Over IP Solutions to > single-site offices or multi-site and worldwide enterprises.?? Make > your virtual office a reality.? Enjoy the freedom to travel while > remaining connected to your office. -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. From adam.kelloway at newpace.ca Tue Mar 6 19:22:02 2012 From: adam.kelloway at newpace.ca (Adam Kelloway) Date: Tue, 06 Mar 2012 12:22:02 -0400 Subject: [Freeswitch-users] Any alternative to repetitive lua session:ready() calls? Message-ID: <4F5639AA.9050900@newpace.ca> Hi there, If I am running a lua script during a call, and the call terminates (caller hangs up, for instance), the lua script continues to execute while the session has already been terminated. This causes ERR entries in the freeswitch log, which I would like to try and minimize. The only way I can see doing this is to frequently call session:ready() before doing anything that could break the script. Is there a better way to exit a script cleanly, without having to add X number of session:ready() calls scattered about the script? Currently, I only call it once (at the beginning of the script). I tried using the hangup hook, but I'm not sure how I can get it to immediately exit the script. Thanks, Adam From peter.olsson at visionutveckling.se Tue Mar 6 19:33:17 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 6 Mar 2012 16:33:17 +0000 Subject: [Freeswitch-users] Any alternative to repetitive lua session:ready() calls? In-Reply-To: <4F5639AA.9050900@newpace.ca> References: <4F5639AA.9050900@newpace.ca> Message-ID: <1FFF97C269757C458224B7C895F35F1504E178@cantor.std.visionutv.se> Maybe this is what you are looking for? commit 09ad887948f7513725ca8b53bdfe721d9008e73b Author: Anthony Minessale Date: Fri Jan 27 19:03:04 2012 -0600 FS-3841 --resolve ok return the string "die" or "exit" from hanguphook to cause an error or call s:destroy("any err message"); either should now halt the script ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Adam Kelloway [adam.kelloway at newpace.ca] Skickat: den 6 mars 2012 17:22 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Any alternative to repetitive lua session:ready() calls? Hi there, If I am running a lua script during a call, and the call terminates (caller hangs up, for instance), the lua script continues to execute while the session has already been terminated. This causes ERR entries in the freeswitch log, which I would like to try and minimize. The only way I can see doing this is to frequently call session:ready() before doing anything that could break the script. Is there a better way to exit a script cleanly, without having to add X number of session:ready() calls scattered about the script? Currently, I only call it once (at the beginning of the script). I tried using the hangup hook, but I'm not sure how I can get it to immediately exit the script. Thanks, Adam _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f56388f32766945510869! From peter.olsson at visionutveckling.se Tue Mar 6 19:35:02 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 6 Mar 2012 16:35:02 +0000 Subject: [Freeswitch-users] Calls drop after 1:48 (kinda) In-Reply-To: References: <1330644898790-7334603.post@n2.nabble.com> , Message-ID: <1FFF97C269757C458224B7C895F35F1504E184@cantor.std.visionutv.se> That is a quite old commit - so of course lots of changes has been commited since then, it doesn't mean there is a bug in latest, it might as well be settings that has changed... Please pastebin a complete log with SIP trace, and we can have a look at it. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Gilad Abada [ga at steadfasttelecom.com] Skickat: den 6 mars 2012 17:16 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Calls drop after 1:48 (kinda) Hey I just wanted to keep you posted on progress. I downgraded to this version hash ID: 77c01bc4b7e8b455cc4c73ec6263b395a1823cbb and its working looks like a bug in the latest. On Tue, Mar 6, 2012 at 9:42 AM, Gilad Abada wrote: > Here is my CLI debug from when the call drops. The extension that is > called out is 1003. I also have pcaps if they are needed. > > 2012-03-06 09:24:27.580387 [DEBUG] switch_core_session.c:875 Send > signal sofia/internal/1003 at example.domain [BREAK] > 2012-03-06 09:24:27.580387 [DEBUG] switch_core_session.c:875 Send > signal sofia/internal/1003 at example.domain [BREAK] > 2012-03-06 09:24:27.580387 [DEBUG] switch_core_session.c:875 Send > signal sofia/internal/1003 at example.domain [BREAK] > 2012-03-06 09:24:27.590418 [DEBUG] sofia.c:5532 Channel > sofia/internal/1003 at example.domain entering state [terminated][200] > 2012-03-06 09:24:27.590418 [DEBUG] switch_channel.c:2848 > (sofia/internal/1003 at example.domain) Callstate Change ACTIVE -> HANGUP > 2012-03-06 09:24:27.590418 [NOTICE] sofia.c:6299 Hangup > sofia/internal/1003 at example.domain [CS_EXECUTE] [NORMAL_CLEARING] > 2012-03-06 09:24:27.590418 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/1003 at example.domain [KILL] > 2012-03-06 09:24:27.590418 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1003 at example.domain [BREAK] > 2012-03-06 09:24:27.590418 [DEBUG] switch_ivr_bridge.c:586 BRIDGE > THREAD DONE [sofia/internal/1003 at example.domain] > 2012-03-06 09:24:27.590418 [DEBUG] switch_ivr_bridge.c:611 Send signal > sofia/external/XXXXXXXXXXX [BREAK] > 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:499 > sofia/internal/1003 at example.domain ending bridge by request from write > function > 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:586 BRIDGE > THREAD DONE [sofia/external/XXXXXXXXXXX] > 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:611 Send signal > sofia/internal/1003 at example.domain [BREAK] > 2012-03-06 09:24:27.610398 [DEBUG] switch_channel.c:2848 > (sofia/external/XXXXXXXXXXX) Callstate Change ACTIVE -> HANGUP > 2012-03-06 09:24:27.610398 [NOTICE] switch_ivr_bridge.c:669 Hangup > sofia/external/XXXXXXXXXXX [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2012-03-06 09:24:27.610398 [DEBUG] switch_channel.c:2871 Send signal > sofia/external/XXXXXXXXXXX [KILL] > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:1180 Send > signal sofia/external/XXXXXXXXXXX [BREAK] > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:420 > (sofia/external/XXXXXXXXXXX) State EXCHANGE_MEDIA going to sleep > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/XXXXXXXXXXX) Running State Change CS_HANGUP > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 > (sofia/external/XXXXXXXXXXX) State HANGUP > 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:463 > sofia/external/XXXXXXXXXXX Overriding SIP cause 480 with 200 from the > other leg > 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:469 Channel > sofia/external/XXXXXXXXXXX hanging up, cause: NORMAL_CLEARING > 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:513 Sending BYE to > sofia/external/XXXXXXXXXXX > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:47 > sofia/external/XXXXXXXXXXX Standard HANGUP, cause: NORMAL_CLEARING > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 > (sofia/external/XXXXXXXXXXX) State HANGUP going to sleep > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:393 > (sofia/external/XXXXXXXXXXX) State Change CS_HANGUP -> CS_REPORTING > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:1180 Send > signal sofia/external/XXXXXXXXXXX [BREAK] > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/XXXXXXXXXXX) Running State Change CS_REPORTING > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:662 > (sofia/external/XXXXXXXXXXX) State REPORTING > 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:1403 > sofia/external/XXXXXXXXXXX skip receive message [UNBRIDGE] (channel is > hungup already) > 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:1406 > sofia/internal/1003 at example.domain skip receive message [UNBRIDGE] > (channel is hungup already) > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:2285 > sofia/internal/1003 at example.domain skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/1003 at example.domain) State EXECUTE going to sleep > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1003 at example.domain) Running State Change CS_HANGUP > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1003 at example.domain) State HANGUP > 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/1003 at example.domain hanging up, cause: NORMAL_CLEARING > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/1003 at example.domain Standard HANGUP, cause: > NORMAL_CLEARING > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1003 at example.domain) State HANGUP going to sleep > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/1003 at example.domain) State Change CS_HANGUP -> > CS_REPORTING > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1003 at example.domain [BREAK] > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1003 at example.domain) Running State Change CS_REPORTING > 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1003 at example.domain) State REPORTING > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:79 > sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:662 > (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:387 > (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_session.c:1180 Send > signal sofia/external/XXXXXXXXXXX [BREAK] > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_session.c:1380 Session > 4 (sofia/external/XXXXXXXXXXX) Locked, Waiting on external entities > 2012-03-06 09:24:27.630498 [NOTICE] switch_core_session.c:1398 Session > 4 (sofia/external/XXXXXXXXXXX) Ended > 2012-03-06 09:24:27.630498 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/external/XXXXXXXXXXX [CS_DESTROY] > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:491 > (sofia/external/XXXXXXXXXXX) Callstate Change HANGUP -> DOWN > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:494 > (sofia/external/XXXXXXXXXXX) Running State Change CS_DESTROY > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:504 > (sofia/external/XXXXXXXXXXX) State DESTROY > 2012-03-06 09:24:27.630498 [DEBUG] mod_sofia.c:374 > sofia/external/XXXXXXXXXXX SOFIA DESTROY > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:86 > sofia/external/XXXXXXXXXXX Standard DESTROY > 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:504 > (sofia/external/XXXXXXXXXXX) State DESTROY going to sleep > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/1003 at example.domain Standard REPORTING, cause: > NORMAL_CLEARING > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1003 at example.domain) State REPORTING going to sleep > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/1003 at example.domain) State Change CS_REPORTING -> > CS_DESTROY > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1003 at example.domain [BREAK] > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_session.c:1380 Session > 3 (sofia/internal/1003 at example.domain) Locked, Waiting on external > entities > 2012-03-06 09:24:27.640384 [NOTICE] switch_core_session.c:1398 Session > 3 (sofia/internal/1003 at example.domain) Ended > 2012-03-06 09:24:27.640384 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/1003 at example.domain [CS_DESTROY] > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/1003 at example.domain) Callstate Change HANGUP -> DOWN > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/1003 at example.domain) Running State Change CS_DESTROY > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1003 at example.domain) State DESTROY > 2012-03-06 09:24:27.640384 [DEBUG] mod_sofia.c:374 > sofia/internal/1003 at example.domain SOFIA DESTROY > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/1003 at example.domain Standard DESTROY > 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1003 at example.domain) State DESTROY going to sleep > 2012-03-06 09:24:28.190377 [DEBUG] switch_core_session.c:875 Send > signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] > 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:499 > sofia/external/+XXXXXXXXXXX at flowroute.com ending bridge by request > from write function > 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:586 BRIDGE > THREAD DONE [sofia/internal/sip:1020 at xxx.xxx.xxx.xxx] > 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:611 Send signal > sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] > 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2848 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Callstate Change ACTIVE -> > HANGUP > 2012-03-06 09:24:28.210377 [NOTICE] switch_ivr_bridge.c:669 Hangup > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [CS_EXCHANGE_MEDIA] > [NORMAL_CLEARING] > 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [KILL] > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:420 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State EXCHANGE_MEDIA going > to sleep > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Running State Change > CS_HANGUP > 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2848 > (sofia/external/+XXXXXXXXXXX at flowroute.com) Callstate Change ACTIVE -> > HANGUP > 2012-03-06 09:24:28.210377 [NOTICE] sofia.c:628 Hangup > sofia/external/+XXXXXXXXXXX at flowroute.com [CS_EXECUTE] > [NORMAL_CLEARING] > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State HANGUP > 2012-03-06 09:24:28.210377 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx hanging up, cause: > NORMAL_CLEARING > 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2871 Send signal > sofia/external/+XXXXXXXXXXX at flowroute.com [KILL] > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send > signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] > 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:586 BRIDGE > THREAD DONE [sofia/external/+XXXXXXXXXXX at flowroute.com] > 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:611 Send signal > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] > 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:1403 > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx skip receive message > [UNBRIDGE] (channel is hungup already) > 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:1406 > sofia/external/+XXXXXXXXXXX at flowroute.com skip receive message > [UNBRIDGE] (channel is hungup already) > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:2285 > sofia/external/+XXXXXXXXXXX at flowroute.com skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:2100 > sofia/external/+XXXXXXXXXXX at flowroute.com Channel is hungup and > application 'bridge' does not have the zombie_exec flag. > 2012-03-06 09:24:28.210377 [DEBUG] switch_cpp.cpp:1007 > sofia/external/+XXXXXXXXXXX at flowroute.com destroy/unlink session from > object > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:2285 > sofia/external/+XXXXXXXXXXX at flowroute.com skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:417 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State EXECUTE going to > sleep > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/+XXXXXXXXXXX at flowroute.com) Running State Change > CS_HANGUP > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State HANGUP > 2012-03-06 09:24:28.210377 [DEBUG] mod_sofia.c:469 Channel > sofia/external/+XXXXXXXXXXX at flowroute.com hanging up, cause: > NORMAL_CLEARING > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:47 > sofia/external/+XXXXXXXXXXX at flowroute.com Standard HANGUP, cause: > NORMAL_CLEARING > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State HANGUP going to > sleep > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:393 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State Change CS_HANGUP -> > CS_REPORTING > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send > signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/+XXXXXXXXXXX at flowroute.com) Running State Change > CS_REPORTING > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:662 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State REPORTING > 2012-03-06 09:24:28.210377 [DEBUG] mod_sofia.c:513 Sending BYE to > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx Standard HANGUP, cause: > NORMAL_CLEARING > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State HANGUP going to sleep > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State Change CS_HANGUP -> > CS_REPORTING > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Running State Change > CS_REPORTING > 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State REPORTING > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:79 > sofia/external/+XXXXXXXXXXX at flowroute.com Standard REPORTING, cause: > NORMAL_CLEARING > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:662 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State REPORTING going to > sleep > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:387 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State Change CS_REPORTING > -> CS_DESTROY > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_session.c:1180 Send > signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_session.c:1380 Session > 5 (sofia/external/+XXXXXXXXXXX at flowroute.com) Locked, Waiting on > external entities > 2012-03-06 09:24:28.230382 [NOTICE] switch_core_session.c:1398 Session > 5 (sofia/external/+XXXXXXXXXXX at flowroute.com) Ended > 2012-03-06 09:24:28.230382 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/external/+XXXXXXXXXXX at flowroute.com [CS_DESTROY] > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:491 > (sofia/external/+XXXXXXXXXXX at flowroute.com) Callstate Change HANGUP -> > DOWN > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:494 > (sofia/external/+XXXXXXXXXXX at flowroute.com) Running State Change > CS_DESTROY > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:504 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State DESTROY > 2012-03-06 09:24:28.230382 [DEBUG] mod_sofia.c:374 > sofia/external/+XXXXXXXXXXX at flowroute.com SOFIA DESTROY > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:86 > sofia/external/+XXXXXXXXXXX at flowroute.com Standard DESTROY > 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:504 > (sofia/external/+XXXXXXXXXXX at flowroute.com) State DESTROY going to > sleep > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx Standard REPORTING, cause: > NORMAL_CLEARING > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State REPORTING going to > sleep > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State Change CS_REPORTING -> > CS_DESTROY > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_session.c:1380 Session > 6 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Locked, Waiting on > external entities > 2012-03-06 09:24:28.240382 [NOTICE] switch_core_session.c:1398 Session > 6 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Ended > 2012-03-06 09:24:28.240382 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [CS_DESTROY] > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Callstate Change HANGUP -> > DOWN > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Running State Change > CS_DESTROY > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State DESTROY > 2012-03-06 09:24:28.240382 [DEBUG] mod_sofia.c:374 > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx SOFIA DESTROY > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/sip:1020 at xxx.xxx.xxx.xxx Standard DESTROY > 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State DESTROY going to sleep > > On Tue, Mar 6, 2012 at 9:40 AM, Gilad Abada wrote: >> Hi >> >> I am having the same exact issue. I am using Grandstream HT 502's >> >> Has anyone come up with a fix? >> >> Thanks >> Gill >> >> On Fri, Mar 2, 2012 at 3:18 AM, Brent Paddon wrote: >>> Have you captured the network traffic between the phone and FS to see whats >>> there? >>> >>> That will probably determine your next step. >>> >>> Brent >>> >>> >>> On Fri, Mar 2, 2012 at 9:34 AM, BloodyIron wrote: >>>> >>>> Hi Folks, >>>> >>>> Okay so this one is a bit tricky to reproduce. One of our extensions will >>>> have their calls dropped preicsely after 1 minutes 48 seconds ( 1:48 ) of >>>> call time. Another extension on the same segment of the network does not >>>> do >>>> the same thing. This extension does this for every single call type, be it >>>> external or internal calls. >>>> >>>> Now you may think, well it may be a busted phone. We reset the phone to >>>> factory defaults and saw no improvement. Furthermore we are seeing it >>>> elsewhere in our freeswitch installation, as in other extensions on other >>>> sites are seeing the same issue. >>>> >>>> These extensions are behind a NAT, however the freeswitch server is >>>> publically facing (as in public IP), with a passive firewall between it >>>> and >>>> the world (no routing, no NAT, etc). >>>> >>>> Just to be clear, we are also using fusionpbx to control the installation, >>>> as typing everything into an xml is not very efficient (but we're not >>>> scared >>>> of working with xml files either). >>>> >>>> Right now, we're gonna try adding " " to our sofia.conf.xml file to >>>> address >>>> it, beyond this we are unsure what to do. >>>> >>>> Can anyone speak on this matter? >>>> >>>> -- >>>> View this message in context: >>>> http://freeswitch-users.2379917.n2.nabble.com/Calls-drop-after-1-48-kinda-tp7334603p7334603.html >>>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Gilad Abada >> >> SteadFast Telecommunications, Inc. >> >> Call us to find out how much you can save with VoIP! >> >> V: 212.589.1001 >> F: 212.589.1011 >> >> >> For 35 years, Steadfast Telecommunications has been providing >> state-of-the-art communications technology to businesses and >> government agencies - large and small. Steadfast Telecommunications >> tailors Unified Communications and Voice-Over IP Solutions to >> single-site offices or multi-site and worldwide enterprises. Make >> your virtual office a reality. Enjoy the freedom to travel while >> remaining connected to your office. > > > > -- > Gilad Abada > > SteadFast Telecommunications, Inc. > > Call us to find out how much you can save with VoIP! > > V: 212.589.1001 > F: 212.589.1011 > > > For 35 years, Steadfast Telecommunications has been providing > state-of-the-art communications technology to businesses and > government agencies - large and small. Steadfast Telecommunications > tailors Unified Communications and Voice-Over IP Solutions to > single-site offices or multi-site and worldwide enterprises. Make > your virtual office a reality. Enjoy the freedom to travel while > remaining connected to your office. -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises. Make your virtual office a reality. Enjoy the freedom to travel while remaining connected to your office. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f5637ab32761170864652! From fs-list at communicatefreely.net Tue Mar 6 19:39:02 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Tue, 06 Mar 2012 11:39:02 -0500 Subject: [Freeswitch-users] MWI does not working anymore? In-Reply-To: References: Message-ID: <4F563DA6.6020400@communicatefreely.net> So how do we set it up so that phones registered to a profile other than the one mapped in the domain can receive MWI? In the registration database, there is an MWI user and an MWI host. These seem to be correct no matter where the phone registers, and the profile is also a field in that database. Is there some way for sofia to listen for MESSAGE_WAITING events and send it to any endpoint that has a matching mwi host and domain? For what it's worth, I did try an explicit MWI subscription from the phone, and it still didn't work properly if the phone wasn't on the profile that the message domain was aliased to. If this is a modification, I would be willing to post a bounty. -Tim Anthony Minessale wrote: > alias allows you to map domains to your profile so the domain can be > used to find that profile in place of the profile name. > so in the case of mwi it looks up the domain name and the alias allows > it to resolve the domain to the correct profile. > > similar to dns only for sofia profiles. > > On Fri, Feb 17, 2012 at 8:02 AM, Yehavi Bourvine > wrote: > >> I played around with the ALIAS definitions and it seems to solve the >> problem. >> >> I've tried searching the WIKI for a description of ALIAS use and did not >> find. Anyone can explain what does the alias do and what is the correct >> usage when you have more than one profile? >> >> Thanks, __Yehavi: >> >> >> 2012/2/17 Yehavi Bourvine >> >>> Hello Anthony, >>> >>> I know, and I've put a comment there that it still does not work. I am >>> downloading the recent GIT every two days to my test system (the last one >>> was yesterday) and it still behaves the same. >>> >>> >>> Thanks, __Yehavi: >>> >>> 2012/2/17 Anthony Minessale >>> >>>> you opened a jira on it and i fixed it and committed it >>>> >>>> >>>> [FS-3866] MWI is not updated >>>> Reporter: Yehavi Bourvine [yehavi] >>>> Assignee: Anthony Minessale II [anthm] >>>> Status: Resolved >>>> http://jira.freeswitch.org/browse/FS-3866 >>>> >>>> commit ff379a97e57a94f11ce75e8894b1ceb79d10a9ed >>>> Author: Anthony Minessale >>>> Date: Fri Feb 10 11:02:41 2012 -0600 >>>> >>>> >>>> >>>> On Thu, Feb 16, 2012 at 2:56 AM, Yehavi Bourvine >>>> wrote: >>>> >>>>> Hello, >>>>> >>>>> I've noticed a problem started in the last two weeks, but I don't see >>>>> anyone else reporting it... >>>>> >>>>> MWI status is not updated anymore. It is only updated at the initial >>>>> boot of >>>>> the phone. Do others that use GIT from the last two weeks notice it? >>>>> >>>>> Thanks, __Yehavi: >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > From anthony.minessale at gmail.com Tue Mar 6 19:44:37 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 6 Mar 2012 10:44:37 -0600 Subject: [Freeswitch-users] Calls drop after 1:48 (kinda) In-Reply-To: <1FFF97C269757C458224B7C895F35F1504E184@cantor.std.visionutv.se> References: <1330644898790-7334603.post@n2.nabble.com> <1FFF97C269757C458224B7C895F35F1504E184@cantor.std.visionutv.se> Message-ID: its going to be session timers we changed the direction of session timers and the phone is probably broken and it worked before cos we were doing it but now we push the uac to do it since it helps fight nat the phone probably just hard coded the header saying they support it like morons On Tue, Mar 6, 2012 at 10:35 AM, Peter Olsson wrote: > That is a quite old commit - so of course lots of changes has been commited since then, it doesn't mean there is a bug in latest, it might as well be settings that has changed... > > Please pastebin a complete log with SIP trace, and we can have a look at it. > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Gilad Abada [ga at steadfasttelecom.com] > Skickat: den 6 mars 2012 17:16 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Calls drop after 1:48 (kinda) > > Hey > > I just wanted to keep you posted on progress. I downgraded to this > version hash ID: 77c01bc4b7e8b455cc4c73ec6263b395a1823cbb > > and its working looks like a bug in the latest. > > > > On Tue, Mar 6, 2012 at 9:42 AM, Gilad Abada wrote: >> Here is my CLI debug from when the call drops. The extension that is >> called out is 1003. I also have pcaps if they are needed. >> >> 2012-03-06 09:24:27.580387 [DEBUG] switch_core_session.c:875 Send >> signal sofia/internal/1003 at example.domain [BREAK] >> 2012-03-06 09:24:27.580387 [DEBUG] switch_core_session.c:875 Send >> signal sofia/internal/1003 at example.domain [BREAK] >> 2012-03-06 09:24:27.580387 [DEBUG] switch_core_session.c:875 Send >> signal sofia/internal/1003 at example.domain [BREAK] >> 2012-03-06 09:24:27.590418 [DEBUG] sofia.c:5532 Channel >> sofia/internal/1003 at example.domain entering state [terminated][200] >> 2012-03-06 09:24:27.590418 [DEBUG] switch_channel.c:2848 >> (sofia/internal/1003 at example.domain) Callstate Change ACTIVE -> HANGUP >> 2012-03-06 09:24:27.590418 [NOTICE] sofia.c:6299 Hangup >> sofia/internal/1003 at example.domain [CS_EXECUTE] [NORMAL_CLEARING] >> 2012-03-06 09:24:27.590418 [DEBUG] switch_channel.c:2871 Send signal >> sofia/internal/1003 at example.domain [KILL] >> 2012-03-06 09:24:27.590418 [DEBUG] switch_core_session.c:1180 Send >> signal sofia/internal/1003 at example.domain [BREAK] >> 2012-03-06 09:24:27.590418 [DEBUG] switch_ivr_bridge.c:586 BRIDGE >> THREAD DONE [sofia/internal/1003 at example.domain] >> 2012-03-06 09:24:27.590418 [DEBUG] switch_ivr_bridge.c:611 Send signal >> sofia/external/XXXXXXXXXXX [BREAK] >> 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:499 >> sofia/internal/1003 at example.domain ending bridge by request from write >> function >> 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:586 BRIDGE >> THREAD DONE [sofia/external/XXXXXXXXXXX] >> 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:611 Send signal >> sofia/internal/1003 at example.domain [BREAK] >> 2012-03-06 09:24:27.610398 [DEBUG] switch_channel.c:2848 >> (sofia/external/XXXXXXXXXXX) Callstate Change ACTIVE -> HANGUP >> 2012-03-06 09:24:27.610398 [NOTICE] switch_ivr_bridge.c:669 Hangup >> sofia/external/XXXXXXXXXXX [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> 2012-03-06 09:24:27.610398 [DEBUG] switch_channel.c:2871 Send signal >> sofia/external/XXXXXXXXXXX [KILL] >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:1180 Send >> signal sofia/external/XXXXXXXXXXX [BREAK] >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:420 >> (sofia/external/XXXXXXXXXXX) State EXCHANGE_MEDIA going to sleep >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 >> (sofia/external/XXXXXXXXXXX) Running State Change CS_HANGUP >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 >> (sofia/external/XXXXXXXXXXX) State HANGUP >> 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:463 >> sofia/external/XXXXXXXXXXX Overriding SIP cause 480 with 200 from the >> other leg >> 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:469 Channel >> sofia/external/XXXXXXXXXXX hanging up, cause: NORMAL_CLEARING >> 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:513 Sending BYE to >> sofia/external/XXXXXXXXXXX >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:47 >> sofia/external/XXXXXXXXXXX Standard HANGUP, cause: NORMAL_CLEARING >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 >> (sofia/external/XXXXXXXXXXX) State HANGUP going to sleep >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:393 >> (sofia/external/XXXXXXXXXXX) State Change CS_HANGUP -> CS_REPORTING >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:1180 Send >> signal sofia/external/XXXXXXXXXXX [BREAK] >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 >> (sofia/external/XXXXXXXXXXX) Running State Change CS_REPORTING >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:662 >> (sofia/external/XXXXXXXXXXX) State REPORTING >> 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:1403 >> sofia/external/XXXXXXXXXXX skip receive message [UNBRIDGE] (channel is >> hungup already) >> 2012-03-06 09:24:27.610398 [DEBUG] switch_ivr_bridge.c:1406 >> sofia/internal/1003 at example.domain skip receive message [UNBRIDGE] >> (channel is hungup already) >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:2285 >> sofia/internal/1003 at example.domain skip receive message >> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:417 >> (sofia/internal/1003 at example.domain) State EXECUTE going to sleep >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 >> (sofia/internal/1003 at example.domain) Running State Change CS_HANGUP >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 >> (sofia/internal/1003 at example.domain) State HANGUP >> 2012-03-06 09:24:27.610398 [DEBUG] mod_sofia.c:469 Channel >> sofia/internal/1003 at example.domain hanging up, cause: NORMAL_CLEARING >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:47 >> sofia/internal/1003 at example.domain Standard HANGUP, cause: >> NORMAL_CLEARING >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:602 >> (sofia/internal/1003 at example.domain) State HANGUP going to sleep >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:393 >> (sofia/internal/1003 at example.domain) State Change CS_HANGUP -> >> CS_REPORTING >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_session.c:1180 Send >> signal sofia/internal/1003 at example.domain [BREAK] >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:362 >> (sofia/internal/1003 at example.domain) Running State Change CS_REPORTING >> 2012-03-06 09:24:27.610398 [DEBUG] switch_core_state_machine.c:662 >> (sofia/internal/1003 at example.domain) State REPORTING >> 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:79 >> sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING >> 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:662 >> (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep >> 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:387 >> (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY >> 2012-03-06 09:24:27.630498 [DEBUG] switch_core_session.c:1180 Send >> signal sofia/external/XXXXXXXXXXX [BREAK] >> 2012-03-06 09:24:27.630498 [DEBUG] switch_core_session.c:1380 Session >> 4 (sofia/external/XXXXXXXXXXX) Locked, Waiting on external entities >> 2012-03-06 09:24:27.630498 [NOTICE] switch_core_session.c:1398 Session >> 4 (sofia/external/XXXXXXXXXXX) Ended >> 2012-03-06 09:24:27.630498 [NOTICE] switch_core_session.c:1400 Close >> Channel sofia/external/XXXXXXXXXXX [CS_DESTROY] >> 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:491 >> (sofia/external/XXXXXXXXXXX) Callstate Change HANGUP -> DOWN >> 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:494 >> (sofia/external/XXXXXXXXXXX) Running State Change CS_DESTROY >> 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:504 >> (sofia/external/XXXXXXXXXXX) State DESTROY >> 2012-03-06 09:24:27.630498 [DEBUG] mod_sofia.c:374 >> sofia/external/XXXXXXXXXXX SOFIA DESTROY >> 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:86 >> sofia/external/XXXXXXXXXXX Standard DESTROY >> 2012-03-06 09:24:27.630498 [DEBUG] switch_core_state_machine.c:504 >> (sofia/external/XXXXXXXXXXX) State DESTROY going to sleep >> 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:79 >> sofia/internal/1003 at example.domain Standard REPORTING, cause: >> NORMAL_CLEARING >> 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:662 >> (sofia/internal/1003 at example.domain) State REPORTING going to sleep >> 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:387 >> (sofia/internal/1003 at example.domain) State Change CS_REPORTING -> >> CS_DESTROY >> 2012-03-06 09:24:27.640384 [DEBUG] switch_core_session.c:1180 Send >> signal sofia/internal/1003 at example.domain [BREAK] >> 2012-03-06 09:24:27.640384 [DEBUG] switch_core_session.c:1380 Session >> 3 (sofia/internal/1003 at example.domain) Locked, Waiting on external >> entities >> 2012-03-06 09:24:27.640384 [NOTICE] switch_core_session.c:1398 Session >> 3 (sofia/internal/1003 at example.domain) Ended >> 2012-03-06 09:24:27.640384 [NOTICE] switch_core_session.c:1400 Close >> Channel sofia/internal/1003 at example.domain [CS_DESTROY] >> 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:491 >> (sofia/internal/1003 at example.domain) Callstate Change HANGUP -> DOWN >> 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:494 >> (sofia/internal/1003 at example.domain) Running State Change CS_DESTROY >> 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:504 >> (sofia/internal/1003 at example.domain) State DESTROY >> 2012-03-06 09:24:27.640384 [DEBUG] mod_sofia.c:374 >> sofia/internal/1003 at example.domain SOFIA DESTROY >> 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:86 >> sofia/internal/1003 at example.domain Standard DESTROY >> 2012-03-06 09:24:27.640384 [DEBUG] switch_core_state_machine.c:504 >> (sofia/internal/1003 at example.domain) State DESTROY going to sleep >> 2012-03-06 09:24:28.190377 [DEBUG] switch_core_session.c:875 Send >> signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] >> 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:499 >> sofia/external/+XXXXXXXXXXX at flowroute.com ending bridge by request >> from write function >> 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:586 BRIDGE >> THREAD DONE [sofia/internal/sip:1020 at xxx.xxx.xxx.xxx] >> 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:611 Send signal >> sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] >> 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2848 >> (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Callstate Change ACTIVE -> >> HANGUP >> 2012-03-06 09:24:28.210377 [NOTICE] switch_ivr_bridge.c:669 Hangup >> sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [CS_EXCHANGE_MEDIA] >> [NORMAL_CLEARING] >> 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2871 Send signal >> sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [KILL] >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send >> signal sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:420 >> (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State EXCHANGE_MEDIA going >> to sleep >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 >> (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Running State Change >> CS_HANGUP >> 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2848 >> (sofia/external/+XXXXXXXXXXX at flowroute.com) Callstate Change ACTIVE -> >> HANGUP >> 2012-03-06 09:24:28.210377 [NOTICE] sofia.c:628 Hangup >> sofia/external/+XXXXXXXXXXX at flowroute.com [CS_EXECUTE] >> [NORMAL_CLEARING] >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 >> (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State HANGUP >> 2012-03-06 09:24:28.210377 [DEBUG] mod_sofia.c:469 Channel >> sofia/internal/sip:1020 at xxx.xxx.xxx.xxx hanging up, cause: >> NORMAL_CLEARING >> 2012-03-06 09:24:28.210377 [DEBUG] switch_channel.c:2871 Send signal >> sofia/external/+XXXXXXXXXXX at flowroute.com [KILL] >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send >> signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] >> 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:586 BRIDGE >> THREAD DONE [sofia/external/+XXXXXXXXXXX at flowroute.com] >> 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:611 Send signal >> sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] >> 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:1403 >> sofia/internal/sip:1020 at xxx.xxx.xxx.xxx skip receive message >> [UNBRIDGE] (channel is hungup already) >> 2012-03-06 09:24:28.210377 [DEBUG] switch_ivr_bridge.c:1406 >> sofia/external/+XXXXXXXXXXX at flowroute.com skip receive message >> [UNBRIDGE] (channel is hungup already) >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:2285 >> sofia/external/+XXXXXXXXXXX at flowroute.com skip receive message >> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:2100 >> sofia/external/+XXXXXXXXXXX at flowroute.com Channel is hungup and >> application 'bridge' does not have the zombie_exec flag. >> 2012-03-06 09:24:28.210377 [DEBUG] switch_cpp.cpp:1007 >> sofia/external/+XXXXXXXXXXX at flowroute.com destroy/unlink session from >> object >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:2285 >> sofia/external/+XXXXXXXXXXX at flowroute.com skip receive message >> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:417 >> (sofia/external/+XXXXXXXXXXX at flowroute.com) State EXECUTE going to >> sleep >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 >> (sofia/external/+XXXXXXXXXXX at flowroute.com) Running State Change >> CS_HANGUP >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 >> (sofia/external/+XXXXXXXXXXX at flowroute.com) State HANGUP >> 2012-03-06 09:24:28.210377 [DEBUG] mod_sofia.c:469 Channel >> sofia/external/+XXXXXXXXXXX at flowroute.com hanging up, cause: >> NORMAL_CLEARING >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:47 >> sofia/external/+XXXXXXXXXXX at flowroute.com Standard HANGUP, cause: >> NORMAL_CLEARING >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 >> (sofia/external/+XXXXXXXXXXX at flowroute.com) State HANGUP going to >> sleep >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:393 >> (sofia/external/+XXXXXXXXXXX at flowroute.com) State Change CS_HANGUP -> >> CS_REPORTING >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send >> signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 >> (sofia/external/+XXXXXXXXXXX at flowroute.com) Running State Change >> CS_REPORTING >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:662 >> (sofia/external/+XXXXXXXXXXX at flowroute.com) State REPORTING >> 2012-03-06 09:24:28.210377 [DEBUG] mod_sofia.c:513 Sending BYE to >> sofia/internal/sip:1020 at xxx.xxx.xxx.xxx >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:47 >> sofia/internal/sip:1020 at xxx.xxx.xxx.xxx Standard HANGUP, cause: >> NORMAL_CLEARING >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:602 >> (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State HANGUP going to sleep >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:393 >> (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State Change CS_HANGUP -> >> CS_REPORTING >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_session.c:1180 Send >> signal sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:362 >> (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Running State Change >> CS_REPORTING >> 2012-03-06 09:24:28.210377 [DEBUG] switch_core_state_machine.c:662 >> (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State REPORTING >> 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:79 >> sofia/external/+XXXXXXXXXXX at flowroute.com Standard REPORTING, cause: >> NORMAL_CLEARING >> 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:662 >> (sofia/external/+XXXXXXXXXXX at flowroute.com) State REPORTING going to >> sleep >> 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:387 >> (sofia/external/+XXXXXXXXXXX at flowroute.com) State Change CS_REPORTING >> -> CS_DESTROY >> 2012-03-06 09:24:28.230382 [DEBUG] switch_core_session.c:1180 Send >> signal sofia/external/+XXXXXXXXXXX at flowroute.com [BREAK] >> 2012-03-06 09:24:28.230382 [DEBUG] switch_core_session.c:1380 Session >> 5 (sofia/external/+XXXXXXXXXXX at flowroute.com) Locked, Waiting on >> external entities >> 2012-03-06 09:24:28.230382 [NOTICE] switch_core_session.c:1398 Session >> 5 (sofia/external/+XXXXXXXXXXX at flowroute.com) Ended >> 2012-03-06 09:24:28.230382 [NOTICE] switch_core_session.c:1400 Close >> Channel sofia/external/+XXXXXXXXXXX at flowroute.com [CS_DESTROY] >> 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:491 >> (sofia/external/+XXXXXXXXXXX at flowroute.com) Callstate Change HANGUP -> >> DOWN >> 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:494 >> (sofia/external/+XXXXXXXXXXX at flowroute.com) Running State Change >> CS_DESTROY >> 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:504 >> (sofia/external/+XXXXXXXXXXX at flowroute.com) State DESTROY >> 2012-03-06 09:24:28.230382 [DEBUG] mod_sofia.c:374 >> sofia/external/+XXXXXXXXXXX at flowroute.com SOFIA DESTROY >> 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:86 >> sofia/external/+XXXXXXXXXXX at flowroute.com Standard DESTROY >> 2012-03-06 09:24:28.230382 [DEBUG] switch_core_state_machine.c:504 >> (sofia/external/+XXXXXXXXXXX at flowroute.com) State DESTROY going to >> sleep >> 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:79 >> sofia/internal/sip:1020 at xxx.xxx.xxx.xxx Standard REPORTING, cause: >> NORMAL_CLEARING >> 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:662 >> (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State REPORTING going to >> sleep >> 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:387 >> (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State Change CS_REPORTING -> >> CS_DESTROY >> 2012-03-06 09:24:28.240382 [DEBUG] switch_core_session.c:1180 Send >> signal sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [BREAK] >> 2012-03-06 09:24:28.240382 [DEBUG] switch_core_session.c:1380 Session >> 6 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Locked, Waiting on >> external entities >> 2012-03-06 09:24:28.240382 [NOTICE] switch_core_session.c:1398 Session >> 6 (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Ended >> 2012-03-06 09:24:28.240382 [NOTICE] switch_core_session.c:1400 Close >> Channel sofia/internal/sip:1020 at xxx.xxx.xxx.xxx [CS_DESTROY] >> 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:491 >> (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Callstate Change HANGUP -> >> DOWN >> 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:494 >> (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) Running State Change >> CS_DESTROY >> 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:504 >> (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State DESTROY >> 2012-03-06 09:24:28.240382 [DEBUG] mod_sofia.c:374 >> sofia/internal/sip:1020 at xxx.xxx.xxx.xxx SOFIA DESTROY >> 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:86 >> sofia/internal/sip:1020 at xxx.xxx.xxx.xxx Standard DESTROY >> 2012-03-06 09:24:28.240382 [DEBUG] switch_core_state_machine.c:504 >> (sofia/internal/sip:1020 at xxx.xxx.xxx.xxx) State DESTROY going to sleep >> >> On Tue, Mar 6, 2012 at 9:40 AM, Gilad Abada wrote: >>> Hi >>> >>> I am having the same exact issue. I am using Grandstream HT 502's >>> >>> Has anyone come up with a fix? >>> >>> Thanks >>> Gill >>> >>> On Fri, Mar 2, 2012 at 3:18 AM, Brent Paddon wrote: >>>> Have you captured the network traffic between the phone and FS to see whats >>>> there? >>>> >>>> That will probably determine your next step. >>>> >>>> Brent >>>> >>>> >>>> On Fri, Mar 2, 2012 at 9:34 AM, BloodyIron wrote: >>>>> >>>>> Hi Folks, >>>>> >>>>> Okay so this one is a bit tricky to reproduce. One of our extensions will >>>>> have their calls dropped preicsely after 1 minutes 48 seconds ( 1:48 ) of >>>>> call time. Another extension on the same segment of the network does not >>>>> do >>>>> the same thing. This extension does this for every single call type, be it >>>>> external or internal calls. >>>>> >>>>> Now you may think, well it may be a busted phone. We reset the phone to >>>>> factory defaults and saw no improvement. Furthermore we are seeing it >>>>> elsewhere in our freeswitch installation, as in other extensions on other >>>>> sites are seeing the same issue. >>>>> >>>>> These extensions are behind a NAT, however the freeswitch server is >>>>> publically facing (as in public IP), with a passive firewall between it >>>>> and >>>>> the world (no routing, no NAT, etc). >>>>> >>>>> Just to be clear, we are also using fusionpbx to control the installation, >>>>> as typing everything into an xml is not very efficient (but we're not >>>>> scared >>>>> of working with xml files either). >>>>> >>>>> Right now, we're gonna try adding " ?" to our sofia.conf.xml file to >>>>> address >>>>> it, beyond this we are unsure what to do. >>>>> >>>>> Can anyone speak on this matter? >>>>> >>>>> -- >>>>> View this message in context: >>>>> http://freeswitch-users.2379917.n2.nabble.com/Calls-drop-after-1-48-kinda-tp7334603p7334603.html >>>>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Gilad Abada >>> >>> SteadFast Telecommunications, Inc. >>> >>> Call us to find out how much you can save with VoIP! >>> >>> V: 212.589.1001 >>> F: 212.589.1011 >>> >>> >>> For 35 years, Steadfast Telecommunications has been providing >>> state-of-the-art communications technology to businesses and >>> government agencies - large and small. Steadfast Telecommunications >>> tailors Unified Communications and Voice-Over IP Solutions to >>> single-site offices or multi-site and worldwide enterprises. ? Make >>> your virtual office a reality. ?Enjoy the freedom to travel while >>> remaining connected to your office. >> >> >> >> -- >> Gilad Abada >> >> SteadFast Telecommunications, Inc. >> >> Call us to find out how much you can save with VoIP! >> >> V: 212.589.1001 >> F: 212.589.1011 >> >> >> For 35 years, Steadfast Telecommunications has been providing >> state-of-the-art communications technology to businesses and >> government agencies - large and small. Steadfast Telecommunications >> tailors Unified Communications and Voice-Over IP Solutions to >> single-site offices or multi-site and worldwide enterprises. ? Make >> your virtual office a reality. ?Enjoy the freedom to travel while >> remaining connected to your office. > > > > -- > Gilad Abada > > SteadFast Telecommunications, Inc. > > Call us to find out how much you can save with VoIP! > > V: 212.589.1001 > F: 212.589.1011 > > > For 35 years, Steadfast Telecommunications has been providing > state-of-the-art communications technology to businesses and > government agencies - large and small. Steadfast Telecommunications > tailors Unified Communications and Voice-Over IP Solutions to > single-site offices or multi-site and worldwide enterprises. ? Make > your virtual office a reality. ?Enjoy the freedom to travel while > remaining connected to your office. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f5637ab32761170864652! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From adam.kelloway at newpace.ca Tue Mar 6 20:36:01 2012 From: adam.kelloway at newpace.ca (Adam Kelloway) Date: Tue, 06 Mar 2012 13:36:01 -0400 Subject: [Freeswitch-users] Any alternative to repetitive lua session:ready() calls? In-Reply-To: <1FFF97C269757C458224B7C895F35F1504E178@cantor.std.visionutv.se> References: <4F5639AA.9050900@newpace.ca> <1FFF97C269757C458224B7C895F35F1504E178@cantor.std.visionutv.se> Message-ID: <4F564B01.7050404@newpace.ca> Kind of, all of those solutions consider the script exiting an 'error'. I am looking for something that exits without considering it an error and thus does not emit an [ERR] log. On 3:59 PM, Peter Olsson wrote: > Maybe this is what you are looking for? > > commit 09ad887948f7513725ca8b53bdfe721d9008e73b > Author: Anthony Minessale > Date: Fri Jan 27 19:03:04 2012 -0600 > > FS-3841 --resolve ok return the string "die" or "exit" from hanguphook to cause an error or call s:destroy("any err message"); either should now halt the script > > > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Adam Kelloway [adam.kelloway at newpace.ca] > Skickat: den 6 mars 2012 17:22 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] Any alternative to repetitive lua session:ready() calls? > > Hi there, > > If I am running a lua script during a call, and the call terminates > (caller hangs up, for instance), the lua script continues to execute > while the session has already been terminated. This causes ERR entries > in the freeswitch log, which I would like to try and minimize. The only > way I can see doing this is to frequently call session:ready() before > doing anything that could break the script. Is there a better way to > exit a script cleanly, without having to add X number of session:ready() > calls scattered about the script? Currently, I only call it once (at the > beginning of the script). I tried using the hangup hook, but I'm not > sure how I can get it to immediately exit the script. > > Thanks, > > Adam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f56388f32766945510869! > > > -- Adam -- NewPace Logo Adam Kelloway Software Engineer, NewPace phone +1 (902) 406--8375 x1031 email Adam.Kelloway at NewPace.com aim /msn Adam.Kelloway @NewPace.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/c11625b9/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Newpace_50x50.png Type: image/png Size: 4620 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/c11625b9/attachment.png From fs-list at communicatefreely.net Tue Mar 6 20:46:17 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Tue, 06 Mar 2012 12:46:17 -0500 Subject: [Freeswitch-users] Channels never hangup? Message-ID: <4F564D69.8090407@communicatefreely.net> Hello, Had a real emergency today, and I'm not sure it's done. All of a sudden, everything is busy. I look at my channel count, and 200 channels (max allowed). We usually don't see more than 40. Some of these calls were two hours old. I know nobody checks voice mail for two hours. I tried a test call, and it never left the channel list. Restarted freeswitch, same thing happened a second time. 200 sessions, 200 stuck channels. Restarted the machine. Same thing again. I haven't touched anything in two days - what did I do? HELP -Tim From benkokakao at gmail.com Tue Mar 6 20:53:39 2012 From: benkokakao at gmail.com (Christian Benke) Date: Tue, 6 Mar 2012 18:53:39 +0100 Subject: [Freeswitch-users] Choopy one-way noise (FreeTDM) In-Reply-To: References: <4F5608BA.4020904@earthspike.net> Message-ID: I've been able to reproduce the issue again in our lab. After starting freeswitch the first few calls where fine, then the choppy noise happened on every call(I repeated about ten times). After restarting wanpipe and freeswitch, the issue is gone and i can't reproduce it for now(Tried for the last 20min)... On 1 March 2012 17:45, Moises Silva wrote: > > Another test you can try to pinpoint the issue and determine once for all > if the problem is below FreeSWITCH is to use the "ftdm trace" command. This > command > will create an input an output file (in the directory you specify) that > contains the raw audio as it is read and written from/to the wanpipe device. I did an ftdm trace when it happened, as Moises suggested - the trace is fine, no noise. Also the RTP-Stream does not show any noise. Since the traces are fine - the problem is happening when the stream leaves freetdm and enters wanpipe according to Moises explanation, right? Best regards, Christian From anthony.minessale at gmail.com Tue Mar 6 21:24:01 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 6 Mar 2012 12:24:01 -0600 Subject: [Freeswitch-users] Channels never hangup? In-Reply-To: <4F564D69.8090407@communicatefreely.net> References: <4F564D69.8090407@communicatefreely.net> Message-ID: update to HEAD with make current try again if you still have the same problem, gcore the running FS and run "thread apply all bt" and capture the output and post to jira. On Tue, Mar 6, 2012 at 11:46 AM, Tim St. Pierre wrote: > Hello, > > Had a real emergency today, and I'm not sure it's done. > > All of a sudden, everything is busy. ?I look at my channel count, and > 200 channels (max allowed). > > We usually don't see more than 40. ?Some of these calls were two hours > old. ?I know nobody checks voice mail for two hours. > > I tried a test call, and it never left the channel list. > > Restarted freeswitch, same thing happened a second time. ?200 sessions, > 200 stuck channels. > > Restarted the machine. ?Same thing again. > > I haven't touched anything in two days - what did I do? > > HELP > > -Tim > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From fs-list at communicatefreely.net Tue Mar 6 21:40:56 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Tue, 06 Mar 2012 13:40:56 -0500 Subject: [Freeswitch-users] Channels never hangup? In-Reply-To: References: <4F564D69.8090407@communicatefreely.net> Message-ID: <4F565A38.1000507@communicatefreely.net> Okay, will do. For what it's worth, this version is: 1.0.head (git-1aa9103 2012-03-01 15-58-48 -0600) Running on FreeBSD 8.2-RELEASE Anthony Minessale wrote: > update to HEAD with make current > try again > if you still have the same problem, gcore the running FS and run > "thread apply all bt" and capture the output and post to jira. > > > On Tue, Mar 6, 2012 at 11:46 AM, Tim St. Pierre > wrote: > >> Hello, >> >> Had a real emergency today, and I'm not sure it's done. >> >> All of a sudden, everything is busy. I look at my channel count, and >> 200 channels (max allowed). >> >> We usually don't see more than 40. Some of these calls were two hours >> old. I know nobody checks voice mail for two hours. >> >> I tried a test call, and it never left the channel list. >> >> Restarted freeswitch, same thing happened a second time. 200 sessions, >> 200 stuck channels. >> >> Restarted the machine. Same thing again. >> >> I haven't touched anything in two days - what did I do? >> >> HELP >> >> -Tim >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > From omortimer at gmail.com Tue Mar 6 22:06:03 2012 From: omortimer at gmail.com (omortimer at gmail.com) Date: Tue, 6 Mar 2012 19:06:03 +0000 Subject: [Freeswitch-users] Channels never hangup? In-Reply-To: <4F564D69.8090407@communicatefreely.net> References: <4F564D69.8090407@communicatefreely.net> Message-ID: <1580617075-1331060763-cardhu_decombobulator_blackberry.rim.net-399988259-@b5.c15.bise7.blackberry> Have you run out of disk space recently? If so your core.db might be corrupt. Try stopping fs, deleting core.db and starting fs back up. Sent from my BlackBerry? smartphone on O2 -----Original Message----- From: "Tim St. Pierre" Sender: freeswitch-users-bounces at lists.freeswitch.org Date: Tue, 06 Mar 2012 12:46:17 To: FreeSWITCH Users Help Reply-To: FreeSWITCH Users Help Subject: [Freeswitch-users] Channels never hangup? Hello, Had a real emergency today, and I'm not sure it's done. All of a sudden, everything is busy. I look at my channel count, and 200 channels (max allowed). We usually don't see more than 40. Some of these calls were two hours old. I know nobody checks voice mail for two hours. I tried a test call, and it never left the channel list. Restarted freeswitch, same thing happened a second time. 200 sessions, 200 stuck channels. Restarted the machine. Same thing again. I haven't touched anything in two days - what did I do? HELP -Tim _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From asilva at wirelessmundi.com Tue Mar 6 22:21:52 2012 From: asilva at wirelessmundi.com (Antonio) Date: Tue, 06 Mar 2012 20:21:52 +0100 Subject: [Freeswitch-users] user dial-string getting variables Message-ID: <1331061712.3526.55.camel@marces.madrid.commsmundi.com> Hi, Is it possible to access a variable that is passed per channel, in some script called in the directory.xml, parameter dial-string? I need to be able to set the var1 dynamically from the dialplan, so i can parse it from my lua script. Using the set command directly it will set the variables for both channels. For example, I dialed two users: then in my directory.xml, i have: So when the return dial-string for the user 100 has it's own var1, and when dial the user 102 has another value. I already try to uuid_dump in dial.lua, but i can't see the variable var1... Using this method, [var1=value], it will set the variable var1 for the B channel, but can i access it from the A channel? Or is there another way to set this variables before processing the dial-string... Thanks, Ant?nio -- Un cordial saludo / Best regards, _________________________ Ant?nio Silva E-mail:asilva at wirelessmundi.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/ddea78dc/attachment.html From fs-list at communicatefreely.net Tue Mar 6 22:39:29 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Tue, 06 Mar 2012 14:39:29 -0500 Subject: [Freeswitch-users] Channels never hangup? In-Reply-To: <1580617075-1331060763-cardhu_decombobulator_blackberry.rim.net-399988259-@b5.c15.bise7.blackberry> References: <4F564D69.8090407@communicatefreely.net> <1580617075-1331060763-cardhu_decombobulator_blackberry.rim.net-399988259-@b5.c15.bise7.blackberry> Message-ID: <4F5667F1.4040103@communicatefreely.net> Lots of disk space, and core DB is in mysql. I did check the db directory, and nothing in there is recent or very big. No problems on the DB server either. Thanks though. -Tim omortimer at gmail.com wrote: > Have you run out of disk space recently? If so your core.db might be corrupt. Try stopping fs, deleting core.db and starting fs back up. > Sent from my BlackBerry? smartphone on O2 > > -----Original Message----- > From: "Tim St. Pierre" > Sender: freeswitch-users-bounces at lists.freeswitch.org > Date: Tue, 06 Mar 2012 12:46:17 > To: FreeSWITCH Users Help > Reply-To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Channels never hangup? > > Hello, > > Had a real emergency today, and I'm not sure it's done. > > All of a sudden, everything is busy. I look at my channel count, and > 200 channels (max allowed). > > We usually don't see more than 40. Some of these calls were two hours > old. I know nobody checks voice mail for two hours. > > I tried a test call, and it never left the channel list. > > Restarted freeswitch, same thing happened a second time. 200 sessions, > 200 stuck channels. > > Restarted the machine. Same thing again. > > I haven't touched anything in two days - what did I do? > > HELP > > -Tim > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From fs-list at communicatefreely.net Tue Mar 6 23:12:03 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Tue, 06 Mar 2012 15:12:03 -0500 Subject: [Freeswitch-users] Channels never hangup? In-Reply-To: References: <4F564D69.8090407@communicatefreely.net> Message-ID: <4F566F93.9010302@communicatefreely.net> Seems easy enough to reproduce now. JIRA (FS-3978) Core file is attached to the Jira. The strange thing is that now I can load one profile, check voice mail three times and all three channels are stuck. Hope you can find something useful. I am going to have to restore an old backup so we can get running safely again. We may have funny BLFs, but at least it handled calls. I will make an image of the system first though, so we can restore it on a spare machine if we need more evidence. Thanks! -Tim Anthony Minessale wrote: > update to HEAD with make current > try again > if you still have the same problem, gcore the running FS and run > "thread apply all bt" and capture the output and post to jira. > > > On Tue, Mar 6, 2012 at 11:46 AM, Tim St. Pierre > wrote: > >> Hello, >> >> Had a real emergency today, and I'm not sure it's done. >> >> All of a sudden, everything is busy. I look at my channel count, and >> 200 channels (max allowed). >> >> We usually don't see more than 40. Some of these calls were two hours >> old. I know nobody checks voice mail for two hours. >> >> I tried a test call, and it never left the channel list. >> >> Restarted freeswitch, same thing happened a second time. 200 sessions, >> 200 stuck channels. >> >> Restarted the machine. Same thing again. >> >> I haven't touched anything in two days - what did I do? >> >> HELP >> >> -Tim >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > From v.wend at hospital-berlin.de Tue Mar 6 17:02:10 2012 From: v.wend at hospital-berlin.de (Volker Wend) Date: Tue, 6 Mar 2012 15:02:10 +0100 Subject: [Freeswitch-users] Voice Mail - Core dump Message-ID: Hi, I try to use the voice mail modul and get always a core dump (first usage). Any Ideas how to proceed ? Best regards, Volker Wend 2012-03-06 14:45:53.016696 [NOTICE] mod_dptools.c:1135 Channel [sofia/sipinterface_1/0173123456 at 172.25.156.1:5060] has been answered 2012-03-06 14:46:09.856705 [NOTICE] sofia.c:628 Hangup sofia/sipinterface_1/01736024861 at 172.25.156.1:5060 [CS_EXECUTE] [NORMAL_CLEARING] Segmentation fault (core dumped) /bin/cat: Schreibfehler: Daten?bergabe unterbrochen (broken pipe) 2012-03-06 14:46:09.956696 [NOTICE] switch_core_session.c:1398 Session 9 (sofia/sipinterface_1/01736024861 at 172.25.156.1:5060) Ended 2012-03-06 14:46:09.956696 [NOTICE] switch_core_session.c:1400 Close Channel sofia/sipinterface_1/01736024861 at 172.25.156.1:5060 [CS_DESTROY] [cid:hospital.jpg] Volker Wend Projektleiter Informationstechnologie, Medizintechnik und Prozessmanagement hospital Dienstleistung + Beratung GmbH Am kleinen Wannsee 5 . 14109 Berlin Tel. +49 (33638) 83261 . Fax +49 (30) 80505 137 Mobil +49 (173) 6024861 . v.wend at hospital-berlin.de hospital Dienstleistung + Beratung GmbH Am Kleinen Wannsee 5A . 14109 Berlin Amtsgericht Charlottenburg HRB 73271 Gesch?ftsf?hrer: Udo Schmidt Steuernummer 27/604/01051 Eine Gesellschaft der Immanuel Diakonie www.immanuel.de DEM LEBEN ZULIEBE. Der Umwelt zuliebe: M?ssen Sie diese Mail wirklich ausdrucken? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/f87d3f12/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: hospital.jpg Type: application/octet-stream Size: 26637 bytes Desc: hospital.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/f87d3f12/attachment-0001.obj From sarahig1985 at gmail.com Tue Mar 6 18:31:39 2012 From: sarahig1985 at gmail.com (Sara Higfler) Date: Tue, 6 Mar 2012 15:31:39 +0000 Subject: [Freeswitch-users] EC2 - One-way Voice / NAT Issue Message-ID: Hi, I'm sure everyone is tired of NAT related questions, but I am really struggling here :( I have freeswitch configured on an EC2 instance, with an elastic IP address. I've followed all of the NAT configuration instructions on the EC2 wiki (http://wiki.freeswitch.org/wiki/Amazon_ec2) and have researched widely on other possible solutions to no avail. Most scenarios work fine, but I'm having issue with one particular call path (see scenario 4 below). Firewall is correctly configured on the EC2 machine. Scenario 1: x-lite to x-lite - Both clients running behind firewall/NAT - Voice path in both directions Scenario 2: x-lite to PSTN - x-lite running behind firewall/NAT - PSTN is connected via SIP-trunk from Freeswitch to VoIPon ITSP - Voice path in both directions Scenario 3: PSTN to PSTN - Both endpoints connected via SIP-trunk (VoIPon ITSP) via Freeswitch - Voice path in both directions Scenario 4: PSTN to x-lite - Exact reverse of scenario 2 - Voice path one way (PSTN to x-lite) Is there something obvious that would prevent this specific scenario from working? I've dumped a SIP trace for the fourth scenario onto pastebin... http://pastebin.com/dg1XqxsJ I'd really appreciate some assistance and guidance! Kind regards, Sara -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/df472375/attachment.html From msc at freeswitch.org Wed Mar 7 00:07:57 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Mar 2012 13:07:57 -0800 Subject: [Freeswitch-users] Voice Mail - Core dump In-Reply-To: References: Message-ID: What is throwing the seg fault? Is it in FreeSWITCH or is it in your email delivery? If you're not sure where to look then start with this page: http://wiki.freeswitch.org/wiki/Reporting_Bugs It has a section on how to run a back trace and open a Jira ticket. -MC 2012/3/6 Volker Wend > Hi,**** > > ** ** > > I try to use the voice mail modul and get always a core dump (first > usage). Any Ideas how to proceed ?**** > > ** ** > > Best regards,**** > > Volker Wend **** > > ** ** > > ** ** > > 2012-03-06 14:45:53.016696 [NOTICE] mod_dptools.c:1135 Channel > [sofia/sipinterface_1/0173123456 at 172.25.156.1:5060] has been answered**** > > 2012-03-06 14:46:09.856705 [NOTICE] sofia.c:628 Hangup > sofia/sipinterface_1/01736024861 at 172.25.156.1:5060 [CS_EXECUTE] > [NORMAL_CLEARING]**** > > Segmentation fault (core dumped)**** > > /bin/cat: Schreibfehler: Daten?bergabe unterbrochen (broken pipe)**** > > 2012-03-06 14:46:09.956696 [NOTICE] switch_core_session.c:1398 Session 9 > (sofia/sipinterface_1/01736024861 at 172.25.156.1:5060) Ended**** > > 2012-03-06 14:46:09.956696 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/sipinterface_1/01736024861 at 172.25.156.1:5060 [CS_DESTROY]*** > * > > ** ** > > ** ** > > > > > *Volker Wend* > Projektleiter > Informationstechnologie, Medizintechnik und Prozessmanagement > > *hospital Dienstleistung + Beratung GmbH* > Am kleinen Wannsee 5 . 14109 Berlin > Tel. +49 (33638) 83261 . Fax +49 (30) 80505 137 > Mobil +49 (173) 6024861 . v.wend at hospital-berlin.de > > hospital Dienstleistung + Beratung GmbH > Am Kleinen Wannsee 5A . 14109 Berlin > Amtsgericht Charlottenburg HRB 73271 > Gesch?ftsf?hrer: Udo Schmidt > Steuernummer 27/604/01051 > > Eine Gesellschaft der Immanuel Diakonie > www.immanuel.de > > DEM LEBEN ZULIEBE. > Der Umwelt zuliebe: M?ssen Sie diese Mail wirklich ausdrucken? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/8ed7ed40/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 26637 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/8ed7ed40/attachment-0001.obj From freeswitch-list at puzzled.xs4all.nl Wed Mar 7 00:18:07 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Tue, 06 Mar 2012 22:18:07 +0100 Subject: [Freeswitch-users] Voice Mail - Core dump In-Reply-To: References: Message-ID: <4F567F0F.9000107@puzzled.xs4all.nl> On 06-03-12 15:02, Volker Wend wrote: > Hi, > > I try to use the voice mail modul and get always a core dump (first > usage). Any Ideas how to proceed ? First you probably want to make sure that this core dump still happens with today's git. If it does then that's a bug which should be reported (including the backtrace) at http://jira.freeswitch.org Before you do that checkout the following link so you provide the developers with all the information they need: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Reporting_A_Bug_With_JIRA Regards, Patrick From brian at freeswitch.org Wed Mar 7 00:21:42 2012 From: brian at freeswitch.org (Brian West) Date: Tue, 6 Mar 2012 15:21:42 -0600 Subject: [Freeswitch-users] Why doesn't continue_on_fail continue? In-Reply-To: References: Message-ID: try setting the variable failure_causes with a comman sep list of failures you wish to consider a failure. /b On Feb 29, 2012, at 7:56 AM, Kevin Snow wrote: > > Did you ever find any information on this? I'm chasing a similar issue. > > In my case, the session is in switch_ivr_originate, an INVITE goes out. > Shortly after, mod_sofia's sofia_handle_sip_i_state() is called with state > nua_callstate_terminated with SIP error 503. It appears this is coming > from the sofia library itself as it did not receive a 503. > Switch_ivr_originate fails with NORMAL_TEMPORARY_FAILURE. > > The problem is the INVITE went out but sofia doesn't send a CANCEL in this > case. We have code that attempts the call again, so another INVITE goes > out. This leaves it receiving 180 RINGING and 200 OKs for both INVITEs. > Essentially two calls but the first is dead. After a couple seconds sofia > sends a 481 Call Does Not Exist for the first Call-Id. > > Of course, I have no idea how to reproduce this and it's pretty rare so I > don't have much to go on. But we do see it occasionally and I'm trying to > understand it. > > > Kevin > > > > > > > > On 2/24/12 3:45 PM, "Harry Coin" wrote: > >> This is a 'why doesn't continue_on_fail' continue question. >> >> session:setVariable("continue_on_fail", "true"); >> session:setVariable("ignore_early_media", "true"); >> session:setVariable("hangup_after_bridge", "true"); >> >> # Try some local registered extensions >> session:execute("bridge", "[leg_timeout=28]loopback/202/default/XML"); >> # Ok, this works, we get here only when nobody picks up. >> # Use the dial plan leading to a bridge >> sofia/internal/xxx at voip2pstn1|sofia/internal/xxx at voip2pstn >> session:execute("bridge", >> "[leg_timeout=30][accountcode=4499090]sofia/internal/5551212@${domain}"); >> # If fails for any reason (all outgoing voip->pstn lines are busy or if >> voip2pstn device is dead/busy) continue here. >> # Problem is -- We never get here if the above fails. >> # The reason is 'NORMAL_TEMPORARY_FAILURE' but it just hangs up the >> calling extension. >> session:execute("bridge", "[leg_timeout=136]loopback/*99008"); #should >> leave voicemail, never gets called. >> >> What am I doing wrong? >> >> Harry >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From v.wend at hospital-berlin.de Wed Mar 7 00:47:43 2012 From: v.wend at hospital-berlin.de (Volker Wend) Date: Tue, 6 Mar 2012 22:47:43 +0100 Subject: [Freeswitch-users] Voice Mail - Core dump In-Reply-To: <4F567F0F.9000107@puzzled.xs4all.nl> References: <4F567F0F.9000107@puzzled.xs4all.nl> Message-ID: <4637C991-7947-45C5-8CC3-3F79261A91CE@immanuel.de> Thanks, I am a Newbie. I didn't know about the bugtracking... I will get back later. I did a checkout today. Regards, Volker Am 06.03.2012 um 22:20 schrieb "Patrick Lists" : > On 06-03-12 15:02, Volker Wend wrote: >> Hi, >> >> I try to use the voice mail modul and get always a core dump (first >> usage). Any Ideas how to proceed ? > > First you probably want to make sure that this core dump still happens > with today's git. If it does then that's a bug which should be reported > (including the backtrace) at http://jira.freeswitch.org > > Before you do that checkout the following link so you provide the > developers with all the information they need: > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Reporting_A_Bug_With_JIRA > > Regards, > Patrick > > Volker Wend Projektleiter Informationstechnologie, Medizintechnik und Prozessmanagement hospital Dienstleistung + Beratung GmbH Am kleinen Wannsee 5 . 14109 Berlin Tel. +49 (33638) 83261 . Fax +49 (30) 80505 137 Mobil +49 (173) 6024861 . v.wend at hospital-berlin.de hospital Dienstleistung + Beratung GmbH Am Kleinen Wannsee 5A . 14109 Berlin Amtsgericht Charlottenburg HRB 73271 Gesch?ftsf?hrer: Udo Schmidt Steuernummer 27/604/01051 Eine Gesellschaft der Immanuel Diakonie www.immanuel.de DEM LEBEN ZULIEBE. _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From johnrose at google.hm Wed Mar 7 01:30:50 2012 From: johnrose at google.hm (John Rose) Date: Tue, 6 Mar 2012 17:30:50 -0500 Subject: [Freeswitch-users] chat API UTF-8 Message-ID: <004701ccfbe8$c9903f40$5cb0bdc0$@google.hm> When using chat API for SIP MESSAGE and sending UTF-8 should the text be URL encoded? John -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/6da0a78c/attachment.html From potxoka at gmail.com Wed Mar 7 01:30:51 2012 From: potxoka at gmail.com (Anto) Date: Tue, 6 Mar 2012 23:30:51 +0100 Subject: [Freeswitch-users] Codec negotiation with carriers Message-ID: Hi I am following the steps in this direction "http://wiki.freeswitch.org/wiki/SBC_Setup" and "http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice", I reread the whole entire wiki (or so I lack), but do not quite assimilate or finding the right formula to operate the bridge :-S. I captured traffic with ngrep, I enabled sip-trace, console logconsole 8, etc., but unless the transcoding error (only two of the hundreds of combinations of settings that I have), I have not seen anything "weird" :-S I have 3 suppliers, each with this codec: 1) 2) 3) G729 G729 G729 G711u G711A G711A G711A G711u G711u G723 G723 G722 GSM I think I understand that when making an outside call, FreeSWITCH follow these steps: USER -> ( Dialplan -> profile (internal) -> bridge (external) -> profile (external) ) -> PROVIDER PROVIDER -> ( Dialplan -> profile (external) -> bridge (internal) -> profile (internal) ) -> USER right? Internal and external I set as follows (and not many changes have done, and not remember it, because I've been testing days). If outbound (outbound-codec-prefs) all codecs specified system does not handle the call, I have to specify these by hand. If active inbound-proxy-media, not the caller. Some of the time I worked, but gave me an error that it can do transcoding G729 codec (I do passthrough), but the proxy does not work half. If the outbound property (outbound-codec-prefs) all codecs specified system does not handle the call, I have to specify these by hand. If active inbound-proxy-media, not the caller. Some of the time I worked, but gave me an error that it can do transcoding G729 codec (I want to make passthrough), but the "proxy media" does not work. Basically, what I do is that local users can use all the codecs allowed (iLBC, GSM, ...) and make an outside call, use the carrier that will indicate the priority but the free codec. With this configuration works for me, but I would like to understand why so if it works and otherwise no. Coming to understand how to configure properly and so as not to disturb the mail list ;-). Thanks ! Best regards Anto vars.xml internal.xml external.xml dialplan/outbound.xml From freeswitch-list at puzzled.xs4all.nl Wed Mar 7 01:34:00 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Tue, 06 Mar 2012 23:34:00 +0100 Subject: [Freeswitch-users] Voice Mail - Core dump In-Reply-To: <4637C991-7947-45C5-8CC3-3F79261A91CE@immanuel.de> References: <4F567F0F.9000107@puzzled.xs4all.nl> <4637C991-7947-45C5-8CC3-3F79261A91CE@immanuel.de> Message-ID: <4F5690D8.6090105@puzzled.xs4all.nl> On 06-03-12 22:47, Volker Wend wrote: > Thanks, I am a Newbie. I didn't know about the bugtracking... I will get back later. I did a checkout today. Welcome to the FreeSWITCH Community! Besides the mailing list there is also an active irc channel (#freeswitch on irc.freenode.net) where chatting about your problem may be easier (and quicker) than sending a lot of email back and forth to the mailing list. The wiki at http://wiki.freeswitch.org has a ton of information. It is recommended to do a quick search on the wiki if you can't figure something out. Finally, I recommend the FreeSWITCH book and the FreeSWITCH cookbook. I have no affiliation, just happy I bought them. And it supports the FreeSWITCH developers who wrote these books. Hope you enjoy FreeSWITCH! Regards, Patrick From dgarcia at anew.com.ve Wed Mar 7 01:38:02 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Tue, 06 Mar 2012 18:08:02 -0430 Subject: [Freeswitch-users] web application to call to fs In-Reply-To: <4F5148F3.6030006@livecall.com> References: <4F5140AE.6000708@anew.com.ve> <4F5148F3.6030006@livecall.com> Message-ID: <4F5691CA.1020609@anew.com.ve> Thanks Jack, I change the login info to XXX@ and works when I dial to an extesion in FS or outside. But if I dial from outside or an extension in FS to the user using flex client, the call is not routed, fail. What I should check? Could be the context name? On 3/2/2012 5:55 PM, Jack wrote: > Make sure You directory .xml file for your user has the same > user_context as your rtmp.config.xml > > On 3/2/2012 1:50 PM, Saugort Dario Garcia Tovar wrote: >> Hi, >> >> You have more info about mod_rtmp? How to put it to work? I got an >> error about could not autenticate when I login in flex client >> >> On 2/29/2012 9:01 PM, Wagner wrote: >>> >>> hello , >>> >>> is there any Web application that i could use to let a user call to >>> my ivr through my website? >>> >>> thanks >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> No virus found in this message. >>> Checked by AVG - www.avg.com >>> Version: 2012.0.1913 / Virus Database: 2114/4844 - Release Date: >>> 03/01/12 >>> >> >> >> -- >> Atentamente, >> *Dario Garc?a* >> Consultor. >> >> CCCT, Nivel C2, Sector Yarey, Mz, >> Ofc. MZ03a. >> Caracas-Venezuela. >> Tel?fono: +58 212 9081842 >> Cel: +58 412 2221515 >> dgarcia at anew.com.ve >> http://www.anew.com.ve >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1913 / Virus Database: 2114/4846 - Release Date: 03/02/12 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/646426de/attachment-0001.html From msc at freeswitch.org Wed Mar 7 02:20:56 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Mar 2012 15:20:56 -0800 Subject: [Freeswitch-users] FreeSWITCH Conf Call Tomorrow - Dialplan Basics Message-ID: Hello all! Here's tomorrow's agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_06 I will be discussing some dialplan basics along with some regex tricks and basic patterns. Other experts may also be chiming in. The idea is that everyone on the call will come away with a more thorough understanding of what happens when a phone call comes into FreeSWITCH. Talk to you tomorrow! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/315888e5/attachment.html From bob.mccarthy at experient.com Wed Mar 7 02:28:08 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Tue, 06 Mar 2012 18:28:08 -0500 Subject: [Freeswitch-users] How to Blind Transfer After a Conference Message-ID: <1331076488.6593.6.camel@CO-999-8> I have some customers that are using SLA but want to Blind transfer the call after the supervisor drops out of the conference. (2 legs left in call) in Asterisk I would use the bridge command which would pull the channels out of the conference and bridge them together. I tried to do the same thing in Freeswitch from a lua script but I get the error: [ERR} switch_cpp.cpp:1277 Channels not ready I parsed the channel names out of the Conference list: Conference 8b09249a-67df-11e1-899d-2dc9f6a91475 (2 members rate: 16000) 95;sofia/internal/sip:CO999x1001.3 at 192.168.57.102;8b0b8262-67df-11e1-89a5-2dc9f6a91475;Outbound Call;303-444-5555;hear|speak|talking|floor;0;0;0;200 94;sofia/internal/101 at 192.168.57.31;8b09249a-67df-11e1-899d-2dc9f6a91475;911 Trunk 01;303-444-5555;hear|speak;0;0;0;200 Some debug output to make sure I am using the correct channel names: channelarray[1]=sofia/internal/sip:CO999x1001.3 at 192.168.57.102 channelarray[2]=sofia/internal/101 at 192.168.57.31 Here is the relevant script commands session1 = freeswitch.Session(channelarray[1]); session1 = freeswitch.Session(channelarray[1]); freeswitch.bridge(session1, session2); -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/7fecd67b/attachment.html From curriegrad2004 at gmail.com Wed Mar 7 03:05:05 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 6 Mar 2012 16:05:05 -0800 Subject: [Freeswitch-users] FreeSWITCH Conf Call Tomorrow - Dialplan Basics In-Reply-To: References: Message-ID: This is a great presentation topic. I find that a lot of new users can't really be bothered to read the wiki or they just find the wiki's too complicated to read in the first place. I hope a presentation/recording like this can really help out the new users out there on how FreeSWITCH should be really configured. On Tue, Mar 6, 2012 at 3:20 PM, Michael Collins wrote: > Hello all! > > Here's tomorrow's agenda page: > http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_06 > > I will be discussing some dialplan basics along with some regex tricks and > basic patterns. Other experts may also be chiming in. The idea is that > everyone on the call will come away with a more thorough understanding of > what happens when a phone call comes into FreeSWITCH. > > Talk to you tomorrow! > > -Michael > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From freeswitch at earthspike.net Wed Mar 7 03:14:58 2012 From: freeswitch at earthspike.net (John) Date: Wed, 07 Mar 2012 00:14:58 +0000 Subject: [Freeswitch-users] Voice Mail - Core dump In-Reply-To: <4637C991-7947-45C5-8CC3-3F79261A91CE@immanuel.de> References: <4F567F0F.9000107@puzzled.xs4all.nl> <4637C991-7947-45C5-8CC3-3F79261A91CE@immanuel.de> Message-ID: <4F56A882.2060804@earthspike.net> On 06/03/12 21:47, Volker Wend wrote: > Thanks, I am a Newbie. I didn't know about the bugtracking... I will get back later. I did a checkout today. > > > Regards, > Volker > > > > > Am 06.03.2012 um 22:20 schrieb "Patrick Lists" : > >> On 06-03-12 15:02, Volker Wend wrote: >>> Hi, >>> >>> I try to use the voice mail modul and get always a core dump (first >>> usage). Any Ideas how to proceed ? >> First you probably want to make sure that this core dump still happens >> with today's git. If it does then that's a bug which should be reported >> (including the backtrace) at http://jira.freeswitch.org >> >> Before you do that checkout the following link so you provide the >> developers with all the information they need: >> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Reporting_A_Bug_With_JIRA >> >> Regards, >> Patrick >> >> Volker Wend > Have you enabled email delivery of voicemails? There's also a section on debugging email delivery within the mod_voicemail wiki page. Segfaults are often caused by some sendmail installations, so many people have used other mail delivery mechanisms (eg nullmailer, ssmtp, sendemail.py). John From mike at geils.com Wed Mar 7 03:28:17 2012 From: mike at geils.com (Mike Patterson) Date: Tue, 6 Mar 2012 19:28:17 -0500 Subject: [Freeswitch-users] Freeswitch as a register server Message-ID: <01c801ccfbf9$32997cd0$97cc7670$@com> I am trying to find out if I could use freeswitch as a register server. Freeswitch would actually be acting more like an SBC. I would like to provide my users a domain or IP address to register with and have Freeswitch pass the registration request to my soft switch. I read on the Freeswitch wiki that there is some SBC functionality but I did not find anything specific to my needs. I would appreciate any suggestions or help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/9f5bf14c/attachment.html From msc at freeswitch.org Wed Mar 7 03:46:49 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Mar 2012 16:46:49 -0800 Subject: [Freeswitch-users] Codec negotiation with carriers In-Reply-To: References: Message-ID: You may want to read up on codec negotiation: http://wiki.freeswitch.org/wiki/Codec_negotiation There are different ways to handle codecs depending on your needs. I'd read that page first and then try out some of the suggestions. If you're still having trouble then I'd recommend getting SIP traces of the traffic and putting them on pastebin.freeswitch.org. The gang here is pretty good at looking over logs and helping with diagnosing problems. :) -MC On Tue, Mar 6, 2012 at 2:30 PM, Anto wrote: > Hi > > I am following the steps in this direction > "http://wiki.freeswitch.org/wiki/SBC_Setup" and > "http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice", > I reread the whole entire wiki (or so I lack), but do not quite > assimilate or finding the right formula to operate the bridge :-S. > > I captured traffic with ngrep, I enabled sip-trace, console logconsole > 8, etc., but unless the transcoding error (only two of the hundreds of > combinations of settings that I have), I have not seen anything > "weird" :-S > > I have 3 suppliers, each with this codec: > > 1) 2) 3) > G729 G729 G729 > G711u G711A G711A > G711A G711u G711u > G723 G723 > G722 > GSM > > I think I understand that when making an outside call, FreeSWITCH > follow these steps: > > USER -> ( Dialplan -> profile (internal) -> bridge (external) -> > profile (external) ) -> PROVIDER > > PROVIDER -> ( Dialplan -> profile (external) -> bridge (internal) -> > profile (internal) ) -> USER > > right? > > Internal and external I set as follows (and not many changes have > done, and not remember it, because I've been testing days). If > outbound (outbound-codec-prefs) all codecs specified system does not > handle the call, I have to specify these by hand. If active > inbound-proxy-media, not the caller. Some of the time I worked, but > gave me an error that it can do transcoding G729 codec (I do > passthrough), but the proxy does not work half. > > If the outbound property (outbound-codec-prefs) all codecs specified > system does not handle the call, I have to specify these by hand. If > active inbound-proxy-media, not the caller. Some of the time I worked, > but gave me an error that it can do transcoding G729 codec (I want to > make passthrough), but the "proxy media" does not work. > > Basically, what I do is that local users can use all the codecs > allowed (iLBC, GSM, ...) and make an outside call, use the carrier > that will indicate the priority but the free codec. > > With this configuration works for me, but I would like to understand > why so if it works and otherwise no. Coming to understand how to > configure properly and so as not to disturb the mail list ;-). Thanks > ! > > Best regards > Anto > > vars.xml > > > data="global_codec_prefs=iLBC,G7221,speex,PCMU,PCMA,BV16,G726-32,GSM,G729,G723,AMR"/> > > data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,iLBC,G7221,speex,BV16,G726-32,GSM"/> > > internal.xml > > > > > > > > > > external.xml > > > > > > > > > > > > dialplan/outbound.xml > > > > > expression="^(\d+)$"> > > > > data="sofia/gateway/provider-2/$1"/> > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/c04de4b0/attachment-0001.html From bdfoster at endigotech.com Wed Mar 7 03:54:40 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 6 Mar 2012 19:54:40 -0500 Subject: [Freeswitch-users] FreeSWITCH Conf Call Tomorrow - Dialplan Basics In-Reply-To: References: Message-ID: I agree with Jeff, great presentation topic. I'll be there for sure. The cool thing is once you learn the concept behind dialplans, it's so much easier when it comes to the more advanced stuff. See you all there! -BDF On Mar 6, 2012 7:06 PM, "curriegrad2004" wrote: > This is a great presentation topic. I find that a lot of new users > can't really be bothered to read the wiki or they just > find the wiki's too complicated to read in the first place. I hope a > presentation/recording like this can really help out > the new users out there on how FreeSWITCH should be really configured. > > On Tue, Mar 6, 2012 at 3:20 PM, Michael Collins > wrote: > > Hello all! > > > > Here's tomorrow's agenda page: > > http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_06 > > > > I will be discussing some dialplan basics along with some regex tricks > and > > basic patterns. Other experts may also be chiming in. The idea is that > > everyone on the call will come away with a more thorough understanding of > > what happens when a phone call comes into FreeSWITCH. > > > > Talk to you tomorrow! > > > > -Michael > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/ef815e5a/attachment.html From msc at freeswitch.org Wed Mar 7 04:07:36 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Mar 2012 17:07:36 -0800 Subject: [Freeswitch-users] How to Blind Transfer After a Conference In-Reply-To: <1331076488.6593.6.camel@CO-999-8> References: <1331076488.6593.6.camel@CO-999-8> Message-ID: if those two calls are all that are in the conference then you just need to uuid_bridge them. Ex from fs_cli uuid_bridge 8b0b8262-67df-11e1-89a5-2dc9f6a91475 8b09249a-67df-11e1-899d- 2dc9f6a91475 You can also do an API from a Lua script: api = freeswitch.API() res = api:execute('uuid_bridge','8b0b8262-67df-11e1-89a5-2dc9f6a91475 8b09249a-67df-11e1-899d-2dc9f6a91475') -MC On Tue, Mar 6, 2012 at 3:28 PM, Bob McCarthy wrote: > ** > I have some customers that are using SLA but want to Blind transfer the > call after the supervisor drops out of the conference. (2 legs left in > call) > > in Asterisk I would use the bridge command which would pull the channels > out of the conference and bridge them together. I tried to do the same > thing in Freeswitch from a lua script but I get the error: [ERR} > switch_cpp.cpp:1277 Channels not ready > > > > I parsed the channel names out of the Conference list: > Conference 8b09249a-67df-11e1-899d-2dc9f6a91475 (2 members rate: 16000) > 95;sofia/internal/sip:CO999x1001.3 at 192.168.57.102;8b0b8262-67df-11e1-89a5-2dc9f6a91475;Outbound > Call;303-444-5555;hear|speak|talking|floor;0;0;0;200 > 94;sofia/internal/101 at 192.168.57.31;8b09249a-67df-11e1-899d-2dc9f6a91475;911 > Trunk 01;303-444-5555;hear|speak;0;0;0;200 > > Some debug output to make sure I am using the correct channel names: > channelarray[1]=sofia/internal/sip:CO999x1001.3 at 192.168.57.102 > channelarray[2]=sofia/internal/101 at 192.168.57.31 > > Here is the relevant script commands > session1 = freeswitch.Session(channelarray[1]); > session1 = freeswitch.Session(channelarray[1]); > freeswitch.bridge(session1, session2); > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/1a949c68/attachment.html From krice at freeswitch.org Wed Mar 7 04:15:19 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 06 Mar 2012 19:15:19 -0600 Subject: [Freeswitch-users] Freeswitch as a register server In-Reply-To: <01c801ccfbf9$32997cd0$97cc7670$@com> Message-ID: Passing registration requests etc on is the Function of a proxy. FreeSWITCH is a B2BUA (back to back user agent) while it can do some of the functions of a proxy it is not one. What you are more specifically looking for in that area is one of the SIP Express Router derivatives like OpenSIPS or Kamilio or you are looking for something more like an ACME Packet SBC... K On 3/6/12 6:28 PM, "Mike Patterson" wrote: > > I am trying to find out if I could use freeswitch as a register server. > Freeswitch would actually be acting more like an SBC. I would like to provide > my users a domain or IP address to register with and have Freeswitch pass the > registration request to my soft switch. I read on the Freeswitch wiki that > there is some SBC functionality but I did not find anything specific to my > needs. I would appreciate any suggestions or help. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120306/5479e7e0/attachment.html From anita.hall at simmortel.com Wed Mar 7 09:17:50 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Wed, 7 Mar 2012 11:47:50 +0530 Subject: [Freeswitch-users] answer when agent connects in mod_callcenter Message-ID: Hi! I tried mod_callcenter and I must say it is an amazing piece of work by Marc Olivier Chouinard! We had built something like this over ESL but nothing beats a loadable module :) The tricky question is how do I configure it so that the caller is answered only after an agent is connected? By default, the call is answered and moh is played. Is it even possible or do I have to look inside mod_callcenter? If that, exactly where does a module answer a call ? Thanks. regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/19b1f6a5/attachment.html From peter.olsson at visionutveckling.se Wed Mar 7 10:11:29 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 7 Mar 2012 07:11:29 +0000 Subject: [Freeswitch-users] answer when agent connects in mod_callcenter In-Reply-To: References: Message-ID: <98AE7CD6-4640-464D-8AD7-58A63A40B2E5@visionutveckling.se> Look in the code for switch_channel_answer(), that's what causes the call to be answered. Anyway, even though an unanswered call seems great in theory, it usually doesn't work, because the providers will only allow that state for a limited time. So in the end, answer is usually what must be done anyway. /Peter ----- Reply message ----- Fr?n: "Anita Hall" Datum: ons, mar 7, 2012 07:24 Rubrik: [Freeswitch-users] answer when agent connects in mod_callcenter Till: "FreeSWITCH Users Help" Hi! I tried mod_callcenter and I must say it is an amazing piece of work by Marc Olivier Chouinard! We had built something like this over ESL but nothing beats a loadable module :) The tricky question is how do I configure it so that the caller is answered only after an agent is connected? By default, the call is answered and moh is played. Is it even possible or do I have to look inside mod_callcenter? If that, exactly where does a module answer a call ? Thanks. regards, Anita !DSPAM:4f56fcb432761851621280! From miha at softnet.si Wed Mar 7 10:18:02 2012 From: miha at softnet.si (Miha Zoubek) Date: Wed, 07 Mar 2012 08:18:02 +0100 Subject: [Freeswitch-users] ENUM Message-ID: <4F570BAA.3000206@softnet.si> Hi, I am having users as registered 018100125.fs1.adsijiiojas.com. In dial plan I am checking if numbers that is registered on FS is ported or not. If it is ported that ${enum_auto_route} if not that do like this: Enum: I am doing this beacuse I would like to have number that enum return in my billing but the call should be made locally like for internal calls without enum. Is there any options to get that kind of behaviour (in cdr to have enum number but call is made like for internal calls)? Thanks! Miha From virbhati at gmail.com Wed Mar 7 10:57:24 2012 From: virbhati at gmail.com (virendra bhati) Date: Wed, 7 Mar 2012 13:27:24 +0530 Subject: [Freeswitch-users] how to create Lua application for Freeswicth Message-ID: FreeSwitch is new for me. I want to make my own small application in lua and connect to mysql DB. Could anyone please suggest me how to start programming in Lua ? I am reading below thread for Lua programming. I want to boost it up with small program so that my confidence will increase. http://www.lua.org/pil/index.html Any book for programming in Lua for freeswitch -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati at gmail.com Skype id:- virbhati2 Hyderabad(India) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/a36cee9f/attachment.html From bob.mccarthy at experient.com Wed Mar 7 11:09:11 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Wed, 7 Mar 2012 01:09:11 -0700 Subject: [Freeswitch-users] How to Blind Transfer After a Conference In-Reply-To: References: <1331076488.6593.6.camel@CO-999-8> Message-ID: <012801ccfc39$95c6c930$c1545b90$@mccarthy@experient.com> worked perfectly thanks ! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, March 06, 2012 6:08 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Blind Transfer After a Conference if those two calls are all that are in the conference then you just need to uuid_bridge them. Ex from fs_cli uuid_bridge 8b0b8262-67df-11e1-89a5-2dc9f6a91475 8b09249a-67df-11e1-899d-2dc9f6a91475 You can also do an API from a Lua script: api = freeswitch.API() res = api:execute('uuid_bridge','8b0b8262-67df-11e1-89a5-2dc9f6a91475 8b09249a-67df-11e1-899d-2dc9f6a91475') -MC On Tue, Mar 6, 2012 at 3:28 PM, Bob McCarthy wrote: I have some customers that are using SLA but want to Blind transfer the call after the supervisor drops out of the conference. (2 legs left in call) in Asterisk I would use the bridge command which would pull the channels out of the conference and bridge them together. I tried to do the same thing in Freeswitch from a lua script but I get the error: [ERR} switch_cpp.cpp:1277 Channels not ready I parsed the channel names out of the Conference list: Conference 8b09249a-67df-11e1-899d-2dc9f6a91475 (2 members rate: 16000) 95;sofia/internal/sip:CO999x1001.3 at 192.168.57.102 ;8b0b8262-67df-11e1-89a5-2dc9f6a91475;Outbound Call;303-444-5555;hear|speak|talking|floor;0;0;0;200 94;sofia/internal/101 at 192.168.57.31;8b09249a-67df-11e1-899d-2dc9f6a91475;911 Trunk 01;303-444-5555;hear|speak;0;0;0;200 Some debug output to make sure I am using the correct channel names: channelarray[1]=sofia/internal/sip:CO999x1001.3 at 192.168.57.102 channelarray[2]=sofia/internal/101 at 192.168.57.31 Here is the relevant script commands session1 = freeswitch.Session(channelarray[1]); session1 = freeswitch.Session(channelarray[1]); freeswitch.bridge(session1, session2); _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/f3bbe0e1/attachment.html From gerald.weber at besharp.at Wed Mar 7 11:14:12 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Wed, 7 Mar 2012 08:14:12 +0000 Subject: [Freeswitch-users] Any alternative to repetitive lua session:ready() calls? In-Reply-To: <4F564B01.7050404@newpace.ca> References: <4F5639AA.9050900@newpace.ca> <1FFF97C269757C458224B7C895F35F1504E178@cantor.std.visionutv.se> <4F564B01.7050404@newpace.ca> Message-ID: Morning, i had the same problem a while ago, reported the jira below and anthm made some changes tu mod_lua. If you put "error()" as last command in your hangup handler (see http://wiki.freeswitch.org/wiki/Mod_lua#session:setHangupHook) No error is dumped in the log. Both 'return "exit"' and 'return "die"' leave an error in the logfiles. A little bit misleading in my opinion, but if you know how to handle it...it works. Tried 15mins ago with latest git FreeSWITCH Version 1.0.head (git-ea975c3 2012-02-28 19-21-04 -0500) Regards gw Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Adam Kelloway Gesendet: Dienstag, 06. M?rz 2012 18:36 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Any alternative to repetitive lua session:ready() calls? Kind of, all of those solutions consider the script exiting an 'error'. I am looking for something that exits without considering it an error and thus does not emit an [ERR] log. On 3:59 PM, Peter Olsson wrote: Maybe this is what you are looking for? commit 09ad887948f7513725ca8b53bdfe721d9008e73b Author: Anthony Minessale Date: Fri Jan 27 19:03:04 2012 -0600 FS-3841 --resolve ok return the string "die" or "exit" from hanguphook to cause an error or call s:destroy("any err message"); either should now halt the script ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Adam Kelloway [adam.kelloway at newpace.ca] Skickat: den 6 mars 2012 17:22 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Any alternative to repetitive lua session:ready() calls? Hi there, If I am running a lua script during a call, and the call terminates (caller hangs up, for instance), the lua script continues to execute while the session has already been terminated. This causes ERR entries in the freeswitch log, which I would like to try and minimize. The only way I can see doing this is to frequently call session:ready() before doing anything that could break the script. Is there a better way to exit a script cleanly, without having to add X number of session:ready() calls scattered about the script? Currently, I only call it once (at the beginning of the script). I tried using the hangup hook, but I'm not sure how I can get it to immediately exit the script. Thanks, Adam _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f56388f32766945510869! -- Adam -- [NewPace Logo] Adam Kelloway Software Engineer, NewPace phone +1 (902) 406-8375 x1031 email Adam.Kelloway at NewPace.com aim/msn Adam.Kelloway@NewPace.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/3bf5ed35/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 4620 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/3bf5ed35/attachment-0001.png From anita.hall at simmortel.com Wed Mar 7 11:21:57 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Wed, 7 Mar 2012 13:51:57 +0530 Subject: [Freeswitch-users] answer when agent connects in mod_callcenter In-Reply-To: <98AE7CD6-4640-464D-8AD7-58A63A40B2E5@visionutveckling.se> References: <98AE7CD6-4640-464D-8AD7-58A63A40B2E5@visionutveckling.se> Message-ID: Hey Peter Much thanks :) It worked like a charm! Now I am able to ring an agent via mod_callcenter and if the agent does not answer, then the caller is not answered. Magic ! Now all I need to do is to hangup the caller if the agent does not answer / cancels .... regards, Anita On Wed, Mar 7, 2012 at 12:41 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Look in the code for switch_channel_answer(), that's what causes the call > to be answered. > > Anyway, even though an unanswered call seems great in theory, it usually > doesn't work, because the providers will only allow that state for a > limited time. So in the end, answer is usually what must be done anyway. > > /Peter > > ----- Reply message ----- > Fr?n: "Anita Hall" > Datum: ons, mar 7, 2012 07:24 > Rubrik: [Freeswitch-users] answer when agent connects in mod_callcenter > Till: "FreeSWITCH Users Help" > > Hi! > > I tried mod_callcenter and I must say it is an amazing piece of work by > Marc Olivier Chouinard! We had built something like this over ESL but > nothing beats a loadable module :) > > The tricky question is how do I configure it so that the caller is > answered only after an agent is connected? By default, the call is answered > and moh is played. Is it even possible or do I have to look inside > mod_callcenter? If that, exactly where does a module answer a call ? > > Thanks. > > regards, > Anita > > !DSPAM:4f56fcb432761851621280! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/6b0bb2ff/attachment.html From brett at launch3.net Wed Mar 7 11:35:36 2012 From: brett at launch3.net (Brett Wilson) Date: Wed, 7 Mar 2012 03:35:36 -0500 Subject: [Freeswitch-users] State of GUIs Message-ID: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> Hey guys, I was wondering if anyone had any info on the current state of FS guis? It seems that most of the gui projects (blue.box, fusionpbx) have been mostly abandoned in terms of development. I would like something simple for end users to self-administer if needed. The problem I have with fusionpbx is that config files are overwritten by whatever is in the database. If you hand-edit a config file, fusion will not parse the file and load those settings into the interface. I realize that takes much more coding to do than using a database to simply write config files. But I feel that the freeswitch interface could be improved anyway. Unfortunately it seems that project has been abandoned. There have not been any releases since mid-2011. Anything new out or around the corner? Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: Brett.Wilson at launch3.net AOL IM: Brett.Wilson at launch3.net www.Launch3.net www.Launch3telecom.com *************************** Description: Description: Blogger-logo Description: Description: FaceBook-Logo Description: Description: Twitter-Logo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/d945c5ab/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: image/jpeg Size: 1715 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/d945c5ab/attachment-0002.jpe From kbdfck at gmail.com Wed Mar 7 12:29:56 2012 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Wed, 7 Mar 2012 13:29:56 +0400 Subject: [Freeswitch-users] DTMF passthrough delay / pass_rfc2833 problems Message-ID: Hi all I'd like to know is there a way to make freeswitch pass RFC2833 dtmf right after it receives first packets as in pass_rfc2833 mode, but still recognize DTMF for bind_meta_app or bind_digit_action? Seems when i enable pass_rfc2833, bind_meta_app stops working. When we use ATA endpoints like Linksys PAP2T or SPA8000 without pass_rfc2833, ATAs sends little piece of inband DTMF followed by RFC2833 packets. While inband piece is immediately forwarded by FS, RFC2833 packets get relayed only after receiving end packets from endpoint, or at least delayed, effectively making double DTMF on other side. We can't use no-media or proxy-media mode as we need to deal with in-call features activated by DTMF :( What can be done to solve this issues? -- Best regards, Dmitry Sytchev, IT Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/ecb468ca/attachment.html From anita.hall at simmortel.com Wed Mar 7 13:10:47 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Wed, 7 Mar 2012 15:40:47 +0530 Subject: [Freeswitch-users] how to create Lua application for Freeswicth In-Reply-To: References: Message-ID: You can start from here http://wiki.freeswitch.org/wiki/Lua_Welcome_IVR_Example and the mod_lua page. regards, Anita On Wed, Mar 7, 2012 at 1:27 PM, virendra bhati wrote: > FreeSwitch is new for me. > I want to make my own small application in lua and connect to mysql DB. > Could anyone please suggest me how to start programming in Lua ? > I am reading below thread for Lua programming. I want to boost it up with > small program so that my confidence will increase. > > http://www.lua.org/pil/index.html > > Any book for programming in Lua for freeswitch > -- > > Thanks and regards > > Virendra Bhati > +91-8885268942 > Software Engineer > E-mail-: virbhati at gmail.com > Skype id:- virbhati2 > Hyderabad(India) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/452db65c/attachment-0001.html From freeswitch at earthspike.net Wed Mar 7 13:15:02 2012 From: freeswitch at earthspike.net (John) Date: Wed, 07 Mar 2012 10:15:02 +0000 Subject: [Freeswitch-users] how to create Lua application for Freeswicth In-Reply-To: References: Message-ID: <4F573526.3000004@earthspike.net> And the FreeSWITCH book ("bridge book") has a chapter as well (or maybe 2 - I've lent my copy to someone at the moment). On 07/03/12 10:10, Anita Hall wrote: > You can start from here > http://wiki.freeswitch.org/wiki/Lua_Welcome_IVR_Example > > and the mod_lua page. > > regards, > Anita > > > > On Wed, Mar 7, 2012 at 1:27 PM, virendra bhati > wrote: > > FreeSwitchis new for me. > I want to make my own small application in luaand connect to > mysqlDB. Could anyone please suggest me how to start programming > in Lua? > I am reading below thread for Luaprogramming. I want to boost it > up with small program so that my confidence will increase. > > http://www.lua.org/pil/index.html > > Any book for programming in Lua for freeswitch > -- > > Thanks and regards > > Virendra Bhati > +91-8885268942 > Software Engineer > E-mail-: virbhati at gmail.com > Skype id:- virbhati2 > Hyderabad(India) > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/63b079a1/attachment.html From virbhati at gmail.com Wed Mar 7 13:18:09 2012 From: virbhati at gmail.com (virendra bhati) Date: Wed, 7 Mar 2012 15:48:09 +0530 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 69, Issue 60 In-Reply-To: References: Message-ID: I saw on vbilling with freeswitch they used vbilling.luac. I want to make such sript in lua. So that programming logic will be safe for outside word. How to comple and make bytecode on lua? Which you provide me it not what I want to implement. On Wed, Mar 7, 2012 at 3:41 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: answer when agent connects in mod_callcenter (Anita Hall) > 2. State of GUIs (Brett Wilson) > 3. DTMF passthrough delay / pass_rfc2833 problems (Dmitry Sytchev) > 4. Re: how to create Lua application for Freeswicth (Anita Hall) > > > ---------- Forwarded message ---------- > From: Anita Hall > To: FreeSWITCH Users Help > Cc: > Date: Wed, 7 Mar 2012 13:51:57 +0530 > Subject: Re: [Freeswitch-users] answer when agent connects in > mod_callcenter > Hey Peter > > Much thanks :) It worked like a charm! > > Now I am able to ring an agent via mod_callcenter and if the agent does > not answer, then the caller is not answered. Magic ! > > Now all I need to do is to hangup the caller if the agent does not answer > / cancels .... > > regards, > Anita > > > > On Wed, Mar 7, 2012 at 12:41 PM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > >> Look in the code for switch_channel_answer(), that's what causes the call >> to be answered. >> >> Anyway, even though an unanswered call seems great in theory, it usually >> doesn't work, because the providers will only allow that state for a >> limited time. So in the end, answer is usually what must be done anyway. >> >> /Peter >> >> ----- Reply message ----- >> Fr?n: "Anita Hall" >> Datum: ons, mar 7, 2012 07:24 >> Rubrik: [Freeswitch-users] answer when agent connects in mod_callcenter >> Till: "FreeSWITCH Users Help" >> >> Hi! >> >> I tried mod_callcenter and I must say it is an amazing piece of work by >> Marc Olivier Chouinard! We had built something like this over ESL but >> nothing beats a loadable module :) >> >> The tricky question is how do I configure it so that the caller is >> answered only after an agent is connected? By default, the call is answered >> and moh is played. Is it even possible or do I have to look inside >> mod_callcenter? If that, exactly where does a module answer a call ? >> >> Thanks. >> >> regards, >> Anita >> >> !DSPAM:4f56fcb432761851621280! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > ---------- Forwarded message ---------- > From: "Brett Wilson" > To: > Cc: > Date: Wed, 7 Mar 2012 03:35:36 -0500 > Subject: [Freeswitch-users] State of GUIs > > Hey guys,**** > > I was wondering if anyone had any info on the current state of FS guis? It > seems that most of the gui projects (blue.box, fusionpbx) have been mostly > abandoned in terms of development. I would like something simple for end > users to self-administer if needed. The problem I have with fusionpbx is > that config files are overwritten by whatever is in the database. If you > hand-edit a config file, fusion will not parse the file and load those > settings into the interface. I realize that takes much more coding to do > than using a database to simply write config files. But I feel that the > freeswitch interface could be improved anyway. Unfortunately it seems that > project has been abandoned. There have not been any releases since mid-2011. > **** > > ** ** > > Anything new out or around the corner?**** > > ** ** > > *Brett Wilson* > > *IT Department* > > *Launch 3 Ventures, LLC***** > > 134 Myer Street**** > > Hackensack, NJ 07601**** > > *Phone:* 877.878.9134 > *Fax:* 646.536.3866**** > > *Email:* Brett.Wilson at launch3.net**** > > *AOL IM:* Brett.Wilson at launch3.net**** > > www.Launch3.net**** > > *www.Launch3telecom.com ***** > > ******************************* > > [image: Description: Description: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-Logo] > **** > > ** ** > > > ---------- Forwarded message ---------- > From: Dmitry Sytchev > To: FreeSWITCH Users Help > Cc: > Date: Wed, 7 Mar 2012 13:29:56 +0400 > Subject: [Freeswitch-users] DTMF passthrough delay / pass_rfc2833 problems > Hi all > > I'd like to know is there a way to make freeswitch pass RFC2833 dtmf right > after it receives first packets as in pass_rfc2833 mode, but still > recognize DTMF for bind_meta_app or bind_digit_action? Seems when i enable > pass_rfc2833, bind_meta_app stops working. > > When we use ATA endpoints like Linksys PAP2T or SPA8000 without > pass_rfc2833, ATAs sends little piece of inband DTMF followed by RFC2833 > packets. While inband piece is immediately forwarded by FS, RFC2833 packets > get relayed only after receiving end packets from endpoint, or at least > delayed, effectively making double DTMF on other side. > > We can't use no-media or proxy-media mode as we need to deal with in-call > features activated by DTMF :( > > What can be done to solve this issues? > > > > -- > Best regards, > > Dmitry Sytchev, > IT Engineer > > > ---------- Forwarded message ---------- > From: Anita Hall > To: FreeSWITCH Users Help > Cc: > Date: Wed, 7 Mar 2012 15:40:47 +0530 > Subject: Re: [Freeswitch-users] how to create Lua application for > Freeswicth > You can start from here > http://wiki.freeswitch.org/wiki/Lua_Welcome_IVR_Example > > and the mod_lua page. > > regards, > Anita > > > > On Wed, Mar 7, 2012 at 1:27 PM, virendra bhati wrote: > >> FreeSwitch is new for me. >> I want to make my own small application in lua and connect to mysql DB. >> Could anyone please suggest me how to start programming in Lua ? >> I am reading below thread for Lua programming. I want to boost it up >> with small program so that my confidence will increase. >> >> http://www.lua.org/pil/index.html >> >> Any book for programming in Lua for freeswitch >> -- >> >> Thanks and regards >> >> Virendra Bhati >> +91-8885268942 >> Software Engineer >> E-mail-: virbhati at gmail.com >> Skype id:- virbhati2 >> Hyderabad(India) >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati at gmail.com Skype id:- virbhati2 Hyderabad(India) -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/jpeg Size: 1815 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/2083bb20/attachment-0005.jpe From bote_radio at botecomm.com Wed Mar 7 13:46:46 2012 From: bote_radio at botecomm.com (Bote Man) Date: Wed, 7 Mar 2012 05:46:46 -0500 Subject: [Freeswitch-users] Voicemail Transcription In-Reply-To: <1FFF97C269757C458224B7C895F35F1504D8C5@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1504D8C5@cantor.std.visionutv.se> Message-ID: <00d801ccfc4f$98695e80$c93c1b80$@com> I use Google Voice to record my voice mail messages and its transcriptions are the source of much laughter at times, even when the speaker is fairly articulate. Don't set your expectations too high. Bote From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: Monday, 05 March, 2012 11:22 To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] Voicemail Transcription I think Nuance (www.nuance.com) is the only serious option for this ? it?s quite expensive though. Even though I don?t think that any solution really work that good... /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Balamurugan Mahendran Skickat: den 5 mars 2012 17:14 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Voicemail Transcription Hi all, We are looking for a tool to transcribe voicemail, I was trying with Sphinx but the quality is not okay to use. Please suggest me the right tool for Voicemail Transcription. Thanks, Bala !DSPAM:4f54e52332761375616741! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/f8253c1e/attachment.html From miha at softnet.si Wed Mar 7 13:48:22 2012 From: miha at softnet.si (Miha Zoubek) Date: Wed, 07 Mar 2012 11:48:22 +0100 Subject: [Freeswitch-users] Destination number Message-ID: <4F573CF6.9000205@softnet.si> Hi, I have user rigstered on FS as exp. 1232456.freeswitch.si. Problem is that I am also getting destination number in CDR as this, but I would like that destiantion number is just number so, in this calse 1232456. IS that possible? Thanks! Miha From shahzad.bhatti at g-r-v.com Wed Mar 7 13:58:58 2012 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Wed, 7 Mar 2012 15:58:58 +0500 Subject: [Freeswitch-users] How to Create PHP Application for Freeswicth using ESL Message-ID: Hi Everyone, i am a new user and want to learn how to create php application using ESL i just want to play a dynamic message using cepstral and get the customer response using dtmf. but i can't unserstand how to handle call using php as their is no mod_php is available like mod_lua, i park the call and now want to speak in the call but unable to do: please guide me i can't solve that problem with playandgetdigits() example of Mr. Michael Collins that what is the code of [handler.php]. ===================================================================================================================== * * *originate user/1500 meeting_request_1500* Then create that extension: =================================================================================================================== is anyone guide me how to do it php, thanks in advance, Shahzad Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/6757e53c/attachment.html From b2m at a-cti.com Wed Mar 7 14:16:58 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Wed, 7 Mar 2012 16:46:58 +0530 Subject: [Freeswitch-users] Voicemail Transcription In-Reply-To: <00d801ccfc4f$98695e80$c93c1b80$@com> References: <1FFF97C269757C458224B7C895F35F1504D8C5@cantor.std.visionutv.se> <00d801ccfc4f$98695e80$c93c1b80$@com> Message-ID: how to setup? do you have any articles to fallow? Thanks, Bala On Wed, Mar 7, 2012 at 4:16 PM, Bote Man wrote: > I use Google Voice to record my voice mail messages and its transcriptions > are the source of much laughter at times, even when the speaker is fairly > articulate. Don't set your expectations too high.**** > > ** ** > > Bote**** > > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Peter Olsson > *Sent:* Monday, 05 March, 2012 11:22 > *To:* 'FreeSWITCH Users Help' > *Subject:* Re: [Freeswitch-users] Voicemail Transcription**** > > ** ** > > I think Nuance (www.nuance.com) is the only serious option for this ? > it?s quite expensive though. Even though I don?t think that any solution > really work that good...**** > > **** > > /Peter**** > > **** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org] > *F?r *Balamurugan Mahendran > *Skickat:* den 5 mars 2012 17:14 > *Till:* FreeSWITCH Users Help > *?mne:* [Freeswitch-users] Voicemail Transcription**** > > **** > > Hi all,**** > > **** > > We are looking for a tool to transcribe voicemail, I was trying with > Sphinx but the quality is not okay to use. Please suggest me the right tool > for Voicemail Transcription.**** > > **** > > Thanks,**** > > Bala**** > > !DSPAM:4f54e52332761375616741! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/e1d01449/attachment-0001.html From andrew at cassidywebservices.co.uk Wed Mar 7 14:45:41 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 7 Mar 2012 11:45:41 +0000 Subject: [Freeswitch-users] State of GUIs In-Reply-To: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> Message-ID: Look at http://www.freepybx.org although I didn't manage to get it installed. I do have fusion installed on a text box although personally I'm not keen on it. On 7 March 2012 08:35, Brett Wilson wrote: > Hey guys,**** > > I was wondering if anyone had any info on the current state of FS guis? It > seems that most of the gui projects (blue.box, fusionpbx) have been mostly > abandoned in terms of development. I would like something simple for end > users to self-administer if needed. The problem I have with fusionpbx is > that config files are overwritten by whatever is in the database. If you > hand-edit a config file, fusion will not parse the file and load those > settings into the interface. I realize that takes much more coding to do > than using a database to simply write config files. But I feel that the > freeswitch interface could be improved anyway. Unfortunately it seems that > project has been abandoned. There have not been any releases since mid-2011. > **** > > ** ** > > Anything new out or around the corner?**** > > ** ** > > *Brett Wilson* > > *IT Department* > > *Launch 3 Ventures, LLC***** > > 134 Myer Street**** > > Hackensack, NJ 07601**** > > *Phone:* 877.878.9134 > *Fax:* 646.536.3866**** > > *Email:* Brett.Wilson at launch3.net**** > > *AOL IM:* Brett.Wilson at launch3.net**** > > www.Launch3.net**** > > *www.Launch3telecom.com ***** > > ******************************* > > [image: Description: Description: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-Logo] > **** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/jpeg Size: 1815 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/0519033e/attachment-0002.jpe From andrew at cassidywebservices.co.uk Wed Mar 7 14:50:30 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 7 Mar 2012 11:50:30 +0000 Subject: [Freeswitch-users] FreeSWITCH Conf Call Tomorrow - Dialplan Basics In-Reply-To: References: Message-ID: I added a UK DID last week as I was having difficulty with my own equipment (which I've now resolved) +44 3300 100 290. It will be left there if there's interest. On 7 March 2012 00:54, Brian Foster wrote: > I agree with Jeff, great presentation topic. I'll be there for sure. The > cool thing is once you learn the concept behind dialplans, it's so much > easier when it comes to the more advanced stuff. See you all there! > > -BDF > On Mar 6, 2012 7:06 PM, "curriegrad2004" wrote: > >> This is a great presentation topic. I find that a lot of new users >> can't really be bothered to read the wiki or they just >> find the wiki's too complicated to read in the first place. I hope a >> presentation/recording like this can really help out >> the new users out there on how FreeSWITCH should be really configured. >> >> On Tue, Mar 6, 2012 at 3:20 PM, Michael Collins >> wrote: >> > Hello all! >> > >> > Here's tomorrow's agenda page: >> > http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_06 >> > >> > I will be discussing some dialplan basics along with some regex tricks >> and >> > basic patterns. Other experts may also be chiming in. The idea is that >> > everyone on the call will come away with a more thorough understanding >> of >> > what happens when a phone call comes into FreeSWITCH. >> > >> > Talk to you tomorrow! >> > >> > -Michael >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/3dc882d4/attachment-0001.html From freeswitch-list at puzzled.xs4all.nl Wed Mar 7 15:16:00 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 07 Mar 2012 13:16:00 +0100 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 69, Issue 60 In-Reply-To: References: Message-ID: <4F575180.7080909@puzzled.xs4all.nl> On 07-03-12 11:18, virendra bhati wrote: > I saw on vbilling with freeswitch they used vbilling.luac. I want to > make such sript in lua. So that programming logic will be safe for > outside word. > How to comple and make bytecode on lua? http://lmgtfy.com/?q=luac Not really FreeSWITCH related isn't it? Not sure what you want to accomplish but you do realize that a luac file can easily be decompiled with luadec? Next time can you at least make some effort to try to find the answer yourself before asking such a generic Lua question? It took a single, one term Google search to find your answer. Patrick From freeswitch-list at puzzled.xs4all.nl Wed Mar 7 15:19:19 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 07 Mar 2012 13:19:19 +0100 Subject: [Freeswitch-users] State of GUIs In-Reply-To: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> Message-ID: <4F575247.9030005@puzzled.xs4all.nl> On 07-03-12 09:35, Brett Wilson wrote: > Hey guys, > > I was wondering if anyone had any info on the current state of FS guis? > It seems that most of the gui projects (blue.box, fusionpbx) have been > mostly abandoned in terms of development. I don't think that is true. Both the 2600hz (blue.box) and fusionpbx irc channels are quite active and I know for a fact that at least fusionpbx gets daily commits. Why don't you hop on irc and ask in both channels (#2600hz: ask pyite and #fusionpbx: ask mcrane). Regards, Patrick From virbhati at gmail.com Wed Mar 7 15:19:43 2012 From: virbhati at gmail.com (virendra bhati) Date: Wed, 7 Mar 2012 17:49:43 +0530 Subject: [Freeswitch-users] .Lua and .Luac are different ?? Message-ID: Hi, .Lua and .Luac are different or only complication difference ? How to make and compile application in .Luac ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati at gmail.com Skype id:- virbhati2 Hyderabad(India) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/fa1c9dbb/attachment.html From drazen.blanusa at nth.ch Wed Mar 7 15:21:28 2012 From: drazen.blanusa at nth.ch (Drazen Blanusa) Date: Wed, 7 Mar 2012 13:21:28 +0100 Subject: [Freeswitch-users] SHUTDOWN event Message-ID: <00bc01ccfc5c$d25cce30$77166a90$@blanusa@nth.ch> Hello, I'm using Java ESL library for communicating with FreeSWITCH. If i subscribe for events, using method setEventSubscriptions, and define list of events, in case of SHUTDOWN event, FreeSWITCH says: [NOTICE] switch_event.c:1889 Event Binding deleted for mod_local_stream:SHUTDOWN If I set subscription for ALL events, than it's working fine. NOTE: it happens only in case of SHUTDOWN event, other events works fine. Anybody knows where is the problem? I've checked ESL subscription string, and it seems to be ok, so I guess it is problem on FreeSWITCH side. Best regards, Drazen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/5a5e8c0d/attachment.html From avi at avimarcus.net Wed Mar 7 15:23:44 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 7 Mar 2012 14:23:44 +0200 Subject: [Freeswitch-users] State of GUIs In-Reply-To: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> Message-ID: On Wed, Mar 7, 2012 at 10:35 AM, Brett Wilson wrote: > Hey guys,**** > > I was wondering if anyone had any info on the current state of FS guis? It > seems that most of the gui projects (blue.box, fusionpbx) have been mostly > abandoned in terms of development. > Hardly. I don't follow the 2600hz / blue.box project, but it could be they are mostly focusing on enterprise management and deployment, rather than actual PBX functionality. They have a pretty slick setup already.. FusionPBX - if you simply watch the SVN commit log - is being refactored with a new layer of database normalization for both compatibility and ease of writing new applications, that can be deployed and/or migrated to any system with your data. (IDs have recently been changed to UUIDs to ease migration.) These projects are hardly abandoned. > I would like something simple for end users to self-administer if needed. > The problem I have with fusionpbx is that config files are overwritten by > whatever is in the database. If you hand-edit a config file, fusion will > not parse the file and load those settings into the interface. > Indeed, such is the case. If you create a file in fusionpbx and want to edit it.. you can rename it (remove the v_ prefix) and fusionpbx won't overwrite it.. you'll have to remove (or disable) it in fusionpbx though or you'll have duplicate entries. If you want it to suck back in changes... it's an edge case, most people don't really need that. If you do, you can write it. I'll even help you test it. > I realize that takes much more coding to do than using a database to > simply write config files. But I feel that the freeswitch interface could > be improved anyway. Unfortunately it seems that project has been abandoned. > There have not been any releases since mid-2011.**** > > ** ** > > Anything new out or around the corner? > http://www.freepybx.org as mentioned was just released - python with flash stuff for a fancy UI. There's also vbilling but that's more a billing management than PBX functionality. None of the PBXs really let you enforce per-minute billing though (there's too many places to lock down if you let them set up the dialplans...) so that's still a major hole. I wrote my own stuff... mostly, without a gui for users to make changes, yet. -Avi Marcus > **** > > ** ** > > *Brett Wilson* > > *IT Department* > > *Launch 3 Ventures, LLC***** > > 134 Myer Street**** > > Hackensack, NJ 07601**** > > *Phone:* 877.878.9134 > *Fax:* 646.536.3866**** > > *Email:* Brett.Wilson at launch3.net**** > > *AOL IM:* Brett.Wilson at launch3.net**** > > www.Launch3.net**** > > *www.Launch3telecom.com ***** > > ******************************* > > [image: Description: Description: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-Logo] > **** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/f9c7f174/attachment-0001.html From benkokakao at gmail.com Wed Mar 7 15:26:47 2012 From: benkokakao at gmail.com (Christian Benke) Date: Wed, 7 Mar 2012 13:26:47 +0100 Subject: [Freeswitch-users] Destination number In-Reply-To: <4F573CF6.9000205@softnet.si> References: <4F573CF6.9000205@softnet.si> Message-ID: On 7 March 2012 11:48, Miha Zoubek wrote: > I would like that destiantion number is just number so, in this calse > 1232456. > > IS that possible? You have several options: You can always manipulate destination_number to whatever you want it to look like and use a different variable in the dialstring, e.g.: You could use accountcode as an indicator: http://wiki.freeswitch.org/wiki/Variable_accountcode Or you can modify the values in the cdrs alltogether in conf/autoload_configs/cdr_csv.conf.xml, see http://wiki.freeswitch.org/wiki/Mod_cdr_csv Best regards, Christian From sos at sokhapkin.dyndns.org Wed Mar 7 15:31:36 2012 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 07 Mar 2012 07:31:36 -0500 Subject: [Freeswitch-users] .Lua and .Luac are different ?? In-Reply-To: References: Message-ID: <3332164.QDuYDpDZfd@sos> .luac is a precompiled .lua file. Run $ luac --help On Wednesday 07 March 2012 17:49:43 virendra bhati wrote: > Hi, > .Lua and .Luac are different or only complication difference ? > How to make and compile application in .Luac ? From miha at softnet.si Wed Mar 7 15:57:56 2012 From: miha at softnet.si (Miha Zoubek) Date: Wed, 07 Mar 2012 13:57:56 +0100 Subject: [Freeswitch-users] Destination number In-Reply-To: References: <4F573CF6.9000205@softnet.si> Message-ID: <4F575B54.3050403@softnet.si> On 03/07/2012 01:26 PM, Christian Benke wrote: > On 7 March 2012 11:48, Miha Zoubek wrote: >> I would like that destiantion number is just number so, in this calse >> 1232456. >> >> IS that possible? > You have several options: > > You can always manipulate destination_number to whatever you want it > to look like and use a different variable in the dialstring, e.g.: > > > > > data="destination_number=${regex(${dialed_extension}|^([0-9]{3,}).freeswitch.si$|%1)}"/> > data="user/${dialed_extension}@${domain_name}"/> > > > > > You could use accountcode as an indicator: > http://wiki.freeswitch.org/wiki/Variable_accountcode > > Or you can modify the values in the cdrs alltogether in > conf/autoload_configs/cdr_csv.conf.xml, see > http://wiki.freeswitch.org/wiki/Mod_cdr_csv > > Best regards, > Christian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Hi @Christian, I have set in my dialplan in my dialplan fs set: 2012-03-07 13:47:11.179122 [NOTICE] switch_core_session.c:2365 Execute set(accountcode=custom) EXECUTE sofia/internal/018108500.fs_kabelvoip1 at fs_kabelvoip1.fs1.softnet.si set(accountcode=custom) In my cdr.conf.xml file: Must I comment this: Because for default template is stil executed? Thanks! Miha From adam.kelloway at newpace.ca Wed Mar 7 16:02:54 2012 From: adam.kelloway at newpace.ca (Adam Kelloway) Date: Wed, 07 Mar 2012 09:02:54 -0400 Subject: [Freeswitch-users] Any alternative to repetitive lua session:ready() calls? In-Reply-To: References: <4F5639AA.9050900@newpace.ca> <1FFF97C269757C458224B7C895F35F1504E178@cantor.std.visionutv.se> <4F564B01.7050404@newpace.ca> Message-ID: <4F575C7E.7050800@newpace.ca> That is indeed what I was looking for, thanks for your help. Adam On 3:59 PM, Gerald Weber wrote: > > Morning, > > i had the same problem a while ago, reported the jira below and anthm > made some changes tu mod_lua. > > If you put "error()" as last command in your hangup handler (see > http://wiki.freeswitch.org/wiki/Mod_lua#session:setHangupHook) > > No error is dumped in the log. > > Both 'return "exit"' and 'return "die"' leave an error in the logfiles. > > A little bit misleading in my opinion, but if you know how to handle > it...it works. > > Tried 15mins ago with latest git FreeSWITCH Version 1.0.head > (git-ea975c3 2012-02-28 19-21-04 -0500) > > Regards > > gw > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von > *Adam Kelloway > *Gesendet:* Dienstag, 06. M?rz 2012 18:36 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] Any alternative to repetitive lua > session:ready() calls? > > Kind of, all of those solutions consider the script exiting an > 'error'. I am looking for something that exits without considering it > an error and thus does not emit an [ERR] log. > > On 3:59 PM, Peter Olsson wrote: > > Maybe this is what you are looking for? > > commit 09ad887948f7513725ca8b53bdfe721d9008e73b > Author: Anthony Minessale > Date: Fri Jan 27 19:03:04 2012 -0600 > > FS-3841 --resolve ok return the string "die" or "exit" from hanguphook to cause an error or call s:destroy("any err message"); either should now halt the script > > > ________________________________________ > Fr?n:freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org ] för Adam Kelloway [adam.kelloway at newpace.ca ] > Skickat: den 6 mars 2012 17:22 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] Any alternative to repetitive lua session:ready() calls? > > Hi there, > > If I am running a lua script during a call, and the call terminates > (caller hangs up, for instance), the lua script continues to execute > while the session has already been terminated. This causes ERR entries > in the freeswitch log, which I would like to try and minimize. The only > way I can see doing this is to frequently call session:ready() before > doing anything that could break the script. Is there a better way to > exit a script cleanly, without having to add X number of session:ready() > calls scattered about the script? Currently, I only call it once (at the > beginning of the script). I tried using the hangup hook, but I'm not > sure how I can get it to immediately exit the script. > > Thanks, > > Adam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f56388f32766945510869! > > > > > -- > Adam > -- > > NewPace Logo > > > > > > > *Adam Kelloway* > > > > > Software Engineer, NewPace > > phone > > > > +1 (902) 406--8375 x1031 > > email > > > > Adam.Kelloway at NewPace.com > > aim /msn > > > > > Adam.Kelloway @NewPace.ca > -- Adam -- NewPace Logo Adam Kelloway Software Engineer, NewPace phone +1 (902) 406--8375 x1031 email Adam.Kelloway at NewPace.com aim /msn Adam.Kelloway @NewPace.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/f39338d3/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Newpace_50x50.png Type: image/png Size: 4620 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/f39338d3/attachment-0001.png From benkokakao at gmail.com Wed Mar 7 16:16:39 2012 From: benkokakao at gmail.com (Christian Benke) Date: Wed, 7 Mar 2012 14:16:39 +0100 Subject: [Freeswitch-users] Destination number In-Reply-To: <4F575B54.3050403@softnet.si> References: <4F573CF6.9000205@softnet.si> <4F575B54.3050403@softnet.si> Message-ID: > I have set in my dialplan data="accountcode=custom"/> > > in my dialplan fs set: > > 2012-03-07 13:47:11.179122 [NOTICE] switch_core_session.c:2365 Execute > set(accountcode=custom) > EXECUTE > sofia/internal/018108500.fs_kabelvoip1 at fs_kabelvoip1.fs1.softnet.si > set(accountcode=custom) > > > In my cdr.conf.xml file: > > > > > Must I comment this: > > Because for default template is stil executed? Hmm, no, don't comment it out - if you want the "custom"-template to be used, set The accountcode-variable on the other hand does not define which template is used, but it is a variable used in the cdr-files. In your "custom"-template it's the variable you defined between "${bleg_uuid}" and "${read_codec}" Regards, Christain From admin at blindi.net Wed Mar 7 16:27:32 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Wed, 7 Mar 2012 14:27:32 +0100 (CET) Subject: [Freeswitch-users] Fs problem: can.t send queue_dtmf in loopback ignore_early_media=true not working In-Reply-To: <4F575B54.3050403@softnet.si> References: <4F573CF6.9000205@softnet.si> <4F575B54.3050403@softnet.si> Message-ID: Hi guys, I use the actual git from today. I create a queue_dtmf extension: fs does not wait until the connection is established. This problem occurs only with loopback. ignore_early_media = true shows no effect. I should route directly over a gateway, it works out. for example: works correctly. Can your looks for this problem please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From Rob.Moore at Aeriandi.com Wed Mar 7 18:58:16 2012 From: Rob.Moore at Aeriandi.com (Rob Moore) Date: Wed, 7 Mar 2012 15:58:16 +0000 Subject: [Freeswitch-users] SIP Invite Contact Header Message-ID: <49C5FCA19A8A114493EBAACA42FE5899104EB9A7@1AERDCEXCHMBX1.AER.AERCO.local> HI All, Hopefully a quick question here. I'm trying to get TLS working with one of our gateway providers and they are asking that I pass an FQDN instead of the external IP address of our Freeswitch server in our contact header within our sip invite. I've tried setting sip_contact_user but that doesn't seem to have any effect on the contents of the contact header. Plus I would really prefer to set this in the gateway profile so it only effects calls passing over a single gateway. Anyone have any advice on how this can be achieved? Thanks Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/169a12e8/attachment.html From msc at freeswitch.org Wed Mar 7 19:24:26 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 7 Mar 2012 08:24:26 -0800 Subject: [Freeswitch-users] [Freeswitch-dev] New Book! In-Reply-To: <1331108482.14853.YahooMailNeo@web120302.mail.ne1.yahoo.com> References: <1331108482.14853.YahooMailNeo@web120302.mail.ne1.yahoo.com> Message-ID: Much appreciated! Thanks, MC On Wed, Mar 7, 2012 at 12:21 AM, Ali R. wrote: > Hello All, > I just received the cookbook from amazon.com today and I just wanted to > say it's really well done. Many thanks for such valuable reference. > In a scale of 1 to 10, I would give this book 11 > > Ali R. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/1a9e3a55/attachment.html From dgarcia at anew.com.ve Wed Mar 7 19:27:40 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Wed, 07 Mar 2012 11:57:40 -0430 Subject: [Freeswitch-users] answer when agent connects in mod_callcenter In-Reply-To: References: <98AE7CD6-4640-464D-8AD7-58A63A40B2E5@visionutveckling.se> Message-ID: <4F578C7C.4040702@anew.com.ve> Hi Anita, I work with different call center solutions like Alcatel, Genesys, Avaya, Nortel, etc. I have not play with mod_callcenter yet. Consider what Peter said about keep the call as unanswered. In a tradicional PBX with an ACD solution the answer state is send to asure to the PSTN or SIP provider that the call is being handled. On the other hand, the "logic" in the ACD the call is park/hold/ in queue until an agent become avalaible or after a timeout the call is rerouted to another point or dropped. It is not a good practice hangup the caller if the agent does not answer, you will "transmit" an idea of "BAD SERVICE". You could instead to recover the call and locate another avalaible agent and put in "Not Ready" or "Logout" the agent who does not answer a call in x seconds. It is a better approach. On 3/7/2012 3:51 AM, Anita Hall wrote: > Hey Peter > > Much thanks :) It worked like a charm! > > Now I am able to ring an agent via mod_callcenter and if the agent > does not answer, then the caller is not answered. Magic ! > > Now all I need to do is to hangup the caller if the agent does not > answer / cancels .... > > regards, > Anita > > > > On Wed, Mar 7, 2012 at 12:41 PM, Peter Olsson > > wrote: > > Look in the code for switch_channel_answer(), that's what causes > the call to be answered. > > Anyway, even though an unanswered call seems great in theory, it > usually doesn't work, because the providers will only allow that > state for a limited time. So in the end, answer is usually what > must be done anyway. > > /Peter > > ----- Reply message ----- > Fr?n: "Anita Hall" > > Datum: ons, mar 7, 2012 07:24 > Rubrik: [Freeswitch-users] answer when agent connects in > mod_callcenter > Till: "FreeSWITCH Users Help" > > > > Hi! > > I tried mod_callcenter and I must say it is an amazing piece of > work by Marc Olivier Chouinard! We had built something like this > over ESL but nothing beats a loadable module :) > > The tricky question is how do I configure it so that the caller is > answered only after an agent is connected? By default, the call is > answered and moh is played. Is it even possible or do I have to > look inside mod_callcenter? If that, exactly where does a module > answer a call ? > > Thanks. > > regards, > Anita > > !DSPAM:4f56fcb432761851621280! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1913 / Virus Database: 2114/4854 - Release Date: 03/06/12 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/0e1f9fbb/attachment-0001.html From msc at freeswitch.org Wed Mar 7 19:31:57 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 7 Mar 2012 08:31:57 -0800 Subject: [Freeswitch-users] how to create Lua application for Freeswicth In-Reply-To: <4F573526.3000004@earthspike.net> References: <4F573526.3000004@earthspike.net> Message-ID: On Wed, Mar 7, 2012 at 2:15 AM, John wrote: > And the FreeSWITCH book ("bridge book") has a chapter as well (or maybe 2 > - I've lent my copy to someone at the moment). > It has a single Lua chapter (#7) written by yours truly. :) It's a really good place to start. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/649cdd54/attachment.html From bdfoster at endigotech.com Wed Mar 7 21:04:11 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 7 Mar 2012 13:04:11 -0500 Subject: [Freeswitch-users] New batch of sound files In-Reply-To: References: Message-ID: Thanks for the heads up. I did download the lastest set of prompts per your instructions, however I don't see the "There are..." prompt. I see the other two for what would be said after the conference count is announced though. I've looked all over the sounds directory trying to find it, even playing back likely candidates but to no avail. Is that prompt missing? -BDF On Mon, Mar 5, 2012 at 2:50 PM, Michael Collins wrote: > Ask and ye shall receive! > > The files have been out on files.freeswitch.org for the past week or so. > Look for version 1.0.18. Also, in my previous email on the subject I > mentioned this trick for forcing your system to get the latest sounds: > edit ${fs_src}/build/sounds_version.txt > change callie to "1.0.18" > save & exit > "make cd-sounds-install" > > New sounds will be downloaded. As far as the prompts you mentioned, I > added several prompts that we can use to piece together some phrase files > or whatnot. I've got two different wordings depending on your preference: > > There is one other person in this conference > There is one other member in this conference > There are... > ...other members in this conference > ...other persons in this conference > > Now you have the pieces you need to build a nice little phrase. > > -MC > > On Sat, Mar 3, 2012 at 3:23 PM, Brian Foster wrote: > >> Hola, >> >> I briefly heard in channel and on conference about a new batch of sound >> files for freeswitch. Is there a download link out there for this? Also, is >> there any notable prompts for conferences? I'm specifically looking for >> some prompts that would take care of telling the incoming participant how >> many other participants are in the conference. >> >> Thanks! >> -BDF >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/083aac25/attachment.html From kris at kriskinc.com Wed Mar 7 21:22:38 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 7 Mar 2012 13:22:38 -0500 Subject: [Freeswitch-users] DTMF passthrough delay / pass_rfc2833 problems In-Reply-To: References: Message-ID: Dmitry, If your ATA devices aren't clamping the DTMF quickly enough you may have to just use inband. On Wed, Mar 7, 2012 at 4:29 AM, Dmitry Sytchev wrote: > Hi all > > I'd like to know is there a way to make freeswitch pass RFC2833 dtmf right > after it receives first packets as in pass_rfc2833 mode, but still recognize > DTMF for bind_meta_app or bind_digit_action? Seems when i enable > pass_rfc2833, bind_meta_app stops working. > > When we use ATA endpoints like Linksys PAP2T or SPA8000 without > pass_rfc2833, ATAs sends little piece of inband DTMF followed by RFC2833 > packets.?While inband piece is immediately forwarded by FS, RFC2833 packets > get relayed only after receiving end packets from endpoint, or at least > delayed, effectively making double DTMF on other side. > > We can't use no-media or proxy-media mode as we need to deal with in-call > features activated by DTMF :( > > What can be done to solve this issues? > > > > -- > Best regards, > > Dmitry Sytchev, > IT Engineer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From anthony.minessale at gmail.com Wed Mar 7 21:47:36 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Mar 2012 12:47:36 -0600 Subject: [Freeswitch-users] DTMF passthrough delay / pass_rfc2833 problems In-Reply-To: References: Message-ID: Add IGNORE_DTMF_DURATION to manual-rtp-bugs sofia profile param or rtp_manual_rtp_bugs channel variable. This should make FS react on receipt of the first packet in the dtmf event. You'll still be subject to waiting for the next "first" event before you get another. On Wed, Mar 7, 2012 at 12:22 PM, Kristian Kielhofner wrote: > Dmitry, > > ?If your ATA devices aren't clamping the DTMF quickly enough you may > have to just use inband. > > On Wed, Mar 7, 2012 at 4:29 AM, Dmitry Sytchev wrote: >> Hi all >> >> I'd like to know is there a way to make freeswitch pass RFC2833 dtmf right >> after it receives first packets as in pass_rfc2833 mode, but still recognize >> DTMF for bind_meta_app or bind_digit_action? Seems when i enable >> pass_rfc2833, bind_meta_app stops working. >> >> When we use ATA endpoints like Linksys PAP2T or SPA8000 without >> pass_rfc2833, ATAs sends little piece of inband DTMF followed by RFC2833 >> packets.?While inband piece is immediately forwarded by FS, RFC2833 packets >> get relayed only after receiving end packets from endpoint, or at least >> delayed, effectively making double DTMF on other side. >> >> We can't use no-media or proxy-media mode as we need to deal with in-call >> features activated by DTMF :( >> >> What can be done to solve this issues? >> >> >> >> -- >> Best regards, >> >> Dmitry Sytchev, >> IT Engineer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From miha at softnet.si Wed Mar 7 23:20:47 2012 From: miha at softnet.si (Miha) Date: Wed, 07 Mar 2012 21:20:47 +0100 Subject: [Freeswitch-users] (no subject) Message-ID: Hi, Where Can I found something about this variable, how to set it, etc? I was searching on wiki, but there is no data. I found on internet that I can use it for attended tranfers (Caller_id_name). THanks Miha From potxoka at gmail.com Wed Mar 7 23:41:50 2012 From: potxoka at gmail.com (Anto) Date: Wed, 7 Mar 2012 21:41:50 +0100 Subject: [Freeswitch-users] Codec negotiation with carriers In-Reply-To: References: Message-ID: Hello Attached file, with the traces of the different tests (with different configurations). http://pastebin.freeswitch.org/18599 I have read the url that you mentioned, the initial guide FreeSWITCH, that of mod_sofia, applications, etc.. I believe that most of the wiki (maybe when do not give the solution, read as much documentation is worse idea :-S, lock me more). I made ??a configuration that works (I have not tested the audio), but earlier (before I started "touch" the profiles) if I could talk to a physical phone (several times). The problem is that I can not understand why it works and sometimes not, and I would like to learn :-). Not only do and forget, so I would like to learn and less disturbing to the mail list and (maybe in the future) to help other newbies like me :-). Thanks ! Best regards Anto 2012/3/7 Michael Collins : > You may want to read up on codec negotiation: > http://wiki.freeswitch.org/wiki/Codec_negotiation > > There are different ways to handle codecs depending on your needs. I'd read > that page first and then try out some of the suggestions. If you're still > having trouble then I'd recommend getting SIP traces of the traffic and > putting them on pastebin.freeswitch.org. The gang here is pretty good at > looking over logs and helping with diagnosing problems. :) > > -MC > > On Tue, Mar 6, 2012 at 2:30 PM, Anto wrote: >> >> Hi >> >> I am following the steps in this direction >> "http://wiki.freeswitch.org/wiki/SBC_Setup" and >> "http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice", >> I reread the whole entire wiki (or so I lack), but do not quite >> assimilate or finding the right formula to operate the bridge :-S. >> >> I captured traffic with ngrep, I enabled sip-trace, console logconsole >> 8, etc., but unless the transcoding error (only two of the hundreds of >> combinations of settings that I have), I have not seen anything >> "weird" :-S >> >> I have 3 suppliers, each with this codec: >> >> 1) ? ? ? ? ? 2) ? ? ? ? ? ? ?3) >> G729 ? ? ? ?G729 ? ? ? ?G729 >> G711u ? ? ?G711A ? ? ?G711A >> G711A ? ? G711u ? ? ? G711u >> ? ? ? ? ? ? ? ?G723 ? ? ? ? G723 >> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?G722 >> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?GSM >> >> I think I understand that when making an outside call, FreeSWITCH >> follow these steps: >> >> USER -> ( ? Dialplan -> profile (internal) -> bridge (external) -> >> profile (external) ? ) -> PROVIDER >> >> PROVIDER -> ( ? Dialplan -> profile (external) -> bridge (internal) -> >> profile (internal) ?) -> USER >> >> right? >> >> Internal and external I set as follows (and not many changes have >> done, and not remember it, because I've been testing days). If >> outbound (outbound-codec-prefs) all codecs specified system does not >> handle the call, I have to specify these by hand. If active >> inbound-proxy-media, not the caller. Some of the time I worked, but >> gave me an error that it can do transcoding G729 codec (I do >> passthrough), but the proxy does not work half. >> >> If the outbound property (outbound-codec-prefs) all codecs specified >> system does not handle the call, I have to specify these by hand. If >> active inbound-proxy-media, not the caller. Some of the time I worked, >> but gave me an error that it can do transcoding G729 codec (I want to >> make passthrough), but the "proxy media" does not work. >> >> Basically, what I do is that local users can use all the codecs >> allowed (iLBC, GSM, ...) and make an outside call, use the carrier >> that will indicate the priority but the free codec. >> >> With this configuration works for me, but I would like to understand >> why so if it works and otherwise no. Coming to understand how to >> configure properly and so as not to disturb the mail list ;-). Thanks >> ! >> >> Best regards >> Anto >> >> vars.xml >> >> > >> data="global_codec_prefs=iLBC,G7221,speex,PCMU,PCMA,BV16,G726-32,GSM,G729,G723,AMR"/> >> > >> data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,iLBC,G7221,speex,BV16,G726-32,GSM"/> >> >> internal.xml >> >> >> >> >> >> >> >> >> >> external.xml >> >> >> >> >> >> >> >> >> >> >> >> dialplan/outbound.xml >> >> >> ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? ?> expression="^(\d+)$"> >> ? ? ? ? ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? ? ? ? ?> data="sofia/gateway/provider-2/$1"/> >> ? ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gabe at gundy.org Thu Mar 8 00:04:16 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 7 Mar 2012 14:04:16 -0700 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: On Wed, Mar 7, 2012 at 1:20 PM, Miha wrote: > Where Can I found something about this variable, how to set > it, etc? > > I was searching on wiki, but there is no data. > > I found on internet that I can use it for attended tranfers > (Caller_id_name). I'm not entirely sure what you're looking for, but here is the link with Caller ID related vars: http://wiki.freeswitch.org/wiki/Channel_Variables#Caller_ID_Related Also, please remember to use a subject when emailing the list. It will help us, help you. Best, Gabe From benkokakao at gmail.com Thu Mar 8 00:13:36 2012 From: benkokakao at gmail.com (Christian Benke) Date: Wed, 7 Mar 2012 22:13:36 +0100 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: On 7 March 2012 21:20, Miha wrote: > Where Can I found something about this variable, how to set > it, etc? > > I was searching on wiki, but there is no data. > > I found on internet that I can use it for attended tranfers > (Caller_id_name). Please rephrase your questiion - what are you trying to achieve? caller_id_name is simply the variable that carries the callers name. It is not required for attended transfers per se. Here are some starting points: http://wiki.freeswitch.org/wiki/Mod_dptools http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set http://wiki.freeswitch.org/wiki/Dialplan_XML http://wiki.freeswitch.org/wiki/Getting_Started_Guide Regards, Christian From rhuddleston at gmail.com Thu Mar 8 00:17:10 2012 From: rhuddleston at gmail.com (Robert) Date: Wed, 07 Mar 2012 16:17:10 -0500 Subject: [Freeswitch-users] OpenSIPS Message-ID: Any OpenSIPS experts on this list? I posted to the OpenSIPS mailing list ? and I don't want to cross post here either. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/37c2e4f8/attachment.html From bdfoster at endigotech.com Thu Mar 8 00:23:34 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 7 Mar 2012 16:23:34 -0500 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: I'm not really sure you should be setting that variable. You might want to try setting effective_caller_id_name and effective_caller_id_number instead of the caller_id_name and caller_id_number. -BDF On Wed, Mar 7, 2012 at 4:13 PM, Christian Benke wrote: > On 7 March 2012 21:20, Miha wrote: > > Where Can I found something about this variable, how to set > > it, etc? > > > > I was searching on wiki, but there is no data. > > > > I found on internet that I can use it for attended tranfers > > (Caller_id_name). > > Please rephrase your questiion - what are you trying to achieve? > > caller_id_name is simply the variable that carries the callers name. > It is not required for attended transfers per se. > > Here are some starting points: > http://wiki.freeswitch.org/wiki/Mod_dptools > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set > http://wiki.freeswitch.org/wiki/Dialplan_XML > http://wiki.freeswitch.org/wiki/Getting_Started_Guide > > Regards, > Christian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/96b49fce/attachment.html From bdfoster at endigotech.com Thu Mar 8 00:24:41 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 7 Mar 2012 16:24:41 -0500 Subject: [Freeswitch-users] OpenSIPS In-Reply-To: References: Message-ID: There might be a few. Unless it's freeswitch related, I wouldn't post here though ;-) -BDF On Wed, Mar 7, 2012 at 4:17 PM, Robert wrote: > Any OpenSIPS experts on this list? > > I posted to the OpenSIPS mailing list ? and I don't want to cross post > here either. > > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/d6bcc675/attachment.html From shaheryarkh at googlemail.com Thu Mar 8 09:08:06 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Thu, 8 Mar 2012 11:08:06 +0500 Subject: [Freeswitch-users] OpenSIPS In-Reply-To: References: Message-ID: I am here. I have FreeSWITCH cluster load balanced and fail over by opensips. :) Thank you. On Thu, Mar 8, 2012 at 2:24 AM, Brian Foster wrote: > There might be a few. Unless it's freeswitch related, I wouldn't post here > though ;-) > > -BDF > > On Wed, Mar 7, 2012 at 4:17 PM, Robert wrote: > >> Any OpenSIPS experts on this list? >> >> I posted to the OpenSIPS mailing list ? and I don't want to cross post >> here either. >> >> Thanks >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/9d83e4f3/attachment-0001.html From miha at softnet.si Thu Mar 8 10:01:50 2012 From: miha at softnet.si (Miha Zoubek) Date: Thu, 08 Mar 2012 08:01:50 +0100 Subject: [Freeswitch-users] set_profile_var Message-ID: <4F58595E.5060001@softnet.si> Hi, how can I use variable set_profile_var to export collerd id name? I can not find any information on wiki. THanks! Miha From peter.olsson at visionutveckling.se Thu Mar 8 11:13:54 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 8 Mar 2012 08:13:54 +0000 Subject: [Freeswitch-users] set_profile_var Message-ID: <1FFF97C269757C458224B7C895F35F1504F78C@cantor.std.visionutv.se> Just export effective_caller_id_name and it will set the variable according to that. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha Zoubek Skickat: den 8 mars 2012 08:02 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] set_profile_var Hi, how can I use variable set_profile_var to export collerd id name? I can not find any information on wiki. THanks! Miha _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f58589f32763254410508! From brett at launch3.net Thu Mar 8 11:17:02 2012 From: brett at launch3.net (Brett Wilson) Date: Thu, 8 Mar 2012 03:17:02 -0500 Subject: [Freeswitch-users] State of GUIs In-Reply-To: References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> Message-ID: <03f301ccfd03$d7777080$86665180$@launch3.net> I checked out freepbx. Freepbx v3, the complete rewrite that was supposed to work with freeswitch, actually got spun off into the blue.box project. Seems that the blue.box project is basically dead. The last commit was November 2011. Fusionpbx looks to be the only game in town. I was wrong, it is still being developed. I just did a new install of it today, and allowed it to auto update. There are indeed some changes that I see from my install from about 4 months ago. So that is a positive thing, but its too bad that the feel is still lacking. I took a look at some code, and what I saw did not impress me. Display code mixed right in with the logic. I did not look for more than about 30 seconds, but IMO it is not 'proper' application design. I am a big proponent of MVC architecture, and what I saw of fusionpbx code does not look all that impressive. I give credit where credit is due. The platform is functioning to an extent, most things actually work. I do realize that developing something of that size requires tons of effort. My ideal solution would be something driven by ExtJS and PHP on the backend. ExtJS provides the most advanced and rich javascript controls i have seen. Plus its sister product, sencha touch, would enable great admin functionality from a smartphone or tablet device. Here I go rambling on. I wish someone would develop an awesome GUI for freeswitch to help get it into the limelight where it deserves to be. If I had the time I would love to create a comprehensive and great looking GUI. Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: Brett.Wilson at launch3.net AOL IM: Brett.Wilson at launch3.net www.Launch3.net www.Launch3telecom.com *************************** Description: Description: Blogger-logo Description: Description: FaceBook-Logo Description: Description: Twitter-Logo From: Andrew Cassidy [mailto:andrew at cassidywebservices.co.uk] Sent: Wednesday, March 07, 2012 6:46 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] State of GUIs Look at http://www.freepybx.org although I didn't manage to get it installed. I do have fusion installed on a text box although personally I'm not keen on it. On 7 March 2012 08:35, Brett Wilson wrote: Hey guys, I was wondering if anyone had any info on the current state of FS guis? It seems that most of the gui projects (blue.box, fusionpbx) have been mostly abandoned in terms of development. I would like something simple for end users to self-administer if needed. The problem I have with fusionpbx is that config files are overwritten by whatever is in the database. If you hand-edit a config file, fusion will not parse the file and load those settings into the interface. I realize that takes much more coding to do than using a database to simply write config files. But I feel that the freeswitch interface could be improved anyway. Unfortunately it seems that project has been abandoned. There have not been any releases since mid-2011. Anything new out or around the corner? Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: Brett.Wilson at launch3.net AOL IM: Brett.Wilson at launch3.net www.Launch3.net www.Launch3telecom.com *************************** Description: Description: Blogger-logo Description: Description: FaceBook-Logo Description: Description: Twitter-Logo _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/jpeg Size: 1715 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/ec73d049/attachment-0005.jpe From avi at avimarcus.net Thu Mar 8 11:33:49 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 8 Mar 2012 10:33:49 +0200 Subject: [Freeswitch-users] State of GUIs In-Reply-To: <03f301ccfd03$d7777080$86665180$@launch3.net> References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> Message-ID: On Thu, Mar 8, 2012 at 10:17 AM, Brett Wilson wrote: > I checked out freepbx. Freepbx v3, the complete rewrite that was supposed > to work with freeswitch, actually got spun off into the blue.box project. > Seems that the blue.box project is basically dead. The last commit was > November 2011. > As mentioned, the blue.box devs are working on deployment rather than the PBX side. I'd figure they think they have a stable product in the PBX.. go ask at #2600hz. > Fusionpbx looks to be the only game in town. > FreePYBX.org was just released. Fusionpbx looks to be the only game in town. I was wrong, it is still being > developed. I just did a new install of it today, and allowed it to auto > update. There are indeed some changes that I see from my install from about > 4 months ago. So that is a positive thing, but its too bad that the feel is > still lacking. I took a look at some code, and what I saw did not impress > me. Display code mixed right in with the logic. I did not look for more > than about 30 seconds, but IMO it is not ?proper? application design. I am > a big proponent of MVC architecture, and what I saw of fusionpbx code does > not look all that impressive. I give credit where credit is due. The > platform is functioning to an extent, most things actually work. I do > realize that developing something of that size requires tons of effort. > As you see.. the cross section of great programmer + motivation + time/availability + interest in a gui for FS that is opensource... is very very rare. > My ideal solution would be something driven by ExtJS and PHP on the > backend. ExtJS provides the most advanced and rich javascript controls i > have seen. Plus its sister product, sencha touch, would enable great admin > functionality from a smartphone or tablet device. Here I go rambling on? I > wish someone would develop an awesome GUI for freeswitch to help get it > into the limelight where it deserves to be. If I had the time I would love > to create a comprehensive and great looking GUI. > Make the time. Or resurrect wikipbx.org.. it's in python/django and has an impressive feature set. Or add what you want to blue.box - it's in php/kohana. Or work on freepybx.. also developed by one person. -Avi > **** > > ** ** > > *Brett Wilson* > > *IT Department* > > *Launch 3 Ventures, LLC***** > > 134 Myer Street**** > > Hackensack, NJ 07601**** > > *Phone:* 877.878.9134 > *Fax:* 646.536.3866**** > > *Email:* Brett.Wilson at launch3.net**** > > *AOL IM:* Brett.Wilson at launch3.net**** > > www.Launch3.net**** > > *www.Launch3telecom.com ***** > > ******************************* > > [image: Description: Description: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-Logo] > **** > > ** ** > > *From:* Andrew Cassidy [mailto:andrew at cassidywebservices.co.uk] > *Sent:* Wednesday, March 07, 2012 6:46 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] State of GUIs**** > > ** ** > > Look at http://www.freepybx.org although I didn't manage to get it > installed. I do have fusion installed on a text box although personally I'm > not keen on it.**** > > ** ** > > On 7 March 2012 08:35, Brett Wilson wrote:**** > > Hey guys,**** > > I was wondering if anyone had any info on the current state of FS guis? It > seems that most of the gui projects (blue.box, fusionpbx) have been mostly > abandoned in terms of development. I would like something simple for end > users to self-administer if needed. The problem I have with fusionpbx is > that config files are overwritten by whatever is in the database. If you > hand-edit a config file, fusion will not parse the file and load those > settings into the interface. I realize that takes much more coding to do > than using a database to simply write config files. But I feel that the > freeswitch interface could be improved anyway. Unfortunately it seems that > project has been abandoned. There have not been any releases since mid-2011. > **** > > **** > > Anything new out or around the corner?**** > > **** > > *Brett Wilson***** > > *IT Department***** > > *Launch 3 Ventures, LLC***** > > 134 Myer Street**** > > Hackensack, NJ 07601**** > > *Phone:* 877.878.9134 > *Fax:* 646.536.3866**** > > *Email:* Brett.Wilson at launch3.net**** > > *AOL IM:* Brett.Wilson at launch3.net**** > > www.Launch3.net**** > > *www.Launch3telecom.com ***** > > ******************************* > > [image: Description: Description: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-Logo] > **** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Andrew Cassidy BSc (Hons) MBCS**** > > Managing Director; Cassidy Web Services Ltd**** > > T: 03300 100 960 F: 03300 100 961**** > > E: andrew at cassidywebservices.co.uk**** > > W: www.cassidywebservices.co.uk**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/30ff89ed/attachment.html From miha at softnet.si Thu Mar 8 12:05:30 2012 From: miha at softnet.si (Miha Zoubek) Date: Thu, 08 Mar 2012 10:05:30 +0100 Subject: [Freeswitch-users] set_profile_var In-Reply-To: <1FFF97C269757C458224B7C895F35F1504F78C@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1504F78C@cantor.std.visionutv.se> Message-ID: <4F58765A.3060700@softnet.si> Hi @Peter, thank you for your quick answer. One option is that where I have now 15623150 that I would have 386981515623150 (the best option). Or in second row, 15623150 should be 18108500 This is my record in sql: 18108500 018108500.fs_kabelvoip1 15623150 fs_kabelvoip1.fs1.softnet.si 03/08/12 09:24 AM 03/08/12 09:24 AM 03/08/12 09:24 AM 7 4 ATTENDED_TRANSFER d476027a-6bb3-4cbf-a87a-94cd3940dfa4 89e7c541-360a-4752-83ea-00ce7f07b2cc 18108500 PCMA PCMA 15623150 386981515623150 386981515623150 default 03/08/12 09:24 AM 03/08/12 09:24 AM 03/08/12 09:24 AM 7 4 NORMAL_CLEARING 89e7c541-360a-4752-83ea-00ce7f07b2cc 56b2e6fe-943f-420c-b7e2-8cfc095cb140 PCMA PCMA Channel-State: [CS_REPORTING] Channel-Call-State: [HANGUP] Channel-State-Number: [11] Channel-Name: [sofia/internal/386981515623150 at 212.13.249.90] Unique-ID: [89e7c541-360a-4752-83ea-00ce7f07b2cc] Call-Direction: [outbound] Presence-Call-Direction: [outbound] Channel-HIT-Dialplan: [true] Channel-Call-UUID: [89e7c541-360a-4752-83ea-00ce7f07b2cc] Answer-State: [hangup] Channel-Read-Codec-Name: [PCMA] Channel-Read-Codec-Rate: [8000] Channel-Read-Codec-Bit-Rate: [64000] Channel-Write-Codec-Name: [PCMA] Channel-Write-Codec-Rate: [8000] Channel-Write-Codec-Bit-Rate: [64000] Caller-Direction: [outbound] Caller-Username: [018108500.fs_kabelvoip1] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [015623150] Caller-Caller-ID-Number: [386981515623150] Caller-Network-Addr: [xxx.xxx.xxx.xxx] Caller-ANI: [018108500.fs_kabelvoip1] Caller-Destination-Number: [386981515623150] ******+ Channel-HIT-Dialplan: [true] Channel-Presence-ID: [018108500.fs_kabelvoip1 at fs_kabelvoip1.fs1.softnet.si] Channel-Call-UUID: [d476027a-6bb3-4cbf-a87a-94cd3940dfa4] Answer-State: [hangup] Channel-Read-Codec-Name: [PCMA] Channel-Read-Codec-Rate: [8000] Channel-Read-Codec-Bit-Rate: [64000] Channel-Write-Codec-Name: [PCMA] Channel-Write-Codec-Rate: [8000] Channel-Write-Codec-Bit-Rate: [64000] Caller-Direction: [inbound] Caller-Username: [018108500.fs_kabelvoip1] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [018108500] Caller-Caller-ID-Number: [018108500.fs_kabelvoip1] Caller-Callee-ID-Name: [015623150] Caller-Callee-ID-Number: [386981515623150] Caller-Network-Addr: [xxx.xxx.xxx.xxx] Caller-ANI: [018108500.fs_kabelvoip1] Caller-Destination-Number: [015623150] I have also put in my dialplan: But still getting Flipping CID from "018108500" <018108500> to "015623150" <386981515623150> Thanks! Miha On 03/08/2012 09:13 AM, Peter Olsson wrote: > Just export effective_caller_id_name and it will set the variable according to that. > > /Peter > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha Zoubek > Skickat: den 8 mars 2012 08:02 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] set_profile_var > > Hi, > > how can I use variable set_profile_var to export collerd id name? > > I can not find any information on wiki. > > THanks! > > Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f58589f32763254410508! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/36ee29d1/attachment-0001.html From miha at softnet.si Thu Mar 8 13:54:29 2012 From: miha at softnet.si (Miha Zoubek) Date: Thu, 08 Mar 2012 11:54:29 +0100 Subject: [Freeswitch-users] stip variable Message-ID: <4F588FE5.3090402@softnet.si> Hi, how can I strip this sofia/internal/18005551212 at tf.voipmich.com, so that I would have only number 18005551212? This return enum. Thanks! Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/0f7d3da7/attachment.html From freeswitch-list at puzzled.xs4all.nl Thu Mar 8 16:18:22 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Thu, 08 Mar 2012 14:18:22 +0100 Subject: [Freeswitch-users] State of GUIs In-Reply-To: <03f301ccfd03$d7777080$86665180$@launch3.net> References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> Message-ID: <4F58B19E.7070101@puzzled.xs4all.nl> On 08-03-12 09:17, Brett Wilson wrote: > I checked out freepbx. Freepbx v3, the complete rewrite that was > supposed to work with freeswitch, actually got spun off into the > blue.box project. Seems that the blue.box project is basically dead. The > last commit was November 2011. Fusionpbx looks to be the only game in > town. I was wrong, it is still being developed. I just did a new install > of it today, and allowed it to auto update. There are indeed some > changes that I see from my install from about 4 months ago. So that is a > positive thing, but its too bad that the feel is still lacking. I took a > look at some code, and what I saw did not impress me. Display code mixed > right in with the logic. I did not look for more than about 30 seconds, > but IMO it is not ?proper? application design. I am a big proponent of > MVC architecture, and what I saw of fusionpbx code does not look all > that impressive. I give credit where credit is due. The platform is > functioning to an extent, most things actually work. I do realize that > developing something of that size requires tons of effort. My ideal > solution would be something driven by ExtJS and PHP on the backend. > ExtJS provides the most advanced and rich javascript controls i have > seen. Plus its sister product, sencha touch, would enable great admin > functionality from a smartphone or tablet device. Here I go rambling on? > I wish someone would develop an awesome GUI for freeswitch to help get > it into the limelight where it deserves to be. If I had the time I would > love to create a comprehensive and great looking GUI. I you like blue.box more than fusionpbx but feel things are missing in blue.box then why not talk to the 2600hz developers and hire them to add that functionality? If you are looking for a slick GUI then have a look at the Cudatel: / Regards, Patrick From roger.castaldo at gmail.com Thu Mar 8 16:46:54 2012 From: roger.castaldo at gmail.com (Roger Castaldo) Date: Thu, 8 Mar 2012 08:46:54 -0500 Subject: [Freeswitch-users] State of GUIs In-Reply-To: <4F58B19E.7070101@puzzled.xs4all.nl> References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> <4F58B19E.7070101@puzzled.xs4all.nl> Message-ID: I wanted to avoid putting myself out there but there is another gui project that I have avoided releasing as open source on account of the fact that I am having issues finding developers to help. It is written in C#, built to run on mono or .net. Runs on windows, debian, slitaz and centos can be designed to run perfectly on others. It contains ability to configure full server stuff as well as controls the call flow of freeswitch through the sockets library. It is designed to be web 2.0 compliant as its all done through javascript and a single web page using ajax calls with json encoding for efficiency. There is still some things needing to be done such as developing a billling module for it, ensuring that mobile browsers can operate it successfully and adding new functionality through modules as well as completing one or two modules. The few people who I have had take a look and try it out have been impressed and slowly, as I have time i have been cleaning up code, making it more efficient and trying to improve functionality. It takes time though when its one person and they are not getting paid to do it, its more of a hobby. On Thu, Mar 8, 2012 at 8:18 AM, Patrick Lists < freeswitch-list at puzzled.xs4all.nl> wrote: > On 08-03-12 09:17, Brett Wilson wrote: > > I checked out freepbx. Freepbx v3, the complete rewrite that was > > supposed to work with freeswitch, actually got spun off into the > > blue.box project. Seems that the blue.box project is basically dead. The > > last commit was November 2011. Fusionpbx looks to be the only game in > > town. I was wrong, it is still being developed. I just did a new install > > of it today, and allowed it to auto update. There are indeed some > > changes that I see from my install from about 4 months ago. So that is a > > positive thing, but its too bad that the feel is still lacking. I took a > > look at some code, and what I saw did not impress me. Display code mixed > > right in with the logic. I did not look for more than about 30 seconds, > > but IMO it is not ?proper? application design. I am a big proponent of > > MVC architecture, and what I saw of fusionpbx code does not look all > > that impressive. I give credit where credit is due. The platform is > > functioning to an extent, most things actually work. I do realize that > > developing something of that size requires tons of effort. My ideal > > solution would be something driven by ExtJS and PHP on the backend. > > ExtJS provides the most advanced and rich javascript controls i have > > seen. Plus its sister product, sencha touch, would enable great admin > > functionality from a smartphone or tablet device. Here I go rambling on? > > I wish someone would develop an awesome GUI for freeswitch to help get > > it into the limelight where it deserves to be. If I had the time I would > > love to create a comprehensive and great looking GUI. > > I you like blue.box more than fusionpbx but feel things are missing in > blue.box then why not talk to the 2600hz developers and hire them to add > that functionality? > > If you are looking for a slick GUI then have a look at the Cudatel: > / > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/e6e78458/attachment.html From jaybinks at gmail.com Thu Mar 8 17:06:50 2012 From: jaybinks at gmail.com (Jay Binks) Date: Fri, 9 Mar 2012 00:06:50 +1000 Subject: [Freeswitch-users] State of GUIs In-Reply-To: References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> <4F58B19E.7070101@puzzled.xs4all.nl> Message-ID: Throw it on github !!! I'm very interested !!! On 08/03/2012, at 11:46 PM, Roger Castaldo wrote: > I wanted to avoid putting myself out there but there is another gui project that I have avoided releasing as open source on account of the fact that I am having issues finding developers to help. It is written in C#, built to run on mono or .net. Runs on windows, debian, slitaz and centos can be designed to run perfectly on others. It contains ability to configure full server stuff as well as controls the call flow of freeswitch through the sockets library. It is designed to be web 2.0 compliant as its all done through javascript and a single web page using ajax calls with json encoding for efficiency. There is still some things needing to be done such as developing a billling module for it, ensuring that mobile browsers can operate it successfully and adding new functionality through modules as well as completing one or two modules. The few people who I have had take a look and try it out have been impressed and slowly, as I have time i have been cleaning up code, making it more efficient and trying to improve functionality. It takes time though when its one person and they are not getting paid to do it, its more of a hobby. > > On Thu, Mar 8, 2012 at 8:18 AM, Patrick Lists wrote: > On 08-03-12 09:17, Brett Wilson wrote: > > I checked out freepbx. Freepbx v3, the complete rewrite that was > > supposed to work with freeswitch, actually got spun off into the > > blue.box project. Seems that the blue.box project is basically dead. The > > last commit was November 2011. Fusionpbx looks to be the only game in > > town. I was wrong, it is still being developed. I just did a new install > > of it today, and allowed it to auto update. There are indeed some > > changes that I see from my install from about 4 months ago. So that is a > > positive thing, but its too bad that the feel is still lacking. I took a > > look at some code, and what I saw did not impress me. Display code mixed > > right in with the logic. I did not look for more than about 30 seconds, > > but IMO it is not ?proper? application design. I am a big proponent of > > MVC architecture, and what I saw of fusionpbx code does not look all > > that impressive. I give credit where credit is due. The platform is > > functioning to an extent, most things actually work. I do realize that > > developing something of that size requires tons of effort. My ideal > > solution would be something driven by ExtJS and PHP on the backend. > > ExtJS provides the most advanced and rich javascript controls i have > > seen. Plus its sister product, sencha touch, would enable great admin > > functionality from a smartphone or tablet device. Here I go rambling on? > > I wish someone would develop an awesome GUI for freeswitch to help get > > it into the limelight where it deserves to be. If I had the time I would > > love to create a comprehensive and great looking GUI. > > I you like blue.box more than fusionpbx but feel things are missing in > blue.box then why not talk to the 2600hz developers and hire them to add > that functionality? > > If you are looking for a slick GUI then have a look at the Cudatel: > / > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/8cc4a8f2/attachment-0001.html From anthony.minessale at gmail.com Thu Mar 8 18:11:07 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 8 Mar 2012 09:11:07 -0600 Subject: [Freeswitch-users] State of GUIs In-Reply-To: <03f301ccfd03$d7777080$86665180$@launch3.net> References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> Message-ID: We encourage people to make GUIs for FreeSWITCH and we have mixed results. Being controlled externally is actually baked into the design of FreeSWITCH. Many of the abstract concepts int the core that people are still unearthing are all about developing external apps. The exercise to build one was intentionally left to the end user. The point was to harness the hard part about telephony so you could focus on your business use case. Really, based on the requirements of the original poster, I think a comercial solution is in order. I also would not judge activity by release dates / commits etc. FreeSWITCH itself has not been released since 2009 yet the C code has grown by 35 megabits since then with commits nearly hourly. We just happen to be more open about our development than others may be comfortable with. Anyway, I know for a fact all of the above list in the original post are still up and running but not everyone has the resources to run an active community. As referred to earlier in this thread, we made the CudaTEL PBX at Barracuda Networks which is an appliance designed to do everything described above and that effort could not be done without a commercial incentive to develop it and a team of dedicated employees. Everything you ever wanted is typically hard to come by when dealing with Free GUIs so the alternative is to invest in one of them and get them to focus on the functionality you seek. On Thu, Mar 8, 2012 at 2:17 AM, Brett Wilson wrote: > I checked out freepbx. Freepbx v3, the complete rewrite that was supposed > to work with freeswitch, actually got spun off into the blue.box project. > Seems that the blue.box project is basically dead. The last commit was > November 2011. Fusionpbx looks to be the only game in town. I was wrong, it > is still being developed. I just did a new install of it today, and allowed > it to auto update. There are indeed some changes that I see from my install > from about 4 months ago. So that is a positive thing, but its too bad that > the feel is still lacking. I took a look at some code, and what I saw did > not impress me. Display code mixed right in with the logic. I did not look > for more than about 30 seconds, but IMO it is not ?proper? application > design. I am a big proponent of MVC architecture, and what I saw of > fusionpbx code does not look all that impressive. I give credit where > credit is due. The platform is functioning to an extent, most things > actually work. I do realize that developing something of that size requires > tons of effort. My ideal solution would be something driven by ExtJS and > PHP on the backend. ExtJS provides the most advanced and rich javascript > controls i have seen. Plus its sister product, sencha touch, would enable > great admin functionality from a smartphone or tablet device. Here I go > rambling on? I wish someone would develop an awesome GUI for freeswitch to > help get it into the limelight where it deserves to be. If I had the time I > would love to create a comprehensive and great looking GUI.**** > > ** ** > > *Brett Wilson* > > *IT Department* > > *Launch 3 Ventures, LLC***** > > 134 Myer Street**** > > Hackensack, NJ 07601**** > > *Phone:* 877.878.9134 > *Fax:* 646.536.3866**** > > *Email:* Brett.Wilson at launch3.net**** > > *AOL IM:* Brett.Wilson at launch3.net**** > > www.Launch3.net**** > > *www.Launch3telecom.com ***** > > ******************************* > > [image: Description: Description: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-Logo] > **** > > ** ** > > *From:* Andrew Cassidy [mailto:andrew at cassidywebservices.co.uk] > *Sent:* Wednesday, March 07, 2012 6:46 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] State of GUIs**** > > ** ** > > Look at http://www.freepybx.org although I didn't manage to get it > installed. I do have fusion installed on a text box although personally I'm > not keen on it.**** > > ** ** > > On 7 March 2012 08:35, Brett Wilson wrote:**** > > Hey guys,**** > > I was wondering if anyone had any info on the current state of FS guis? It > seems that most of the gui projects (blue.box, fusionpbx) have been mostly > abandoned in terms of development. I would like something simple for end > users to self-administer if needed. The problem I have with fusionpbx is > that config files are overwritten by whatever is in the database. If you > hand-edit a config file, fusion will not parse the file and load those > settings into the interface. I realize that takes much more coding to do > than using a database to simply write config files. But I feel that the > freeswitch interface could be improved anyway. Unfortunately it seems that > project has been abandoned. There have not been any releases since mid-2011. > **** > > **** > > Anything new out or around the corner?**** > > **** > > *Brett Wilson***** > > *IT Department***** > > *Launch 3 Ventures, LLC***** > > 134 Myer Street**** > > Hackensack, NJ 07601**** > > *Phone:* 877.878.9134 > *Fax:* 646.536.3866**** > > *Email:* Brett.Wilson at launch3.net**** > > *AOL IM:* Brett.Wilson at launch3.net**** > > www.Launch3.net**** > > *www.Launch3telecom.com ***** > > ******************************* > > [image: Description: Description: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-Logo] > **** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Andrew Cassidy BSc (Hons) MBCS**** > > Managing Director; Cassidy Web Services Ltd**** > > T: 03300 100 960 F: 03300 100 961**** > > E: andrew at cassidywebservices.co.uk**** > > W: www.cassidywebservices.co.uk**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/jpeg Size: 1815 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/6339b741/attachment-0005.jpe From benkokakao at gmail.com Thu Mar 8 18:17:40 2012 From: benkokakao at gmail.com (Christian Benke) Date: Thu, 8 Mar 2012 16:17:40 +0100 Subject: [Freeswitch-users] stip variable In-Reply-To: <4F588FE5.3090402@softnet.si> References: <4F588FE5.3090402@softnet.si> Message-ID: On 8 March 2012 11:54, Miha Zoubek wrote: > how can I strip this sofia/internal/18005551212 at tf.voipmich.com, so that I > would have only number 18005551212? See http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_regex - also take a look at the example i've given you in your other thread yesterday. Regards Christian From fluixab at bellsouth.net Thu Mar 8 18:24:38 2012 From: fluixab at bellsouth.net (Bernard Fluixa) Date: Thu, 8 Mar 2012 07:24:38 -0800 (PST) Subject: [Freeswitch-users] mod shell stream In-Reply-To: References: <5BA3C64DA53B4EAC9B1DA367F8FFD398@gmail.com> Message-ID: <1331220278.14097.YahooMailRC@web180216.mail.gq1.yahoo.com> Michael, Thanks for your reply. Yes it works in this case. Any thoughts? Bernard ________________________________ From: Michael Collins To: FreeSWITCH Users Help Sent: Mon, March 5, 2012 10:18:44 PM Subject: Re: [Freeswitch-users] mod shell stream On Sun, Mar 4, 2012 at 4:30 AM, Bernard Fluixa wrote: That is already done with the -t option (tuples only) > > >Thanks anyway > >Bernard > > Does your script work if you do it in two steps, i.e. read from the database and drop into a file, then cat the file into sox? I'm just curious if something unexpected is happening on the command line when piping from psql to sox. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/5728f436/attachment.html From brian at freeswitch.org Thu Mar 8 18:10:56 2012 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Mar 2012 09:10:56 -0600 Subject: [Freeswitch-users] stip variable In-Reply-To: <4F588FE5.3090402@softnet.si> References: <4F588FE5.3090402@softnet.si> Message-ID: <73D50A0D-B8C3-4EB2-91A5-6B2C76C82907@freeswitch.org> or modify enum.conf to only give you the numer? /b On Mar 8, 2012, at 4:54 AM, Miha Zoubek wrote: > Hi, > > how can I strip this sofia/internal/18005551212 at tf.voipmich.com, so that I would have only number 18005551212? > > This return enum. > > > Thanks! > Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/3539aa88/attachment.html From bob.mccarthy at experient.com Thu Mar 8 19:09:44 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Thu, 8 Mar 2012 09:09:44 -0700 Subject: [Freeswitch-users] Putting in a Hangup Hook onto an SLA Barge In Message-ID: <020e01ccfd45$e1b0c190$a51244b0$@mccarthy@experient.com> where in the configs do I add the api_hangup_hook for SLA when a shared line member barges in ? I want to execute a lua script when any member of a SLA conference hangs up. Ive got leg A and B covered just not Leg C. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/19b264fa/attachment.html From benkokakao at gmail.com Thu Mar 8 21:31:18 2012 From: benkokakao at gmail.com (Christian Benke) Date: Thu, 8 Mar 2012 19:31:18 +0100 Subject: [Freeswitch-users] session-api - How to continue with A-leg after bridge-timeout? Message-ID: Hello! I have written a DB-based routing-script to be able to configure complex and dynamic call-behaviours but i'm currently stuck with scenarious where different call-channels are merged(att_xfer, intercept, et al), as my script fails with these scenarious. I'm testing using a simplified version of the script(nxo_test.py) to understand the mechanism behind those scenarious and now i'm at a point where i no longer know the proper tools to handle these situations. Here's what is happening: This is the test-script in pseudocode: > SRC=A > DEST=B > log("Test1") > sleep(1000) > log("Test2") > sleep(1000) > log("Test3") > bridge([origination_caller_id_number="${SRC}"]user/${DEST}@${domain_name}) > log("Test4") > sleep(1000) > "log" "Test5" > sleep(1000) > "log" "Test6" A dials B. The call enters the default context where the test-script is called, the first steps are executed and the bridge is initiated. B picks up the call, the following (truncated) channel-uuids exist at this point: "a7c4"(A->) "a7cc"(->B) B then initiates an attended transfer to C, which follows the same path and script as the first call(But the call is bridged to C instead of B this time). Two new channels are set up - uuid's "a7d5"(B->) and "a7dd"(->C) B immediately transfers the call before C has picked up. Channel "a7cc"(->B) is killed by att_xfer while Channel "a7c4"(A->) finishes the script(The remaining Steps Test3-Test5 are executed) and is then automatically parked in endless_playback by att_xfer, waiting for C to pick up. C lets it ring until the timeout of 15s ends, "a7c4" and "a7dd" are hung up while a7d5 continues as a ZOMBIE-channel, trying to execute the remaining steps in the script(The "sleep"-commands fail due to the zombie-state). And this is where i'm stuck - i would like channel "a7c4" to not get hung up but to continue with the steps which "a7d5" could no longer execute. Is there some way to prevent a7c4 from getting hung up when the ATTENDED_TRANSFER to C fails? Algorithmically i can figure out how the remaining steps can get memorised and executed, i just don't know how to prevent "a7c4" from getting hung up and beeing transfer to a place where i can still work with it :-/ I can't use the HangupHook either, as a7c4 is already in zombie-state at that point. For reference, i've attached the test-script and a freeswitch-log of the call above. I hope someone can lead me on the right track... Best regards Christian -------------- next part -------------- ------------------------------------------------------------------------ recv 861 bytes from udp/[10.3.0.22]:5060 at 17:34:05.594085: ------------------------------------------------------------------------ INVITE sip:30 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bK8ad4759287A067FB From: "A" ;tag=B0C14C18-E7DC6BB1 To: CSeq: 1 INVITE Call-ID: c64e932c-3eca2965-da54ee6 at 10.3.0.22 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 241 v=0 o=- 1167627581 1167627581 IN IP4 10.3.0.22 s=Polycom IP Phone c=IN IP4 10.3.0.22 t=0 0 a=sendrecv m=audio 2262 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ------------------------------------------------------------------------ send 333 bytes to udp/[10.3.0.22]:5060 at 17:34:05.595235: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bK8ad4759287A067FB From: "A" ;tag=B0C14C18-E7DC6BB1 To: Call-ID: c64e932c-3eca2965-da54ee6 at 10.3.0.22 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:05.592923 [DEBUG] sofia.c:7559 IP 10.3.0.22 Rejected by acl "domains". Falling back to Digest auth. 2012-03-08 18:34:05.592923 [WARNING] sofia_reg.c:1422 SIP auth challenge (INVITE) on sofia profile 'internal' for [30 at 10.3.0.4] from ip 10.3.0.22 send 816 bytes to udp/[10.3.0.22]:5060 at 17:34:05.598758: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bK8ad4759287A067FB From: "A" ;tag=B0C14C18-E7DC6BB1 To: ;tag=Bvv69Ze2ttHyB Call-ID: c64e932c-3eca2965-da54ee6 at 10.3.0.22 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="10.3.0.4", nonce="e73b92aa-6944-11e1-a7c3-a55c2ebde3ff", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 535 bytes from udp/[10.3.0.22]:5060 at 17:34:05.608337: ------------------------------------------------------------------------ ACK sip:30 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bK8ad4759287A067FB From: "A" ;tag=B0C14C18-E7DC6BB1 To: ;tag=Bvv69Ze2ttHyB CSeq: 1 ACK Call-ID: c64e932c-3eca2965-da54ee6 at 10.3.0.22 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ recv 1115 bytes from udp/[10.3.0.22]:5060 at 17:34:05.611505: ------------------------------------------------------------------------ INVITE sip:30 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bKacba9efF22172C0 From: "A" ;tag=B0C14C18-E7DC6BB1 To: CSeq: 2 INVITE Call-ID: c64e932c-3eca2965-da54ee6 at 10.3.0.22 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="10", realm="10.3.0.4", nonce="e73b92aa-6944-11e1-a7c3-a55c2ebde3ff", qop=auth, cnonce="eRH+crsyq6cDpwa", nc=00000001, uri="sip:30 at 10.3.0.4:5060;user=phone", response="a3d381aceaa4551a5eabcb39b426928d", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 241 v=0 o=- 1167627581 1167627581 IN IP4 10.3.0.22 s=Polycom IP Phone c=IN IP4 10.3.0.22 t=0 0 a=sendrecv m=audio 2262 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ------------------------------------------------------------------------ send 332 bytes to udp/[10.3.0.22]:5060 at 17:34:05.612619: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bKacba9efF22172C0 From: "A" ;tag=B0C14C18-E7DC6BB1 To: Call-ID: c64e932c-3eca2965-da54ee6 at 10.3.0.22 CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:05.612916 [DEBUG] sofia.c:7559 IP 10.3.0.22 Rejected by acl "domains". Falling back to Digest auth. 2012-03-08 18:34:05.612916 [NOTICE] switch_channel.c:926 New Channel sofia/internal/10 at 10.3.0.4 [e73f07fa-6944-11e1-a7c4-a55c2ebde3ff] 2012-03-08 18:34:05.612916 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_NEW 2012-03-08 18:34:05.612916 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/10 at 10.3.0.4) State NEW 2012-03-08 18:34:05.612916 [DEBUG] sofia.c:5526 Channel sofia/internal/10 at 10.3.0.4 entering state [received][100] 2012-03-08 18:34:05.612916 [DEBUG] sofia.c:5537 Remote SDP: v=0 o=- 1167627581 1167627581 IN IP4 10.3.0.22 s=Polycom IP Phone c=IN IP4 10.3.0.22 t=0 0 a=sendrecv m=audio 2262 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 2012-03-08 18:34:05.612916 [DEBUG] sofia_glue.c:4865 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 2012-03-08 18:34:05.612916 [DEBUG] sofia_glue.c:2982 Set Codec sofia/internal/10 at 10.3.0.4 G722/8000 20 ms 160 samples 64000 bits 2012-03-08 18:34:05.612916 [DEBUG] switch_core_codec.c:111 sofia/internal/10 at 10.3.0.4 Original read codec set to G722:9 2012-03-08 18:34:05.612916 [DEBUG] sofia_glue.c:4986 Set 2833 dtmf send/recv payload to 127 2012-03-08 18:34:05.612916 [DEBUG] sofia.c:5749 (sofia/internal/10 at 10.3.0.4) State Change CS_NEW -> CS_INIT 2012-03-08 18:34:05.612916 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:05.612916 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_INIT 2012-03-08 18:34:05.612916 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/10 at 10.3.0.4) State INIT 2012-03-08 18:34:05.612916 [DEBUG] mod_sofia.c:85 sofia/internal/10 at 10.3.0.4 SOFIA INIT 2012-03-08 18:34:05.612916 [DEBUG] mod_sofia.c:125 (sofia/internal/10 at 10.3.0.4) State Change CS_INIT -> CS_ROUTING 2012-03-08 18:34:05.612916 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:05.612916 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/10 at 10.3.0.4) State INIT going to sleep 2012-03-08 18:34:05.612916 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_ROUTING 2012-03-08 18:34:05.612916 [DEBUG] switch_channel.c:1886 (sofia/internal/10 at 10.3.0.4) Callstate Change DOWN -> RINGING 2012-03-08 18:34:05.612916 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/10 at 10.3.0.4) State ROUTING 2012-03-08 18:34:05.612916 [DEBUG] mod_sofia.c:148 sofia/internal/10 at 10.3.0.4 SOFIA ROUTING 2012-03-08 18:34:05.612916 [DEBUG] switch_core_state_machine.c:104 sofia/internal/10 at 10.3.0.4 Standard ROUTING 2012-03-08 18:34:05.612916 [INFO] mod_dialplan_xml.c:485 Processing A <10>->30 in context default Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->nxo_enable_chefsec] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [nxo_enable_chefsec] destination_number(30) =~ /^\*95(\d{0,7})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [group-intercept] destination_number(30) =~ /^\*82$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [intercept-ext] destination_number(30) =~ /^\*81(\d+)$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->nxo_single_intercom_with_two_way_audio] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [nxo_single_intercom_with_two_way_audio] destination_number(30) =~ /^\*01(\d{2,7})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->nxo_group_intercom] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [nxo_group_intercom] destination_number(30) =~ /^\*02(\d{0,7})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->Outbound to PSTN 11 Digits] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [Outbound to PSTN 11 Digits] destination_number(30) =~ /^(1[2-9][0-9]{2}[2-9]{7})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->del-group] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [del-group] destination_number(30) =~ /^\*\*50(\d{2})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->add-group] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [add-group] destination_number(30) =~ /^\*\*51(\d{2})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [call-group-simo] destination_number(30) =~ /^\*52(\d{2,4})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->call-group-order] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [call-group-order] destination_number(30) =~ /^\*53(\d{2,4})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->nb_conferences] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [nb_conferences] destination_number(30) =~ /^\*(30\d{2})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->wb_conferences] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [wb_conferences] destination_number(30) =~ /^\*(31\d{2})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->uwb_conferences] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [uwb_conferences] destination_number(30) =~ /^\*(32\d{2})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->cdquality_conferences] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [cdquality_conferences] destination_number(30) =~ /^\*(33\d{2})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->global_directory] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [global_directory] destination_number(30) =~ /^\*77(\d{1,3})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->redirect_now] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [redirect_now] destination_number(30) =~ /^\*21(\d{0,20})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->redirect_timeout] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [redirect_timeout] destination_number(30) =~ /^\*22(\d{0,20})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->redirect_busy] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [redirect_busy] destination_number(30) =~ /^\*23(\d{0,20})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->call_privacy] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [call_privacy] destination_number(30) =~ /^\*67(\d+)$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->call_privacy] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [call_privacy] destination_number(30) =~ /^\*60$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->send_to_voicemail] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [send_to_voicemail] destination_number(30) =~ /^\*99(\d{2,7})$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->vmain] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [vmain] destination_number(30) =~ /^vmain$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->vmain1] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [vmain1] destination_number(30) =~ /^vmain1$|^\*97$|^97$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->vmain2] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [vmain2] destination_number(30) =~ /^vmain2$|^\*98$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->redirect_now] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [redirect_now] destination_number(30) =~ /^\*35$/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->Local_Extension_test] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (PASS) [Local_Extension_test] destination_number(30) =~ /(^\d{2,7}$)/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 Action set(dialed_extension=30) Dialplan: sofia/internal/10 at 10.3.0.4 Action export(dialed_extension=30) Dialplan: sofia/internal/10 at 10.3.0.4 Action python(nxo_test) 2012-03-08 18:34:05.612916 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/10 at 10.3.0.4) State Change CS_ROUTING -> CS_EXECUTE 2012-03-08 18:34:05.612916 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:05.612916 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/10 at 10.3.0.4) State ROUTING going to sleep 2012-03-08 18:34:05.612916 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_EXECUTE 2012-03-08 18:34:05.612916 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/10 at 10.3.0.4) State EXECUTE 2012-03-08 18:34:05.612916 [DEBUG] mod_sofia.c:241 sofia/internal/10 at 10.3.0.4 SOFIA EXECUTE 2012-03-08 18:34:05.612916 [DEBUG] switch_core_state_machine.c:192 sofia/internal/10 at 10.3.0.4 Standard EXECUTE EXECUTE sofia/internal/10 at 10.3.0.4 set(dialed_extension=30) 2012-03-08 18:34:05.633001 [DEBUG] mod_dptools.c:1281 sofia/internal/10 at 10.3.0.4 SET [dialed_extension]=[30] EXECUTE sofia/internal/10 at 10.3.0.4 export(dialed_extension=30) 2012-03-08 18:34:05.633001 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [dialed_extension]=[30] EXECUTE sofia/internal/10 at 10.3.0.4 python(nxo_test) 2012-03-08 18:34:05.633001 [NOTICE] mod_python.c:212 Invoking py module: nxo_test 2012-03-08 18:34:05.652944 [DEBUG] mod_python.c:281 Call python script 2012-03-08 18:34:05.652944 [INFO] switch_cpp.cpp:1227 TEST: Destination-extension 30 send 1326 bytes to udp/[10.3.0.25]:5060 at 17:34:05.792382: ------------------------------------------------------------------------ NOTIFY sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKptUaH731cyvgm Max-Forwards: 70 From: ;tag=JElIsHAUjri7 To: "C" ;tag=432C4A25-2C483674 Call-ID: a941c408-2c54f3a7-82c5a8d6 at 10.3.0.25 CSeq: 185332666 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=1156 Content-Type: application/dialog-info+xml Content-Length: 513 confirmed sip:10 at 10.3.0.4 sip:30 at 10.3.0.4 ------------------------------------------------------------------------ send 1334 bytes to udp/[10.3.0.26]:5060 at 17:34:05.793468: ------------------------------------------------------------------------ NOTIFY sip:30 at 10.3.0.26 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKQ3m3j2m596j3F Max-Forwards: 70 From: ;tag=hzYaAHIZscZq To: "B" ;tag=A52D4836-AD38FE89 Call-ID: 3f77033d-f3b24ec8-f0479fab at 10.3.0.26 CSeq: 185332667 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=1156 Content-Type: application/dialog-info+xml Content-Length: 513 confirmed sip:10 at 10.3.0.4 sip:30 at 10.3.0.4 ------------------------------------------------------------------------ recv 406 bytes from udp/[10.3.0.25]:5060 at 17:34:05.799132: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKptUaH731cyvgm From: ;tag=JElIsHAUjri7 To: "C" ;tag=432C4A25-2C483674 CSeq: 185332666 NOTIFY Call-ID: a941c408-2c54f3a7-82c5a8d6 at 10.3.0.25 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 414 bytes from udp/[10.3.0.26]:5060 at 17:34:05.799603: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKQ3m3j2m596j3F From: ;tag=hzYaAHIZscZq To: "B" ;tag=A52D4836-AD38FE89 CSeq: 185332667 NOTIFY Call-ID: 3f77033d-f3b24ec8-f0479fab at 10.3.0.26 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ EXECUTE sofia/internal/10 at 10.3.0.4 set(continue_on_fail=true) 2012-03-08 18:34:06.232952 [DEBUG] mod_dptools.c:1281 sofia/internal/10 at 10.3.0.4 SET [continue_on_fail]=[true] EXECUTE sofia/internal/10 at 10.3.0.4 set(hangup_after_bridge=true) 2012-03-08 18:34:06.232952 [DEBUG] mod_dptools.c:1281 sofia/internal/10 at 10.3.0.4 SET [hangup_after_bridge]=[true] 2012-03-08 18:34:06.232952 [WARNING] switch_cpp.cpp:1227 TEST 1 for UUID e73f07fa-6944-11e1-a7c4-a55c2ebde3ff, src 10 dest 30 EXECUTE sofia/internal/10 at 10.3.0.4 sleep(1000) 2012-03-08 18:34:07.232931 [WARNING] switch_cpp.cpp:1227 TEST 2 for UUID e73f07fa-6944-11e1-a7c4-a55c2ebde3ff, src 10 dest 30 EXECUTE sofia/internal/10 at 10.3.0.4 sleep(1000) 2012-03-08 18:34:08.232934 [WARNING] switch_cpp.cpp:1227 TEST 3 for UUID e73f07fa-6944-11e1-a7c4-a55c2ebde3ff, src 10 dest 30 EXECUTE sofia/internal/10 at 10.3.0.4 sleep(1000) EXECUTE sofia/internal/10 at 10.3.0.4 info() 2012-03-08 18:34:09.252934 [INFO] mod_dptools.c:1439 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [RINGING] Channel-State-Number: [4] Channel-Name: [sofia/internal/10 at 10.3.0.4] Unique-ID: [e73f07fa-6944-11e1-a7c4-a55c2ebde3ff] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-HIT-Dialplan: [true] Channel-Presence-ID: [10 at 10.3.0.4] Channel-Call-UUID: [e73f07fa-6944-11e1-a7c4-a55c2ebde3ff] Answer-State: [ringing] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Read-Codec-Bit-Rate: [64000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Channel-Write-Codec-Bit-Rate: [64000] Caller-Direction: [inbound] Caller-Username: [10] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [A] Caller-Caller-ID-Number: [10] Caller-Network-Addr: [10.3.0.22] Caller-ANI: [10] Caller-Destination-Number: [30] Caller-Unique-ID: [e73f07fa-6944-11e1-a7c4-a55c2ebde3ff] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/10 at 10.3.0.4] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1331228045612916] Caller-Channel-Created-Time: [1331228045612916] 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] variable_direction: [inbound] variable_uuid: [e73f07fa-6944-11e1-a7c4-a55c2ebde3ff] variable_session_id: [9] variable_sip_local_network_addr: [10.3.0.4] variable_sip_network_ip: [10.3.0.22] variable_sip_network_port: [5060] variable_sip_received_ip: [10.3.0.22] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_authorized: [true] variable_sip_number_alias: [10] variable_sip_auth_username: [10] variable_sip_auth_realm: [10.3.0.4] variable_number_alias: [10] variable_user_name: [10] variable_domain_name: [10.3.0.4] variable_record_stereo: [true] variable_default_gateway: [fonira] variable_default_areacode: [01] variable_transfer_fallback_extension: [operator] variable_toll_allow: [local,domestic,international,vas] variable_accountcode: [10] variable_user_context: [default] variable_effective_caller_id_name: [A ] variable_effective_caller_id_number: [10] variable_outbound_caller_id_name: [A ] variable_outbound_caller_id_number: [1997156010] variable_callgroup: [intercept] variable_sip_from_user: [10] variable_sip_from_uri: [10 at 10.3.0.4] variable_sip_from_host: [10.3.0.4] variable_sip_from_user_stripped: [10] variable_sip_from_tag: [B0C14C18-E7DC6BB1] variable_sofia_profile_name: [internal] variable_sip_full_via: [SIP/2.0/UDP 10.3.0.22;branch=z9hG4bKacba9efF22172C0] variable_sip_from_display: [A] variable_sip_full_from: ["A" ;tag=B0C14C18-E7DC6BB1] variable_sip_full_to: [] variable_sip_req_params: [user=phone] variable_sip_req_user: [30] variable_sip_req_port: [5060] variable_sip_req_uri: [30 at 10.3.0.4:5060] variable_sip_req_host: [10.3.0.4] variable_sip_to_params: [user=phone] variable_sip_to_user: [30] variable_sip_to_uri: [30 at 10.3.0.4] variable_sip_to_host: [10.3.0.4] variable_sip_contact_user: [10] variable_sip_contact_uri: [10 at 10.3.0.22] variable_sip_contact_host: [10.3.0.22] variable_channel_name: [sofia/internal/10 at 10.3.0.4] variable_sip_call_id: [c64e932c-3eca2965-da54ee6 at 10.3.0.22] variable_sip_user_agent: [PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933] variable_sip_via_host: [10.3.0.22] variable_max_forwards: [70] variable_presence_id: [10 at 10.3.0.4] variable_switch_r_sdp: [v=0 o=- 1167627581 1167627581 IN IP4 10.3.0.22 s=Polycom IP Phone c=IN IP4 10.3.0.22 t=0 0 a=sendrecv m=audio 2262 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ] variable_remote_media_ip: [10.3.0.22] variable_remote_media_port: [2262] variable_sip_audio_recv_pt: [9] variable_sip_use_codec_name: [G722] variable_sip_use_codec_rate: [8000] variable_sip_use_codec_ptime: [20] variable_read_codec: [G722] variable_read_rate: [16000] variable_write_codec: [G722] variable_write_rate: [16000] variable_endpoint_disposition: [RECEIVED] variable_DP_MATCH: [ARRAY::30|:30] variable_call_uuid: [e73f07fa-6944-11e1-a7c4-a55c2ebde3ff] variable_dialed_extension: [30] variable_export_vars: [dialed_extension] variable_continue_on_fail: [true] variable_hangup_after_bridge: [true] variable_current_application: [info] EXECUTE sofia/internal/10 at 10.3.0.4 bridge([leg_timeout=15][origination_caller_id_number=10]user/30 at 10.3.0.4) 2012-03-08 18:34:09.252934 [DEBUG] switch_channel.c:1047 sofia/internal/10 at 10.3.0.4 EXPORTING[export_vars] [dialed_extension]=[30] to event 2012-03-08 18:34:09.252934 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-08 18:34:09.252934 [DEBUG] switch_ivr_originate.c:2299 Parsing session specific variables 2012-03-08 18:34:09.252934 [DEBUG] switch_event.c:1522 Parsing variable [leg_timeout]=[15] 2012-03-08 18:34:09.252934 [DEBUG] switch_event.c:1522 Parsing variable [origination_caller_id_number]=[10] 2012-03-08 18:34:09.252934 [DEBUG] switch_channel.c:1047 sofia/internal/10 at 10.3.0.4 EXPORTING[export_vars] [dialed_extension]=[30] to event 2012-03-08 18:34:09.252934 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-08 18:34:09.252934 [DEBUG] switch_event.c:1522 Parsing variable [presence_id]=[30 at 10.3.0.4] 2012-03-08 18:34:09.252934 [NOTICE] switch_channel.c:926 New Channel sofia/internal/sip:30 at 10.3.0.26 [e96bae48-6944-11e1-a7cc-a55c2ebde3ff] 2012-03-08 18:34:09.252934 [DEBUG] mod_sofia.c:4673 (sofia/internal/sip:30 at 10.3.0.26) State Change CS_NEW -> CS_INIT 2012-03-08 18:34:09.252934 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:09.252934 [DEBUG] switch_ivr_originate.c:2561 sofia/internal/sip:30 at 10.3.0.26 Setting leg timeout to 15 2012-03-08 18:34:09.252934 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:30 at 10.3.0.26) Running State Change CS_INIT 2012-03-08 18:34:09.252934 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:30 at 10.3.0.26) State INIT 2012-03-08 18:34:09.252934 [DEBUG] mod_sofia.c:85 sofia/internal/sip:30 at 10.3.0.26 SOFIA INIT 2012-03-08 18:34:09.252934 [DEBUG] mod_sofia.c:125 (sofia/internal/sip:30 at 10.3.0.26) State Change CS_INIT -> CS_ROUTING 2012-03-08 18:34:09.252934 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:09.252934 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:30 at 10.3.0.26) State INIT going to sleep 2012-03-08 18:34:09.252934 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:30 at 10.3.0.26) Running State Change CS_ROUTING 2012-03-08 18:34:09.252934 [DEBUG] switch_channel.c:1886 (sofia/internal/sip:30 at 10.3.0.26) Callstate Change DOWN -> RINGING send 1255 bytes to udp/[10.3.0.26]:5060 at 17:34:09.271864: ------------------------------------------------------------------------ INVITE sip:30 at 10.3.0.26 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKrcevmX586F9NB Max-Forwards: 69 From: "A " ;tag=DeFrDpg9mcy3j To: Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 CSeq: 25277960 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 369 X-FS-Support: update_display,send_info Remote-Party-ID: "A " ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1331211599 1331211600 IN IP4 10.3.0.4 s=FreeSWITCH c=IN IP4 10.3.0.4 t=0 0 m=audio 16450 RTP/AVP 9 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 16418 RTP/AVP 98 99 100 34 31 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-2000/90000 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 ------------------------------------------------------------------------ 2012-03-08 18:34:09.252934 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:30 at 10.3.0.26) State ROUTING 2012-03-08 18:34:09.252934 [DEBUG] mod_sofia.c:148 sofia/internal/sip:30 at 10.3.0.26 SOFIA ROUTING 2012-03-08 18:34:09.252934 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:30 at 10.3.0.26) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-03-08 18:34:09.252934 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:09.252934 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:30 at 10.3.0.26) State ROUTING going to sleep 2012-03-08 18:34:09.252934 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:30 at 10.3.0.26) Running State Change CS_CONSUME_MEDIA 2012-03-08 18:34:09.252934 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:30 at 10.3.0.26) State CONSUME_MEDIA 2012-03-08 18:34:09.252934 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:30 at 10.3.0.26) State CONSUME_MEDIA going to sleep 2012-03-08 18:34:09.252934 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:09.252934 [DEBUG] sofia.c:5526 Channel sofia/internal/sip:30 at 10.3.0.26 entering state [calling][0] recv 413 bytes from udp/[10.3.0.26]:5060 at 17:34:09.279527: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKrcevmX586F9NB From: "A" ;tag=DeFrDpg9mcy3j To: "B" ;tag=7463F28A-2FFBD77D CSeq: 25277960 INVITE Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ send 1323 bytes to udp/[10.3.0.22]:5060 at 17:34:09.314994: ------------------------------------------------------------------------ NOTIFY sip:10 at 10.3.0.22 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKSN7mprpc4rZ8p Max-Forwards: 70 From: ;tag=BL1yX2TI1X2c To: "A" ;tag=7B5DFEAE-48CDC3F7 Call-ID: ef7cb8bd-dc381cfe-bf64ccc7 at 10.3.0.22 CSeq: 185332668 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3315 Content-Type: application/dialog-info+xml Content-Length: 509 early sip:30 at 10.3.0.4 sip:10 at 10.3.0.4 ------------------------------------------------------------------------ send 1321 bytes to udp/[10.3.0.25]:5060 at 17:34:09.318770: ------------------------------------------------------------------------ NOTIFY sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKty0DrK7F11NUj Max-Forwards: 70 From: ;tag=zVD97bWqz9Ei To: "C" ;tag=E729C66F-499D331E Call-ID: 6e8428e-4012accd-98b6429c at 10.3.0.25 CSeq: 185332669 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3315 Content-Type: application/dialog-info+xml Content-Length: 509 early sip:30 at 10.3.0.4 sip:10 at 10.3.0.4 ------------------------------------------------------------------------ recv 407 bytes from udp/[10.3.0.22]:5060 at 17:34:09.320972: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKSN7mprpc4rZ8p From: ;tag=BL1yX2TI1X2c To: "A" ;tag=7B5DFEAE-48CDC3F7 CSeq: 185332668 NOTIFY Call-ID: ef7cb8bd-dc381cfe-bf64ccc7 at 10.3.0.22 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ send 1333 bytes to udp/[10.3.0.27]:5060 at 17:34:09.323875: ------------------------------------------------------------------------ NOTIFY sip:31 at 10.3.0.27 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKU7S6SerKyacee Max-Forwards: 70 From: ;tag=j2FaQyd4KlIW To: "Patrick SXXXXXXl" ;tag=26B3115D-24F8F110 Call-ID: 11c82624-cdfad25f-e9f5882 at 10.3.0.27 CSeq: 185332670 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3315 Content-Type: application/dialog-info+xml Content-Length: 509 early sip:30 at 10.3.0.4 sip:10 at 10.3.0.4 ------------------------------------------------------------------------ recv 405 bytes from udp/[10.3.0.25]:5060 at 17:34:09.324463: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKty0DrK7F11NUj From: ;tag=zVD97bWqz9Ei To: "C" ;tag=E729C66F-499D331E CSeq: 185332669 NOTIFY Call-ID: 6e8428e-4012accd-98b6429c at 10.3.0.25 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 417 bytes from udp/[10.3.0.27]:5060 at 17:34:09.328816: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKU7S6SerKyacee From: ;tag=j2FaQyd4KlIW To: "Patrick SXXXXXXl" ;tag=26B3115D-24F8F110 CSeq: 185332670 NOTIFY Call-ID: 11c82624-cdfad25f-e9f5882 at 10.3.0.27 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 450 bytes from udp/[10.3.0.26]:5060 at 17:34:09.330485: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKrcevmX586F9NB From: "A" ;tag=DeFrDpg9mcy3j To: "B" ;tag=7463F28A-2FFBD77D CSeq: 25277960 INVITE Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Allow-Events: talk,hold,conference Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:09.312952 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:09.312952 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:09.352961 [DEBUG] sofia.c:5526 Channel sofia/internal/sip:30 at 10.3.0.26 entering state [proceeding][180] 2012-03-08 18:34:09.352961 [NOTICE] sofia.c:5618 Ring-Ready sofia/internal/sip:30 at 10.3.0.26! 2012-03-08 18:34:09.352961 [NOTICE] mod_sofia.c:2514 Ring-Ready sofia/internal/10 at 10.3.0.4! send 797 bytes to udp/[10.3.0.22]:5060 at 17:34:09.373141: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bKacba9efF22172C0 From: "A" ;tag=B0C14C18-E7DC6BB1 To: ;tag=c5NZBUZ5Q37gQ Call-ID: c64e932c-3eca2965-da54ee6 at 10.3.0.22 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 Remote-Party-ID: "Outbound Call" <30>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ 2012-03-08 18:34:09.352961 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:09.374533 [DEBUG] sofia.c:5526 Channel sofia/internal/10 at 10.3.0.4 entering state [early][180] 2012-03-08 18:34:09.374533 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:09.374533 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/internal/10 at 10.3.0.4! recv 915 bytes from udp/[10.3.0.26]:5060 at 17:34:10.190061: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKrcevmX586F9NB From: "A" ;tag=DeFrDpg9mcy3j To: "B" ;tag=7463F28A-2FFBD77D CSeq: 25277960 INVITE Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Type: application/sdp Content-Length: 347 v=0 o=- 1167627586 1167627586 IN IP4 10.3.0.26 s=Polycom IP Phone c=IN IP4 10.3.0.26 t=0 0 m=audio 2234 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 m=video 0 RTP/AVP 98 99 100 34 31 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-2000/90000 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 ------------------------------------------------------------------------ 2012-03-08 18:34:10.172931 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:10.172931 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:10.192945 [DEBUG] sofia.c:5526 Channel sofia/internal/sip:30 at 10.3.0.26 entering state [completing][200] 2012-03-08 18:34:10.192945 [DEBUG] sofia.c:5537 Remote SDP: v=0 o=- 1167627586 1167627586 IN IP4 10.3.0.26 s=Polycom IP Phone c=IN IP4 10.3.0.26 t=0 0 m=audio 2234 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 m=video 0 RTP/AVP 98 99 100 34 31 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-2000/90000 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 send 336 bytes to udp/[10.3.0.26]:5060 at 17:34:10.201818: ------------------------------------------------------------------------ ACK sip:30 at 10.3.0.26 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKvgKZU98pUK20S Max-Forwards: 70 From: "A " ;tag=DeFrDpg9mcy3j To: ;tag=7463F28A-2FFBD77D Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 CSeq: 25277960 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:10.192945 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:10.192945 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:10.192945 [DEBUG] sofia.c:5526 Channel sofia/internal/sip:30 at 10.3.0.26 entering state [ready][200] 2012-03-08 18:34:10.192945 [DEBUG] sofia_glue.c:4865 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 2012-03-08 18:34:10.192945 [DEBUG] sofia_glue.c:2982 Set Codec sofia/internal/sip:30 at 10.3.0.26 G722/8000 20 ms 160 samples 64000 bits 2012-03-08 18:34:10.192945 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:30 at 10.3.0.26 Original read codec set to G722:9 2012-03-08 18:34:10.192945 [DEBUG] sofia_glue.c:4979 Set 2833 dtmf send payload to 127 2012-03-08 18:34:10.192945 [DEBUG] sofia_glue.c:3234 AUDIO RTP [sofia/internal/sip:30 at 10.3.0.26] 10.3.0.4 port 16450 -> 10.3.0.26 port 2234 codec: 9 ms: 20 2012-03-08 18:34:10.192945 [DEBUG] switch_rtp.c:1661 Starting timer [soft] 160 bytes per 20ms 2012-03-08 18:34:10.192945 [DEBUG] sofia_glue.c:3498 Set 2833 dtmf send payload to 127 2012-03-08 18:34:10.192945 [DEBUG] sofia_glue.c:3504 Set 2833 dtmf receive payload to 101 2012-03-08 18:34:10.192945 [DEBUG] switch_channel.c:3190 (sofia/internal/sip:30 at 10.3.0.26) Callstate Change RINGING -> ACTIVE 2012-03-08 18:34:10.192945 [DEBUG] switch_channel.c:3202 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:10.192945 [NOTICE] sofia.c:6238 Channel [sofia/internal/sip:30 at 10.3.0.26] has been answered 2012-03-08 18:34:10.212964 [DEBUG] sofia_glue.c:3234 AUDIO RTP [sofia/internal/10 at 10.3.0.4] 10.3.0.4 port 16420 -> 10.3.0.22 port 2262 codec: 9 ms: 20 2012-03-08 18:34:10.212964 [DEBUG] switch_rtp.c:1661 Starting timer [soft] 160 bytes per 20ms 2012-03-08 18:34:10.212964 [DEBUG] sofia_glue.c:3498 Set 2833 dtmf send payload to 127 2012-03-08 18:34:10.212964 [DEBUG] sofia_glue.c:3504 Set 2833 dtmf receive payload to 127 2012-03-08 18:34:10.212964 [DEBUG] mod_sofia.c:750 Local SDP sofia/internal/10 at 10.3.0.4: v=0 o=FreeSWITCH 1331211630 1331211631 IN IP4 10.3.0.4 s=FreeSWITCH c=IN IP4 10.3.0.4 t=0 0 m=audio 16420 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2012-03-08 18:34:10.212964 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:10.212964 [DEBUG] switch_channel.c:3190 (sofia/internal/10 at 10.3.0.4) Callstate Change RINGING -> ACTIVE send 1069 bytes to udp/[10.3.0.22]:5060 at 17:34:10.220245: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bKacba9efF22172C0 From: "A" ;tag=B0C14C18-E7DC6BB1 To: ;tag=c5NZBUZ5Q37gQ Call-ID: c64e932c-3eca2965-da54ee6 at 10.3.0.22 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 239 Remote-Party-ID: "Outbound Call" <30>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1331211630 1331211631 IN IP4 10.3.0.4 s=FreeSWITCH c=IN IP4 10.3.0.4 t=0 0 m=audio 16420 RTP/AVP 9 127 a=rtpmap:9 G722/80002012-03-08 18:34:10.212964 [NOTICE] switch_ivr_originate.c:3209 Channel [sofia/internal/10 at 10.3.0.4] has been answered a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2012-03-08 18:34:10.212964 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:10.212964 [DEBUG] sofia.c:5526 Channel sofia/internal/10 at 10.3.0.4 entering state [completed][200] 2012-03-08 18:34:10.212964 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [sofia/internal/sip:30 at 10.3.0.26] 2012-03-08 18:34:10.212964 [DEBUG] switch_ivr_originate.c:2561 sofia/internal/sip:30 at 10.3.0.26 Setting leg timeout to 15 2012-03-08 18:34:10.212964 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [sofia/internal/sip:30 at 10.3.0.26] recv 538 bytes from udp/[10.3.0.22]:5060 at 17:34:10.229645: ------------------------------------------------------------------------ ACK sip:30 at 10.3.0.4:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bKe8d47063D608BAD4 From: "A" ;tag=B0C14C18-E7DC6BB1 To: ;tag=c5NZBUZ5Q37gQ CSeq: 2 ACK Call-ID: c64e932c-3eca2965-da54ee6 at 10.3.0.22 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:10.212964 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:10.212964 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:10.212964 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:10.212964 [DEBUG] sofia.c:5526 Channel sofia/internal/10 at 10.3.0.4 entering state [ready][200] 2012-03-08 18:34:10.232988 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:10.232988 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:10.232988 [DEBUG] switch_ivr_bridge.c:1328 (sofia/internal/sip:30 at 10.3.0.26) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2012-03-08 18:34:10.232988 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:10.232988 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:30 at 10.3.0.26) Running State Change CS_EXCHANGE_MEDIA 2012-03-08 18:34:10.232988 [DEBUG] switch_core_state_machine.c:420 (sofia/internal/sip:30 at 10.3.0.26) State EXCHANGE_MEDIA 2012-03-08 18:34:10.232988 [DEBUG] mod_sofia.c:578 SOFIA EXCHANGE_MEDIA 2012-03-08 18:34:10.232988 [DEBUG] switch_core_session.c:791 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:10.232988 [DEBUG] switch_core_session.c:791 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] send 1325 bytes to udp/[10.3.0.25]:5060 at 17:34:10.284271: ------------------------------------------------------------------------ NOTIFY sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKXScrX4StrvrKN Max-Forwards: 70 From: ;tag=zVD97bWqz9Ei To: "C" ;tag=E729C66F-499D331E Call-ID: 6e8428e-4012accd-98b6429c at 10.3.0.25 CSeq: 185332671 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3314 Content-Type: application/dialog-info+xml Content-Length: 513 confirmed sip:30 at 10.3.0.4 sip:10 at 10.3.0.4 ------------------------------------------------------------------------ send 1337 bytes to udp/[10.3.0.27]:5060 at 17:34:10.287919: ------------------------------------------------------------------------ NOTIFY sip:31 at 10.3.0.27 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKy25gZZayN5e6g Max-Forwards: 70 From: ;tag=j2FaQyd4KlIW To: "Patrick SXXXXXXl" ;tag=26B3115D-24F8F110 Call-ID: 11c82624-cdfad25f-e9f5882 at 10.3.0.27 CSeq: 185332672 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3314 Content-Type: application/dialog-info+xml Content-Length: 513 confirmed sip:30 at 10.3.0.4 sip:10 at 10.3.0.4 ------------------------------------------------------------------------ recv 405 bytes from udp/[10.3.0.25]:5060 at 17:34:10.289984: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKXScrX4StrvrKN From: ;tag=zVD97bWqz9Ei To: "C" ;tag=E729C66F-499D331E CSeq: 185332671 NOTIFY Call-ID: 6e8428e-4012accd-98b6429c at 10.3.0.25 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ send 1327 bytes to udp/[10.3.0.22]:5060 at 17:34:10.292085: ------------------------------------------------------------------------ NOTIFY sip:10 at 10.3.0.22 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKZBZ90tU1je5rc Max-Forwards: 70 From: ;tag=BL1yX2TI1X2c To: "A" ;tag=7B5DFEAE-48CDC3F7 Call-ID: ef7cb8bd-dc381cfe-bf64ccc7 at 10.3.0.22 CSeq: 185332673 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3314 Content-Type: application/dialog-info+xml Content-Length: 513 confirmed sip:30 at 10.3.0.4 sip:10 at 10.3.0.4 ------------------------------------------------------------------------ recv 417 bytes from udp/[10.3.0.27]:5060 at 17:34:10.294071: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKy25gZZayN5e6g From: ;tag=j2FaQyd4KlIW To: "Patrick SXXXXXXl" ;tag=26B3115D-24F8F110 CSeq: 185332672 NOTIFY Call-ID: 11c82624-cdfad25f-e9f5882 at 10.3.0.27 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:10.295560 [DEBUG] switch_rtp.c:3205 Correct ip/port confirmed. recv 407 bytes from udp/[10.3.0.22]:5060 at 17:34:10.306207: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKZBZ90tU1je5rc From: ;tag=BL1yX2TI1X2c To: "A" ;tag=7B5DFEAE-48CDC3F7 CSeq: 185332673 NOTIFY Call-ID: ef7cb8bd-dc381cfe-bf64ccc7 at 10.3.0.22 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ send 1326 bytes to udp/[10.3.0.25]:5060 at 17:34:10.321415: ------------------------------------------------------------------------ NOTIFY sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK0mr22Nc5FQUBr Max-Forwards: 70 From: ;tag=JElIsHAUjri7 To: "C" ;tag=432C4A25-2C483674 Call-ID: a941c408-2c54f3a7-82c5a8d6 at 10.3.0.25 CSeq: 185332674 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=1151 Content-Type: application/dialog-info+xml Content-Length: 513 confirmed sip:10 at 10.3.0.4 sip:30 at 10.3.0.4 ------------------------------------------------------------------------ send 1334 bytes to udp/[10.3.0.26]:5060 at 17:34:10.325110: ------------------------------------------------------------------------ NOTIFY sip:30 at 10.3.0.26 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK1XHU4gX8c0HyK Max-Forwards: 70 From: ;tag=hzYaAHIZscZq To: "B" ;tag=A52D4836-AD38FE89 Call-ID: 3f77033d-f3b24ec8-f0479fab at 10.3.0.26 CSeq: 185332675 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=1151 Content-Type: application/dialog-info+xml Content-Length: 513 confirmed sip:10 at 10.3.0.4 sip:30 at 10.3.0.4 ------------------------------------------------------------------------ recv 406 bytes from udp/[10.3.0.25]:5060 at 17:34:10.327746: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK0mr22Nc5FQUBr From: ;tag=JElIsHAUjri7 To: "C" ;tag=432C4A25-2C483674 CSeq: 185332674 NOTIFY Call-ID: a941c408-2c54f3a7-82c5a8d6 at 10.3.0.25 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 414 bytes from udp/[10.3.0.26]:5060 at 17:34:10.330727: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK1XHU4gX8c0HyK From: ;tag=hzYaAHIZscZq To: "B" ;tag=A52D4836-AD38FE89 CSeq: 185332675 NOTIFY Call-ID: 3f77033d-f3b24ec8-f0479fab at 10.3.0.26 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:10.332940 [DEBUG] switch_rtp.c:3205 Correct ip/port confirmed. recv 986 bytes from udp/[10.3.0.26]:5060 at 17:34:12.188618: ------------------------------------------------------------------------ INVITE sip:mod_sofia at 10.3.0.4:5060 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bKb0c085088B8047EB From: "B" ;tag=7463F28A-2FFBD77D To: "A" ;tag=DeFrDpg9mcy3j CSeq: 1 INVITE Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 345 v=0 o=- 1167627586 1167627587 IN IP4 10.3.0.26 s=Polycom IP Phone c=IN IP4 0.0.0.0 t=0 0 m=audio 2234 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 m=video 0 RTP/AVP 98 99 100 34 31 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-2000/90000 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 ------------------------------------------------------------------------ send 358 bytes to udp/[10.3.0.26]:5060 at 17:34:12.189427: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bKb0c085088B8047EB From: "B" ;tag=7463F28A-2FFBD77D To: "A" ;tag=DeFrDpg9mcy3j Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:12.172933 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:12.172933 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:12.192941 [DEBUG] sofia.c:5526 Channel sofia/internal/sip:30 at 10.3.0.26 entering state [received][100] 2012-03-08 18:34:12.192941 [DEBUG] sofia.c:5537 Remote SDP: v=0 o=- 1167627586 1167627587 IN IP4 10.3.0.26 s=Polycom IP Phone c=IN IP4 0.0.0.0 t=0 0 m=audio 2234 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=sendonly m=video 0 RTP/AVP 98 99 100 34 31 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-2000/90000 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 2012-03-08 18:34:12.192941 [DEBUG] switch_channel.c:1560 (sofia/internal/sip:30 at 10.3.0.26) Callstate Change ACTIVE -> HELD 2012-03-08 18:34:12.192941 [DEBUG] switch_core_session.c:1012 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:12.212935 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] send 1322 bytes to udp/[10.3.0.22]:5060 at 17:34:12.251628: ------------------------------------------------------------------------ NOTIFY sip:10 at 10.3.0.22 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK26am6Beca97gF Max-Forwards: 70 From: ;tag=BL1yX2TI1X2c To: "A" ;tag=7B5DFEAE-48CDC3F7 Call-ID: ef7cb8bd-dc381cfe-bf64ccc7 at 10.3.0.22 CSeq: 185332676 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3312 Content-Type: application/dialog-info+xml Content-Length: 508 early sip:30 at 10.3.0.4 sip:10 at 10.3.0.4 ------------------------------------------------------------------------ send 1320 bytes to udp/[10.3.0.25]:5060 at 17:34:12.252624: ------------------------------------------------------------------------ NOTIFY sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK3F4c86yF7Hy3a Max-Forwards: 70 From: ;tag=zVD97bWqz9Ei To: "C" ;tag=E729C66F-499D331E Call-ID: 6e8428e-4012accd-98b6429c at 10.3.0.25 CSeq: 185332677 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3312 Content-Type: application/dialog-info+xml Content-Length: 508 early sip:30 at 10.3.0.4 sip:10 at 10.3.0.4 ------------------------------------------------------------------------ send 1332 bytes to udp/[10.3.0.27]:5060 at 17:34:12.257830: ------------------------------------------------------------------------ NOTIFY sip:31 at 10.3.0.27 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK4rX591FK4tmpp Max-Forwards: 70 From: ;tag=j2FaQyd4KlIW To: "Patrick SXXXXXXl" ;tag=26B3115D-24F8F110 Call-ID: 11c82624-cdfad25f-e9f5882 at 10.3.0.27 CSeq: 185332678 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3312 Content-Type: application/dialog-info+xml Content-Length: 508 early sip:30 at 10.3.0.4 sip:10 at 10.3.0.4 ------------------------------------------------------------------------ recv 405 bytes from udp/[10.3.0.25]:5060 at 17:34:12.259955: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK3F4c86yF7Hy3a From: ;tag=zVD97bWqz9Ei To: "C" ;tag=E729C66F-499D331E CSeq: 185332677 NOTIFY Call-ID: 6e8428e-4012accd-98b6429c at 10.3.0.25 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 407 bytes from udp/[10.3.0.22]:5060 at 17:34:12.260542: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK26am6Beca97gF From: ;tag=BL1yX2TI1X2c To: "A" ;tag=7B5DFEAE-48CDC3F7 CSeq: 185332676 NOTIFY Call-ID: ef7cb8bd-dc381cfe-bf64ccc7 at 10.3.0.22 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 417 bytes from udp/[10.3.0.27]:5060 at 17:34:12.264011: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK4rX591FK4tmpp From: ;tag=j2FaQyd4KlIW To: "Patrick SXXXXXXl" ;tag=26B3115D-24F8F110 CSeq: 185332678 NOTIFY Call-ID: 11c82624-cdfad25f-e9f5882 at 10.3.0.27 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:12.392939 [DEBUG] switch_ivr.c:591 sofia/internal/10 at 10.3.0.4 Command Execute playback(local_stream://moh) EXECUTE sofia/internal/10 at 10.3.0.4 playback(local_stream://moh) 2012-03-08 18:34:12.392939 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/16000] 16000hz 2012-03-08 18:34:12.392939 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16 at 16000hz 1 channels 20ms 2012-03-08 18:34:12.432933 [DEBUG] sofia_glue.c:4865 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 2012-03-08 18:34:12.432933 [DEBUG] sofia_glue.c:2916 Already using G722 2012-03-08 18:34:12.432933 [DEBUG] sofia_glue.c:4979 Set 2833 dtmf send payload to 101 2012-03-08 18:34:12.432933 [DEBUG] sofia_glue.c:3223 Audio params changed for sofia/internal/sip:30 at 10.3.0.26 from 10.3.0.26:2234 to 0.0.0.0:2234 2012-03-08 18:34:12.432933 [DEBUG] sofia_glue.c:3234 AUDIO RTP [sofia/internal/sip:30 at 10.3.0.26] 10.3.0.4 port 16450 -> 0.0.0.0 port 2234 codec: 9 ms: 20 2012-03-08 18:34:12.432933 [DEBUG] sofia_glue.c:3275 AUDIO RTP CHANGING DEST TO: [0.0.0.0:2234] 2012-03-08 18:34:12.432933 [DEBUG] sofia.c:6034 Processing updated SDP send 937 bytes to udp/[10.3.0.26]:5060 at 17:34:12.446972: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bKb0c085088B8047EB From: "B" ;tag=7463F28A-2FFBD77D To: "A" ;tag=DeFrDpg9mcy3j Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 CSeq: 1 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 297 v=0 o=FreeSWITCH 1331211599 1331211601 IN IP4 10.3.0.4 s=FreeSWITCH c=IN IP4 10.3.0.4 t=0 0 m=audio 16450 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=recvonly a=silenceSupp:off - - - - a=ptime:20 m=video 0 RTP/AVP 98 a=rtpmap:98 H264/90000 ------------------------------------------------------------------------ 2012-03-08 18:34:12.432933 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:12.452935 [DEBUG] sofia.c:5526 Channel sofia/internal/sip:30 at 10.3.0.26 entering state [completed][200] recv 537 bytes from udp/[10.3.0.26]:5060 at 17:34:12.455260: ------------------------------------------------------------------------ ACK sip:mod_sofia at 10.3.0.4:5060 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bKe85649e7EEFB8C2 From: "B" ;tag=7463F28A-2FFBD77D To: "A" ;tag=DeFrDpg9mcy3j CSeq: 1 ACK Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:12.452935 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:12.452935 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:12.452935 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:12.472932 [DEBUG] sofia.c:5526 Channel sofia/internal/sip:30 at 10.3.0.26 entering state [ready][200] recv 869 bytes from udp/[10.3.0.26]:5060 at 17:34:14.406913: ------------------------------------------------------------------------ INVITE sip:20 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK460db7754D408CC0 From: "B" ;tag=4AB6AEB9-5DA56A24 To: CSeq: 1 INVITE Call-ID: da10f563-8c35d41e-ad0761b1 at 10.3.0.26 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 241 v=0 o=- 1167627590 1167627590 IN IP4 10.3.0.26 s=Polycom IP Phone c=IN IP4 10.3.0.26 t=0 0 a=sendrecv m=audio 2236 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ------------------------------------------------------------------------ send 341 bytes to udp/[10.3.0.26]:5060 at 17:34:14.407711: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK460db7754D408CC0 From: "B" ;tag=4AB6AEB9-5DA56A24 To: Call-ID: da10f563-8c35d41e-ad0761b1 at 10.3.0.26 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:14.392941 [DEBUG] sofia.c:7559 IP 10.3.0.26 Rejected by acl "domains". Falling back to Digest auth. 2012-03-08 18:34:14.392941 [WARNING] sofia_reg.c:1422 SIP auth challenge (INVITE) on sofia profile 'internal' for [20 at 10.3.0.4] from ip 10.3.0.26 send 824 bytes to udp/[10.3.0.26]:5060 at 17:34:14.412500: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK460db7754D408CC0 From: "B" ;tag=4AB6AEB9-5DA56A24 To: ;tag=eQ8gFH1cjNmpe Call-ID: da10f563-8c35d41e-ad0761b1 at 10.3.0.26 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="10.3.0.4", nonce="ec7c3b52-6944-11e1-a7d4-a55c2ebde3ff", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 543 bytes from udp/[10.3.0.26]:5060 at 17:34:14.420731: ------------------------------------------------------------------------ ACK sip:20 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK460db7754D408CC0 From: "B" ;tag=4AB6AEB9-5DA56A24 To: ;tag=eQ8gFH1cjNmpe CSeq: 1 ACK Call-ID: da10f563-8c35d41e-ad0761b1 at 10.3.0.26 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ recv 1124 bytes from udp/[10.3.0.26]:5060 at 17:34:14.423632: ------------------------------------------------------------------------ INVITE sip:20 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK909bdcdc8659BA5F From: "B" ;tag=4AB6AEB9-5DA56A24 To: CSeq: 2 INVITE Call-ID: da10f563-8c35d41e-ad0761b1 at 10.3.0.26 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="30", realm="10.3.0.4", nonce="ec7c3b52-6944-11e1-a7d4-a55c2ebde3ff", qop=auth, cnonce="XJz4HrL/160QNgt", nc=00000001, uri="sip:20 at 10.3.0.4:5060;user=phone", response="5aa06aa91b7e6941539ee02b96a22789", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 241 v=0 o=- 1167627590 1167627590 IN IP4 10.3.0.26 s=Polycom IP Phone c=IN IP4 10.3.0.26 t=0 0 a=sendrecv m=audio 2236 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ------------------------------------------------------------------------ send 341 bytes to udp/[10.3.0.26]:5060 at 17:34:14.424685: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK909bdcdc8659BA5F From: "B" ;tag=4AB6AEB9-5DA56A24 To: Call-ID: da10f563-8c35d41e-ad0761b1 at 10.3.0.26 CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:14.412921 [DEBUG] sofia.c:7559 IP 10.3.0.26 Rejected by acl "domains". Falling back to Digest auth. 2012-03-08 18:34:14.432937 [NOTICE] switch_channel.c:926 New Channel sofia/internal/30 at 10.3.0.4 [ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff] 2012-03-08 18:34:14.432937 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/30 at 10.3.0.4) Running State Change CS_NEW 2012-03-08 18:34:14.432937 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/30 at 10.3.0.4) State NEW 2012-03-08 18:34:14.432937 [DEBUG] sofia.c:5526 Channel sofia/internal/30 at 10.3.0.4 entering state [received][100] 2012-03-08 18:34:14.432937 [DEBUG] sofia.c:5537 Remote SDP: v=0 o=- 1167627590 1167627590 IN IP4 10.3.0.26 s=Polycom IP Phone c=IN IP4 10.3.0.26 t=0 0 a=sendrecv m=audio 2236 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 2012-03-08 18:34:14.432937 [DEBUG] sofia_glue.c:4865 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 2012-03-08 18:34:14.432937 [DEBUG] sofia_glue.c:2982 Set Codec sofia/internal/30 at 10.3.0.4 G722/8000 20 ms 160 samples 64000 bits 2012-03-08 18:34:14.432937 [DEBUG] switch_core_codec.c:111 sofia/internal/30 at 10.3.0.4 Original read codec set to G722:9 2012-03-08 18:34:14.432937 [DEBUG] sofia_glue.c:4986 Set 2833 dtmf send/recv payload to 127 2012-03-08 18:34:14.432937 [DEBUG] sofia.c:5749 (sofia/internal/30 at 10.3.0.4) State Change CS_NEW -> CS_INIT 2012-03-08 18:34:14.432937 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/30 at 10.3.0.4 [BREAK] 2012-03-08 18:34:14.432937 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/30 at 10.3.0.4) Running State Change CS_INIT 2012-03-08 18:34:14.432937 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/30 at 10.3.0.4) State INIT 2012-03-08 18:34:14.432937 [DEBUG] mod_sofia.c:85 sofia/internal/30 at 10.3.0.4 SOFIA INIT 2012-03-08 18:34:14.432937 [DEBUG] mod_sofia.c:125 (sofia/internal/30 at 10.3.0.4) State Change CS_INIT -> CS_ROUTING 2012-03-08 18:34:14.432937 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/30 at 10.3.0.4 [BREAK] 2012-03-08 18:34:14.432937 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/30 at 10.3.0.4) State INIT going to sleep 2012-03-08 18:34:14.432937 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/30 at 10.3.0.4) Running State Change CS_ROUTING 2012-03-08 18:34:14.432937 [DEBUG] switch_channel.c:1886 (sofia/internal/30 at 10.3.0.4) Callstate Change DOWN -> RINGING 2012-03-08 18:34:14.432937 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/30 at 10.3.0.4) State ROUTING 2012-03-08 18:34:14.432937 [DEBUG] mod_sofia.c:148 sofia/internal/30 at 10.3.0.4 SOFIA ROUTING 2012-03-08 18:34:14.432937 [DEBUG] switch_core_state_machine.c:104 sofia/internal/30 at 10.3.0.4 Standard ROUTING 2012-03-08 18:34:14.432937 [INFO] mod_dialplan_xml.c:485 Processing B <30>->20 in context default Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->nxo_enable_chefsec] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [nxo_enable_chefsec] destination_number(20) =~ /^\*95(\d{0,7})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [group-intercept] destination_number(20) =~ /^\*82$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [intercept-ext] destination_number(20) =~ /^\*81(\d+)$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->nxo_single_intercom_with_two_way_audio] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [nxo_single_intercom_with_two_way_audio] destination_number(20) =~ /^\*01(\d{2,7})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->nxo_group_intercom] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [nxo_group_intercom] destination_number(20) =~ /^\*02(\d{0,7})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->Outbound to PSTN 11 Digits] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [Outbound to PSTN 11 Digits] destination_number(20) =~ /^(1[2-9][0-9]{2}[2-9]{7})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->del-group] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [del-group] destination_number(20) =~ /^\*\*50(\d{2})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->add-group] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [add-group] destination_number(20) =~ /^\*\*51(\d{2})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [call-group-simo] destination_number(20) =~ /^\*52(\d{2,4})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->call-group-order] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [call-group-order] destination_number(20) =~ /^\*53(\d{2,4})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->nb_conferences] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [nb_conferences] destination_number(20) =~ /^\*(30\d{2})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->wb_conferences] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [wb_conferences] destination_number(20) =~ /^\*(31\d{2})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->uwb_conferences] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [uwb_conferences] destination_number(20) =~ /^\*(32\d{2})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->cdquality_conferences] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [cdquality_conferences] destination_number(20) =~ /^\*(33\d{2})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->global_directory] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [global_directory] destination_number(20) =~ /^\*77(\d{1,3})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->redirect_now] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [redirect_now] destination_number(20) =~ /^\*21(\d{0,20})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->redirect_timeout] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [redirect_timeout] destination_number(20) =~ /^\*22(\d{0,20})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->redirect_busy] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [redirect_busy] destination_number(20) =~ /^\*23(\d{0,20})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->call_privacy] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [call_privacy] destination_number(20) =~ /^\*67(\d+)$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->call_privacy] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [call_privacy] destination_number(20) =~ /^\*60$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->send_to_voicemail] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [send_to_voicemail] destination_number(20) =~ /^\*99(\d{2,7})$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->vmain] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [vmain] destination_number(20) =~ /^vmain$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->vmain1] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [vmain1] destination_number(20) =~ /^vmain1$|^\*97$|^97$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->vmain2] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [vmain2] destination_number(20) =~ /^vmain2$|^\*98$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->redirect_now] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (FAIL) [redirect_now] destination_number(20) =~ /^\*35$/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 parsing [default->Local_Extension_test] continue=false Dialplan: sofia/internal/30 at 10.3.0.4 Regex (PASS) [Local_Extension_test] destination_number(20) =~ /(^\d{2,7}$)/ break=on-false Dialplan: sofia/internal/30 at 10.3.0.4 Action set(dialed_extension=20) Dialplan: sofia/internal/30 at 10.3.0.4 Action export(dialed_extension=20) Dialplan: sofia/internal/30 at 10.3.0.4 Action python(nxo_test) 2012-03-08 18:34:14.432937 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/30 at 10.3.0.4) State Change CS_ROUTING -> CS_EXECUTE 2012-03-08 18:34:14.432937 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/30 at 10.3.0.4 [BREAK] 2012-03-08 18:34:14.432937 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/30 at 10.3.0.4) State ROUTING going to sleep 2012-03-08 18:34:14.432937 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/30 at 10.3.0.4) Running State Change CS_EXECUTE 2012-03-08 18:34:14.432937 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/30 at 10.3.0.4) State EXECUTE 2012-03-08 18:34:14.432937 [DEBUG] mod_sofia.c:241 sofia/internal/30 at 10.3.0.4 SOFIA EXECUTE 2012-03-08 18:34:14.432937 [DEBUG] switch_core_state_machine.c:192 sofia/internal/30 at 10.3.0.4 Standard EXECUTE EXECUTE sofia/internal/30 at 10.3.0.4 set(dialed_extension=20) 2012-03-08 18:34:14.432937 [DEBUG] mod_dptools.c:1281 sofia/internal/30 at 10.3.0.4 SET [dialed_extension]=[20] EXECUTE sofia/internal/30 at 10.3.0.4 export(dialed_extension=20) 2012-03-08 18:34:14.432937 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [dialed_extension]=[20] EXECUTE sofia/internal/30 at 10.3.0.4 python(nxo_test) 2012-03-08 18:34:14.452987 [NOTICE] mod_python.c:212 Invoking py module: nxo_test 2012-03-08 18:34:14.452987 [DEBUG] mod_python.c:281 Call python script 2012-03-08 18:34:14.452987 [INFO] switch_cpp.cpp:1227 TEST: Destination-extension 20 send 1327 bytes to udp/[10.3.0.22]:5060 at 17:34:14.522894: ------------------------------------------------------------------------ NOTIFY sip:10 at 10.3.0.22 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK51pyBX0p13a9H Max-Forwards: 70 From: ;tag=BL1yX2TI1X2c To: "A" ;tag=7B5DFEAE-48CDC3F7 Call-ID: ef7cb8bd-dc381cfe-bf64ccc7 at 10.3.0.22 CSeq: 185332679 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3310 Content-Type: application/dialog-info+xml Content-Length: 513 confirmed sip:30 at 10.3.0.4 sip:20 at 10.3.0.4 ------------------------------------------------------------------------ send 1325 bytes to udp/[10.3.0.25]:5060 at 17:34:14.523907: ------------------------------------------------------------------------ NOTIFY sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK6agQDrHtyc1UD Max-Forwards: 70 From: ;tag=zVD97bWqz9Ei To: "C" ;tag=E729C66F-499D331E Call-ID: 6e8428e-4012accd-98b6429c at 10.3.0.25 CSeq: 185332680 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3310 Content-Type: application/dialog-info+xml Content-Length: 513 confirmed sip:30 at 10.3.0.4 sip:20 at 10.3.0.4 ------------------------------------------------------------------------ send 1337 bytes to udp/[10.3.0.27]:5060 at 17:34:14.524901: ------------------------------------------------------------------------ NOTIFY sip:31 at 10.3.0.27 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK7K9FFK2XUNQeS Max-Forwards: 70 From: ;tag=j2FaQyd4KlIW To: "Patrick SXXXXXXl" ;tag=26B3115D-24F8F110 Call-ID: 11c82624-cdfad25f-e9f5882 at 10.3.0.27 CSeq: 185332681 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3310 Content-Type: application/dialog-info+xml Content-Length: 513 confirmed sip:30 at 10.3.0.4 sip:20 at 10.3.0.4 ------------------------------------------------------------------------ recv 407 bytes from udp/[10.3.0.22]:5060 at 17:34:14.530514: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK51pyBX0p13a9H From: ;tag=BL1yX2TI1X2c To: "A" ;tag=7B5DFEAE-48CDC3F7 CSeq: 185332679 NOTIFY Call-ID: ef7cb8bd-dc381cfe-bf64ccc7 at 10.3.0.22 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 417 bytes from udp/[10.3.0.27]:5060 at 17:34:14.530992: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK7K9FFK2XUNQeS From: ;tag=j2FaQyd4KlIW To: "Patrick SXXXXXXl" ;tag=26B3115D-24F8F110 CSeq: 185332681 NOTIFY Call-ID: 11c82624-cdfad25f-e9f5882 at 10.3.0.27 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 405 bytes from udp/[10.3.0.25]:5060 at 17:34:14.531364: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK6agQDrHtyc1UD From: ;tag=zVD97bWqz9Ei To: "C" ;tag=E729C66F-499D331E CSeq: 185332680 NOTIFY Call-ID: 6e8428e-4012accd-98b6429c at 10.3.0.25 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ EXECUTE sofia/internal/30 at 10.3.0.4 set(continue_on_fail=true) 2012-03-08 18:34:14.952961 [DEBUG] mod_dptools.c:1281 sofia/internal/30 at 10.3.0.4 SET [continue_on_fail]=[true] EXECUTE sofia/internal/30 at 10.3.0.4 set(hangup_after_bridge=true) 2012-03-08 18:34:14.952961 [DEBUG] mod_dptools.c:1281 sofia/internal/30 at 10.3.0.4 SET [hangup_after_bridge]=[true] 2012-03-08 18:34:14.952961 [WARNING] switch_cpp.cpp:1227 TEST 1 for UUID ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff, src 30 dest 20 EXECUTE sofia/internal/30 at 10.3.0.4 sleep(1000) 2012-03-08 18:34:15.952932 [WARNING] switch_cpp.cpp:1227 TEST 2 for UUID ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff, src 30 dest 20 EXECUTE sofia/internal/30 at 10.3.0.4 sleep(1000) 2012-03-08 18:34:16.952933 [WARNING] switch_cpp.cpp:1227 TEST 3 for UUID ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff, src 30 dest 20 EXECUTE sofia/internal/30 at 10.3.0.4 sleep(1000) EXECUTE sofia/internal/30 at 10.3.0.4 info() 2012-03-08 18:34:17.952932 [INFO] mod_dptools.c:1439 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [RINGING] Channel-State-Number: [4] Channel-Name: [sofia/internal/30 at 10.3.0.4] Unique-ID: [ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-HIT-Dialplan: [true] Channel-Presence-ID: [30 at 10.3.0.4] Channel-Call-UUID: [ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff] Answer-State: [ringing] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Read-Codec-Bit-Rate: [64000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Channel-Write-Codec-Bit-Rate: [64000] Caller-Direction: [inbound] Caller-Username: [30] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [B] Caller-Caller-ID-Number: [30] Caller-Network-Addr: [10.3.0.26] Caller-ANI: [30] Caller-Destination-Number: [20] Caller-Unique-ID: [ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/30 at 10.3.0.4] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1331228054432937] Caller-Channel-Created-Time: [1331228054432937] 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] variable_direction: [inbound] variable_uuid: [ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff] variable_session_id: [11] variable_sip_local_network_addr: [10.3.0.4] variable_sip_network_ip: [10.3.0.26] variable_sip_network_port: [5060] variable_sip_received_ip: [10.3.0.26] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_authorized: [true] variable_sip_number_alias: [30] variable_sip_auth_username: [30] variable_sip_auth_realm: [10.3.0.4] variable_number_alias: [30] variable_user_name: [30] variable_domain_name: [10.3.0.4] variable_record_stereo: [true] variable_default_gateway: [fonira] variable_default_areacode: [01] variable_transfer_fallback_extension: [operator] variable_toll_allow: [local,domestic,international,vas] variable_accountcode: [30] variable_user_context: [default] variable_effective_caller_id_name: [B] variable_effective_caller_id_number: [30] variable_outbound_caller_id_name: [B] variable_outbound_caller_id_number: [1997156030] variable_callgroup: [intercept] variable_sip_from_user: [30] variable_sip_from_uri: [30 at 10.3.0.4] variable_sip_from_host: [10.3.0.4] variable_sip_from_user_stripped: [30] variable_sip_from_tag: [4AB6AEB9-5DA56A24] variable_sofia_profile_name: [internal] variable_sip_full_via: [SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK909bdcdc8659BA5F] variable_sip_from_display: [B] variable_sip_full_from: ["B" ;tag=4AB6AEB9-5DA56A24] variable_sip_full_to: [] variable_sip_req_params: [user=phone] variable_sip_req_user: [20] variable_sip_req_port: [5060] variable_sip_req_uri: [20 at 10.3.0.4:5060] variable_sip_req_host: [10.3.0.4] variable_sip_to_params: [user=phone] variable_sip_to_user: [20] variable_sip_to_uri: [20 at 10.3.0.4] variable_sip_to_host: [10.3.0.4] variable_sip_contact_user: [30] variable_sip_contact_uri: [30 at 10.3.0.26] variable_sip_contact_host: [10.3.0.26] variable_channel_name: [sofia/internal/30 at 10.3.0.4] variable_sip_call_id: [da10f563-8c35d41e-ad0761b1 at 10.3.0.26] variable_sip_user_agent: [PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933] variable_sip_via_host: [10.3.0.26] variable_max_forwards: [70] variable_presence_id: [30 at 10.3.0.4] variable_switch_r_sdp: [v=0 o=- 1167627590 1167627590 IN IP4 10.3.0.26 s=Polycom IP Phone c=IN IP4 10.3.0.26 t=0 0 a=sendrecv m=audio 2236 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ] variable_remote_media_ip: [10.3.0.26] variable_remote_media_port: [2236] variable_sip_audio_recv_pt: [9] variable_sip_use_codec_name: [G722] variable_sip_use_codec_rate: [8000] variable_sip_use_codec_ptime: [20] variable_read_codec: [G722] variable_read_rate: [16000] variable_write_codec: [G722] variable_write_rate: [16000] variable_endpoint_disposition: [RECEIVED] variable_DP_MATCH: [ARRAY::20|:20] variable_call_uuid: [ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff] variable_dialed_extension: [20] variable_export_vars: [dialed_extension] variable_continue_on_fail: [true] variable_hangup_after_bridge: [true] variable_current_application: [info] EXECUTE sofia/internal/30 at 10.3.0.4 bridge([leg_timeout=15][origination_caller_id_number=30]user/20 at 10.3.0.4) 2012-03-08 18:34:17.972915 [DEBUG] switch_channel.c:1047 sofia/internal/30 at 10.3.0.4 EXPORTING[export_vars] [dialed_extension]=[20] to event 2012-03-08 18:34:17.972915 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-08 18:34:17.972915 [DEBUG] switch_ivr_originate.c:2299 Parsing session specific variables 2012-03-08 18:34:17.972915 [DEBUG] switch_event.c:1522 Parsing variable [leg_timeout]=[15] 2012-03-08 18:34:17.972915 [DEBUG] switch_event.c:1522 Parsing variable [origination_caller_id_number]=[30] 2012-03-08 18:34:17.972915 [DEBUG] switch_channel.c:1047 sofia/internal/30 at 10.3.0.4 EXPORTING[export_vars] [dialed_extension]=[20] to event 2012-03-08 18:34:17.972915 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-08 18:34:17.972915 [DEBUG] switch_event.c:1522 Parsing variable [presence_id]=[20 at 10.3.0.4] 2012-03-08 18:34:17.972915 [NOTICE] switch_channel.c:926 New Channel sofia/internal/sip:20 at 10.3.0.25 [ee9ed232-6944-11e1-a7dd-a55c2ebde3ff] 2012-03-08 18:34:17.972915 [DEBUG] mod_sofia.c:4673 (sofia/internal/sip:20 at 10.3.0.25) State Change CS_NEW -> CS_INIT 2012-03-08 18:34:17.972915 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-08 18:34:17.992947 [DEBUG] switch_ivr_originate.c:2561 sofia/internal/sip:20 at 10.3.0.25 Setting leg timeout to 15 2012-03-08 18:34:17.992947 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:20 at 10.3.0.25) Running State Change CS_INIT 2012-03-08 18:34:17.992947 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:20 at 10.3.0.25) State INIT 2012-03-08 18:34:17.992947 [DEBUG] mod_sofia.c:85 sofia/internal/sip:20 at 10.3.0.25 SOFIA INIT 2012-03-08 18:34:17.992947 [DEBUG] mod_sofia.c:125 (sofia/internal/sip:20 at 10.3.0.25) State Change CS_INIT -> CS_ROUTING 2012-03-08 18:34:17.992947 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-08 18:34:17.992947 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:20 at 10.3.0.25) State INIT going to sleep 2012-03-08 18:34:17.992947 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:20 at 10.3.0.25) Running State Change CS_ROUTING 2012-03-08 18:34:17.992947 [DEBUG] switch_channel.c:1886 (sofia/internal/sip:20 at 10.3.0.25) Callstate Change DOWN -> RINGING send 1267 bytes to udp/[10.3.0.25]:5060 at 17:34:17.995749: ------------------------------------------------------------------------ INVITE sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK8v28geK1ryD1m Max-Forwards: 69 From: "B" ;tag=g9t2j72Kc70UN To: Call-ID: c5fdcf90-e3e7-122f-41a2-00900b1be504 CSeq: 25277964 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 369 X-FS-Support: update_display,send_info Remote-Party-ID: "B" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1331211673 1331211674 IN IP4 10.3.0.4 s=FreeSWITCH c=IN IP4 10.3.0.4 t=0 0 m=audio 16384 RTP/AVP 9 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 16422 RTP/AVP 98 99 100 34 31 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-2000/90000 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 ------------------------------------------------------------------------ 2012-03-08 18:34:17.992947 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:20 at 10.3.0.25) State ROUTING 2012-03-08 18:34:17.992947 [DEBUG] mod_sofia.c:148 sofia/internal/sip:20 at 10.3.0.25 SOFIA ROUTING 2012-03-08 18:34:17.992947 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:20 at 10.3.0.25) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-03-08 18:34:17.992947 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-08 18:34:17.992947 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:20 at 10.3.0.25) State ROUTING going to sleep 2012-03-08 18:34:17.992947 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:20 at 10.3.0.25) Running State Change CS_CONSUME_MEDIA 2012-03-08 18:34:17.992947 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:20 at 10.3.0.25) State CONSUME_MEDIA 2012-03-08 18:34:17.992947 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:20 at 10.3.0.25) State CONSUME_MEDIA going to sleep 2012-03-08 18:34:17.992947 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-08 18:34:17.992947 [DEBUG] sofia.c:5526 Channel sofia/internal/sip:20 at 10.3.0.25 entering state [calling][0] recv 412 bytes from udp/[10.3.0.25]:5060 at 17:34:18.004636: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK8v28geK1ryD1m From: "B" ;tag=g9t2j72Kc70UN To: "C" ;tag=D3DA58FE-5FE1943D CSeq: 25277964 INVITE Call-ID: c5fdcf90-e3e7-122f-41a2-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ send 1323 bytes to udp/[10.3.0.22]:5060 at 17:34:18.044978: ------------------------------------------------------------------------ NOTIFY sip:10 at 10.3.0.22 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK95U1j934N73Kg Max-Forwards: 70 From: ;tag=nYGREnwV9Yw1 To: "A" ;tag=3AFAE503-964EC474 Call-ID: 5e0bdbb0-b7399c09-c8206caa at 10.3.0.22 CSeq: 185332682 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3307 Content-Type: application/dialog-info+xml Content-Length: 509 early sip:20 at 10.3.0.4 sip:30 at 10.3.0.4 ------------------------------------------------------------------------ send 1330 bytes to udp/[10.3.0.26]:5060 at 17:34:18.048242: ------------------------------------------------------------------------ NOTIFY sip:30 at 10.3.0.26 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKaFNtm4m8jgt6B Max-Forwards: 70 From: ;tag=eMGDLbL5kcdj To: "B" ;tag=5D666AB7-2B76D012 Call-ID: 467e0be4-bb0339a7-ce4d6682 at 10.3.0.26 CSeq: 185332683 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3306 Content-Type: application/dialog-info+xml Content-Length: 509 early sip:20 at 10.3.0.4 sip:30 at 10.3.0.4 ------------------------------------------------------------------------ recv 407 bytes from udp/[10.3.0.22]:5060 at 17:34:18.052291: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK95U1j934N73Kg From: ;tag=nYGREnwV9Yw1 To: "A" ;tag=3AFAE503-964EC474 CSeq: 185332682 NOTIFY Call-ID: 5e0bdbb0-b7399c09-c8206caa at 10.3.0.22 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 414 bytes from udp/[10.3.0.26]:5060 at 17:34:18.053999: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKaFNtm4m8jgt6B From: ;tag=eMGDLbL5kcdj To: "B" ;tag=5D666AB7-2B76D012 CSeq: 185332683 NOTIFY Call-ID: 467e0be4-bb0339a7-ce4d6682 at 10.3.0.26 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 449 bytes from udp/[10.3.0.25]:5060 at 17:34:18.060861: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK8v28geK1ryD1m From: "B" ;tag=g9t2j72Kc70UN To: "C" ;tag=D3DA58FE-5FE1943D CSeq: 25277964 INVITE Call-ID: c5fdcf90-e3e7-122f-41a2-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Allow-Events: talk,hold,conference Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:18.052945 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-08 18:34:18.052945 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-08 18:34:18.052945 [DEBUG] sofia.c:5526 Channel sofia/internal/sip:20 at 10.3.0.25 entering state [proceeding][180] 2012-03-08 18:34:18.052945 [NOTICE] sofia.c:5618 Ring-Ready sofia/internal/sip:20 at 10.3.0.25! 2012-03-08 18:34:18.052945 [NOTICE] mod_sofia.c:2514 Ring-Ready sofia/internal/30 at 10.3.0.4! send 806 bytes to udp/[10.3.0.26]:5060 at 17:34:18.071638: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK909bdcdc8659BA5F From: "B" ;tag=4AB6AEB9-5DA56A24 To: ;tag=F019gcjgFya9S Call-ID: da10f563-8c35d41e-ad0761b1 at 10.3.0.26 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 Remote-Party-ID: "Outbound Call" <20>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ 2012-03-08 18:34:18.052945 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/30 at 10.3.0.4 [BREAK] 2012-03-08 18:34:18.072922 [DEBUG] sofia.c:5526 Channel sofia/internal/30 at 10.3.0.4 entering state [early][180] 2012-03-08 18:34:18.072922 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/30 at 10.3.0.4 [BREAK] 2012-03-08 18:34:18.072922 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/internal/30 at 10.3.0.4! recv 625 bytes from udp/[10.3.0.26]:5060 at 17:34:20.691350: ------------------------------------------------------------------------ REFER sip:mod_sofia at 10.3.0.4:5060 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bKb6a74a78C1F008DB From: "B" ;tag=7463F28A-2FFBD77D To: "A" ;tag=DeFrDpg9mcy3j CSeq: 2 REFER Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Refer-To: Referred-By: Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:20.672931 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:20.692943 [DEBUG] sofia.c:6514 Process REFER to [20 at 10.3.0.4] 2012-03-08 18:34:20.692943 [DEBUG] sofia.c:6532 Replaces: [da10f563-8c35d41e-ad0761b1 at 10.3.0.26] send 711 bytes to udp/[10.3.0.26]:5060 at 17:34:20.693751: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bKb6a74a78C1F008DB From: "B" ;tag=7463F28A-2FFBD77D To: "A" ;tag=DeFrDpg9mcy3j Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 CSeq: 2 REFER Contact: Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:20.692943 [NOTICE] sofia.c:6615 Attended Transfer on originating session ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff 2012-03-08 18:34:20.692943 [DEBUG] switch_ivr.c:1711 (sofia/internal/10 at 10.3.0.4) State Change CS_EXECUTE -> CS_ROUTING 2012-03-08 18:34:20.692943 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:20.692943 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:20.692943 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/10 at 10.3.0.4 to inline[endless_playback:local_stream://moh,park at default] send 851 bytes to udp/[10.3.0.26]:5060 at 17:34:20.695326: ------------------------------------------------------------------------ NOTIFY sip:30 at 10.3.0.26 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKBreKpZ5BgSgSQ Max-Forwards: 70 From: "A " ;tag=DeFrDpg9mcy3j To: ;tag=7463F28A-2FFBD77D Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 CSeq: 25277961 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: refer;id=2 Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 16 SIP/2.0 200 OK ------------------------------------------------------------------------ send 645 bytes to udp/[10.3.0.26]:5060 at 17:34:20.696067: ------------------------------------------------------------------------ BYE sip:30 at 10.3.0.26 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKc17BrtpFD26BK Max-Forwards: 70 From: ;tag=F019gcjgFya9S To: "B" ;tag=4AB6AEB9-5DA56A24 Call-ID: da10f563-8c35d41e-ad0761b1 at 10.3.0.26 CSeq: 25277966 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="normal_clearing" Content-Length: 0 ------------------------------------------------------------------------ send 661 bytes to udp/[10.3.0.26]:5060 at 17:34:20.696525: ------------------------------------------------------------------------ SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK909bdcdc8659BA5F From: "B" ;tag=4AB6AEB9-5DA56A24 To: ;tag=F019gcjgFya9S Call-ID: da10f563-8c35d41e-ad0761b1 at 10.3.0.26 CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 428 bytes from udp/[10.3.0.26]:5060 at 17:34:20.703100: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKBreKpZ5BgSgSQ From: "A" ;tag=DeFrDpg9mcy3j To: "B" ;tag=7463F28A-2FFBD77D CSeq: 25277961 NOTIFY Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:20.692943 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] recv 440 bytes from udp/[10.3.0.26]:5060 at 17:34:20.705492: ------------------------------------------------------------------------ BYE sip:mod_sofia at 10.3.0.4:5060 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK415b2b56FCCA5AA9 From: "B" ;tag=7463F28A-2FFBD77D To: "A" ;tag=DeFrDpg9mcy3j CSeq: 3 BYE Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:20.692943 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] recv 744 bytes from udp/[10.3.0.26]:5060 at 17:34:20.709331: ------------------------------------------------------------------------ CANCEL sip:20 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK909bdcdc8659BA5F From: "B" ;tag=4AB6AEB9-5DA56A24 To: CSeq: 2 CANCEL Call-ID: da10f563-8c35d41e-ad0761b1 at 10.3.0.26 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Proxy-Authorization: Digest username="30", realm="10.3.0.4", nonce="ec7c3b52-6944-11e1-a7d4-a55c2ebde3ff", qop=auth, cnonce="XJz4HrL/160QNgt", nc=00000002, uri="sip:20 at 10.3.0.4:5060;user=phone", response="05c11ccdcec345f1fca83aa780770b28", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 274 bytes to udp/[10.3.0.26]:5060 at 17:34:20.709881: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK909bdcdc8659BA5F From: "B" ;tag=4AB6AEB9-5DA56A24 To: ;tag=F019gcjgFya9S Call-ID: da10f563-8c35d41e-ad0761b1 at 10.3.0.26 CSeq: 2 CANCEL Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:20.712938 [DEBUG] switch_ivr_play_say.c:1678 done playing file local_stream://moh 2012-03-08 18:34:20.712938 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/sip:30 at 10.3.0.26] 2012-03-08 18:34:20.712938 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:20.712938 [DEBUG] switch_channel.c:2848 (sofia/internal/sip:30 at 10.3.0.26) Callstate Change HELD -> HANGUP 2012-03-08 18:34:20.712938 [NOTICE] switch_ivr_bridge.c:669 Hangup sofia/internal/sip:30 at 10.3.0.26 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2012-03-08 18:34:20.712938 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/sip:30 at 10.3.0.26 [KILL] 2012-03-08 18:34:20.712938 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:20.712938 [DEBUG] switch_core_state_machine.c:420 (sofia/internal/sip:30 at 10.3.0.26) State EXCHANGE_MEDIA going to sleep 2012-03-08 18:34:20.712938 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:30 at 10.3.0.26) Running State Change CS_HANGUP 2012-03-08 18:34:20.712938 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:20.712938 [DEBUG] switch_ivr_bridge.c:329 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:20.712938 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:30 at 10.3.0.26) State HANGUP 2012-03-08 18:34:20.712938 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:30 at 10.3.0.26 hanging up, cause: NORMAL_CLEARING 2012-03-08 18:34:20.712938 [DEBUG] mod_sofia.c:513 Sending BYE to sofia/internal/sip:30 at 10.3.0.26 send 622 bytes to udp/[10.3.0.26]:5060 at 17:34:20.724016: ------------------------------------------------------------------------ BYE sip:30 at 10.3.0.26 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKDa14SN7jaBXye Max-Forwards: 70 From: "A " ;tag=DeFrDpg9mcy3j To: ;tag=7463F28A-2FFBD77D Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 CSeq: 25277962 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:20.712938 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:30 at 10.3.0.26 Standard HANGUP, cause: NORMAL_CLEARING 2012-03-08 18:34:20.712938 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:30 at 10.3.0.26) State HANGUP going to sleep 2012-03-08 18:34:20.712938 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/sip:30 at 10.3.0.26) State Change CS_HANGUP -> CS_REPORTING 2012-03-08 18:34:20.712938 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:20.712938 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:30 at 10.3.0.26) Running State Change CS_REPORTING 2012-03-08 18:34:20.712938 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:30 at 10.3.0.26) State REPORTING 2012-03-08 18:34:20.712938 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:30 at 10.3.0.26 Standard REPORTING, cause: NORMAL_CLEARING 2012-03-08 18:34:20.712938 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:30 at 10.3.0.26) State REPORTING going to sleep 2012-03-08 18:34:20.712938 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/sip:30 at 10.3.0.26) State Change CS_REPORTING -> CS_DESTROY 2012-03-08 18:34:20.712938 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:20.712938 [DEBUG] switch_core_session.c:1380 Session 10 (sofia/internal/sip:30 at 10.3.0.26) Locked, Waiting on external entities recv 411 bytes from udp/[10.3.0.26]:5060 at 17:34:20.729848: ------------------------------------------------------------------------ SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKc17BrtpFD26BK From: ;tag=F019gcjgFya9S To: "B" ;tag=4AB6AEB9-5DA56A24 CSeq: 25277966 BYE Call-ID: da10f563-8c35d41e-ad0761b1 at 10.3.0.26 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:20.712938 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/30 at 10.3.0.4 [BREAK] 2012-03-08 18:34:20.712938 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/30 at 10.3.0.4 [BREAK] 2012-03-08 18:34:20.712938 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/30 at 10.3.0.4 [BREAK] 2012-03-08 18:34:20.712938 [DEBUG] sofia.c:5526 Channel sofia/internal/30 at 10.3.0.4 entering state [terminated][481] 2012-03-08 18:34:20.712938 [DEBUG] switch_channel.c:2848 (sofia/internal/30 at 10.3.0.4) Callstate Change RINGING -> HANGUP 2012-03-08 18:34:20.712938 [NOTICE] sofia.c:6293 Hangup sofia/internal/30 at 10.3.0.4 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2012-03-08 18:34:20.712938 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/30 at 10.3.0.4 [KILL] 2012-03-08 18:34:20.712938 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/30 at 10.3.0.4 [BREAK] 2012-03-08 18:34:20.732954 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/10 at 10.3.0.4] 2012-03-08 18:34:20.732954 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-08 18:34:20.732954 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] recv 543 bytes from udp/[10.3.0.26]:5060 at 17:34:20.734163: ------------------------------------------------------------------------ ACK sip:20 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK909bdcdc8659BA5F From: "B" ;tag=4AB6AEB9-5DA56A24 To: ;tag=F019gcjgFya9S CSeq: 2 ACK Call-ID: da10f563-8c35d41e-ad0761b1 at 10.3.0.26 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:20.732954 [INFO] switch_cpp.cpp:1227 hangup hook for transfer!! EXECUTE sofia/internal/10 at 10.3.0.4 info() 2012-03-08 18:34:20.732954 [NOTICE] switch_core_session.c:1398 Session 10 (sofia/internal/sip:30 at 10.3.0.26) Ended 2012-03-08 18:34:20.732954 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/sip:30 at 10.3.0.26 [CS_DESTROY] 2012-03-08 18:34:20.732954 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/sip:30 at 10.3.0.26) Callstate Change HANGUP -> DOWN 2012-03-08 18:34:20.732954 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/sip:30 at 10.3.0.26) Running State Change CS_DESTROY 2012-03-08 18:34:20.732954 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:30 at 10.3.0.26) State DESTROY 2012-03-08 18:34:20.732954 [DEBUG] mod_sofia.c:374 sofia/internal/sip:30 at 10.3.0.26 SOFIA DESTROY 2012-03-08 18:34:20.732954 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:30 at 10.3.0.26 Standard DESTROY 2012-03-08 18:34:20.732954 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:30 at 10.3.0.26) State DESTROY going to sleep 2012-03-08 18:34:20.732954 [INFO] mod_dptools.c:1439 CHANNEL_DATA: Channel-State: [CS_ROUTING] Channel-Call-State: [ACTIVE] Channel-State-Number: [2] Channel-Name: [sofia/internal/10 at 10.3.0.4] Unique-ID: [e73f07fa-6944-11e1-a7c4-a55c2ebde3ff] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-HIT-Dialplan: [true] Channel-Presence-ID: [10 at 10.3.0.4] Channel-Call-UUID: [e73f07fa-6944-11e1-a7c4-a55c2ebde3ff] Answer-State: [answered] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Read-Codec-Bit-Rate: [64000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Channel-Write-Codec-Bit-Rate: [64000] Caller-Direction: [inbound] Caller-Username: [10] Caller-Dialplan: [inline] Caller-Caller-ID-Name: [A] Caller-Caller-ID-Number: [10] Caller-Callee-ID-Name: [Outbound Call] Caller-Callee-ID-Number: [30] Caller-Network-Addr: [10.3.0.22] Caller-ANI: [10] Caller-Destination-Number: [endless_playback:local_stream://moh,park] Caller-Unique-ID: [e73f07fa-6944-11e1-a7c4-a55c2ebde3ff] Caller-Source: [mod_sofia] Caller-Transfer-Source: [1331228060:f03b71c2-6944-11e1-a7e1-a55c2ebde3ff:bl_xfer:endless_playback:local_stream://moh,park/default/inline] Caller-Context: [default] Caller-RDNIS: [30] Caller-Channel-Name: [sofia/internal/10 at 10.3.0.4] Caller-Profile-Index: [2] Caller-Profile-Created-Time: [1331228060692943] Caller-Channel-Created-Time: [1331228045612916] Caller-Channel-Answered-Time: [1331228050212964] Caller-Channel-Progress-Time: [1331228049352961] 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] variable_direction: [inbound] variable_uuid: [e73f07fa-6944-11e1-a7c4-a55c2ebde3ff] variable_session_id: [9] variable_sip_local_network_addr: [10.3.0.4] variable_sip_network_ip: [10.3.0.22] variable_sip_network_port: [5060] variable_sip_received_ip: [10.3.0.22] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_authorized: [true] variable_sip_number_alias: [10] variable_sip_auth_username: [10] variable_sip_auth_realm: [10.3.0.4] variable_number_alias: [10] variable_user_name: [10] variable_domain_name: [10.3.0.4] variable_record_stereo: [true] variable_default_gateway: [fonira] variable_default_areacode: [01] variable_transfer_fallback_extension: [operator] variable_toll_allow: [local,domestic,international,vas] variable_accountcode: [10] variable_user_context: [default] variable_effective_caller_id_name: [A ] variable_effective_caller_id_number: [10] variable_outbound_caller_id_name: [A ] variable_outbound_caller_id_number: [1997156010] variable_callgroup: [intercept] variable_sip_from_user: [10] variable_sip_from_uri: [10 at 10.3.0.4] variable_sip_from_host: [10.3.0.4] variable_sip_from_user_stripped: [10] variable_sofia_profile_name: [internal] variable_sip_req_params: [user=phone] variable_sip_req_user: [30] variable_sip_req_port: [5060] variable_sip_req_uri: [30 at 10.3.0.4:5060] variable_sip_req_host: [10.3.0.4] variable_sip_to_params: [user=phone] variable_sip_to_user: [30] variable_sip_to_uri: [30 at 10.3.0.4] variable_sip_to_host: [10.3.0.4] variable_sip_contact_user: [10] variable_sip_contact_uri: [10 at 10.3.0.22] variable_sip_contact_host: [10.3.0.22] variable_channel_name: [sofia/internal/10 at 10.3.0.4] variable_sip_user_agent: [PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933] variable_sip_via_host: [10.3.0.22] variable_presence_id: [10 at 10.3.0.4] variable_switch_r_sdp: [v=0 o=- 1167627581 1167627581 IN IP4 10.3.0.22 s=Polycom IP Phone c=IN IP4 10.3.0.22 t=0 0 a=sendrecv m=audio 2262 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ] variable_sip_audio_recv_pt: [9] variable_sip_use_codec_name: [G722] variable_sip_use_codec_rate: [8000] variable_sip_use_codec_ptime: [20] variable_read_codec: [G722] variable_read_rate: [16000] variable_write_codec: [G722] variable_write_rate: [16000] variable_DP_MATCH: [ARRAY::30|:30] variable_call_uuid: [e73f07fa-6944-11e1-a7c4-a55c2ebde3ff] variable_dialed_extension: [30] variable_export_vars: [dialed_extension] variable_continue_on_fail: [true] variable_hangup_after_bridge: [true] variable_dialed_user: [30] variable_dialed_domain: [10.3.0.4] variable_sip_local_sdp_str: [v=0 o=FreeSWITCH 1331211630 1331211631 IN IP4 10.3.0.4 s=FreeSWITCH c=IN IP4 10.3.0.4 t=0 0 m=audio 16420 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ] variable_local_media_ip: [10.3.0.4] variable_local_media_port: [16420] variable_advertised_media_ip: [10.3.0.4] variable_sip_use_pt: [9] variable_rtp_use_ssrc: [2056504069] variable_sip_2833_send_payload: [127] variable_sip_2833_recv_payload: [127] variable_remote_media_ip: [10.3.0.22] variable_remote_media_port: [2262] variable_endpoint_disposition: [ANSWER] variable_originate_disposition: [SUCCESS] variable_DIALSTATUS: [SUCCESS] variable_last_bridge_to: [e96bae48-6944-11e1-a7cc-a55c2ebde3ff] variable_sip_to_tag: [c5NZBUZ5Q37gQ] variable_sip_from_tag: [B0C14C18-E7DC6BB1] variable_sip_cseq: [2] variable_sip_call_id: [c64e932c-3eca2965-da54ee6 at 10.3.0.22] variable_sip_full_via: [SIP/2.0/UDP 10.3.0.22;branch=z9hG4bKe8d47063D608BAD4] variable_sip_from_display: [A] variable_sip_full_from: ["A" ;tag=B0C14C18-E7DC6BB1] variable_sip_full_to: [;tag=c5NZBUZ5Q37gQ] variable_bridge_channel: [sofia/internal/sip:30 at 10.3.0.26] variable_bridge_uuid: [e96bae48-6944-11e1-a7cc-a55c2ebde3ff] variable_switch_m_sdp: [v=0 o=- 1167627586 1167627587 IN IP4 10.3.0.26 s=Polycom IP Phone c=IN IP4 0.0.0.0 t=0 0 m=audio 2234 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=sendonly m=video 0 RTP/AVP 98 99 100 34 31 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-2000/90000 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 ] variable_att_xfer_kill_uuid: [ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff] variable_max_forwards: [69] variable_transfer_history: [ARRAY::1331228060:f03b71c2-6944-11e1-a7e1-a55c2ebde3ff:bl_xfer:endless_playback:local_stream://moh,park/default/inline] variable_transfer_source: [1331228060:f03b71c2-6944-11e1-a7e1-a55c2ebde3ff:bl_xfer:endless_playback:local_stream://moh,park/default/inline] variable_playback_seconds: [16] variable_playback_ms: [16720] variable_playback_samples: [133760] variable_bridge_hangup_cause: [NORMAL_CLEARING] variable_current_application: [info] 2012-03-08 18:34:20.732954 [WARNING] switch_cpp.cpp:1227 TEST X for UUID e73f07fa-6944-11e1-a7c4-a55c2ebde3ff, src 10 dest 30 2012-03-08 18:34:20.732954 [WARNING] switch_cpp.cpp:1227 TEST: Originate Disposition after bridge is SUCCESS 2012-03-08 18:34:20.732954 [WARNING] switch_cpp.cpp:1227 TEST 4 for UUID e73f07fa-6944-11e1-a7c4-a55c2ebde3ff, src 10 dest 30 EXECUTE sofia/internal/10 at 10.3.0.4 sleep(1000) recv 394 bytes from udp/[10.3.0.26]:5060 at 17:34:20.741808: ------------------------------------------------------------------------ SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKDa14SN7jaBXye From: "A" ;tag=DeFrDpg9mcy3j To: ;tag=7463F28A-2FFBD77D CSeq: 25277962 BYE Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ send 270 bytes to udp/[10.3.0.26]:5060 at 17:34:20.745132: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK415b2b56FCCA5AA9 From: "B" ;tag=7463F28A-2FFBD77D To: "A" ;tag=DeFrDpg9mcy3j Call-ID: c0caa6b7-e3e7-122f-41a2-00900b1be504 CSeq: 3 BYE Content-Length: 0 ------------------------------------------------------------------------ send 1070 bytes to udp/[10.3.0.25]:5060 at 17:34:20.776577: ------------------------------------------------------------------------ NOTIFY sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKeKtXUgrp7KKHa Max-Forwards: 70 From: ;tag=zVD97bWqz9Ei To: "C" ;tag=E729C66F-499D331E Call-ID: 6e8428e-4012accd-98b6429c at 10.3.0.25 CSeq: 185332684 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3304 Content-Type: application/dialog-info+xml Content-Length: 258 terminated ------------------------------------------------------------------------ send 1082 bytes to udp/[10.3.0.27]:5060 at 17:34:20.780085: ------------------------------------------------------------------------ NOTIFY sip:31 at 10.3.0.27 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKFvKpXB9S4v93N Max-Forwards: 70 From: ;tag=j2FaQyd4KlIW To: "Patrick SXXXXXXl" ;tag=26B3115D-24F8F110 Call-ID: 11c82624-cdfad25f-e9f5882 at 10.3.0.27 CSeq: 185332685 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3304 Content-Type: application/dialog-info+xml Content-Length: 258 terminated ------------------------------------------------------------------------ recv 405 bytes from udp/[10.3.0.25]:5060 at 17:34:20.783041: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKeKtXUgrp7KKHa From: ;tag=zVD97bWqz9Ei To: "C" ;tag=E729C66F-499D331E CSeq: 185332684 NOTIFY Call-ID: 6e8428e-4012accd-98b6429c at 10.3.0.25 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ send 1072 bytes to udp/[10.3.0.22]:5060 at 17:34:20.783873: ------------------------------------------------------------------------ NOTIFY sip:10 at 10.3.0.22 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKg5cFZ6SX15ZpH Max-Forwards: 70 From: ;tag=BL1yX2TI1X2c To: "A" ;tag=7B5DFEAE-48CDC3F7 Call-ID: ef7cb8bd-dc381cfe-bf64ccc7 at 10.3.0.22 CSeq: 185332686 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3304 Content-Type: application/dialog-info+xml Content-Length: 258 terminated ------------------------------------------------------------------------ recv 417 bytes from udp/[10.3.0.27]:5060 at 17:34:20.787412: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKFvKpXB9S4v93N From: ;tag=j2FaQyd4KlIW To: "Patrick SXXXXXXl" ;tag=26B3115D-24F8F110 CSeq: 185332685 NOTIFY Call-ID: 11c82624-cdfad25f-e9f5882 at 10.3.0.27 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 407 bytes from udp/[10.3.0.22]:5060 at 17:34:20.790635: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKg5cFZ6SX15ZpH From: ;tag=BL1yX2TI1X2c To: "A" ;tag=7B5DFEAE-48CDC3F7 CSeq: 185332686 NOTIFY Call-ID: ef7cb8bd-dc381cfe-bf64ccc7 at 10.3.0.22 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:21.732935 [WARNING] switch_cpp.cpp:1227 TEST 5 for UUID e73f07fa-6944-11e1-a7c4-a55c2ebde3ff, src 10 dest 30 EXECUTE sofia/internal/10 at 10.3.0.4 sleep(1000) 2012-03-08 18:34:22.732931 [WARNING] switch_cpp.cpp:1227 TEST 6 for UUID e73f07fa-6944-11e1-a7c4-a55c2ebde3ff, src 10 dest 30 EXECUTE sofia/internal/10 at 10.3.0.4 sleep(1000) 2012-03-08 18:34:23.732934 [DEBUG] mod_python.c:284 Finished calling python script 2012-03-08 18:34:23.732934 [DEBUG] switch_cpp.cpp:1007 sofia/internal/10 at 10.3.0.4 destroy/unlink session from object 2012-03-08 18:34:23.732934 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/10 at 10.3.0.4) State EXECUTE going to sleep 2012-03-08 18:34:23.732934 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_ROUTING 2012-03-08 18:34:23.732934 [DEBUG] switch_channel.c:1886 (sofia/internal/10 at 10.3.0.4) Callstate Change ACTIVE -> RINGING 2012-03-08 18:34:23.753118 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/10 at 10.3.0.4) State ROUTING 2012-03-08 18:34:23.753118 [DEBUG] mod_sofia.c:148 sofia/internal/10 at 10.3.0.4 SOFIA ROUTING 2012-03-08 18:34:23.753118 [DEBUG] switch_core_state_machine.c:104 sofia/internal/10 at 10.3.0.4 Standard ROUTING 2012-03-08 18:34:23.753118 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/10 at 10.3.0.4) State Change CS_ROUTING -> CS_EXECUTE 2012-03-08 18:34:23.753118 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:23.753118 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/10 at 10.3.0.4) State ROUTING going to sleep 2012-03-08 18:34:23.753118 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_EXECUTE 2012-03-08 18:34:23.753118 [DEBUG] switch_channel.c:1888 (sofia/internal/10 at 10.3.0.4) Callstate Change RINGING -> ACTIVE 2012-03-08 18:34:23.753118 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/10 at 10.3.0.4) State EXECUTE 2012-03-08 18:34:23.753118 [DEBUG] mod_sofia.c:241 sofia/internal/10 at 10.3.0.4 SOFIA EXECUTE 2012-03-08 18:34:23.753118 [DEBUG] switch_core_state_machine.c:192 sofia/internal/10 at 10.3.0.4 Standard EXECUTE EXECUTE sofia/internal/10 at 10.3.0.4 endless_playback(local_stream://moh) 2012-03-08 18:34:23.753118 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/16000] 16000hz 2012-03-08 18:34:23.753118 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16 at 16000hz 1 channels 20ms send 1402 bytes to udp/[10.3.0.25]:5060 at 17:34:23.860078: ------------------------------------------------------------------------ NOTIFY sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKHe6701a1yep9c Max-Forwards: 70 From: ;tag=JElIsHAUjri7 To: "C" ;tag=432C4A25-2C483674 Call-ID: a941c408-2c54f3a7-82c5a8d6 at 10.3.0.25 CSeq: 185332687 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=1138 Content-Type: application/dialog-info+xml Content-Length: 589 confirmed sip:10 at 10.3.0.4 sip:endless_playback:local_stream://moh,park at 10.3.0.4 ------------------------------------------------------------------------ send 1410 bytes to udp/[10.3.0.26]:5060 at 17:34:23.862933: ------------------------------------------------------------------------ NOTIFY sip:30 at 10.3.0.26 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKjQZ02vU4UQcvr Max-Forwards: 70 From: ;tag=hzYaAHIZscZq To: "B" ;tag=A52D4836-AD38FE89 Call-ID: 3f77033d-f3b24ec8-f0479fab at 10.3.0.26 CSeq: 185332688 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=1138 Content-Type: application/dialog-info+xml Content-Length: 589 confirmed sip:10 at 10.3.0.4 sip:endless_playback:local_stream://moh,park at 10.3.0.4 ------------------------------------------------------------------------ recv 406 bytes from udp/[10.3.0.25]:5060 at 17:34:23.867468: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKHe6701a1yep9c From: ;tag=JElIsHAUjri7 To: "C" ;tag=432C4A25-2C483674 CSeq: 185332687 NOTIFY Call-ID: a941c408-2c54f3a7-82c5a8d6 at 10.3.0.25 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 414 bytes from udp/[10.3.0.26]:5060 at 17:34:23.870281: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKjQZ02vU4UQcvr From: ;tag=hzYaAHIZscZq To: "B" ;tag=A52D4836-AD38FE89 CSeq: 185332688 NOTIFY Call-ID: 3f77033d-f3b24ec8-f0479fab at 10.3.0.26 Contact: Event: dialog User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:32.012932 [DEBUG] switch_channel.c:2598 (sofia/internal/10 at 10.3.0.4) State Change CS_EXECUTE -> CS_RESET 2012-03-08 18:34:32.012932 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:32.032936 [DEBUG] switch_ivr_play_say.c:1678 done playing file local_stream://moh 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/10 at 10.3.0.4) State EXECUTE going to sleep 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_RESET 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:413 (sofia/internal/10 at 10.3.0.4) State RESET 2012-03-08 18:34:32.032936 [DEBUG] switch_channel.c:2600 (sofia/internal/10 at 10.3.0.4) State Change CS_RESET -> CS_EXECUTE 2012-03-08 18:34:32.032936 [DEBUG] mod_sofia.c:166 sofia/internal/10 at 10.3.0.4 SOFIA RESET 2012-03-08 18:34:32.032936 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:32.032936 [DEBUG] switch_channel.c:2848 (sofia/internal/sip:20 at 10.3.0.25) Callstate Change RINGING -> HANGUP 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:413 (sofia/internal/10 at 10.3.0.4) State RESET going to sleep 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_EXECUTE 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/10 at 10.3.0.4) State EXECUTE 2012-03-08 18:34:32.032936 [DEBUG] mod_sofia.c:241 sofia/internal/10 at 10.3.0.4 SOFIA EXECUTE 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:192 sofia/internal/10 at 10.3.0.4 Standard EXECUTE 2012-03-08 18:34:32.032936 [NOTICE] switch_ivr_originate.c:3075 Hangup sofia/internal/sip:20 at 10.3.0.25 [CS_CONSUME_MEDIA] [ATTENDED_TRANSFER] 2012-03-08 18:34:32.032936 [NOTICE] switch_core_state_machine.c:226 sofia/internal/10 at 10.3.0.4 has executed the last dialplan instruction, hanging up. 2012-03-08 18:34:32.032936 [DEBUG] switch_channel.c:2848 (sofia/internal/10 at 10.3.0.4) Callstate Change ACTIVE -> HANGUP 2012-03-08 18:34:32.032936 [NOTICE] switch_core_state_machine.c:228 Hangup sofia/internal/10 at 10.3.0.4 [CS_EXECUTE] [NORMAL_CLEARING] 2012-03-08 18:34:32.032936 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/sip:20 at 10.3.0.25 [KILL] 2012-03-08 18:34:32.032936 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-08 18:34:32.032936 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [ATTENDED_TRANSFER] 2012-03-08 18:34:32.032936 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 601 [ATTENDED_TRANSFER] 2012-03-08 18:34:32.032936 [INFO] mod_dptools.c:2922 Originate Failed. Cause: ATTENDED_TRANSFER 2012-03-08 18:34:32.032936 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/10 at 10.3.0.4 [KILL] 2012-03-08 18:34:32.032936 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/10 at 10.3.0.4) State EXECUTE going to sleep 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_HANGUP 2012-03-08 18:34:32.032936 [DEBUG] switch_core_session.c:2285 sofia/internal/30 at 10.3.0.4 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-03-08 18:34:32.032936 [INFO] switch_cpp.cpp:1227 hangup hook for hangup!! 2012-03-08 18:34:32.032936 [DEBUG] switch_core_session.c:2095 sofia/internal/30 at 10.3.0.4 ZOMBIE EXEC info() EXECUTE sofia/internal/30 at 10.3.0.4 info() 2012-03-08 18:34:32.032936 [DEBUG] switch_core_session.c:2271 sofia/internal/30 at 10.3.0.4 skip receive message [APPLICATION_EXEC] (channel is hungup already) 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/10 at 10.3.0.4) State HANGUP 2012-03-08 18:34:32.032936 [DEBUG] mod_sofia.c:469 Channel sofia/internal/10 at 10.3.0.4 hanging up, cause: NORMAL_CLEARING 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:20 at 10.3.0.25) Running State Change CS_HANGUP 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:20 at 10.3.0.25) State HANGUP 2012-03-08 18:34:32.032936 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:20 at 10.3.0.25 hanging up, cause: ATTENDED_TRANSFER 2012-03-08 18:34:32.032936 [INFO] mod_dptools.c:1439 CHANNEL_DATA: Channel-State: [CS_HANGUP] Channel-Call-State: [HANGUP] Channel-State-Number: [10] Channel-Name: [sofia/internal/30 at 10.3.0.4] Unique-ID: [ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-HIT-Dialplan: [true] Channel-Presence-ID: [30 at 10.3.0.4] Channel-Call-UUID: [ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff] Answer-State: [hangup] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Read-Codec-Bit-Rate: [64000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Channel-Write-Codec-Bit-Rate: [64000] Caller-Direction: [inbound] Caller-Username: [30] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [B] Caller-Caller-ID-Number: [30] Caller-Callee-ID-Name: [Outbound Call] Caller-Callee-ID-Number: [20] Caller-Network-Addr: [10.3.0.26] Caller-ANI: [30] Caller-Destination-Number: [20] Caller-Unique-ID: [ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/30 at 10.3.0.4] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1331228054432937] Caller-Channel-Created-Time: [1331228054432937] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [1331228058052945] 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] variable_direction: [inbound] variable_uuid: [ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff] variable_session_id: [11] variable_sip_local_network_addr: [10.3.0.4] variable_sip_network_ip: [10.3.0.26] variable_sip_network_port: [5060] variable_sip_received_ip: [10.3.0.26] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_authorized: [true] variable_sip_number_alias: [30] variable_sip_auth_username: [30] variable_sip_auth_realm: [10.3.0.4] variable_number_alias: [30] variable_user_name: [30] variable_domain_name: [10.3.0.4] variable_record_stereo: [true] variable_default_gateway: [fonira] variable_default_areacode: [01] variable_transfer_fallback_extension: [operator] variable_toll_allow: [local,domestic,international,vas] variable_accountcode: [30] variable_user_context: [default] variable_effective_caller_id_name: [B] variable_effective_caller_id_number: [30] variable_outbound_caller_id_name: [B] variable_outbound_caller_id_number: [1997156030] variable_callgroup: [intercept] variable_sip_from_user: [30] variable_sip_from_uri: [30 at 10.3.0.4] variable_sip_from_host: [10.3.0.4] variable_sip_from_user_stripped: [30] variable_sip_from_tag: [4AB6AEB9-5DA56A24] variable_sofia_profile_name: [internal] variable_sip_full_via: [SIP/2.0/UDP 10.3.0.26;branch=z9hG4bK909bdcdc8659BA5F] variable_sip_from_display: [B] variable_sip_full_from: ["B" ;tag=4AB6AEB9-5DA56A24] variable_sip_full_to: [] variable_sip_req_params: [user=phone] variable_sip_req_user: [20] variable_sip_req_port: [5060] variable_sip_req_uri: [20 at 10.3.0.4:5060] variable_sip_req_host: [10.3.0.4] variable_sip_to_params: [user=phone] variable_sip_to_user: [20] variable_sip_to_uri: [20 at 10.3.0.4] variable_sip_to_host: [10.3.0.4] variable_sip_contact_user: [30] variable_sip_contact_uri: [30 at 10.3.0.26] variable_sip_contact_host: [10.3.0.26] variable_channel_name: [sofia/internal/30 at 10.3.0.4] variable_sip_call_id: [da10f563-8c35d41e-ad0761b1 at 10.3.0.26] variable_sip_user_agent: [PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933] variable_sip_via_host: [10.3.0.26] variable_max_forwards: [70] variable_presence_id: [30 at 10.3.0.4] variable_switch_r_sdp: [v=0 o=- 1167627590 1167627590 IN IP4 10.3.0.26 s=Polycom IP Phone c=IN IP4 10.3.0.26 t=0 0 a=sendrecv m=audio 2236 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ] variable_remote_media_ip: [10.3.0.26] variable_remote_media_port: [2236] variable_sip_audio_recv_pt: [9] variable_sip_use_codec_name: [G722] variable_sip_use_codec_rate: [8000] variable_sip_use_codec_ptime: [20] variable_read_codec: [G722] variable_read_rate: [16000] variable_write_codec: [G722] variable_write_rate: [16000] variable_DP_MATCH: [ARRAY::20|:20] variable_call_uuid: [ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff] variable_dialed_extension: [20] variable_export_vars: [dialed_extension] variable_continue_on_fail: [true] variable_hangup_after_bridge: [true] variable_dialed_user: [20] variable_dialed_domain: [10.3.0.4] variable_endpoint_disposition: [ATTENDED_TRANSFER] variable_sip_term_status: [481] variable_proto_specific_hangup_cause: [sip:481] variable_sip_term_cause: [41] variable_originate_disposition: [ATTENDED_TRANSFER] variable_DIALSTATUS: [ATTENDED_TRANSFER] variable_current_application: [info] 2012-03-08 18:34:32.032936 [DEBUG] switch_core_session.c:2285 sofia/internal/30 at 10.3.0.4 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-03-08 18:34:32.032936 [WARNING] switch_cpp.cpp:1227 TEST X for UUID ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff, src 30 dest 20 2012-03-08 18:34:32.032936 [WARNING] switch_cpp.cpp:1227 TEST: Originate Disposition after bridge is ATTENDED_TRANSFER 2012-03-08 18:34:32.032936 [WARNING] switch_cpp.cpp:1227 TEST 4 for UUID ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff, src 30 dest 20 2012-03-08 18:34:32.032936 [DEBUG] switch_core_session.c:2100 sofia/internal/30 at 10.3.0.4 Channel is hungup and application 'sleep' does not have the zombie_exec flag. 2012-03-08 18:34:32.032936 [WARNING] switch_cpp.cpp:1227 TEST 5 for UUID ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff, src 30 dest 20 2012-03-08 18:34:32.032936 [DEBUG] switch_core_session.c:2100 sofia/internal/30 at 10.3.0.4 Channel is hungup and application 'sleep' does not have the zombie_exec flag. 2012-03-08 18:34:32.032936 [WARNING] switch_cpp.cpp:1227 TEST 6 for UUID ec80d1f8-6944-11e1-a7d5-a55c2ebde3ff, src 30 dest 20 2012-03-08 18:34:32.032936 [DEBUG] switch_core_session.c:2100 sofia/internal/30 at 10.3.0.4 Channel is hungup and application 'sleep' does not have the zombie_exec flag. 2012-03-08 18:34:32.032936 [DEBUG] mod_python.c:284 Finished calling python script 2012-03-08 18:34:32.032936 [DEBUG] switch_cpp.cpp:1007 sofia/internal/30 at 10.3.0.4 destroy/unlink session from object 2012-03-08 18:34:32.032936 [DEBUG] switch_core_session.c:2285 sofia/internal/30 at 10.3.0.4 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/30 at 10.3.0.4) State EXECUTE going to sleep 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/30 at 10.3.0.4) Running State Change CS_HANGUP 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/30 at 10.3.0.4) State HANGUP 2012-03-08 18:34:32.032936 [DEBUG] mod_sofia.c:469 Channel sofia/internal/30 at 10.3.0.4 hanging up, cause: NORMAL_TEMPORARY_FAILURE 2012-03-08 18:34:32.032936 [DEBUG] mod_sofia.c:513 Sending BYE to sofia/internal/10 at 10.3.0.4 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:47 sofia/internal/10 at 10.3.0.4 Standard HANGUP, cause: NORMAL_CLEARING 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/10 at 10.3.0.4) State HANGUP going to sleep 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/10 at 10.3.0.4) State Change CS_HANGUP -> CS_REPORTING send 637 bytes to udp/[10.3.0.22]:5060 at 17:34:32.048986: ------------------------------------------------------------------------ BYE sip:10 at 10.3.0.22 SIP/2.02012-03-08 18:34:32.032936 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKK0rS4Qc8r02em Max-Forwards: 70 From: ;tag=c5NZBUZ5Q37gQ To: "A" ;tag=B0C14C18-E7DC6BB12012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_REPORTING Call-ID: c64e932c-3eca2965-da54ee6 at 10.3.0.22 CSeq: 25277972 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fbe4e64 2012-02-22 23-08-19 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/10 at 10.3.0.4) State REPORTING 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:79 sofia/internal/10 at 10.3.0.4 Standard REPORTING, cause: NORMAL_CLEARING 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/10 at 10.3.0.4) State REPORTING going to sleep 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/10 at 10.3.0.4) State Change CS_REPORTING -> CS_DESTROY 2012-03-08 18:34:32.032936 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-08 18:34:32.032936 [DEBUG] switch_core_session.c:1380 Session 9 (sofia/internal/10 at 10.3.0.4) Locked, Waiting on external entities 2012-03-08 18:34:32.032936 [NOTICE] switch_core_session.c:1398 Session 9 (sofia/internal/10 at 10.3.0.4) Ended 2012-03-08 18:34:32.032936 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/10 at 10.3.0.4 [CS_DESTROY] 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/10 at 10.3.0.4) Callstate Change HANGUP -> DOWN 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/10 at 10.3.0.4) Running State Change CS_DESTROY 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/10 at 10.3.0.4) State DESTROY 2012-03-08 18:34:32.032936 [DEBUG] mod_sofia.c:374 sofia/internal/10 at 10.3.0.4 SOFIA DESTROY 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:86 sofia/internal/10 at 10.3.0.4 Standard DESTROY 2012-03-08 18:34:32.032936 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/10 at 10.3.0.4) State DESTROY going to sleep recv 399 bytes from udp/[10.3.0.22]:5060 at 17:34:32.055973: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKK0rS4Qc8r02em From: ;tag=c5NZBUZ5Q37gQ To: "A" ;tag=B0C14C18-E7DC6BB1 CSeq: 25277972 BYE Call-ID: c64e932c-3eca2965-da54ee6 at 10.3.0.22 Contact: 2012-03-08 18:34:32.052952 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/sip:20 at 10.3.0.25 User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:20 at 10.3.0.25 Standard HANGUP, cause: ATTENDED_TRANSFER 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:20 at 10.3.0.25) State HANGUP going to sleep 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/sip:20 at 10.3.0.25) State Change CS_HANGUP -> CS_REPORTING 2012-03-08 18:34:32.052952 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:20 at 10.3.0.25) Running State Change CS_REPORTING 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:20 at 10.3.0.25) State REPORTING 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:20 at 10.3.0.25 Standard REPORTING, cause: ATTENDED_TRANSFER 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:20 at 10.3.0.25) State REPORTING going to sleep send 341 bytes to udp/[10.3.0.25]:5060 at 17:34:32.056855: ------------------------------------------------------------------------ CANCEL sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK8v28geK1ryD1m Max-Forwards: 69 From: "B" ;tag=g9t2j72Kc70UN To: Call-ID: c5fdcf90-e3e7-122f-41a2-00900b1be504 CSeq: 25277964 CANCEL Reason: FreeSWITCH;cause=601;text="ATTENDED_TRANSFER" Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/sip:20 at 10.3.0.25) State Change CS_REPORTING -> CS_DESTROY 2012-03-08 18:34:32.052952 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-08 18:34:32.052952 [DEBUG] switch_core_session.c:1380 Session 12 (sofia/internal/sip:20 at 10.3.0.25) Locked, Waiting on external entities 2012-03-08 18:34:32.052952 [NOTICE] switch_core_session.c:1398 Session 12 (sofia/internal/sip:20 at 10.3.0.25) Ended 2012-03-08 18:34:32.052952 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/sip:20 at 10.3.0.25 [CS_DESTROY] 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/sip:20 at 10.3.0.25) Callstate Change HANGUP -> DOWN 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/sip:20 at 10.3.0.25) Running State Change CS_DESTROY 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:20 at 10.3.0.25) State DESTROY 2012-03-08 18:34:32.052952 [DEBUG] mod_sofia.c:374 sofia/internal/sip:20 at 10.3.0.25 SOFIA DESTROY 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:20 at 10.3.0.25 Standard DESTROY 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:20 at 10.3.0.25) State DESTROY going to sleep recv 386 bytes from udp/[10.3.0.25]:5060 at 17:34:32.061492: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK8v28geK1ryD1m From: "B" ;tag=g9t2j72Kc70UN To: "C" CSeq: 25277964 CANCEL Call-ID: c5fdcf90-e3e7-122f-41a2-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:47 sofia/internal/30 at 10.3.0.4 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/30 at 10.3.0.4) State HANGUP going to sleep 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/30 at 10.3.0.4) State Change CS_HANGUP -> CS_REPORTING 2012-03-08 18:34:32.052952 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/30 at 10.3.0.4 [BREAK] 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/30 at 10.3.0.4) Running State Change CS_REPORTING 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/30 at 10.3.0.4) State REPORTING 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:79 sofia/internal/30 at 10.3.0.4 Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/30 at 10.3.0.4) State REPORTING going to sleep 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/30 at 10.3.0.4) State Change CS_REPORTING -> CS_DESTROY 2012-03-08 18:34:32.052952 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/30 at 10.3.0.4 [BREAK] 2012-03-08 18:34:32.052952 [DEBUG] switch_core_session.c:1380 Session 11 (sofia/internal/30 at 10.3.0.4) Locked, Waiting on external entities 2012-03-08 18:34:32.052952 [NOTICE] switch_core_session.c:1398 Session 11 (sofia/internal/30 at 10.3.0.4) Ended 2012-03-08 18:34:32.052952 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/30 at 10.3.0.4 [CS_DESTROY] 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/30 at 10.3.0.4) Callstate Change HANGUP -> DOWN 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/30 at 10.3.0.4) Running State Change CS_DESTROY 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/30 at 10.3.0.4) State DESTROY 2012-03-08 18:34:32.052952 [DEBUG] mod_sofia.c:374 sofia/internal/30 at 10.3.0.4 SOFIA DESTROY 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:86 sofia/internal/30 at 10.3.0.4 Standard DESTROY 2012-03-08 18:34:32.052952 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/30 at 10.3.0.4) State DESTROY going to sleep -------------- next part -------------- import os import re import sys from freeswitch import * import logging,sys _REVISION_ = "$Revision: 1891 $" # HANGUP HOOK # # session is a session object # what is "hangup" or "transfer" # if you pass an extra arg to setInputCallback then append 'arg' to get that value # def hangup_hook(session, what, arg): def hangup_hook(session, what): consoleLog("INFO", "hangup hook for %s!!\n\n" % what) return # INPUT CALLBACK # # session is a session object # what is "dtmf" or "event" # obj is a dtmf object or an event object depending on the 'what' var. # if you pass an extra arg to setInputCallback then append 'arg' to get that value # def input_callback(session, what, obj, arg): def input_callback(session, what, obj): if (what == "dtmf"): consoleLog("INFO", what + " " + obj.digit + "\n") else: consoleLog("INFO", what + " " + obj.serialize() + "\n") return "pause" # APPLICATION # # default name for apps is "handler" it can be overridden with :: # session is a session object # args is all the args passed after the module name def handler(session, args): # try: session.setHangupHook(hangup_hook) session.setInputCallback(input_callback) destination_extension = int(session.getVariable("dialed_extension")) caller_number = session.getVariable("caller_id_number") domain = session.getVariable("domain_name") consoleLog("INFO", "TEST: Destination-extension "+str(destination_extension)+"\n") session.execute("set","continue_on_fail=true") session.execute("set","hangup_after_bridge=true") uuid = session.getVariable("uuid") consoleLog("WARNING", "TEST 1 for UUID "+str(uuid)+", src "+str(caller_number)+" dest "+str(destination_extension)+"\n") session.execute("sleep","1000") consoleLog("WARNING", "TEST 2 for UUID "+str(uuid)+", src "+str(caller_number)+" dest "+str(destination_extension)+"\n") session.execute("sleep","1000") consoleLog("WARNING", "TEST 3 for UUID "+str(uuid)+", src "+str(caller_number)+" dest "+str(destination_extension)+"\n") session.execute("sleep","1000") session.execute('info') session.execute("bridge","[leg_timeout=%s][origination_caller_id_number=%s]user/%s@${domain_name}" % ('15',str(caller_number), str(destination_extension))) session.execute('info') originate_disposition = session.getVariable("originate_disposition") consoleLog("WARNING", "TEST X for UUID "+str(uuid)+", src "+str(caller_number)+" dest "+str(destination_extension)+"\n") consoleLog("WARNING", "TEST: Originate Disposition after bridge is "+str(originate_disposition)+"\n") #if originate_disposition == "PICKED_OFF" or originate_disposition == "ORIGINATOR_CANCEL" or\ # originate_disposition == "SUCCESS": #or originate_disposition == 'ATTENDED_TRANSFER' or \ # #originate_disposition == 'BLIND_TRANSFER': # consoleLog("WARNING", "Sending call to park\n") # session.execute("park") #else: consoleLog("WARNING", "TEST 4 for UUID "+str(uuid)+", src "+str(caller_number)+" dest "+str(destination_extension)+"\n") session.execute("sleep","1000") consoleLog("WARNING", "TEST 5 for UUID "+str(uuid)+", src "+str(caller_number)+" dest "+str(destination_extension)+"\n") session.execute("sleep","1000") consoleLog("WARNING", "TEST 6 for UUID "+str(uuid)+", src "+str(caller_number)+" dest "+str(destination_extension)+"\n") session.execute("sleep","1000") # FSAPI CALL FROM CLI, DP HTTP etc # # default name for python FSAPI is "fsapi" it can be overridden with :: # stream is a switch_stream, anything written with stream.write() is returned to the caller # env is a switch_event # args is all the args passed after the module name # session is a session object when called from the dial plan or the string "na" when not. def fsapi(session, stream, env, args): stream.write("w00t!\n" + env.serialize()) # RUN A FUNCTION IN A THREAD # # default name for pyrun is "runtime" it can be overridden with :: # args is all the args passed after the module name def runtime(args): print args + "\n" From basit.engg at gmail.com Thu Mar 8 21:56:05 2012 From: basit.engg at gmail.com (Abdul Basit) Date: Thu, 8 Mar 2012 23:56:05 +0500 Subject: [Freeswitch-users] sipt-t support in freeswitch Message-ID: FS fellows, Does FreeSWITCH support sip-t protocol? http://www.dialogic.com/webhelp/BorderNet2020/1.1.0/WebHelp/sip_sipt_ov.htm can anyone point me in right direction? -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/e2818c20/attachment.html From basit.engg at gmail.com Thu Mar 8 22:05:29 2012 From: basit.engg at gmail.com (Abdul Basit) Date: Fri, 9 Mar 2012 00:05:29 +0500 Subject: [Freeswitch-users] sipt-t support in freeswitch In-Reply-To: References: Message-ID: for reference what i m looking for is SIP for Telephones (SIP-T): Context and Architecture http://tools.ietf.org/html/draft-ietf-sipping-sipt-04 On Thu, Mar 8, 2012 at 11:56 PM, Abdul Basit wrote: > FS fellows, > > Does FreeSWITCH support sip-t protocol? > > http://www.dialogic.com/webhelp/BorderNet2020/1.1.0/WebHelp/sip_sipt_ov.htm > > can anyone point me in right direction? > > -- > Regards, > > Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 > -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/99526d44/attachment.html From anita.hall at simmortel.com Thu Mar 8 22:10:39 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Fri, 9 Mar 2012 00:40:39 +0530 Subject: [Freeswitch-users] Sangoma Card Originate Not Working on Forwarded Call Numbers In-Reply-To: References: <4F550583.4060105@earthspike.net> Message-ID: Hi Stephen Thanks a TON, pun unintended :) I tried setting the TON to 2 or 4 and NPI to 1 and on Q.931 trace it shows that the numbering type and plan have been set correctly. However, it still show the same UNALLOCATED_NUMBER hangup cause. I have attached the pcap trace of an incoming call which then sets the channel variables as given on the link you shared and bridges a call to the said number 07305880719 . The reason why I have to do this is because I could not figure out how to set these profile variables from the originate command, so I set them up in the dialplan. This number 07305880719 is valid, I am able to call it from my mobile and also from another card + freeswitch I suspected that the issue could be with sangoma isdn library, so I tried with libpri instead. Even with correct TON and NPI, the call is not being set-up. Could this be an issue with my PRI line or Sangoma card itself ? regards, Anita On Tue, Mar 6, 2012 at 1:58 AM, Stephen Wilde wrote: > In your log I see an "UNALLOCATED_NUMBER" release cause. > If the number is ok, this can be due to the ISDN numbering plan/number > type of the called party. > What are required by your isdn provider? > You can try with different parameters (for your called number I think that > you have to specify numbering plan=ISDN and number type=National). > These parameter can be specified in profile or dialplan. > > http://wiki.sangoma.com/wanpipe-freeswitch-config-appendix#ton-npi-per-call > > Stephen > > > On Mon, Mar 5, 2012 at 7:55 PM, Anita Hall wrote: > >> Thanks John. >> >> I could take a pcap trace using >> wanpipemon -i w4g1 -pcap -pcap_file sng_isdn_fwd.pc -prot ISDN -full >> -systime -c trd >> >> The q931 for a call which is not maturing is as follows. This is the case >> when the called number is set to unconditional forwarding. >> 3 6.057463 Local User Remote Network Q.931 SETUP >> 5 6.142822 Remote Network Local User Q.931 CALL PROCEEDING >> 7 Remote Network Local User Q.931 DISCONNECT >> 9 Local User Remote Network Q.931 RELEASE >> 11 Remote Network Local User Q.931 RELEASE COMPLETE >> >> This happens only with Sangoma Card. I am using the sangoma_isdn stack >> and FreeTDM. Somehow, it is not able to understand that the call is >> getting forwarded. >> >> The q931 for a call which is maturing is as follows >> 11 Local User Remote Network Q.931 SETUP >> 13 Remote Network Local User Q.931 CALL PROCEEDING >> 15 Remote Network Local User Q.931 ALERTING >> 17 Remote Network Local User Q.931 CONNECT >> 23 Remote Network Local User Q.931 DISCONNECT >> 25 Local User Remote Network Q.931 RELEASE >> 27 Remote Network Local User Q.931 RELEASE COMPLETE >> >> >> >> regards, >> Anita >> >> >> >> >> On Mon, Mar 5, 2012 at 11:57 PM, John wrote: >> >>> On 05/03/12 16:31, Anita Hall wrote: >>> > Hi >>> > >>> > There are some numbers on which I am not able to originate calls from >>> > a Sangoma Card, while I am able to do the same from a Digium like >>> > card. I am using FreeTDM. >>> > >>> > >originate freetdm/4/a/07305880672 &park() >>> > >>> > Log is here http://pastebin.freeswitch.org/18578 >>> > >>> > Note that I am able to call other numbers from this system and I have >>> > a working PRI line and FreeTDM. >>> > >>> > Also the number 07305880672 is working fine. It unconditionally >>> > forwards the call to another number. I am problem with calling all >>> > such numbers. >>> > >>> > I am able to do q931 traces using sangoma_isdn but how to view them ? >>> > >>> > regards, >>> > Anita >>> > >>> Anita, >>> >>> You can use wireshark to decode the q.931 traces; the output files are >>> .pcap format. >>> >>> John >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/0985bfa6/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: ton2npi1.pcap Type: application/cap Size: 894 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/0985bfa6/attachment.bin From kris at kriskinc.com Thu Mar 8 23:27:59 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 8 Mar 2012 15:27:59 -0500 Subject: [Freeswitch-users] sipt-t support in freeswitch In-Reply-To: References: Message-ID: SIP-T requires multipart body support which FreeSWITCH does have. FreeSWITCH can pass multipart bodies from one endpoint to the next (bridge). However, at this point in time FreeSWITCH cannot parse or create the encapsulated SIP-T bodies. On Thu, Mar 8, 2012 at 2:05 PM, Abdul Basit wrote: > for reference what i m looking for is > > SIP for Telephones (SIP-T): Context and Architecture > > http://tools.ietf.org/html/draft-ietf-sipping-sipt-04 > > > > > On Thu, Mar 8, 2012 at 11:56 PM, Abdul Basit wrote: >> >> FS fellows, >> >> Does FreeSWITCH support sip-t protocol? >> >> >> http://www.dialogic.com/webhelp/BorderNet2020/1.1.0/WebHelp/sip_sipt_ov.htm >> >> can anyone point me in right direction? >> >> -- >> Regards, >> >> Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 > > > > > -- > Regards, > > Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From wstephen80 at gmail.com Fri Mar 9 00:01:30 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Thu, 8 Mar 2012 22:01:30 +0100 Subject: [Freeswitch-users] Sangoma Card Originate Not Working on Forwarded Call Numbers In-Reply-To: References: <4F550583.4060105@earthspike.net> Message-ID: Hi, in your trace I see that the incoming call has numbering plan ISDN and number type subscriber and the called without 0. Have you tried same setting to 7305880719? The format of the number can also affect the call so you can try with countrycode+number, 00+countrycode+number, numer or the number without the 0. Typically the countrycode+number is used when you specify the TON=international, 00+countrycode+number for TON=unknown and number (with or without 0) for TON=National. Stephen On Thu, Mar 8, 2012 at 8:10 PM, Anita Hall wrote: > Hi Stephen > > Thanks a TON, pun unintended :) > > I tried setting the TON to 2 or 4 and NPI to 1 and on Q.931 trace it shows > that the numbering type and plan have been set correctly. However, it still > show the same UNALLOCATED_NUMBER hangup cause. > > I have attached the pcap trace of an incoming call which then sets the > channel variables as given on the link you shared and bridges a call to the > said number 07305880719 . The reason why I have to do this is because I > could not figure out how to set these profile variables from the originate > command, so I set them up in the dialplan. > > This number 07305880719 is valid, I am able to call it from my mobile and > also from another card + freeswitch > > I suspected that the issue could be with sangoma isdn library, so I tried > with libpri instead. Even with correct TON and NPI, the call is not being > set-up. > > Could this be an issue with my PRI line or Sangoma card itself ? > > regards, > Anita > > > > > On Tue, Mar 6, 2012 at 1:58 AM, Stephen Wilde wrote: > >> In your log I see an "UNALLOCATED_NUMBER" release cause. >> If the number is ok, this can be due to the ISDN numbering plan/number >> type of the called party. >> What are required by your isdn provider? >> You can try with different parameters (for your called number I think >> that you have to specify numbering plan=ISDN and number type=National). >> These parameter can be specified in profile or dialplan. >> >> >> http://wiki.sangoma.com/wanpipe-freeswitch-config-appendix#ton-npi-per-call >> >> Stephen >> >> >> On Mon, Mar 5, 2012 at 7:55 PM, Anita Hall wrote: >> >>> Thanks John. >>> >>> I could take a pcap trace using >>> wanpipemon -i w4g1 -pcap -pcap_file sng_isdn_fwd.pc -prot ISDN -full >>> -systime -c trd >>> >>> The q931 for a call which is not maturing is as follows. This is the >>> case when the called number is set to unconditional forwarding. >>> 3 6.057463 Local User Remote Network Q.931 SETUP >>> 5 6.142822 Remote Network Local User Q.931 CALL PROCEEDING >>> 7 Remote Network Local User Q.931 DISCONNECT >>> 9 Local User Remote Network Q.931 RELEASE >>> 11 Remote Network Local User Q.931 RELEASE COMPLETE >>> >>> This happens only with Sangoma Card. I am using the sangoma_isdn stack >>> and FreeTDM. Somehow, it is not able to understand that the call is >>> getting forwarded. >>> >>> The q931 for a call which is maturing is as follows >>> 11 Local User Remote Network Q.931 SETUP >>> 13 Remote Network Local User Q.931 CALL PROCEEDING >>> 15 Remote Network Local User Q.931 ALERTING >>> 17 Remote Network Local User Q.931 CONNECT >>> 23 Remote Network Local User Q.931 DISCONNECT >>> 25 Local User Remote Network Q.931 RELEASE >>> 27 Remote Network Local User Q.931 RELEASE COMPLETE >>> >>> >>> >>> regards, >>> Anita >>> >>> >>> >>> >>> On Mon, Mar 5, 2012 at 11:57 PM, John wrote: >>> >>>> On 05/03/12 16:31, Anita Hall wrote: >>>> > Hi >>>> > >>>> > There are some numbers on which I am not able to originate calls from >>>> > a Sangoma Card, while I am able to do the same from a Digium like >>>> > card. I am using FreeTDM. >>>> > >>>> > >originate freetdm/4/a/07305880672 &park() >>>> > >>>> > Log is here http://pastebin.freeswitch.org/18578 >>>> > >>>> > Note that I am able to call other numbers from this system and I have >>>> > a working PRI line and FreeTDM. >>>> > >>>> > Also the number 07305880672 is working fine. It unconditionally >>>> > forwards the call to another number. I am problem with calling all >>>> > such numbers. >>>> > >>>> > I am able to do q931 traces using sangoma_isdn but how to view them ? >>>> > >>>> > regards, >>>> > Anita >>>> > >>>> Anita, >>>> >>>> You can use wireshark to decode the q.931 traces; the output files are >>>> .pcap format. >>>> >>>> John >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/81d40c4d/attachment.html From basit.engg at gmail.com Fri Mar 9 00:44:28 2012 From: basit.engg at gmail.com (Abdul Basit) Date: Fri, 9 Mar 2012 02:44:28 +0500 Subject: [Freeswitch-users] sipt-t support in freeswitch In-Reply-To: References: Message-ID: Is there any patch will available in near future? anyone working on it? -- regards, Abdul Basit On Fri, Mar 9, 2012 at 1:27 AM, Kristian Kielhofner wrote: > SIP-T requires multipart body support which FreeSWITCH does have. > FreeSWITCH can pass multipart bodies from one endpoint to the next > (bridge). However, at this point in time FreeSWITCH cannot parse or > create the encapsulated SIP-T bodies. > > On Thu, Mar 8, 2012 at 2:05 PM, Abdul Basit wrote: > > for reference what i m looking for is > > > > SIP for Telephones (SIP-T): Context and Architecture > > > > http://tools.ietf.org/html/draft-ietf-sipping-sipt-04 > > > > > > > > > > On Thu, Mar 8, 2012 at 11:56 PM, Abdul Basit > wrote: > >> > >> FS fellows, > >> > >> Does FreeSWITCH support sip-t protocol? > >> > >> > >> > http://www.dialogic.com/webhelp/BorderNet2020/1.1.0/WebHelp/sip_sipt_ov.htm > >> > >> can anyone point me in right direction? > >> > >> -- > >> Regards, > >> > >> Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 > > > > > > > > > > -- > > Regards, > > > > Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/b632aa43/attachment-0001.html From kris at kriskinc.com Fri Mar 9 01:39:30 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 8 Mar 2012 17:39:30 -0500 Subject: [Freeswitch-users] sipt-t support in freeswitch In-Reply-To: References: Message-ID: Not that I know of. It would be a major effort. FYI I believe Yate supports SIP-T. On Thu, Mar 8, 2012 at 4:44 PM, Abdul Basit wrote: > Is there any patch will available in near future? anyone working on it? > > -- > regards, > > Abdul Basit > > > On Fri, Mar 9, 2012 at 1:27 AM, Kristian Kielhofner > wrote: >> >> SIP-T requires multipart body support which FreeSWITCH does have. >> FreeSWITCH can pass multipart bodies from one endpoint to the next >> (bridge). ?However, at this point in time FreeSWITCH cannot parse or >> create the encapsulated SIP-T bodies. >> >> On Thu, Mar 8, 2012 at 2:05 PM, Abdul Basit wrote: >> > for reference what i m looking for is >> > >> > SIP for Telephones (SIP-T): Context and Architecture >> > >> > http://tools.ietf.org/html/draft-ietf-sipping-sipt-04 >> > >> > >> > >> > >> > On Thu, Mar 8, 2012 at 11:56 PM, Abdul Basit >> > wrote: >> >> >> >> FS fellows, >> >> >> >> Does FreeSWITCH support sip-t protocol? >> >> >> >> >> >> >> >> http://www.dialogic.com/webhelp/BorderNet2020/1.1.0/WebHelp/sip_sipt_ov.htm >> >> >> >> can anyone point me in right direction? >> >> >> >> -- >> >> Regards, >> >> >> >> Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 >> > >> > >> > >> > >> > -- >> > Regards, >> > >> > Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Kristian Kielhofner >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From basit.engg at gmail.com Fri Mar 9 01:56:51 2012 From: basit.engg at gmail.com (Abdul Basit) Date: Fri, 9 Mar 2012 03:56:51 +0500 Subject: [Freeswitch-users] sipt-t support in freeswitch In-Reply-To: References: Message-ID: Thank you Kristian. Our major work is on asterisk and now trying to move to FreeSWITCH. Now there is a new requirement of sip-t support. I will explore yate to see how this is fruitful. -- regards, Abdul Basit On Fri, Mar 9, 2012 at 3:39 AM, Kristian Kielhofner wrote: > Not that I know of. It would be a major effort. > > FYI I believe Yate supports SIP-T. > > On Thu, Mar 8, 2012 at 4:44 PM, Abdul Basit wrote: > > Is there any patch will available in near future? anyone working on it? > > > > -- > > regards, > > > > Abdul Basit > > > > > > On Fri, Mar 9, 2012 at 1:27 AM, Kristian Kielhofner > > wrote: > >> > >> SIP-T requires multipart body support which FreeSWITCH does have. > >> FreeSWITCH can pass multipart bodies from one endpoint to the next > >> (bridge). However, at this point in time FreeSWITCH cannot parse or > >> create the encapsulated SIP-T bodies. > >> > >> On Thu, Mar 8, 2012 at 2:05 PM, Abdul Basit > wrote: > >> > for reference what i m looking for is > >> > > >> > SIP for Telephones (SIP-T): Context and Architecture > >> > > >> > http://tools.ietf.org/html/draft-ietf-sipping-sipt-04 > >> > > >> > > >> > > >> > > >> > On Thu, Mar 8, 2012 at 11:56 PM, Abdul Basit > >> > wrote: > >> >> > >> >> FS fellows, > >> >> > >> >> Does FreeSWITCH support sip-t protocol? > >> >> > >> >> > >> >> > >> >> > http://www.dialogic.com/webhelp/BorderNet2020/1.1.0/WebHelp/sip_sipt_ov.htm > >> >> > >> >> can anyone point me in right direction? > >> >> > >> >> -- > >> >> Regards, > >> >> > >> >> Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 > >> > > >> > > >> > > >> > > >> > -- > >> > Regards, > >> > > >> > Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > >> > >> > >> -- > >> Kristian Kielhofner > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/dc9a1f33/attachment.html From msc at freeswitch.org Fri Mar 9 02:27:18 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 8 Mar 2012 15:27:18 -0800 Subject: [Freeswitch-users] New batch of sound files In-Reply-To: References: Message-ID: I'm showing a file named "ivr-there_are.wav" in all my directories... do you not have that one? -MC On Wed, Mar 7, 2012 at 10:04 AM, Brian Foster wrote: > Thanks for the heads up. I did download the lastest set of prompts per > your instructions, however I don't see the "There are..." prompt. I see the > other two for what would be said after the conference count is announced > though. I've looked all over the sounds directory trying to find it, even > playing back likely candidates but to no avail. Is that prompt missing? > > -BDF > > On Mon, Mar 5, 2012 at 2:50 PM, Michael Collins wrote: > >> Ask and ye shall receive! >> >> The files have been out on files.freeswitch.org for the past week or so. >> Look for version 1.0.18. Also, in my previous email on the subject I >> mentioned this trick for forcing your system to get the latest sounds: >> edit ${fs_src}/build/sounds_version.txt >> change callie to "1.0.18" >> save & exit >> "make cd-sounds-install" >> >> New sounds will be downloaded. As far as the prompts you mentioned, I >> added several prompts that we can use to piece together some phrase files >> or whatnot. I've got two different wordings depending on your preference: >> >> There is one other person in this conference >> There is one other member in this conference >> There are... >> ...other members in this conference >> ...other persons in this conference >> >> Now you have the pieces you need to build a nice little phrase. >> >> -MC >> >> On Sat, Mar 3, 2012 at 3:23 PM, Brian Foster wrote: >> >>> Hola, >>> >>> I briefly heard in channel and on conference about a new batch of sound >>> files for freeswitch. Is there a download link out there for this? Also, is >>> there any notable prompts for conferences? I'm specifically looking for >>> some prompts that would take care of telling the incoming participant how >>> many other participants are in the conference. >>> >>> Thanks! >>> -BDF >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-429-1069 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/41021d4b/attachment-0001.html From anthony.minessale at gmail.com Fri Mar 9 02:31:22 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 8 Mar 2012 17:31:22 -0600 Subject: [Freeswitch-users] Channels never hangup? In-Reply-To: <4F566F93.9010302@communicatefreely.net> References: <4F564D69.8090407@communicatefreely.net> <4F566F93.9010302@communicatefreely.net> Message-ID: did you read the followup comments on the bug? On Tue, Mar 6, 2012 at 2:12 PM, Tim St. Pierre wrote: > Seems easy enough to reproduce now. > > JIRA (FS-3978) > > Core file is attached to the Jira. > > The strange thing is that now I can load one profile, check voice mail > three times and all three channels are stuck. > > Hope you can find something useful. > > I am going to have to restore an old backup so we can get running safely > again. ?We may have funny BLFs, but at least it handled calls. > > I will make an image of the system first though, so we can restore it on > a spare machine if we need more evidence. > > Thanks! > > -Tim > > Anthony Minessale wrote: >> update to HEAD with make current >> try again >> if you still have the same problem, gcore the running FS and run >> "thread apply all bt" and capture the output and post to jira. >> >> >> On Tue, Mar 6, 2012 at 11:46 AM, Tim St. Pierre >> wrote: >> >>> Hello, >>> >>> Had a real emergency today, and I'm not sure it's done. >>> >>> All of a sudden, everything is busy. ?I look at my channel count, and >>> 200 channels (max allowed). >>> >>> We usually don't see more than 40. ?Some of these calls were two hours >>> old. ?I know nobody checks voice mail for two hours. >>> >>> I tried a test call, and it never left the channel list. >>> >>> Restarted freeswitch, same thing happened a second time. ?200 sessions, >>> 200 stuck channels. >>> >>> Restarted the machine. ?Same thing again. >>> >>> I haven't touched anything in two days - what did I do? >>> >>> HELP >>> >>> -Tim >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From nestor at tiendalinux.com Fri Mar 9 02:38:51 2012 From: nestor at tiendalinux.com (Nestor A Diaz) Date: Thu, 08 Mar 2012 18:38:51 -0500 Subject: [Freeswitch-users] EVENT_TRAP: IP change detected -> no Write unlock -> Error Creating SIP UA for profile: internal In-Reply-To: References: <4F4D5F07.4040301@tiendalinux.com> Message-ID: <4F59430B.8040004@tiendalinux.com> On 02/28/2012 11:26 PM, Anthony Minessale wrote: > it happens when the route to the internet changes to another device. > you an disable this: > > edit your sofia profile file internal.xml > > and comment this: > > > Hi Anthony , but the route didn't change, i just added aliases to the external interface (mistake: i wrote internal interface on the first mail but it was the external interface), the funny thing is that no matter if i disabled the aliases, after some calls the freeswitch trigger the event: 2012-03-08 16:04:56.186572 [INFO] mod_sofia.c:5155 EVENT_TRAP: IP change detected and no internal profile anymore, i have already disabled the aliases when this happened. The external interface was a network card with native vlan and the internal interface was a vlan over the same hardware interface. (just because i used to separate voice from data on different vlans) I just put your fix on the internal profile, I hope the problem does not happen again. Hope that will give you some hints. Thanks for such awesome piece of software ! Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-485-3020 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211 at tiendalinux.com Email/MSN: nestor at tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia From asteriskcoding at gmail.com Wed Mar 7 21:54:49 2012 From: asteriskcoding at gmail.com (Ast Coder) Date: Wed, 7 Mar 2012 13:54:49 -0500 Subject: [Freeswitch-users] FreeSWITCH Conf Call Tomorrow - Dialplan Basics In-Reply-To: References: Message-ID: Hi, What is the screen share URL for this call? I am on Skype and the remote.voipprovider.net is broken. Also, irc channel is not on the Wiki. Thanks, On Tue, Mar 6, 2012 at 6:20 PM, Michael Collins wrote: > Hello all! > > Here's tomorrow's agenda page: > http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_06 > > I will be discussing some dialplan basics along with some regex tricks and > basic patterns. Other experts may also be chiming in. The idea is that > everyone on the call will come away with a more thorough understanding of > what happens when a phone call comes into FreeSWITCH. > > Talk to you tomorrow! > > -Michael > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120307/20c02412/attachment.html From bedgar at vseinc.com Thu Mar 8 20:35:38 2012 From: bedgar at vseinc.com (bedgar at vseinc.com) Date: Thu, 8 Mar 2012 12:35:38 -0500 Subject: [Freeswitch-users] Adding 1khz or high resolution timer to CentOS for FS virtualization Message-ID: <333789DE5C38474EB3A478A538F4EBAB0A32C20D3C@prod-exch01.corp.vseinc.com> We have kernel 2.6.18-194.8.1.el5 with FS 1.0.6 and are testing virtualizing FS on ESX5. I know there is constant hashing out in the community. Please help in locating any past postings in guiding us in this direction. Also any rumblings welcome. Brian C. Edgar, Jr. Senior Systems Administrator Voice Systems Engineering Inc. Email: bedgar at vseinc.com Phone: 215-953-8568 x278 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/39ad6249/attachment.html From curriegrad2004 at gmail.com Fri Mar 9 02:43:10 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Thu, 8 Mar 2012 15:43:10 -0800 Subject: [Freeswitch-users] Adding 1khz or high resolution timer to CentOS for FS virtualization In-Reply-To: <333789DE5C38474EB3A478A538F4EBAB0A32C20D3C@prod-exch01.corp.vseinc.com> References: <333789DE5C38474EB3A478A538F4EBAB0A32C20D3C@prod-exch01.corp.vseinc.com> Message-ID: 1KHz should be fine on that kernel On Thu, Mar 8, 2012 at 9:35 AM, wrote: > We have kernel 2.6.18-194.8.1.el5 with FS 1.0.6 and are testing virtualizing > FS on ESX5.? I know there is constant hashing out in the community.? Please > help in locating any past postings in guiding us in this direction. > > > > Also any rumblings welcome. > > > > Brian C. Edgar, Jr. > > Senior Systems Administrator > > Voice Systems Engineering Inc. > > Email: bedgar at vseinc.com > > Phone: 215-953-8568 x278 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Fri Mar 9 02:51:59 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 8 Mar 2012 15:51:59 -0800 Subject: [Freeswitch-users] Voicemail Transcription In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1504D8C5@cantor.std.visionutv.se> <00d801ccfc4f$98695e80$c93c1b80$@com> Message-ID: On Wed, Mar 7, 2012 at 3:16 AM, Balamurugan Mahendran wrote: > how to setup? do you have any articles to fallow? http://wiki.freeswitch.org/wiki/Transcribing_Voicemail -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/de6ad2cb/attachment.html From msc at freeswitch.org Fri Mar 9 02:53:19 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 8 Mar 2012 15:53:19 -0800 Subject: [Freeswitch-users] mod shell stream In-Reply-To: <1331220278.14097.YahooMailRC@web180216.mail.gq1.yahoo.com> References: <5BA3C64DA53B4EAC9B1DA367F8FFD398@gmail.com> <1331220278.14097.YahooMailRC@web180216.mail.gq1.yahoo.com> Message-ID: I'm afraid I'll have to defer to those more knowledgeable than I in such matters. Any sox and/or cmd line gurus out there with some ideas? -MC On Thu, Mar 8, 2012 at 7:24 AM, Bernard Fluixa wrote: > Michael, > > Thanks for your reply. Yes it works in this case. > > Any thoughts? > > Bernard > > ------------------------------ > *From:* Michael Collins > *To:* FreeSWITCH Users Help > *Sent:* Mon, March 5, 2012 10:18:44 PM > *Subject:* Re: [Freeswitch-users] mod shell stream > > > > On Sun, Mar 4, 2012 at 4:30 AM, Bernard Fluixa wrote: > >> That is already done with the -t option (tuples only) >> >> Thanks anyway >> >> Bernard >> >> > Does your script work if you do it in two steps, i.e. read from the > database and drop into a file, then cat the file into sox? I'm just curious > if something unexpected is happening on the command line when piping from > psql to sox. > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/20345f8e/attachment-0001.html From msc at freeswitch.org Fri Mar 9 02:56:52 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 8 Mar 2012 15:56:52 -0800 Subject: [Freeswitch-users] Fs problem: can.t send queue_dtmf in loopback ignore_early_media=true not working In-Reply-To: References: <4F573CF6.9000205@softnet.si> <4F575B54.3050403@softnet.si> Message-ID: Why do you need to use loopback? -MC On Wed, Mar 7, 2012 at 5:27 AM, Thomas Hoellriegel wrote: > Hi guys, > I use the actual git from today. > I create a queue_dtmf extension: > > > > > > > > > > fs does not wait until the connection is established. > This problem occurs only with loopback. > ignore_early_media = true shows no effect. > I should route directly over a gateway, it works out. > for example: > data="{ignore_early_media=**true}sofia/gateway/landline/**015703730412"/> > works correctly. > Can your looks for this problem please? > thanks. > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/178369cf/attachment.html From freeswitch-list at puzzled.xs4all.nl Fri Mar 9 03:02:00 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Fri, 09 Mar 2012 01:02:00 +0100 Subject: [Freeswitch-users] Adding 1khz or high resolution timer to CentOS for FS virtualization In-Reply-To: <333789DE5C38474EB3A478A538F4EBAB0A32C20D3C@prod-exch01.corp.vseinc.com> References: <333789DE5C38474EB3A478A538F4EBAB0A32C20D3C@prod-exch01.corp.vseinc.com> Message-ID: <4F594878.9060003@puzzled.xs4all.nl> On 08-03-12 18:35, bedgar at vseinc.com wrote: > We have kernel 2.6.18-194.8.1.el5 with FS 1.0.6 and are testing > virtualizing FS on ESX5. I know there is constant hashing out in the > community. Please help in locating any past postings in guiding us in > this direction. For starters I would use git MASTER as 1.0.6 is ancient. Both CentOS 5 & 6 have 1000 Hz timers which is what you want. Just search the mailinglist for "virtual", "virtualized" and "virtualisation" and you'll get a lot of hits. Regards, Patrick From david at styleflare.com Fri Mar 9 03:05:18 2012 From: david at styleflare.com (David J) Date: Thu, 8 Mar 2012 19:05:18 -0500 Subject: [Freeswitch-users] Adding 1khz or high resolution timer to CentOS for FS virtualization In-Reply-To: <4F594878.9060003@puzzled.xs4all.nl> References: <333789DE5C38474EB3A478A538F4EBAB0A32C20D3C@prod-exch01.corp.vseinc.com> <4F594878.9060003@puzzled.xs4all.nl> Message-ID: Should be 1khz by default on that kernel On Mar 8, 2012 7:02 PM, "Patrick Lists" wrote: > On 08-03-12 18:35, bedgar at vseinc.com wrote: > > We have kernel 2.6.18-194.8.1.el5 with FS 1.0.6 and are testing > > virtualizing FS on ESX5. I know there is constant hashing out in the > > community. Please help in locating any past postings in guiding us in > > this direction. > > For starters I would use git MASTER as 1.0.6 is ancient. Both CentOS 5 & > 6 have 1000 Hz timers which is what you want. > > Just search the mailinglist for "virtual", "virtualized" and > "virtualisation" and you'll get a lot of hits. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/0a142ec6/attachment.html From aksrini at hotmail.com Fri Mar 9 03:38:35 2012 From: aksrini at hotmail.com (Srini K) Date: Thu, 8 Mar 2012 16:38:35 -0800 Subject: [Freeswitch-users] How to get session obj from uuid Message-ID: Hi,How to get the Session object if I know the uuid using mod_managed. ThanksSrini -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/97736c4a/attachment.html From bdfoster at endigotech.com Fri Mar 9 03:43:48 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 8 Mar 2012 19:43:48 -0500 Subject: [Freeswitch-users] New batch of sound files In-Reply-To: References: Message-ID: MC, That's what I figured it was called... must be something on my end. I'll download manually and add it in. -BDF On Mar 8, 2012 6:28 PM, "Michael Collins" wrote: > I'm showing a file named "ivr-there_are.wav" in all my directories... do > you not have that one? > -MC > > On Wed, Mar 7, 2012 at 10:04 AM, Brian Foster wrote: > >> Thanks for the heads up. I did download the lastest set of prompts per >> your instructions, however I don't see the "There are..." prompt. I see the >> other two for what would be said after the conference count is announced >> though. I've looked all over the sounds directory trying to find it, even >> playing back likely candidates but to no avail. Is that prompt missing? >> >> -BDF >> >> On Mon, Mar 5, 2012 at 2:50 PM, Michael Collins wrote: >> >>> Ask and ye shall receive! >>> >>> The files have been out on files.freeswitch.org for the past week or >>> so. Look for version 1.0.18. Also, in my previous email on the subject I >>> mentioned this trick for forcing your system to get the latest sounds: >>> edit ${fs_src}/build/sounds_version.txt >>> change callie to "1.0.18" >>> save & exit >>> "make cd-sounds-install" >>> >>> New sounds will be downloaded. As far as the prompts you mentioned, I >>> added several prompts that we can use to piece together some phrase files >>> or whatnot. I've got two different wordings depending on your preference: >>> >>> There is one other person in this conference >>> There is one other member in this conference >>> There are... >>> ...other members in this conference >>> ...other persons in this conference >>> >>> Now you have the pieces you need to build a nice little phrase. >>> >>> -MC >>> >>> On Sat, Mar 3, 2012 at 3:23 PM, Brian Foster wrote: >>> >>>> Hola, >>>> >>>> I briefly heard in channel and on conference about a new batch of sound >>>> files for freeswitch. Is there a download link out there for this? Also, is >>>> there any notable prompts for conferences? I'm specifically looking for >>>> some prompts that would take care of telling the incoming participant how >>>> many other participants are in the conference. >>>> >>>> Thanks! >>>> -BDF >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-429-1069 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/07412107/attachment-0001.html From bdfoster at endigotech.com Fri Mar 9 03:49:08 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 8 Mar 2012 19:49:08 -0500 Subject: [Freeswitch-users] FreeSWITCH Conf Call Tomorrow - Dialplan Basics In-Reply-To: References: Message-ID: IRC: freenode/#freeswitch The screenshare is not up when there is nothing to display :-) On Mar 8, 2012 6:40 PM, "Ast Coder" wrote: > Hi, > > What is the screen share URL for this call? I am on Skype and the > remote.voipprovider.net is broken. > > Also, irc channel is not on the Wiki. > > > Thanks, > > On Tue, Mar 6, 2012 at 6:20 PM, Michael Collins wrote: > >> Hello all! >> >> Here's tomorrow's agenda page: >> http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_06 >> >> I will be discussing some dialplan basics along with some regex tricks >> and basic patterns. Other experts may also be chiming in. The idea is that >> everyone on the call will come away with a more thorough understanding of >> what happens when a phone call comes into FreeSWITCH. >> >> Talk to you tomorrow! >> >> -Michael >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/9049c44f/attachment.html From B.Tietz at pinguin.ag Fri Mar 9 08:19:22 2012 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Fri, 9 Mar 2012 06:19:22 +0100 Subject: [Freeswitch-users] uuid_kill Message-ID: <07BF4904977CC645B485E970424193AD0FF1077265@localhost> Hi, sometimes I have some stuck calls in FS. "show calls" is showing them to me. If I try to kill the call with "uuid_kill " the call won't be killed... I have to restart freeswitch, but I suppose this is not the best way for this problem. Deleting from database doesn't help too... Are there any other ideas? regards, Benjamin T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/70b03445/attachment.html From asteriskcoding at gmail.com Fri Mar 9 08:47:21 2012 From: asteriskcoding at gmail.com (Ast Coder) Date: Fri, 9 Mar 2012 00:47:21 -0500 Subject: [Freeswitch-users] uuid_kill In-Reply-To: <07BF4904977CC645B485E970424193AD0FF1077265@localhost> References: <07BF4904977CC645B485E970424193AD0FF1077265@localhost> Message-ID: I am just learning FreeSwitch and I see a lot of stuck calls. Why is that? Does that really mean there are active channels to PSTN even though my SIP end-point hanged up minutes ago? There shouldn't be any need for uuid_kill. Is this common to FreeSwitch to see calls getting stuck? Thanks, On Fri, Mar 9, 2012 at 12:19 AM, wrote: > Hi,**** > > ** ** > > sometimes I have some stuck calls in FS. ?show calls? is showing them to > me.**** > > If I try to kill the call with ?uuid_kill ? the call won?t be > killed? I have to restart freeswitch, but I suppose this is not the best > way for this problem. Deleting from database doesn?t help too?**** > > ** ** > > Are there any other ideas?**** > > ** ** > > regards,**** > > Benjamin T.**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/04333cd2/attachment.html From peter.olsson at visionutveckling.se Fri Mar 9 08:52:08 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 9 Mar 2012 05:52:08 +0000 Subject: [Freeswitch-users] uuid_kill In-Reply-To: <07BF4904977CC645B485E970424193AD0FF1077265@localhost> References: <07BF4904977CC645B485E970424193AD0FF1077265@localhost> Message-ID: <5B60DA68-2A64-440C-AD82-55C11AA9CA37@visionutveckling.se> First of all - make sure you're on latest git :) /Peter ----- Reply message ----- Fr?n: "B.Tietz at pinguin.ag" Datum: fre, mar 9, 2012 06:28 Rubrik: [Freeswitch-users] uuid_kill Till: "freeswitch-users at lists.freeswitch.org" Hi, sometimes I have some stuck calls in FS. ?show calls? is showing them to me. If I try to kill the call with ?uuid_kill ? the call won?t be killed? I have to restart freeswitch, but I suppose this is not the best way for this problem. Deleting from database doesn?t help too? Are there any other ideas? regards, Benjamin T. !DSPAM:4f59927032768497121946! From asteriskcoding at gmail.com Fri Mar 9 09:11:16 2012 From: asteriskcoding at gmail.com (Ast Coder) Date: Fri, 9 Mar 2012 01:11:16 -0500 Subject: [Freeswitch-users] uuid_kill In-Reply-To: <5B60DA68-2A64-440C-AD82-55C11AA9CA37@visionutveckling.se> References: <07BF4904977CC645B485E970424193AD0FF1077265@localhost> <5B60DA68-2A64-440C-AD82-55C11AA9CA37@visionutveckling.se> Message-ID: Was installed two days ago. Has there been a bug fix? I don't see any Changelog file in the freeswitch download directory. Usually, where should I be able to find revision numbers and changelogs of the new version? Thanks On Fri, Mar 9, 2012 at 12:52 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > First of all - make sure you're on latest git :) > > /Peter > > ----- Reply message ----- > Fr?n: "B.Tietz at pinguin.ag" > Datum: fre, mar 9, 2012 06:28 > Rubrik: [Freeswitch-users] uuid_kill > Till: "freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > > Hi, > > sometimes I have some stuck calls in FS. ?show calls? is showing them to > me. > If I try to kill the call with ?uuid_kill ? the call won?t be > killed? I have to restart freeswitch, but I suppose this is not the best > way for this problem. Deleting from database doesn?t help too? > > Are there any other ideas? > > regards, > Benjamin T. > > !DSPAM:4f59927032768497121946! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/cda0e396/attachment-0001.html From bdfoster at endigotech.com Fri Mar 9 09:40:00 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 9 Mar 2012 01:40:00 -0500 Subject: [Freeswitch-users] uuid_kill In-Reply-To: References: <07BF4904977CC645B485E970424193AD0FF1077265@localhost> <5B60DA68-2A64-440C-AD82-55C11AA9CA37@visionutveckling.se> Message-ID: Bug fixes are done on a daily, almost hourly basis. The best place to look and see what has changed is at http://jira.freeswitch.org. On Mar 9, 2012 1:12 AM, "Ast Coder" wrote: > Was installed two days ago. Has there been a bug fix? > > I don't see any Changelog file in the freeswitch download directory. > Usually, where should I be able to find revision numbers and changelogs of > the new version? > > Thanks > > On Fri, Mar 9, 2012 at 12:52 AM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > >> First of all - make sure you're on latest git :) >> >> /Peter >> >> ----- Reply message ----- >> Fr?n: "B.Tietz at pinguin.ag" >> Datum: fre, mar 9, 2012 06:28 >> Rubrik: [Freeswitch-users] uuid_kill >> Till: "freeswitch-users at lists.freeswitch.org" < >> freeswitch-users at lists.freeswitch.org> >> >> Hi, >> >> sometimes I have some stuck calls in FS. ?show calls? is showing them to >> me. >> If I try to kill the call with ?uuid_kill ? the call won?t be >> killed? I have to restart freeswitch, but I suppose this is not the best >> way for this problem. Deleting from database doesn?t help too? >> >> Are there any other ideas? >> >> regards, >> Benjamin T. >> >> !DSPAM:4f59927032768497121946! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/8f47dc56/attachment.html From sdevoy at bizfocused.com Fri Mar 9 06:46:49 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 8 Mar 2012 22:46:49 -0500 Subject: [Freeswitch-users] FS_CLI and Log levels -sofia help cant stop unwanted log lines Message-ID: <098501ccfda7$423e0bb0$c6ba2310$@com> Hi, I must be missing some commands to filter down the messages displayed when I run fs_cli. I have tried (with some success): Console loglevel n (n = 1, 2, 3 helps very little) Sofia loglevel all n does not appear to have any affect. Sofia global siptrace on and "off" do what I expected!!!! I still get reams of log lines when any call comes in or goes out. Samples: 2012-03-08 14:34:28.132435 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14436046159 at 64.136.174.30 [BREAK] 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:362 (sofia/external_noauth/14436046159 at 64.136.174.30) Running State Change CS_REPORTING 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:662 (sofia/external_noauth/14436046159 at 64.136.174.30) State REPORTING 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:387 (sofia/external/240) State Change CS_REPORTING -> CS_DESTROY 2012-03-08 14:34:28.132435 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/240 [BREAK] 2012-03-08 14:34:28.132435 [DEBUG] switch_core_session.c:1380 Session 403 (sofia/external/240) Locked, Waiting on external entities 2012-03-08 14:34:28.132435 [NOTICE] switch_core_session.c:1398 Session 403 (sofia/external/240) Ended 2012-03-08 14:34:28.132435 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external/240 [CS_DESTROY] 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:79 sofia/external_noauth/14436046159 at 64.136.174.30 Standard REPORTING, cause: NORMAL_CLEARING 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:662 (sofia/external_noauth/14436046159 at 64.136.174.30) State REPORTING going to sleep 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:491 (sofia/external/240) Callstate Change HANGUP -> DOWN 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:387 (sofia/external_noauth/14436046159 at 64.136.174.30) State Change CS_REPORTING -> CS_DESTROY 2012-03-08 14:34:28.132435 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14436046159 at 64.136.174.30 [BREAK] What am I missing? Thanks in advance, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/9b9261cb/attachment-0001.html From sdevoy at bizfocused.com Fri Mar 9 06:54:17 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 8 Mar 2012 22:54:17 -0500 Subject: [Freeswitch-users] DIALPLAN: Bridge, Hang Up and Voice mail Message-ID: <099001ccfda8$4d552320$e7ff6960$@com> Hi, I have a couple of questions regarding XML dialplan functionality. First, using this simple dial plan: I can call 220, if there is no answer it goes to voicemail. GOOD. However, if that user answers and then hangs up, I still go to voice mail. I thought bridge terminated the call. Question 1: Why does the dial plan continue to voicemail after bridge answers, talks and hangs up? Question 2: If this user's phone is unplugged, the bridge kicks an error and it DOES NOT go to voice mail. How can I detect "extension unavailable" in the dial plan and perhaps forward to a cell phone? Thanks in advance, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120308/edd0aa79/attachment-0001.html From avi at avimarcus.net Fri Mar 9 10:26:35 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 9 Mar 2012 09:26:35 +0200 Subject: [Freeswitch-users] DIALPLAN: Bridge, Hang Up and Voice mail In-Reply-To: <099001ccfda8$4d552320$e7ff6960$@com> References: <099001ccfda8$4d552320$e7ff6960$@com> Message-ID: 1) It's an option that defaults to false. See: http://wiki.freeswitch.org/wiki/Variable_hangup_after_bridge 2) Same, it's an option that default to "don't continue on error" To change, see: http://wiki.freeswitch.org/wiki/Variable_continue_on_fail I'm pretty sure these are both in the default configurations. Did you take a look at them? They are pretty good example of many of the most common use cases. -Avi On Fri, Mar 9, 2012 at 5:54 AM, Sean Devoy wrote: > Hi,**** > > ** ** > > I have a couple of questions regarding XML dialplan functionality.**** > > ** ** > > First, using this simple dial plan:**** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > * > *** > > **** > > **** > > **** > > ** ** > > I can call 220, if there is no answer it goes to voicemail. GOOD.**** > > However, if that user answers and then hangs up, I still go to voice mail.. > I thought bridge terminated the call.**** > > Question 1: Why does the dial plan continue to voicemail after bridge > answers, talks and hangs up?**** > > ** ** > > Question 2: If this user?s phone is unplugged, the bridge kicks an error > and it DOES NOT go to voice mail. How can I detect ?extension unavailable? > in the dial plan and perhaps forward to a cell phone?**** > > ** ** > > ** ** > > Thanks in advance,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/e99d097c/attachment.html From avi at avimarcus.net Fri Mar 9 10:28:28 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 9 Mar 2012 09:28:28 +0200 Subject: [Freeswitch-users] FS_CLI and Log levels -sofia help cant stop unwanted log lines In-Reply-To: <098501ccfda7$423e0bb0$c6ba2310$@com> References: <098501ccfda7$423e0bb0$c6ba2310$@com> Message-ID: Try F8 which is "/log debug" or "/log 7". (This has to be in the fs_cli, not the actual freeswitch binary. Run FS with -nc for "no console" and then launch the fs_cli) -Avi On Fri, Mar 9, 2012 at 5:46 AM, Sean Devoy wrote: > Hi,**** > > ** ** > > I must be missing some commands to filter down the messages displayed when > I run fs_cli. I have tried (with some success):**** > > Console loglevel n (n = 1, 2, 3 helps very little)*** > * > > Sofia loglevel all n does not appear to have any affect. > **** > > ** ** > > Sofia global siptrace on and ?off? do what I expected!!!!**** > > ** ** > > I still get reams of log lines when any call comes in or goes out. **** > > ** ** > > Samples:**** > > 2012-03-08 14:34:28.132435 [DEBUG] switch_core_session.c:1180 Send signal > sofia/external_noauth/14436046159 at 64.136.174.30 [BREAK]**** > > 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:362 > (sofia/external_noauth/14436046159 at 64.136.174.30) Running State Change > CS_REPORTING**** > > 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:662 > (sofia/external_noauth/14436046159 at 64.136.174.30) State REPORTING**** > > 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:387 > (sofia/external/240) State Change CS_REPORTING -> CS_DESTROY**** > > 2012-03-08 14:34:28.132435 [DEBUG] switch_core_session.c:1180 Send signal > sofia/external/240 [BREAK]**** > > 2012-03-08 14:34:28.132435 [DEBUG] switch_core_session.c:1380 Session 403 > (sofia/external/240) Locked, Waiting on external entities**** > > 2012-03-08 14:34:28.132435 [NOTICE] switch_core_session.c:1398 Session 403 > (sofia/external/240) Ended**** > > 2012-03-08 14:34:28.132435 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/external/240 [CS_DESTROY]**** > > 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:79 > sofia/external_noauth/14436046159 at 64.136.174.30 Standard REPORTING, > cause: NORMAL_CLEARING**** > > 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:662 > (sofia/external_noauth/14436046159 at 64.136.174.30) State REPORTING going > to sleep**** > > 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:491 > (sofia/external/240) Callstate Change HANGUP -> DOWN**** > > 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:387 > (sofia/external_noauth/14436046159 at 64.136.174.30) State Change > CS_REPORTING -> CS_DESTROY**** > > 2012-03-08 14:34:28.132435 [DEBUG] switch_core_session.c:1180 Send signal > sofia/external_noauth/14436046159 at 64.136.174.30 [BREAK]**** > > ** ** > > What am I missing?**** > > ** ** > > Thanks in advance,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/e78771f2/attachment-0001.html From miha at softnet.si Fri Mar 9 11:34:01 2012 From: miha at softnet.si (Miha Zoubek) Date: Fri, 09 Mar 2012 09:34:01 +0100 Subject: [Freeswitch-users] stip variable In-Reply-To: References: <4F588FE5.3090402@softnet.si> Message-ID: <4F59C079.2070309@softnet.si> On 03/08/2012 04:17 PM, Christian Benke wrote: > On 8 March 2012 11:54, Miha Zoubek wrote: >> how can I strip this sofia/internal/18005551212 at tf.voipmich.com, so that I >> would have only number 18005551212? > See http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_regex - also > take a look at the example i've given you in your other thread > yesterday. > > Regards > Christian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Thank you Christian. Regards, Miha From miha at softnet.si Fri Mar 9 11:38:19 2012 From: miha at softnet.si (Miha Zoubek) Date: Fri, 09 Mar 2012 09:38:19 +0100 Subject: [Freeswitch-users] UUID variables Message-ID: <4F59C17B.1030502@softnet.si> Hi, is it possible to get variables from different chanel (UUID)? When I make attended transfer different caller_id_name/number appears in my CDR that should. For exp: A calls B, B maks xfer to C. At the end A is talking to C. In my CDR I am getting. A to B. B to C. And B to C. The last should be A to C. Thank you! Miha From miha at softnet.si Fri Mar 9 12:31:10 2012 From: miha at softnet.si (Miha Zoubek) Date: Fri, 09 Mar 2012 10:31:10 +0100 Subject: [Freeswitch-users] UUID variables In-Reply-To: <4F59C17B.1030502@softnet.si> References: <4F59C17B.1030502@softnet.si> Message-ID: <4F59CDDE.7080205@softnet.si> On 03/09/2012 09:38 AM, Miha Zoubek wrote: > Hi, > > is it possible to get variables from different chanel (UUID)? > > When I make attended transfer different caller_id_name/number appears in > my CDR that should. > > > For exp: > > A calls B, B maks xfer to C. > At the end A is talking to C. > > In my CDR I am getting. A to B. B to C. And B to C. The last should be A > to C. > > Thank you! > Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Forget this. Regards, Miha From shahzad.bhatti at g-r-v.com Fri Mar 9 12:41:49 2012 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Fri, 9 Mar 2012 14:41:49 +0500 Subject: [Freeswitch-users] How to Create PHP Application for Freeswicth using ESL In-Reply-To: References: Message-ID: Hi Everyone, i am a new user and want to learn how to create php application using ESL i just want to play a dynamic message using cepstral and get the customer response using dtmf. but i can't unserstand how to handle call using php as their is no mod_php is available like mod_lua, i park the call and now want to speak in the call but unable to do: please guide me i can't solve that problem with playandgetdigits() example of Mr. Michael Collins that what is the code of [handler.php]. =============================================================================================== * * *originate user/1500 meeting_request_1500* Then create that extension: ============================================================================================= is anyone guide me how to do it php, thanks in advance, Shahzad Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/44748b06/attachment.html From nasida at live.ru Fri Mar 9 14:44:46 2012 From: nasida at live.ru (Yuriy Nasida) Date: Fri, 9 Mar 2012 15:44:46 +0400 Subject: [Freeswitch-users] reserve mysql database using odbc in the core Message-ID: Hello guys! If anybody know how can I use reserve mysql database using odbc in the core? So if my first mysql server was down I would like that FS would use second mysql server for core. http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/fd2db7ef/attachment.html From motosota at gmail.com Fri Mar 9 15:09:33 2012 From: motosota at gmail.com (Mike) Date: Fri, 9 Mar 2012 12:09:33 +0000 Subject: [Freeswitch-users] mod_callcenter setting caller name on a per call basis Message-ID: Hi, I'd like to use mod_callcenter to have the agents answer calls on behalf of multiple clients. The simplest way? - and the way that I'm doing it at the moment (without mod_callcenter) is setting the caller_id_name in the dialplan based on the dialled number. So it's a bit of a fudge, (as it should be the called_id_name as it were),? but at least it displays on the phone so that the agent can answer properly: 123 calling 555(ACME) displays: From: ACME 123 Agent answers "Hello ACME, how can I help you" 456 calling 666(Bedrock) displays: From: Bedrock 666 Agent answers "Hello Bedrock, how can I help you" With mod_callcenter the wiki says: > sleep(1000) >> log("Test2") >> sleep(1000) >> log("Test3") >> bridge([origination_caller_id_number="${SRC}"]user/${DEST}@${domain_name}) >> log("Test4") >> sleep(1000) >> "log" "Test5" >> sleep(1000) >> "log" "Test6" > > A dials B. The call enters the default context where the test-script > is called, the first steps are executed and the bridge is initiated. > > B picks up the call, the following (truncated) channel-uuids exist at > this point: > "a7c4"(A->) > "a7cc"(->B) > > B then initiates an attended transfer to C, which follows the same > path and script as the first call(But the call is bridged to C instead > of B this time). Two new channels are set up - uuid's "a7d5"(B->) and > "a7dd"(->C) > > B immediately transfers the call before C has picked up. > > Channel "a7cc"(->B) is killed by att_xfer while Channel "a7c4"(A->) > finishes the script(The remaining Steps Test3-Test5 are executed) and > is then automatically parked in endless_playback by att_xfer, waiting > for C to pick up. > > C lets it ring until the timeout of 15s ends, "a7c4" and "a7dd" are > hung up while a7d5 continues as a ZOMBIE-channel, trying to execute > the remaining steps in the script(The "sleep"-commands fail due to the > zombie-state). > > And this is where i'm stuck - i would like channel "a7c4" to not get > hung up but to continue with the steps which "a7d5" could no longer > execute. Is there some way to prevent a7c4 from getting hung up when > the ATTENDED_TRANSFER to C fails? Algorithmically i can figure out how > the remaining steps can get memorised and executed, i just don't know > how to prevent "a7c4" from getting hung up and beeing transfer to a > place where i can still work with it :-/ I can't use the HangupHook > either, as a7c4 is already in zombie-state at that point. > > For reference, i've attached the test-script and a freeswitch-log of > the call above. > > I hope someone can lead me on the right track... > > Best regards > Christian From peter.olsson at visionutveckling.se Fri Mar 9 15:32:11 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 9 Mar 2012 12:32:11 +0000 Subject: [Freeswitch-users] reserve mysql database using odbc in the core Message-ID: <1FFF97C269757C458224B7C895F35F150505D4@cantor.std.visionutv.se> You need to setup MySQL to be able to failover. Most people use MySQL + DRBD + heartbeat for this kind of operation. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Yuriy Nasida Skickat: den 9 mars 2012 12:45 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] reserve mysql database using odbc in the core Hello guys! If anybody know how can I use reserve mysql database using odbc in the core? So if my first mysql server was down I would like that FS would use second mysql server for core. http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core Thanks. !DSPAM:4f59ec5e32765716517795! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/7a10b46a/attachment-0001.html From motosota at gmail.com Fri Mar 9 16:58:30 2012 From: motosota at gmail.com (Mike) Date: Fri, 9 Mar 2012 13:58:30 +0000 Subject: [Freeswitch-users] mod_callcenter setting caller name on a per call basis In-Reply-To: References: Message-ID: In case it's of use to anyone else: Thanks to the guys on the IRC - the solution to this is to set the caller_id_name variable in the dialplan before calling the callcenter application thuswise: This works a treat. The other methods we tried - using effective_caller_id_name or orignination_caller_id_name with either export, set or bridge_export didn't work. Mike On Fri, Mar 9, 2012 at 12:09 PM, Mike wrote: > Hi, > > I'd like to use mod_callcenter to have the agents answer calls on > behalf of multiple clients. > > The simplest way? - and the way that I'm doing it at the moment > (without mod_callcenter) is setting the caller_id_name in the dialplan > based on the dialled number. So it's a bit of a fudge, (as it should > be the called_id_name as it were),? but at least it displays on the > phone so that the agent can answer properly: > > 123 calling 555(ACME) displays: > From: ACME > 123 > Agent answers "Hello ACME, how can I help you" > > 456 calling 666(Bedrock) displays: > From: Bedrock > 666 > Agent answers "Hello Bedrock, how can I help you" > > With mod_callcenter the wiki says: > > > contact="[origination_caller_id_name='Queue Caller']user/1001 at default > > > Which works fine with a fixed value, but trying to include a variable > set earlier in the dialplan like: > contact="[origination_caller_id_name=${the_name_I_set}]user/1001 at default > > > Throws an error: > 2012-03-09 10:59:25.810882 [CRIT] switch_channel.c:1183 Invalid data > (${sip_from_display} contains a variable) > 2012-03-09 10:59:25.810882 [CRIT] switch_channel.c:1183 Invalid data > (${sip_full_from} contains a variable) > > I understand the security reasons for not allowing variables within > variables - as explained by Brian here > http://jira.freeswitch.org/browse/FS-3015 > > So - given all that - any suggestions on how to get this to work with > mod_callcenter? > > Mike From nasida at live.ru Fri Mar 9 17:03:00 2012 From: nasida at live.ru (Yuriy Nasida) Date: Fri, 9 Mar 2012 18:03:00 +0400 Subject: [Freeswitch-users] reserve mysql database using odbc in the core In-Reply-To: <1FFF97C269757C458224B7C895F35F150505D4@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F150505D4@cantor.std.visionutv.se> Message-ID: Peter, Thanks for your answer. Yes I know about using MySQL + DRBD + heartbeat but I search the best way.I plan to use 2 box only. each box will have (FS+mysql). Both FS has to use same mysql. It is necessary for feature - the recovering of current calls in FS2 if FS1 (i mean application level only, i.e. not full box1) will be broken. http://wiki.freeswitch.org/wiki/Freeswitch_HA If full box1 will broken, FS2 from box2 has to use own mysql server (mysql2) which will be synchronized with mysql1 from box1 (by means of mysql replication). I will lose current calls but new calls will work fine. I am not sure if it is possible. So I search possibility of using of backup mysql server if main server down. It may looks like second DNS or second SIP-proxy. I just need to put the second ip address of the standby mysql server on FS2. So FS2 will have two IP address of mysql servers: 1) ip adress of mysql1 (box1) 2) own ip. Thanks. From: peter.olsson at visionutveckling.se To: freeswitch-users at lists.freeswitch.org Date: Fri, 9 Mar 2012 12:32:11 +0000 Subject: Re: [Freeswitch-users] reserve mysql database using odbc in the core You need to setup MySQL to be able to failover. Most people use MySQL + DRBD + heartbeat for this kind of operation. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Yuriy Nasida Skickat: den 9 mars 2012 12:45 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] reserve mysql database using odbc in the core Hello guys! If anybody know how can I use reserve mysql database using odbc in the core? So if my first mysql server was down I would like that FS would use second mysql server for core. http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core Thanks. !DSPAM:4f59ec5e32765716517795! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/94a61b19/attachment.html From krice at freeswitch.org Fri Mar 9 17:22:26 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 09 Mar 2012 08:22:26 -0600 Subject: [Freeswitch-users] uuid_kill In-Reply-To: Message-ID: There was a recent bug where this happened... Update to current git, and check it again if you are still getting these hung calls, please help us out by using gcore on the still running freeswitch with the hung calls to get a core dump, then open that with gdb and follow the bug reporting info at http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb _.28Linux.2FUnix.29 and open a Jira. Since I am throw this out on the mailing list for this bug please check to make sure someone hasn?t already opened a bug on it before opening a new one. Stop but #freeswitch on irc.freenode.net (our irc channel) if you have a box in this state and we can get a core dump but you need help on it. (note, we?ll have to crash it to get the right coredump so you don?t want to gcore it on a production system with live calls) K On 3/8/12 11:47 PM, "Ast Coder" wrote: > I am just learning FreeSwitch and I see a lot of stuck calls.? > > Why is that? Does that really mean there are active channels to PSTN even > though my SIP end-point hanged up minutes ago??There shouldn't be any need for > uuid_kill. Is this common to FreeSwitch to see calls getting stuck? > > Thanks, > > On Fri, Mar 9, 2012 at 12:19 AM, wrote: >> Hi, >> ? >> sometimes I have some stuck calls in FS. ?show calls? is showing them to me. >> If I try to kill the call with ?uuid_kill ? the call won?t be killed? I >> have to restart freeswitch, but I suppose this is not the best way for this >> problem. Deleting from database doesn?t help too? >> ? >> Are there any other ideas? >> ? >> regards, >> Benjamin T. >> ? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/cd98c1bc/attachment.html From miha at softnet.si Fri Mar 9 17:37:05 2012 From: miha at softnet.si (Miha Zoubek) Date: Fri, 09 Mar 2012 15:37:05 +0100 Subject: [Freeswitch-users] blocking destination number Message-ID: <4F5A1591.8090703@softnet.si> Hi, what is the bast way to block destination number for certain user. Is it possible to do it in user/dir? Regards, Miha From maqsood100 at gmail.com Fri Mar 9 13:34:59 2012 From: maqsood100 at gmail.com (Mohamed Alam) Date: Fri, 9 Mar 2012 16:04:59 +0530 Subject: [Freeswitch-users] Please help me would like to use PHP ESL Message-ID: I am new to this. I would like to use PHP ESL for text chat,voice chat. How to create database. Where to add users. How voice chat will be generated. Please help me -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/b2be5dd0/attachment-0001.html From roger.castaldo at gmail.com Fri Mar 9 18:07:57 2012 From: roger.castaldo at gmail.com (Roger Castaldo) Date: Fri, 9 Mar 2012 10:07:57 -0500 Subject: [Freeswitch-users] State of GUIs In-Reply-To: References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> Message-ID: Jay am I to take your response as you are interested in helping with the development of the project and have some background in C#? On Thu, Mar 8, 2012 at 10:11 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > We encourage people to make GUIs for FreeSWITCH and we have mixed results. > Being controlled externally is actually baked into the design of > FreeSWITCH. Many of the abstract concepts int the core that people are > still unearthing are all about developing external apps. The exercise to > build one was intentionally left to the end user. The point was to harness > the hard part about telephony so you could focus on your business use case. > Really, based on the requirements of the original poster, I think a > comercial solution is in order. I also would not judge activity by release > dates / commits etc. FreeSWITCH itself has not been released since 2009 > yet the C code has grown by 35 megabits since then with commits nearly > hourly. We just happen to be more open about our development than others > may be comfortable with. > > Anyway, I know for a fact all of the above list in the original post are > still up and running but not everyone has the resources to run an active > community. > > As referred to earlier in this thread, we made the CudaTEL PBX at > Barracuda Networks which is an appliance designed to do everything > described above and that effort could not be done without > a commercial incentive to develop it and a team of dedicated employees. > > Everything you ever wanted is typically hard to come by when dealing with > Free GUIs so the alternative is to invest in one of them and get them to > focus on the functionality you seek. > > > > > > > > On Thu, Mar 8, 2012 at 2:17 AM, Brett Wilson wrote: > >> I checked out freepbx. Freepbx v3, the complete rewrite that was supposed >> to work with freeswitch, actually got spun off into the blue.box project. >> Seems that the blue.box project is basically dead. The last commit was >> November 2011. Fusionpbx looks to be the only game in town. I was wrong, it >> is still being developed. I just did a new install of it today, and allowed >> it to auto update. There are indeed some changes that I see from my install >> from about 4 months ago. So that is a positive thing, but its too bad that >> the feel is still lacking. I took a look at some code, and what I saw did >> not impress me. Display code mixed right in with the logic. I did not look >> for more than about 30 seconds, but IMO it is not ?proper? application >> design. I am a big proponent of MVC architecture, and what I saw of >> fusionpbx code does not look all that impressive. I give credit where >> credit is due. The platform is functioning to an extent, most things >> actually work. I do realize that developing something of that size requires >> tons of effort. My ideal solution would be something driven by ExtJS and >> PHP on the backend. ExtJS provides the most advanced and rich javascript >> controls i have seen. Plus its sister product, sencha touch, would enable >> great admin functionality from a smartphone or tablet device. Here I go >> rambling on? I wish someone would develop an awesome GUI for freeswitch to >> help get it into the limelight where it deserves to be. If I had the time I >> would love to create a comprehensive and great looking GUI.**** >> >> ** ** >> >> *Brett Wilson* >> >> *IT Department* >> >> *Launch 3 Ventures, LLC***** >> >> 134 Myer Street**** >> >> Hackensack, NJ 07601**** >> >> *Phone:* 877.878.9134 >> *Fax:* 646.536.3866**** >> >> *Email:* Brett.Wilson at launch3.net**** >> >> *AOL IM:* Brett.Wilson at launch3.net**** >> >> www.Launch3.net**** >> >> *www.Launch3telecom.com ***** >> >> ******************************* >> >> [image: Description: Description: Blogger-logo][image: >> Description: Description: FaceBook-Logo][image: >> Description: Description: Twitter-Logo] >> **** >> >> ** ** >> >> *From:* Andrew Cassidy [mailto:andrew at cassidywebservices.co.uk] >> *Sent:* Wednesday, March 07, 2012 6:46 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] State of GUIs**** >> >> ** ** >> >> Look at http://www.freepybx.org although I didn't manage to get it >> installed. I do have fusion installed on a text box although personally I'm >> not keen on it.**** >> >> ** ** >> >> On 7 March 2012 08:35, Brett Wilson wrote:**** >> >> Hey guys,**** >> >> I was wondering if anyone had any info on the current state of FS guis? >> It seems that most of the gui projects (blue.box, fusionpbx) have been >> mostly abandoned in terms of development. I would like something simple for >> end users to self-administer if needed. The problem I have with fusionpbx >> is that config files are overwritten by whatever is in the database. If you >> hand-edit a config file, fusion will not parse the file and load those >> settings into the interface. I realize that takes much more coding to do >> than using a database to simply write config files. But I feel that the >> freeswitch interface could be improved anyway. Unfortunately it seems that >> project has been abandoned. There have not been any releases since mid-2011. >> **** >> >> **** >> >> Anything new out or around the corner?**** >> >> **** >> >> *Brett Wilson***** >> >> *IT Department***** >> >> *Launch 3 Ventures, LLC***** >> >> 134 Myer Street**** >> >> Hackensack, NJ 07601**** >> >> *Phone:* 877.878.9134 >> *Fax:* 646.536.3866**** >> >> *Email:* Brett.Wilson at launch3.net**** >> >> *AOL IM:* Brett.Wilson at launch3.net**** >> >> www.Launch3.net**** >> >> *www.Launch3telecom.com ***** >> >> ******************************* >> >> [image: Description: Description: Blogger-logo][image: >> Description: Description: FaceBook-Logo][image: >> Description: Description: Twitter-Logo] >> **** >> >> **** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> >> >> **** >> >> ** ** >> >> -- >> Andrew Cassidy BSc (Hons) MBCS**** >> >> Managing Director; Cassidy Web Services Ltd**** >> >> T: 03300 100 960 F: 03300 100 961**** >> >> E: andrew at cassidywebservices.co.uk**** >> >> W: www.cassidywebservices.co.uk**** >> >> ** ** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/jpeg Size: 1815 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/32f41482/attachment-0005.jpe From admin at blindi.net Fri Mar 9 18:14:56 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Fri, 9 Mar 2012 16:14:56 +0100 (CET) Subject: [Freeswitch-users] Fs problem: can.t send queue_dtmf in loopback ignore_early_media=true not working In-Reply-To: References: <4F573CF6.9000205@softnet.si> <4F575B54.3050403@softnet.si> Message-ID: Hi Michael, > Why do you need to use loopback? I have a Dialplan to route originated / DISA calls to different gateways. For example: 02 - 09 beginning areacodes to german Landlines. (landlinegateway) 032 is the Areacode for Voip-numers, "Voipnumbersgateway) 00 x is the longdistanceprifix, (internationalgateway) and 015 - 017 beginning all cellphone numbers. (cellphonegateway) I route these prefixes to different gateways. The problem: Unfortunately, I can only use loopback so the calls are distributed over these gateways. Due to cost, I can not run anything over a gateway. -------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From vipkilla at gmail.com Fri Mar 9 18:42:01 2012 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 9 Mar 2012 10:42:01 -0500 Subject: [Freeswitch-users] State of GUIs In-Reply-To: References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> Message-ID: I'm currently developing a multi-tenant scalable FS GUI based on the PHP Framework YII and intralanman's PHP CURL library. The main objective is to use XML_CURL to configure as much as possible. I plan to implement a "snapshot" feature which will write all the XML to files and load it into FS incase XML_CURL fails. Also, it is designed around a "routing" table so FS does not load the entire dialplan from XML_CURL each time there is a call transaction. Instead the dialed number or call will match to a route in the database and only give FS the XML it needs (which usually ends up being only 5-10 lines of XML). I plan to put it on GITHUB when I get further along. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/2ec2de9d/attachment.html From andrew at cassidywebservices.co.uk Fri Mar 9 19:04:09 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Fri, 9 Mar 2012 16:04:09 +0000 Subject: [Freeswitch-users] State of GUIs In-Reply-To: References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> Message-ID: As with a number of other people, I'm writing one to suit my needs too. Mine is being written in Python/Django and is aimed at being multi tenant in a cloud setting or single user on a manged device setting sharing as much of the code from both scenarios as possible. I'm working on using a lot of jQuery in the user interface and am currently using MySQL as the data store although that is changable. The Django application also produces files for XML CURL, and is aimed at only basic functionality, so only adding users, extensions, groups, dids, conferences and queues will be implemented. On 9 March 2012 15:42, Vik Killa wrote: > I'm currently developing a multi-tenant scalable FS GUI based on the PHP > Framework YII and intralanman's PHP CURL library. The main objective is to > use XML_CURL to configure as much as possible. I plan to implement a > "snapshot" feature which will write all the XML to files and load it into > FS incase XML_CURL fails. Also, it is designed around a "routing" table so > FS does not load the entire dialplan from XML_CURL each time there is a > call transaction. Instead the dialed number or call will match to a route > in the database and only give FS the XML it needs (which usually ends up > being only 5-10 lines of XML). I plan to put it on GITHUB when I get > further along. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/65f92c99/attachment.html From jmesquita at freeswitch.org Fri Mar 9 19:05:20 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Fri, 9 Mar 2012 13:05:20 -0300 Subject: [Freeswitch-users] session-api - How to continue with A-leg after bridge-timeout? In-Reply-To: References: Message-ID: <748938CE1BF2401E86D9253FAB16066D@freeswitch.org> I honest believe that you won't be able to accomplish what you want like that. I don't think you can modify the core state machine from a script and by the time you get the control back on the script, the channel is already gone because of the att_xfer inner working? But I am really not an expert and I can barely understand how the att_xfer code works. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday, March 9, 2012 at 9:13 AM, Christian Benke wrote: > *sigh* No answer yet :-( > > Was my explanation comprehensive? > > On 8 March 2012 19:31, Christian Benke wrote: > > Hello! > > > > I have written a DB-based routing-script to be able to configure > > complex and dynamic call-behaviours but i'm currently stuck with > > scenarious where different call-channels are merged(att_xfer, > > intercept, et al), as my script fails with these scenarious. > > > > I'm testing using a simplified version of the script(nxo_test.py) to > > understand the mechanism behind those scenarious and now i'm at a > > point where i no longer know the proper tools to handle these > > situations. Here's what is happening: > > > > This is the test-script in pseudocode: > > > > > SRC=A > > > DEST=B > > > log("Test1") > > > sleep(1000) > > > log("Test2") > > > sleep(1000) > > > log("Test3") > > > bridge([origination_caller_id_number="${SRC}"]user/${DEST}@${domain_name}) > > > log("Test4") > > > sleep(1000) > > > "log" "Test5" > > > sleep(1000) > > > "log" "Test6" > > > > > > > > > A dials B. The call enters the default context where the test-script > > is called, the first steps are executed and the bridge is initiated. > > > > B picks up the call, the following (truncated) channel-uuids exist at > > this point: > > "a7c4"(A->) > > "a7cc"(->B) > > > > B then initiates an attended transfer to C, which follows the same > > path and script as the first call(But the call is bridged to C instead > > of B this time). Two new channels are set up - uuid's "a7d5"(B->) and > > "a7dd"(->C) > > > > B immediately transfers the call before C has picked up. > > > > Channel "a7cc"(->B) is killed by att_xfer while Channel "a7c4"(A->) > > finishes the script(The remaining Steps Test3-Test5 are executed) and > > is then automatically parked in endless_playback by att_xfer, waiting > > for C to pick up. > > > > C lets it ring until the timeout of 15s ends, "a7c4" and "a7dd" are > > hung up while a7d5 continues as a ZOMBIE-channel, trying to execute > > the remaining steps in the script(The "sleep"-commands fail due to the > > zombie-state). > > > > And this is where i'm stuck - i would like channel "a7c4" to not get > > hung up but to continue with the steps which "a7d5" could no longer > > execute. Is there some way to prevent a7c4 from getting hung up when > > the ATTENDED_TRANSFER to C fails? Algorithmically i can figure out how > > the remaining steps can get memorised and executed, i just don't know > > how to prevent "a7c4" from getting hung up and beeing transfer to a > > place where i can still work with it :-/ I can't use the HangupHook > > either, as a7c4 is already in zombie-state at that point. > > > > For reference, i've attached the test-script and a freeswitch-log of > > the call above. > > > > I hope someone can lead me on the right track... > > > > Best regards > > Christian > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/b88e3c22/attachment.html From nasida at live.ru Fri Mar 9 19:08:45 2012 From: nasida at live.ru (Yuriy Nasida) Date: Fri, 9 Mar 2012 20:08:45 +0400 Subject: [Freeswitch-users] reserve mysql database using odbc in the core In-Reply-To: References: <1FFF97C269757C458224B7C895F35F150505D4@cantor.std.visionutv.se>, Message-ID: Ok, I have recieved really good advice in IRC chat from Andee. "install mysqlproxy on the freeswitch box, tell odbc to look at localhost and configure mysqlproxy to look to the two actual mysql servers+ ip failover and as long as the replication is live you can failover without dropping calls" So if FS1 will down but mysql1 is live, I will save all current calls. If full box1 down, I will save some part of current calls (it is depends from replication timeout) So I plan to install two mysqlproxy on each box.Thus each FS will have the main and the reserve mysql server. I will continue and upgrade this thread. Thanks. From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Fri, 9 Mar 2012 18:03:00 +0400 Subject: Re: [Freeswitch-users] reserve mysql database using odbc in the core Peter, Thanks for your answer. Yes I know about using MySQL + DRBD + heartbeat but I search the best way.I plan to use 2 box only. each box will have (FS+mysql). Both FS has to use same mysql. It is necessary for feature - the recovering of current calls in FS2 if FS1 (i mean application level only, i.e. not full box1) will be broken. http://wiki.freeswitch.org/wiki/Freeswitch_HA If full box1 will broken, FS2 from box2 has to use own mysql server (mysql2) which will be synchronized with mysql1 from box1 (by means of mysql replication). I will lose current calls but new calls will work fine. I am not sure if it is possible. So I search possibility of using of backup mysql server if main server down. It may looks like second DNS or second SIP-proxy. I just need to put the second ip address of the standby mysql server on FS2. So FS2 will have two IP address of mysql servers: 1) ip adress of mysql1 (box1) 2) own ip. Thanks. From: peter.olsson at visionutveckling.se To: freeswitch-users at lists.freeswitch.org Date: Fri, 9 Mar 2012 12:32:11 +0000 Subject: Re: [Freeswitch-users] reserve mysql database using odbc in the core You need to setup MySQL to be able to failover. Most people use MySQL + DRBD + heartbeat for this kind of operation. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Yuriy Nasida Skickat: den 9 mars 2012 12:45 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] reserve mysql database using odbc in the core Hello guys! If anybody know how can I use reserve mysql database using odbc in the core? So if my first mysql server was down I would like that FS would use second mysql server for core. http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core Thanks. !DSPAM:4f59ec5e32765716517795! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/8cd22fa9/attachment-0001.html From lloyd.aloysius at gmail.com Fri Mar 9 19:45:04 2012 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Fri, 9 Mar 2012 11:45:04 -0500 Subject: [Freeswitch-users] reserve mysql database using odbc in the core In-Reply-To: References: <1FFF97C269757C458224B7C895F35F150505D4@cantor.std.visionutv.se> Message-ID: I have similar setup, But two freeswitch server's and two mysql server's freeswitch server 1 & server 2 connect mysql server1 all the time. My question is if freeswitch server 1 down. How to recover the calls from freeswitch 2? What are the settings ? How reliable this solution ... there is IP , Ports involves in the call path? Thanks Lloyd * * On Fri, Mar 9, 2012 at 11:08 AM, Yuriy Nasida wrote: > Ok, I have recieved really good advice in IRC chat from Andee. > > "install mysqlproxy on the freeswitch box, tell odbc to look at localhost > and configure mysqlproxy to look to the two actual mysql servers > + ip failover and as long as the replication is live you can failover > without dropping calls" > > So if FS1 will down but mysql1 is live, I will save all current calls. If > full box1 down, I will save some part of current calls (it is depends from > replication timeout) > > So I plan to install two mysqlproxy on each box. > Thus each FS will have the main and the reserve mysql server. > > I will continue and upgrade this thread. > > Thanks. > > ------------------------------ > From: nasida at live.ru > To: freeswitch-users at lists.freeswitch.org > Date: Fri, 9 Mar 2012 18:03:00 +0400 > > Subject: Re: [Freeswitch-users] reserve mysql database using odbc in the > core > > Peter, > > Thanks for your answer. Yes I know about using MySQL + DRBD + heartbeat > but I search the best way. > I plan to use 2 box only. each box will have (FS+mysql). Both FS has to > use same mysql. It is necessary for feature - the recovering of current > calls in FS2 if FS1 (i mean application level only, i.e. not full box1) > will be broken. http://wiki.freeswitch.org/wiki/Freeswitch_HA > > If full box1 will broken, FS2 from box2 has to use own mysql server > (mysql2) which will be synchronized with mysql1 from box1 (by means of > mysql replication). I will lose current calls but new calls will work fine. > > I am not sure if it is possible. So I search possibility of using of > backup mysql server if main server down. It may looks like second DNS or > second SIP-proxy. I just need to put the second ip address of the standby > mysql server on FS2. So FS2 will have two IP address of mysql servers: 1) > ip adress of mysql1 (box1) 2) own ip. > > > Thanks. > > ------------------------------ > From: peter.olsson at visionutveckling.se > To: freeswitch-users at lists.freeswitch.org > Date: Fri, 9 Mar 2012 12:32:11 +0000 > Subject: Re: [Freeswitch-users] reserve mysql database using odbc in the > core > > You need to setup MySQL to be able to failover. Most people use MySQL + > DRBD + heartbeat for this kind of operation. > > > > /Peter > > > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Yuriy Nasida > *Skickat:* den 9 mars 2012 12:45 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* [Freeswitch-users] reserve mysql database using odbc in the core > > > > Hello guys! > > > > If anybody know how can I use reserve mysql database using odbc in the > core? So if my first mysql server was down I would like that FS would use > second mysql server for core. > > > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core > > > > Thanks. > > !DSPAM:4f59ec5e32765716517795! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/e078cc50/attachment.html From freeswitch at simpot.com Fri Mar 9 19:46:58 2012 From: freeswitch at simpot.com (Dmitry Saratsky) Date: Fri, 9 Mar 2012 18:46:58 +0200 Subject: [Freeswitch-users] Error installing mod com g729 Message-ID: <000001ccfe14$3f421c40$bdc654c0$@com> Hi all, I have an issue similar to this: http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070981.htm l Currently I work successfully with: fsg729-167-installer After I run: fsg729-194-installer and then /usr/local/freeswitch/bin/validator - I got: 18:37 fs:/usr/src/g729# /usr/local/freeswitch/bin/validator G.729A licencing tool You will require one or more sales codes to activate G.729A Enter a sales code, or a blank line to end: Enter a sales code, or a blank line to end: Sales codes to be activated: OK (Y/N)? y ERROR [(null)] 18:37 fs:/usr/src/g729# And Null size of licence.zip: -rw-r--r-- 1 root root 0 Mar 9 18:37 licences.zip If I run again: fsg729-167-installer - All works fine. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/dce856a8/attachment.html From benkokakao at gmail.com Fri Mar 9 20:21:09 2012 From: benkokakao at gmail.com (Christian Benke) Date: Fri, 9 Mar 2012 18:21:09 +0100 Subject: [Freeswitch-users] session-api - How to continue with A-leg after bridge-timeout? In-Reply-To: <748938CE1BF2401E86D9253FAB16066D@freeswitch.org> References: <748938CE1BF2401E86D9253FAB16066D@freeswitch.org> Message-ID: On 9 March 2012 17:05, Jo?o Mesquita wrote: > I honest believe that you won't be able to accomplish what you want like > that. I don't think you can modify the core state machine from a script and > by the time you get the control back on the script, the channel is already > gone because of the att_xfer inner working? Hmm, what i don't understand - when i rebuild the same route-logic in XML, the initiating channel is not hung up but continues the steps in the extension it has been transfered to - which is exactly what i'm trying to achieve. See attached XML-dialplan. Can someone with a better understand of the core_state_machine please shed some light on this? Regards, Christian -------------- next part -------------- From nasida at live.ru Fri Mar 9 20:21:32 2012 From: nasida at live.ru (Yuriy Nasida) Date: Fri, 9 Mar 2012 21:21:32 +0400 Subject: [Freeswitch-users] reserve mysql database using odbc in the core In-Reply-To: References: <1FFF97C269757C458224B7C895F35F150505D4@cantor.std.visionutv.se>, , , Message-ID: FS can write the date about all current calls in special db. Both FS can use this db.I think it will help you.http://wiki.freeswitch.org/wiki/Freeswitch_HA From: lloyd.aloysius at gmail.com Date: Fri, 9 Mar 2012 11:45:04 -0500 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] reserve mysql database using odbc in the core I have similar setup, But two freeswitch server's and two mysql server's freeswitch server 1 & server 2 connect mysql server1 all the time. My question is if freeswitch server 1 down. How to recover the calls from freeswitch 2? What are the settings ? How reliable this solution ... there is IP , Ports involves in the call path? ThanksLloyd On Fri, Mar 9, 2012 at 11:08 AM, Yuriy Nasida wrote: Ok, I have recieved really good advice in IRC chat from Andee. "install mysqlproxy on the freeswitch box, tell odbc to look at localhost and configure mysqlproxy to look to the two actual mysql servers + ip failover and as long as the replication is live you can failover without dropping calls" So if FS1 will down but mysql1 is live, I will save all current calls. If full box1 down, I will save some part of current calls (it is depends from replication timeout) So I plan to install two mysqlproxy on each box.Thus each FS will have the main and the reserve mysql server. I will continue and upgrade this thread. Thanks. From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Fri, 9 Mar 2012 18:03:00 +0400 Subject: Re: [Freeswitch-users] reserve mysql database using odbc in the core Peter, Thanks for your answer. Yes I know about using MySQL + DRBD + heartbeat but I search the best way.I plan to use 2 box only. each box will have (FS+mysql). Both FS has to use same mysql. It is necessary for feature - the recovering of current calls in FS2 if FS1 (i mean application level only, i.e. not full box1) will be broken. http://wiki.freeswitch.org/wiki/Freeswitch_HA If full box1 will broken, FS2 from box2 has to use own mysql server (mysql2) which will be synchronized with mysql1 from box1 (by means of mysql replication). I will lose current calls but new calls will work fine. I am not sure if it is possible. So I search possibility of using of backup mysql server if main server down. It may looks like second DNS or second SIP-proxy. I just need to put the second ip address of the standby mysql server on FS2. So FS2 will have two IP address of mysql servers: 1) ip adress of mysql1 (box1) 2) own ip. Thanks. From: peter.olsson at visionutveckling.se To: freeswitch-users at lists.freeswitch.org Date: Fri, 9 Mar 2012 12:32:11 +0000 Subject: Re: [Freeswitch-users] reserve mysql database using odbc in the core You need to setup MySQL to be able to failover. Most people use MySQL + DRBD + heartbeat for this kind of operation. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Yuriy Nasida Skickat: den 9 mars 2012 12:45 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] reserve mysql database using odbc in the core Hello guys! If anybody know how can I use reserve mysql database using odbc in the core? So if my first mysql server was down I would like that FS would use second mysql server for core. http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core Thanks. !DSPAM:4f59ec5e32765716517795! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/7ba39d07/attachment.html From benkokakao at gmail.com Fri Mar 9 20:51:30 2012 From: benkokakao at gmail.com (Christian Benke) Date: Fri, 9 Mar 2012 18:51:30 +0100 Subject: [Freeswitch-users] session-api - How to continue with A-leg after bridge-timeout? In-Reply-To: References: <748938CE1BF2401E86D9253FAB16066D@freeswitch.org> Message-ID: > Hmm, what i don't understand - when i rebuild the same route-logic in > XML, the initiating channel is not hung up but continues the steps in > the extension it has been transfered to Attached is a log of this scenario, i've only added an additional "info" to the XML-dialplan posted before. The initial channel bf969cf6-6a0f-11e1-8e5c-299021af10b6 is still there at the very end of the transfered call - there must be some way to get the same result in my script-based routing and be able to reroute this initial channel? Regards, Christian -------------- next part -------------- ------------------------------------------------------------------------ recv 864 bytes from udp/[10.3.0.22]:5060 at 17:46:06.881969: ------------------------------------------------------------------------ INVITE sip:820 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bKb780533fA6980116 From: "A" ;tag=A59BE1A5-8A2B1314 To: CSeq: 1 INVITE Call-ID: 73ec8749-ef90cb48-95fd5843 at 10.3.0.22 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 241 v=0 o=- 1167610354 1167610354 IN IP4 10.3.0.22 s=Polycom IP Phone c=IN IP4 10.3.0.22 t=0 0 a=sendrecv m=audio 2226 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ------------------------------------------------------------------------ send 335 bytes to udp/[10.3.0.22]:5060 at 17:46:06.883097: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bKb780533fA6980116 From: "A" ;tag=A59BE1A5-8A2B1314 To: Call-ID: 73ec8749-ef90cb48-95fd5843 at 10.3.0.22 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:06.873921 [DEBUG] sofia.c:7567 IP 10.3.0.22 Rejected by acl "domains". Falling back to Digest auth. 2012-03-09 18:46:06.873921 [WARNING] sofia_reg.c:1422 SIP auth challenge (INVITE) on sofia profile 'internal' for [820 at 10.3.0.4] from ip 10.3.0.22 send 818 bytes to udp/[10.3.0.22]:5060 at 17:46:06.887952: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bKb780533fA6980116 From: "A" ;tag=A59BE1A5-8A2B1314 To: ;tag=v24X3rSHa9e9S Call-ID: 73ec8749-ef90cb48-95fd5843 at 10.3.0.22 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="10.3.0.4", nonce="bf913b3a-6a0f-11e1-8e5b-299021af10b6", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 538 bytes from udp/[10.3.0.22]:5060 at 17:46:06.903231: ------------------------------------------------------------------------ ACK sip:820 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bKb780533fA6980116 From: "A" ;tag=A59BE1A5-8A2B1314 To: ;tag=v24X3rSHa9e9S CSeq: 1 ACK Call-ID: 73ec8749-ef90cb48-95fd5843 at 10.3.0.22 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ recv 1120 bytes from udp/[10.3.0.22]:5060 at 17:46:06.910997: ------------------------------------------------------------------------ INVITE sip:820 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bK747f362a84831CAD From: "A" ;tag=A59BE1A5-8A2B1314 To: CSeq: 2 INVITE Call-ID: 73ec8749-ef90cb48-95fd5843 at 10.3.0.22 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="10", realm="10.3.0.4", nonce="bf913b3a-6a0f-11e1-8e5b-299021af10b6", qop=auth, cnonce="+FNV0YSM4tEnYzr", nc=00000001, uri="sip:820 at 10.3.0.4:5060;user=phone", response="75f359abb81875ac5b6b9a587665ce8d", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 241 v=0 o=- 1167610354 1167610354 IN IP4 10.3.0.22 s=Polycom IP Phone c=IN IP4 10.3.0.22 t=0 0 a=sendrecv m=audio 2226 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ------------------------------------------------------------------------ send 335 bytes to udp/[10.3.0.22]:5060 at 17:46:06.912114: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bK747f362a84831CAD From: "A" ;tag=A59BE1A5-8A2B1314 To: Call-ID: 73ec8749-ef90cb48-95fd5843 at 10.3.0.22 CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:06.913289 [DEBUG] sofia.c:7567 IP 10.3.0.22 Rejected by acl "domains". Falling back to Digest auth. 2012-03-09 18:46:06.913289 [NOTICE] switch_channel.c:926 New Channel sofia/internal/10 at 10.3.0.4 [bf969cf6-6a0f-11e1-8e5c-299021af10b6] 2012-03-09 18:46:06.913289 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_NEW 2012-03-09 18:46:06.913289 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/10 at 10.3.0.4) State NEW 2012-03-09 18:46:06.913289 [DEBUG] sofia.c:5532 Channel sofia/internal/10 at 10.3.0.4 entering state [received][100] 2012-03-09 18:46:06.913289 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=- 1167610354 1167610354 IN IP4 10.3.0.22 s=Polycom IP Phone c=IN IP4 10.3.0.22 t=0 0 a=sendrecv m=audio 2226 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 2012-03-09 18:46:06.913289 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 2012-03-09 18:46:06.913289 [DEBUG] sofia_glue.c:2991 Set Codec sofia/internal/10 at 10.3.0.4 G722/8000 20 ms 160 samples 64000 bits 2012-03-09 18:46:06.913289 [DEBUG] switch_core_codec.c:111 sofia/internal/10 at 10.3.0.4 Original read codec set to G722:9 2012-03-09 18:46:06.913289 [DEBUG] sofia_glue.c:4995 Set 2833 dtmf send/recv payload to 127 2012-03-09 18:46:06.913289 [DEBUG] sofia.c:5757 (sofia/internal/10 at 10.3.0.4) State Change CS_NEW -> CS_INIT 2012-03-09 18:46:06.913289 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:06.913289 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_INIT 2012-03-09 18:46:06.913289 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/10 at 10.3.0.4) State INIT 2012-03-09 18:46:06.913289 [DEBUG] mod_sofia.c:85 sofia/internal/10 at 10.3.0.4 SOFIA INIT 2012-03-09 18:46:06.913289 [DEBUG] mod_sofia.c:125 (sofia/internal/10 at 10.3.0.4) State Change CS_INIT -> CS_ROUTING 2012-03-09 18:46:06.913289 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:06.913289 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/10 at 10.3.0.4) State INIT going to sleep 2012-03-09 18:46:06.913289 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_ROUTING 2012-03-09 18:46:06.913289 [DEBUG] switch_channel.c:1886 (sofia/internal/10 at 10.3.0.4) Callstate Change DOWN -> RINGING 2012-03-09 18:46:06.913289 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/10 at 10.3.0.4) State ROUTING 2012-03-09 18:46:06.913289 [DEBUG] mod_sofia.c:148 sofia/internal/10 at 10.3.0.4 SOFIA ROUTING 2012-03-09 18:46:06.913289 [DEBUG] switch_core_state_machine.c:104 sofia/internal/10 at 10.3.0.4 Standard ROUTING 2012-03-09 18:46:06.913289 [INFO] mod_dialplan_xml.c:485 Processing A <10>->820 in context default Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->test1] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (FAIL) [test1] destination_number(820) =~ /810/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 parsing [default->test3] continue=false Dialplan: sofia/internal/10 at 10.3.0.4 Regex (PASS) [test3] destination_number(820) =~ /820/ break=on-false Dialplan: sofia/internal/10 at 10.3.0.4 Action info() Dialplan: sofia/internal/10 at 10.3.0.4 Action log(WARNING TEST1 for UUID ${uuid}, src ${caller_id_number}, dest ${destination_number}) Dialplan: sofia/internal/10 at 10.3.0.4 Action sleep(1000) Dialplan: sofia/internal/10 at 10.3.0.4 Action log(WARNING TEST2 for UUID ${uuid}, src ${caller_id_number}, dest ${destination_number}) Dialplan: sofia/internal/10 at 10.3.0.4 Action sleep(1000) Dialplan: sofia/internal/10 at 10.3.0.4 Action log(WARNING TEST3 for UUID ${uuid}, src ${caller_id_number}, dest ${destination_number}) Dialplan: sofia/internal/10 at 10.3.0.4 Action sleep(1000) Dialplan: sofia/internal/10 at 10.3.0.4 Action bridge([leg_timeout=10][origination_caller_id_number=${caller_id_number}]user/20@${domain_name}) Dialplan: sofia/internal/10 at 10.3.0.4 Action info() Dialplan: sofia/internal/10 at 10.3.0.4 Action log(WARNING TEST4 for UUID ${uuid}, src ${caller_id_number}, dest ${destination_number}) Dialplan: sofia/internal/10 at 10.3.0.4 Action sleep(1000) Dialplan: sofia/internal/10 at 10.3.0.4 Action log(WARNING TEST5 for UUID ${uuid}, src ${caller_id_number}, dest ${destination_number}) Dialplan: sofia/internal/10 at 10.3.0.4 Action sleep(1000) Dialplan: sofia/internal/10 at 10.3.0.4 Action log(WARNING TEST6 for UUID ${uuid}, src ${caller_id_number}, dest ${destination_number}) Dialplan: sofia/internal/10 at 10.3.0.4 Action sleep(1000) 2012-03-09 18:46:06.934453 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/10 at 10.3.0.4) State Change CS_ROUTING -> CS_EXECUTE 2012-03-09 18:46:06.934453 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:06.934453 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/10 at 10.3.0.4) State ROUTING going to sleep 2012-03-09 18:46:06.934453 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_EXECUTE 2012-03-09 18:46:06.934453 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/10 at 10.3.0.4) State EXECUTE 2012-03-09 18:46:06.934453 [DEBUG] mod_sofia.c:241 sofia/internal/10 at 10.3.0.4 SOFIA EXECUTE 2012-03-09 18:46:06.934453 [DEBUG] switch_core_state_machine.c:192 sofia/internal/10 at 10.3.0.4 Standard EXECUTE EXECUTE sofia/internal/10 at 10.3.0.4 info() 2012-03-09 18:46:06.934453 [INFO] mod_dptools.c:1439 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [RINGING] Channel-State-Number: [4] Channel-Name: [sofia/internal/10 at 10.3.0.4] Unique-ID: [bf969cf6-6a0f-11e1-8e5c-299021af10b6] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-HIT-Dialplan: [true] Channel-Presence-ID: [10 at 10.3.0.4] Channel-Call-UUID: [bf969cf6-6a0f-11e1-8e5c-299021af10b6] Answer-State: [ringing] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Read-Codec-Bit-Rate: [64000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Channel-Write-Codec-Bit-Rate: [64000] Caller-Direction: [inbound] Caller-Username: [10] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [A] Caller-Caller-ID-Number: [10] Caller-Network-Addr: [10.3.0.22] Caller-ANI: [10] Caller-Destination-Number: [820] Caller-Unique-ID: [bf969cf6-6a0f-11e1-8e5c-299021af10b6] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/10 at 10.3.0.4] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1331315166913289] Caller-Channel-Created-Time: [1331315166913289] 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] variable_direction: [inbound] variable_uuid: [bf969cf6-6a0f-11e1-8e5c-299021af10b6] variable_session_id: [181] variable_sip_local_network_addr: [10.3.0.4] variable_sip_network_ip: [10.3.0.22] variable_sip_network_port: [5060] variable_sip_received_ip: [10.3.0.22] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_authorized: [true] variable_sip_number_alias: [10] variable_sip_auth_username: [10] variable_sip_auth_realm: [10.3.0.4] variable_number_alias: [10] variable_user_name: [10] variable_domain_name: [10.3.0.4] variable_record_stereo: [true] variable_default_gateway: [fonira] variable_default_areacode: [01] variable_transfer_fallback_extension: [operator] variable_toll_allow: [local,domestic,international,vas] variable_accountcode: [10] variable_user_context: [default] variable_effective_caller_id_name: [A ] variable_effective_caller_id_number: [10] variable_outbound_caller_id_name: [A ] variable_outbound_caller_id_number: [1997156010] variable_callgroup: [intercept] variable_sip_from_user: [10] variable_sip_from_uri: [10 at 10.3.0.4] variable_sip_from_host: [10.3.0.4] variable_sip_from_user_stripped: [10] variable_sip_from_tag: [A59BE1A5-8A2B1314] variable_sofia_profile_name: [internal] variable_sip_full_via: [SIP/2.0/UDP 10.3.0.22;branch=z9hG4bK747f362a84831CAD] variable_sip_from_display: [A] variable_sip_full_from: ["A" ;tag=A59BE1A5-8A2B1314] variable_sip_full_to: [] variable_sip_req_params: [user=phone] variable_sip_req_user: [820] variable_sip_req_port: [5060] variable_sip_req_uri: [820 at 10.3.0.4:5060] variable_sip_req_host: [10.3.0.4] variable_sip_to_params: [user=phone] variable_sip_to_user: [820] variable_sip_to_uri: [820 at 10.3.0.4] variable_sip_to_host: [10.3.0.4] variable_sip_contact_user: [10] variable_sip_contact_uri: [10 at 10.3.0.22] variable_sip_contact_host: [10.3.0.22] variable_channel_name: [sofia/internal/10 at 10.3.0.4] variable_sip_call_id: [73ec8749-ef90cb48-95fd5843 at 10.3.0.22] variable_sip_user_agent: [PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933] variable_sip_via_host: [10.3.0.22] variable_max_forwards: [70] variable_presence_id: [10 at 10.3.0.4] variable_switch_r_sdp: [v=0 o=- 1167610354 1167610354 IN IP4 10.3.0.22 s=Polycom IP Phone c=IN IP4 10.3.0.22 t=0 0 a=sendrecv m=audio 2226 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ] variable_remote_media_ip: [10.3.0.22] variable_remote_media_port: [2226] variable_sip_audio_recv_pt: [9] variable_sip_use_codec_name: [G722] variable_sip_use_codec_rate: [8000] variable_sip_use_codec_ptime: [20] variable_read_codec: [G722] variable_read_rate: [16000] variable_write_codec: [G722] variable_write_rate: [16000] variable_endpoint_disposition: [RECEIVED] variable_call_uuid: [bf969cf6-6a0f-11e1-8e5c-299021af10b6] variable_current_application: [info] EXECUTE sofia/internal/10 at 10.3.0.4 log(WARNING TEST1 for UUID bf969cf6-6a0f-11e1-8e5c-299021af10b6, src 10, dest 820) 2012-03-09 18:46:06.934453 [WARNING] mod_dptools.c:1420 TEST1 for UUID bf969cf6-6a0f-11e1-8e5c-299021af10b6, src 10, dest 820 EXECUTE sofia/internal/10 at 10.3.0.4 sleep(1000) EXECUTE sofia/internal/10 at 10.3.0.4 log(WARNING TEST2 for UUID bf969cf6-6a0f-11e1-8e5c-299021af10b6, src 10, dest 820) 2012-03-09 18:46:07.932932 [WARNING] mod_dptools.c:1420 TEST2 for UUID bf969cf6-6a0f-11e1-8e5c-299021af10b6, src 10, dest 820 EXECUTE sofia/internal/10 at 10.3.0.4 sleep(1000) EXECUTE sofia/internal/10 at 10.3.0.4 log(WARNING TEST3 for UUID bf969cf6-6a0f-11e1-8e5c-299021af10b6, src 10, dest 820) 2012-03-09 18:46:08.913902 [WARNING] mod_dptools.c:1420 TEST3 for UUID bf969cf6-6a0f-11e1-8e5c-299021af10b6, src 10, dest 820 EXECUTE sofia/internal/10 at 10.3.0.4 sleep(1000) EXECUTE sofia/internal/10 at 10.3.0.4 bridge([leg_timeout=10][origination_caller_id_number=10]user/20 at 10.3.0.4) 2012-03-09 18:46:09.933915 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-09 18:46:09.933915 [DEBUG] switch_ivr_originate.c:2299 Parsing session specific variables 2012-03-09 18:46:09.933915 [DEBUG] switch_event.c:1522 Parsing variable [leg_timeout]=[10] 2012-03-09 18:46:09.933915 [DEBUG] switch_event.c:1522 Parsing variable [origination_caller_id_number]=[10] 2012-03-09 18:46:09.933915 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-09 18:46:09.933915 [DEBUG] switch_event.c:1522 Parsing variable [presence_id]=[20 at 10.3.0.4] 2012-03-09 18:46:09.933915 [NOTICE] switch_channel.c:926 New Channel sofia/internal/sip:20 at 10.3.0.25 [c1665710-6a0f-11e1-8e64-299021af10b6] 2012-03-09 18:46:09.933915 [DEBUG] mod_sofia.c:4691 (sofia/internal/sip:20 at 10.3.0.25) State Change CS_NEW -> CS_INIT 2012-03-09 18:46:09.933915 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:09.933915 [DEBUG] switch_ivr_originate.c:2561 sofia/internal/sip:20 at 10.3.0.25 Setting leg timeout to 10 2012-03-09 18:46:09.933915 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:20 at 10.3.0.25) Running State Change CS_INIT 2012-03-09 18:46:09.933915 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:20 at 10.3.0.25) State INIT 2012-03-09 18:46:09.933915 [DEBUG] mod_sofia.c:85 sofia/internal/sip:20 at 10.3.0.25 SOFIA INIT 2012-03-09 18:46:09.933915 [DEBUG] mod_sofia.c:125 (sofia/internal/sip:20 at 10.3.0.25) State Change CS_INIT -> CS_ROUTING 2012-03-09 18:46:09.933915 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:09.933915 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:20 at 10.3.0.25) State INIT going to sleep 2012-03-09 18:46:09.933915 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:20 at 10.3.0.25) Running State Change CS_ROUTING 2012-03-09 18:46:09.933915 [DEBUG] switch_channel.c:1886 (sofia/internal/sip:20 at 10.3.0.25) Callstate Change DOWN -> RINGING send 1255 bytes to udp/[10.3.0.25]:5060 at 17:46:09.964563: ------------------------------------------------------------------------ INVITE sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK6rSXB0ZDat57m Max-Forwards: 69 From: "A " ;tag=ymQF7eUr4tUeH To: Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 CSeq: 25321520 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 369 X-FS-Support: update_display,send_info Remote-Party-ID: "A " ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1331298751 1331298752 IN IP4 10.3.0.4 s=FreeSWITCH c=IN IP4 10.3.0.4 t=0 0 m=audio 16418 RTP/AVP 9 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 16404 RTP/AVP 98 99 100 34 31 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-2000/90000 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 ------------------------------------------------------------------------ 2012-03-09 18:46:09.933915 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:20 at 10.3.0.25) State ROUTING 2012-03-09 18:46:09.933915 [DEBUG] mod_sofia.c:148 sofia/internal/sip:20 at 10.3.0.25 SOFIA ROUTING 2012-03-09 18:46:09.933915 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:20 at 10.3.0.25) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-03-09 18:46:09.933915 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:09.933915 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:20 at 10.3.0.25) State ROUTING going to sleep 2012-03-09 18:46:09.933915 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:20 at 10.3.0.25) Running State Change CS_CONSUME_MEDIA 2012-03-09 18:46:09.933915 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:20 at 10.3.0.25) State CONSUME_MEDIA 2012-03-09 18:46:09.933915 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:20 at 10.3.0.25) State CONSUME_MEDIA going to sleep 2012-03-09 18:46:09.933915 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:09.933915 [DEBUG] sofia.c:5532 Channel sofia/internal/sip:20 at 10.3.0.25 entering state [calling][0] recv 405 bytes from udp/[10.3.0.25]:5060 at 17:46:09.975584: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK6rSXB0ZDat57m From: "A" ;tag=ymQF7eUr4tUeH To: "B" ;tag=C3930705-7AD5A1D0 CSeq: 25321520 INVITE Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 442 bytes from udp/[10.3.0.25]:5060 at 17:46:10.037810: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK6rSXB0ZDat57m From: "A" ;tag=ymQF7eUr4tUeH To: "B" ;tag=C3930705-7AD5A1D0 CSeq: 25321520 INVITE Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Allow-Events: talk,hold,conference Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:10.032935 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:10.032935 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:10.032935 [DEBUG] sofia.c:5532 Channel sofia/internal/sip:20 at 10.3.0.25 entering state [proceeding][180] 2012-03-09 18:46:10.032935 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/sip:20 at 10.3.0.25! 2012-03-09 18:46:10.032935 [NOTICE] mod_sofia.c:2514 Ring-Ready sofia/internal/10 at 10.3.0.4! send 801 bytes to udp/[10.3.0.22]:5060 at 17:46:10.049093: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bK747f362a84831CAD From: "A" ;tag=A59BE1A5-8A2B1314 To: ;tag=XByp5KaN7H5UN Call-ID: 73ec8749-ef90cb48-95fd5843 at 10.3.0.22 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 Remote-Party-ID: "Outbound Call" <20>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ 2012-03-09 18:46:10.032935 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:10.032935 [DEBUG] sofia.c:5532 Channel sofia/internal/10 at 10.3.0.4 entering state [early][180] 2012-03-09 18:46:10.032935 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:10.032935 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/internal/10 at 10.3.0.4! recv 907 bytes from udp/[10.3.0.25]:5060 at 17:46:11.270917: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK6rSXB0ZDat57m From: "A" ;tag=ymQF7eUr4tUeH To: "B" ;tag=C3930705-7AD5A1D0 CSeq: 25321520 INVITE Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Type: application/sdp Content-Length: 347 v=0 o=- 1167610359 1167610359 IN IP4 10.3.0.25 s=Polycom IP Phone c=IN IP4 10.3.0.25 t=0 0 m=audio 2230 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 m=video 0 RTP/AVP 98 99 100 34 31 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-2000/90000 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 ------------------------------------------------------------------------ 2012-03-09 18:46:11.253958 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:11.253958 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:11.253958 [DEBUG] sofia.c:5532 Channel sofia/internal/sip:20 at 10.3.0.25 entering state [completing][200] 2012-03-09 18:46:11.253958 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=- 1167610359 1167610359 IN IP4 10.3.0.25 s=Polycom IP Phone c=IN IP4 10.3.0.25 t=0 0 m=audio 2230 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 m=video 0 RTP/AVP 98 99 100 34 31 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-2000/90000 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 send 336 bytes to udp/[10.3.0.25]:5060 at 17:46:11.282795: ------------------------------------------------------------------------ ACK sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK71jpDUgH72Utg Max-Forwards: 70 From: "A " ;tag=ymQF7eUr4tUeH To: ;tag=C3930705-7AD5A1D0 Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 CSeq: 25321520 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:11.253958 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:11.253958 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:11.253958 [DEBUG] sofia.c:5532 Channel sofia/internal/sip:20 at 10.3.0.25 entering state [ready][200] 2012-03-09 18:46:11.253958 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 2012-03-09 18:46:11.253958 [DEBUG] sofia_glue.c:2991 Set Codec sofia/internal/sip:20 at 10.3.0.25 G722/8000 20 ms 160 samples 64000 bits 2012-03-09 18:46:11.253958 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:20 at 10.3.0.25 Original read codec set to G722:9 2012-03-09 18:46:11.253958 [DEBUG] sofia_glue.c:4988 Set 2833 dtmf send payload to 127 2012-03-09 18:46:11.253958 [DEBUG] sofia_glue.c:3243 AUDIO RTP [sofia/internal/sip:20 at 10.3.0.25] 10.3.0.4 port 16418 -> 10.3.0.25 port 2230 codec: 9 ms: 20 2012-03-09 18:46:11.253958 [DEBUG] switch_rtp.c:1661 Starting timer [soft] 160 bytes per 20ms 2012-03-09 18:46:11.253958 [DEBUG] sofia_glue.c:3507 Set 2833 dtmf send payload to 127 2012-03-09 18:46:11.253958 [DEBUG] sofia_glue.c:3513 Set 2833 dtmf receive payload to 101 2012-03-09 18:46:11.253958 [DEBUG] switch_channel.c:3190 (sofia/internal/sip:20 at 10.3.0.25) Callstate Change RINGING -> ACTIVE 2012-03-09 18:46:11.253958 [DEBUG] switch_channel.c:3202 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:11.253958 [NOTICE] sofia.c:6246 Channel [sofia/internal/sip:20 at 10.3.0.25] has been answered 2012-03-09 18:46:11.292944 [DEBUG] sofia_glue.c:3243 AUDIO RTP [sofia/internal/10 at 10.3.0.4] 10.3.0.4 port 16424 -> 10.3.0.22 port 2226 codec: 9 ms: 20 2012-03-09 18:46:11.292944 [DEBUG] switch_rtp.c:1661 Starting timer [soft] 160 bytes per 20ms 2012-03-09 18:46:11.292944 [DEBUG] sofia_glue.c:3507 Set 2833 dtmf send payload to 127 2012-03-09 18:46:11.292944 [DEBUG] sofia_glue.c:3513 Set 2833 dtmf receive payload to 127 2012-03-09 18:46:11.292944 [DEBUG] mod_sofia.c:750 Local SDP sofia/internal/10 at 10.3.0.4: v=0 o=FreeSWITCH 1331298747 1331298748 IN IP4 10.3.0.4 s=FreeSWITCH c=IN IP4 10.3.0.4 t=0 0 m=audio 16424 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2012-03-09 18:46:11.292944 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:11.292944 [DEBUG] switch_channel.c:3190 (sofia/internal/10 at 10.3.0.4) Callstate Change RINGING -> ACTIVE send 1073 bytes to udp/[10.3.0.22]:5060 at 17:46:11.302824: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bK747f362a84831CAD From: "A" ;tag=A59BE1A5-8A2B1314 To: ;tag=XByp5KaN7H5UN Call-ID: 73ec8749-ef90cb48-95fd5843 at 10.3.0.22 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces2012-03-09 18:46:11.292944 [NOTICE] switch_ivr_originate.c:3209 Channel [sofia/internal/10 at 10.3.0.4] has been answered Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 239 Remote-Party-ID: "Outbound Call" <20>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1331298747 1331298748 IN IP4 10.3.0.4 s=FreeSWITCH c=IN IP4 10.3.0.4 t=0 0 m=audio 16424 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2012-03-09 18:46:11.292944 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:11.292944 [DEBUG] sofia.c:5532 Channel sofia/internal/10 at 10.3.0.4 entering state [completed][200] 2012-03-09 18:46:11.292944 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [sofia/internal/sip:20 at 10.3.0.25] 2012-03-09 18:46:11.292944 [DEBUG] switch_ivr_originate.c:2561 sofia/internal/sip:20 at 10.3.0.25 Setting leg timeout to 10 2012-03-09 18:46:11.292944 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [sofia/internal/sip:20 at 10.3.0.25] 2012-03-09 18:46:11.292944 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:11.292944 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:11.292944 [DEBUG] switch_ivr_bridge.c:1328 (sofia/internal/sip:20 at 10.3.0.25) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2012-03-09 18:46:11.292944 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:11.292944 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:20 at 10.3.0.25) Running State Change CS_EXCHANGE_MEDIA 2012-03-09 18:46:11.292944 [DEBUG] switch_core_state_machine.c:420 (sofia/internal/sip:20 at 10.3.0.25) State EXCHANGE_MEDIA 2012-03-09 18:46:11.292944 [DEBUG] mod_sofia.c:578 SOFIA EXCHANGE_MEDIA recv 541 bytes from udp/[10.3.0.22]:5060 at 17:46:11.314619: ------------------------------------------------------------------------ ACK sip:820 at 10.3.0.4:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.3.0.22;branch=z9hG4bK6e693bfe5C6D05D1 From: "A" ;tag=A59BE1A5-8A2B1314 To: ;tag=XByp5KaN7H5UN CSeq: 2 ACK Call-ID: 73ec8749-ef90cb48-95fd5843 at 10.3.0.22 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:11.314245 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:11.314245 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:11.314245 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:11.352934 [DEBUG] sofia.c:5532 Channel sofia/internal/10 at 10.3.0.4 entering state [ready][200] 2012-03-09 18:46:11.352934 [DEBUG] switch_core_session.c:791 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:11.352934 [DEBUG] switch_core_session.c:791 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:11.412951 [DEBUG] switch_rtp.c:3210 Correct ip/port confirmed. 2012-03-09 18:46:11.412951 [DEBUG] switch_rtp.c:3210 Correct ip/port confirmed. recv 978 bytes from udp/[10.3.0.25]:5060 at 17:46:12.152357: ------------------------------------------------------------------------ INVITE sip:mod_sofia at 10.3.0.4:5060 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bKf57df0b37D8DB26E From: "B" ;tag=C3930705-7AD5A1D0 To: "A" ;tag=ymQF7eUr4tUeH CSeq: 1 INVITE Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 345 v=0 o=- 1167610359 1167610360 IN IP4 10.3.0.25 s=Polycom IP Phone c=IN IP4 0.0.0.0 t=0 0 m=audio 2230 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 m=video 0 RTP/AVP 98 99 100 34 31 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-2000/90000 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 ------------------------------------------------------------------------ send 350 bytes to udp/[10.3.0.25]:5060 at 17:46:12.153437: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bKf57df0b37D8DB26E From: "B" ;tag=C3930705-7AD5A1D0 To: "A" ;tag=ymQF7eUr4tUeH Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:12.153783 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:12.153783 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:12.155039 [DEBUG] sofia.c:5532 Channel sofia/internal/sip:20 at 10.3.0.25 entering state [received][100] 2012-03-09 18:46:12.155039 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=- 1167610359 1167610360 IN IP4 10.3.0.25 s=Polycom IP Phone c=IN IP4 0.0.0.0 t=0 0 m=audio 2230 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=sendonly m=video 0 RTP/AVP 98 99 100 34 31 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-2000/90000 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 2012-03-09 18:46:12.155039 [DEBUG] switch_channel.c:1560 (sofia/internal/sip:20 at 10.3.0.25) Callstate Change ACTIVE -> HELD 2012-03-09 18:46:12.155039 [DEBUG] switch_core_session.c:1012 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:12.192940 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:12.273953 [DEBUG] switch_ivr.c:591 sofia/internal/10 at 10.3.0.4 Command Execute playback(local_stream://moh) EXECUTE sofia/internal/10 at 10.3.0.4 playback(local_stream://moh) 2012-03-09 18:46:12.273953 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/16000] 16000hz 2012-03-09 18:46:12.273953 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16 at 16000hz 1 channels 20ms 2012-03-09 18:46:12.393911 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 2012-03-09 18:46:12.393911 [DEBUG] sofia_glue.c:2925 Already using G722 2012-03-09 18:46:12.393911 [DEBUG] sofia_glue.c:4988 Set 2833 dtmf send payload to 101 2012-03-09 18:46:12.393911 [DEBUG] sofia_glue.c:3232 Audio params changed for sofia/internal/sip:20 at 10.3.0.25 from 10.3.0.25:2230 to 0.0.0.0:2230 2012-03-09 18:46:12.393911 [DEBUG] sofia_glue.c:3243 AUDIO RTP [sofia/internal/sip:20 at 10.3.0.25] 10.3.0.4 port 16418 -> 0.0.0.0 port 2230 codec: 9 ms: 20 2012-03-09 18:46:12.393911 [DEBUG] sofia_glue.c:3284 AUDIO RTP CHANGING DEST TO: [0.0.0.0:2230] 2012-03-09 18:46:12.393911 [DEBUG] sofia.c:6042 Processing updated SDP send 929 bytes to udp/[10.3.0.25]:5060 at 17:46:12.409614: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bKf57df0b37D8DB26E From: "B" ;tag=C3930705-7AD5A1D0 To: "A" ;tag=ymQF7eUr4tUeH Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 CSeq: 1 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 297 v=0 o=FreeSWITCH 1331298751 1331298753 IN IP4 10.3.0.4 s=FreeSWITCH c=IN IP4 10.3.0.4 t=0 0 m=audio 16418 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=recvonly a=silenceSupp:off - - - - a=ptime:20 m=video 0 RTP/AVP 98 a=rtpmap:98 H264/90000 ------------------------------------------------------------------------ 2012-03-09 18:46:12.393911 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] recv 529 bytes from udp/[10.3.0.25]:5060 at 17:46:12.428189: ------------------------------------------------------------------------ ACK sip:mod_sofia at 10.3.0.4:5060 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bK262feca5B7DADFD From: "B" ;tag=C3930705-7AD5A1D0 To: "A" ;tag=ymQF7eUr4tUeH CSeq: 1 ACK Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:12.393911 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:12.393911 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:12.393911 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:12.432936 [DEBUG] sofia.c:5532 Channel sofia/internal/sip:20 at 10.3.0.25 entering state [completed][200] 2012-03-09 18:46:12.432936 [DEBUG] sofia.c:5532 Channel sofia/internal/sip:20 at 10.3.0.25 entering state [ready][200] recv 862 bytes from udp/[10.3.0.25]:5060 at 17:46:13.885657: ------------------------------------------------------------------------ INVITE sip:830 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bKdf353688FD5B112B From: "B" ;tag=FAB3376C-15AEB62F To: CSeq: 1 INVITE Call-ID: 12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 241 v=0 o=- 1167610362 1167610362 IN IP4 10.3.0.25 s=Polycom IP Phone c=IN IP4 10.3.0.25 t=0 0 a=sendrecv m=audio 2232 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ------------------------------------------------------------------------ send 333 bytes to udp/[10.3.0.25]:5060 at 17:46:13.886560: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bKdf353688FD5B112B From: "B" ;tag=FAB3376C-15AEB62F To: Call-ID: 12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:13.872937 [DEBUG] sofia.c:7567 IP 10.3.0.25 Rejected by acl "domains". Falling back to Digest auth. 2012-03-09 18:46:13.872937 [WARNING] sofia_reg.c:1422 SIP auth challenge (INVITE) on sofia profile 'internal' for [830 at 10.3.0.4] from ip 10.3.0.25 send 816 bytes to udp/[10.3.0.25]:5060 at 17:46:13.892050: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bKdf353688FD5B112B From: "B" ;tag=FAB3376C-15AEB62F To: ;tag=ZXg889Bv13H1c Call-ID: 12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="10.3.0.4", nonce="c3bdc00c-6a0f-11e1-8e6c-299021af10b6", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 536 bytes from udp/[10.3.0.25]:5060 at 17:46:13.903758: ------------------------------------------------------------------------ ACK sip:830 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bKdf353688FD5B112B From: "B" ;tag=FAB3376C-15AEB62F To: ;tag=ZXg889Bv13H1c CSeq: 1 ACK Call-ID: 12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ recv 1118 bytes from udp/[10.3.0.25]:5060 at 17:46:13.909222: ------------------------------------------------------------------------ INVITE sip:830 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bK924971a7A89CEC02 From: "B" ;tag=FAB3376C-15AEB62F To: CSeq: 2 INVITE Call-ID: 12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="20", realm="10.3.0.4", nonce="c3bdc00c-6a0f-11e1-8e6c-299021af10b6", qop=auth, cnonce="xa7J+697IQEBbl7", nc=00000001, uri="sip:830 at 10.3.0.4:5060;user=phone", response="c217f9da396e6a6d2e36f677c1d585b6", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 241 v=0 o=- 1167610362 1167610362 IN IP4 10.3.0.25 s=Polycom IP Phone c=IN IP4 10.3.0.25 t=0 0 a=sendrecv m=audio 2232 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ------------------------------------------------------------------------ send 333 bytes to udp/[10.3.0.25]:5060 at 17:46:13.910452: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bK924971a7A89CEC02 From: "B" ;tag=FAB3376C-15AEB62F To: Call-ID: 12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25 CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:13.894125 [DEBUG] sofia.c:7567 IP 10.3.0.25 Rejected by acl "domains". Falling back to Digest auth. 2012-03-09 18:46:13.894125 [NOTICE] switch_channel.c:926 New Channel sofia/internal/20 at 10.3.0.4 [c3c26f12-6a0f-11e1-8e6d-299021af10b6] 2012-03-09 18:46:13.894125 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/20 at 10.3.0.4) Running State Change CS_NEW 2012-03-09 18:46:13.894125 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/20 at 10.3.0.4) State NEW 2012-03-09 18:46:13.894125 [DEBUG] sofia.c:5532 Channel sofia/internal/20 at 10.3.0.4 entering state [received][100] 2012-03-09 18:46:13.894125 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=- 1167610362 1167610362 IN IP4 10.3.0.25 s=Polycom IP Phone c=IN IP4 10.3.0.25 t=0 0 a=sendrecv m=audio 2232 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 2012-03-09 18:46:13.894125 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 2012-03-09 18:46:13.894125 [DEBUG] sofia_glue.c:2991 Set Codec sofia/internal/20 at 10.3.0.4 G722/8000 20 ms 160 samples 64000 bits 2012-03-09 18:46:13.894125 [DEBUG] switch_core_codec.c:111 sofia/internal/20 at 10.3.0.4 Original read codec set to G722:9 2012-03-09 18:46:13.894125 [DEBUG] sofia_glue.c:4995 Set 2833 dtmf send/recv payload to 127 2012-03-09 18:46:13.894125 [DEBUG] sofia.c:5757 (sofia/internal/20 at 10.3.0.4) State Change CS_NEW -> CS_INIT 2012-03-09 18:46:13.894125 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/20 at 10.3.0.4 [BREAK] 2012-03-09 18:46:13.894125 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/20 at 10.3.0.4) Running State Change CS_INIT 2012-03-09 18:46:13.894125 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/20 at 10.3.0.4) State INIT 2012-03-09 18:46:13.894125 [DEBUG] mod_sofia.c:85 sofia/internal/20 at 10.3.0.4 SOFIA INIT 2012-03-09 18:46:13.894125 [DEBUG] mod_sofia.c:125 (sofia/internal/20 at 10.3.0.4) State Change CS_INIT -> CS_ROUTING 2012-03-09 18:46:13.894125 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/20 at 10.3.0.4 [BREAK] 2012-03-09 18:46:13.894125 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/20 at 10.3.0.4) State INIT going to sleep 2012-03-09 18:46:13.894125 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/20 at 10.3.0.4) Running State Change CS_ROUTING 2012-03-09 18:46:13.894125 [DEBUG] switch_channel.c:1886 (sofia/internal/20 at 10.3.0.4) Callstate Change DOWN -> RINGING 2012-03-09 18:46:13.894125 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/20 at 10.3.0.4) State ROUTING 2012-03-09 18:46:13.894125 [DEBUG] mod_sofia.c:148 sofia/internal/20 at 10.3.0.4 SOFIA ROUTING 2012-03-09 18:46:13.894125 [DEBUG] switch_core_state_machine.c:104 sofia/internal/20 at 10.3.0.4 Standard ROUTING 2012-03-09 18:46:13.894125 [INFO] mod_dialplan_xml.c:485 Processing B <20>->830 in context default Dialplan: sofia/internal/20 at 10.3.0.4 parsing [default->test1] continue=false Dialplan: sofia/internal/20 at 10.3.0.4 Regex (FAIL) [test1] destination_number(830) =~ /810/ break=on-false Dialplan: sofia/internal/20 at 10.3.0.4 parsing [default->test3] continue=false Dialplan: sofia/internal/20 at 10.3.0.4 Regex (FAIL) [test3] destination_number(830) =~ /820/ break=on-false Dialplan: sofia/internal/20 at 10.3.0.4 parsing [default->test2] continue=false Dialplan: sofia/internal/20 at 10.3.0.4 Regex (PASS) [test2] destination_number(830) =~ /830/ break=on-false Dialplan: sofia/internal/20 at 10.3.0.4 Action info() Dialplan: sofia/internal/20 at 10.3.0.4 Action log(WARNING TEST1 for UUID ${uuid}, src ${caller_id_number}, dest ${destination_number}) Dialplan: sofia/internal/20 at 10.3.0.4 Action sleep(1000) Dialplan: sofia/internal/20 at 10.3.0.4 Action log(WARNING TEST2 for UUID ${uuid}, src ${caller_id_number}, dest ${destination_number}) Dialplan: sofia/internal/20 at 10.3.0.4 Action sleep(1000) Dialplan: sofia/internal/20 at 10.3.0.4 Action log(WARNING TEST3 for UUID ${uuid}, src ${caller_id_number}, dest ${destination_number}) Dialplan: sofia/internal/20 at 10.3.0.4 Action sleep(1000) Dialplan: sofia/internal/20 at 10.3.0.4 Action bridge([leg_timeout=10][origination_caller_id_number=${caller_id_number}]user/30@${domain_name}) Dialplan: sofia/internal/20 at 10.3.0.4 Action info() Dialplan: sofia/internal/20 at 10.3.0.4 Action log(WARNING TEST4 for UUID ${uuid}, src ${caller_id_number}, dest ${destination_number}) Dialplan: sofia/internal/20 at 10.3.0.4 Action sleep(1000) Dialplan: sofia/internal/20 at 10.3.0.4 Action log(WARNING TEST5 for UUID ${uuid}, src ${caller_id_number}, dest ${destination_number}) Dialplan: sofia/internal/20 at 10.3.0.4 Action sleep(1000) Dialplan: sofia/internal/20 at 10.3.0.4 Action log(WARNING TEST6 for UUID ${uuid}, src ${caller_id_number}, dest ${destination_number}) Dialplan: sofia/internal/20 at 10.3.0.4 Action sleep(1000) 2012-03-09 18:46:13.894125 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/20 at 10.3.0.4) State Change CS_ROUTING -> CS_EXECUTE 2012-03-09 18:46:13.894125 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/20 at 10.3.0.4 [BREAK] 2012-03-09 18:46:13.894125 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/20 at 10.3.0.4) State ROUTING going to sleep 2012-03-09 18:46:13.894125 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/20 at 10.3.0.4) Running State Change CS_EXECUTE 2012-03-09 18:46:13.894125 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/20 at 10.3.0.4) State EXECUTE 2012-03-09 18:46:13.894125 [DEBUG] mod_sofia.c:241 sofia/internal/20 at 10.3.0.4 SOFIA EXECUTE 2012-03-09 18:46:13.894125 [DEBUG] switch_core_state_machine.c:192 sofia/internal/20 at 10.3.0.4 Standard EXECUTE EXECUTE sofia/internal/20 at 10.3.0.4 info() 2012-03-09 18:46:13.933936 [INFO] mod_dptools.c:1439 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [RINGING] Channel-State-Number: [4] Channel-Name: [sofia/internal/20 at 10.3.0.4] Unique-ID: [c3c26f12-6a0f-11e1-8e6d-299021af10b6] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-HIT-Dialplan: [true] Channel-Presence-ID: [20 at 10.3.0.4] Channel-Call-UUID: [c3c26f12-6a0f-11e1-8e6d-299021af10b6] Answer-State: [ringing] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Read-Codec-Bit-Rate: [64000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Channel-Write-Codec-Bit-Rate: [64000] Caller-Direction: [inbound] Caller-Username: [20] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [B] Caller-Caller-ID-Number: [20] Caller-Network-Addr: [10.3.0.25] Caller-ANI: [20] Caller-Destination-Number: [830] Caller-Unique-ID: [c3c26f12-6a0f-11e1-8e6d-299021af10b6] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/20 at 10.3.0.4] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1331315173894125] Caller-Channel-Created-Time: [1331315173894125] 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] variable_direction: [inbound] variable_uuid: [c3c26f12-6a0f-11e1-8e6d-299021af10b6] variable_session_id: [183] variable_sip_local_network_addr: [10.3.0.4] variable_sip_network_ip: [10.3.0.25] variable_sip_network_port: [5060] variable_sip_received_ip: [10.3.0.25] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_authorized: [true] variable_sip_number_alias: [20] variable_sip_auth_username: [20] variable_sip_auth_realm: [10.3.0.4] variable_number_alias: [20] variable_user_name: [20] variable_domain_name: [10.3.0.4] variable_record_stereo: [true] variable_default_gateway: [fonira] variable_default_areacode: [01] variable_transfer_fallback_extension: [operator] variable_toll_allow: [local,domestic,international,vas] variable_accountcode: [20] variable_user_context: [default] variable_effective_caller_id_name: [B ] variable_effective_caller_id_number: [20] variable_outbound_caller_id_name: [B ] variable_outbound_caller_id_number: [1997156020] variable_callgroup: [intercept] variable_sip_from_user: [20] variable_sip_from_uri: [20 at 10.3.0.4] variable_sip_from_host: [10.3.0.4] variable_sip_from_user_stripped: [20] variable_sip_from_tag: [FAB3376C-15AEB62F] variable_sofia_profile_name: [internal] variable_sip_full_via: [SIP/2.0/UDP 10.3.0.25;branch=z9hG4bK924971a7A89CEC02] variable_sip_from_display: [B] variable_sip_full_from: ["B" ;tag=FAB3376C-15AEB62F] variable_sip_full_to: [] variable_sip_req_params: [user=phone] variable_sip_req_user: [830] variable_sip_req_port: [5060] variable_sip_req_uri: [830 at 10.3.0.4:5060] variable_sip_req_host: [10.3.0.4] variable_sip_to_params: [user=phone] variable_sip_to_user: [830] variable_sip_to_uri: [830 at 10.3.0.4] variable_sip_to_host: [10.3.0.4] variable_sip_contact_user: [20] variable_sip_contact_uri: [20 at 10.3.0.25] variable_sip_contact_host: [10.3.0.25] variable_channel_name: [sofia/internal/20 at 10.3.0.4] variable_sip_call_id: [12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25] variable_sip_user_agent: [PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933] variable_sip_via_host: [10.3.0.25] variable_max_forwards: [70] variable_presence_id: [20 at 10.3.0.4] variable_switch_r_sdp: [v=0 o=- 1167610362 1167610362 IN IP4 10.3.0.25 s=Polycom IP Phone c=IN IP4 10.3.0.25 t=0 0 a=sendrecv m=audio 2232 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ] variable_remote_media_ip: [10.3.0.25] variable_remote_media_port: [2232] variable_sip_audio_recv_pt: [9] variable_sip_use_codec_name: [G722] variable_sip_use_codec_rate: [8000] variable_sip_use_codec_ptime: [20] variable_read_codec: [G722] variable_read_rate: [16000] variable_write_codec: [G722] variable_write_rate: [16000] variable_endpoint_disposition: [RECEIVED] variable_call_uuid: [c3c26f12-6a0f-11e1-8e6d-299021af10b6] variable_current_application: [info] EXECUTE sofia/internal/20 at 10.3.0.4 log(WARNING TEST1 for UUID c3c26f12-6a0f-11e1-8e6d-299021af10b6, src 20, dest 830) 2012-03-09 18:46:13.933936 [WARNING] mod_dptools.c:1420 TEST1 for UUID c3c26f12-6a0f-11e1-8e6d-299021af10b6, src 20, dest 830 EXECUTE sofia/internal/20 at 10.3.0.4 sleep(1000) EXECUTE sofia/internal/20 at 10.3.0.4 log(WARNING TEST2 for UUID c3c26f12-6a0f-11e1-8e6d-299021af10b6, src 20, dest 830) 2012-03-09 18:46:14.913910 [WARNING] mod_dptools.c:1420 TEST2 for UUID c3c26f12-6a0f-11e1-8e6d-299021af10b6, src 20, dest 830 EXECUTE sofia/internal/20 at 10.3.0.4 sleep(1000) EXECUTE sofia/internal/20 at 10.3.0.4 log(WARNING TEST3 for UUID c3c26f12-6a0f-11e1-8e6d-299021af10b6, src 20, dest 830) 2012-03-09 18:46:15.933910 [WARNING] mod_dptools.c:1420 TEST3 for UUID c3c26f12-6a0f-11e1-8e6d-299021af10b6, src 20, dest 830 EXECUTE sofia/internal/20 at 10.3.0.4 sleep(1000) EXECUTE sofia/internal/20 at 10.3.0.4 bridge([leg_timeout=10][origination_caller_id_number=20]user/30 at 10.3.0.4) 2012-03-09 18:46:16.932926 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-09 18:46:16.932926 [DEBUG] switch_ivr_originate.c:2299 Parsing session specific variables 2012-03-09 18:46:16.932926 [DEBUG] switch_event.c:1522 Parsing variable [leg_timeout]=[10] 2012-03-09 18:46:16.932926 [DEBUG] switch_event.c:1522 Parsing variable [origination_caller_id_number]=[20] 2012-03-09 18:46:16.954363 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-09 18:46:16.954363 [DEBUG] switch_event.c:1522 Parsing variable [presence_id]=[30 at 10.3.0.4] 2012-03-09 18:46:16.954363 [NOTICE] switch_channel.c:926 New Channel sofia/internal/sip:30 at 10.3.0.26 [c5926360-6a0f-11e1-8e75-299021af10b6] 2012-03-09 18:46:16.954363 [DEBUG] mod_sofia.c:4691 (sofia/internal/sip:30 at 10.3.0.26) State Change CS_NEW -> CS_INIT 2012-03-09 18:46:16.954363 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-09 18:46:16.954363 [DEBUG] switch_ivr_originate.c:2561 sofia/internal/sip:30 at 10.3.0.26 Setting leg timeout to 10 2012-03-09 18:46:16.954363 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:30 at 10.3.0.26) Running State Change CS_INIT 2012-03-09 18:46:16.954363 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:30 at 10.3.0.26) State INIT 2012-03-09 18:46:16.954363 [DEBUG] mod_sofia.c:85 sofia/internal/sip:30 at 10.3.0.26 SOFIA INIT 2012-03-09 18:46:16.954363 [DEBUG] mod_sofia.c:125 (sofia/internal/sip:30 at 10.3.0.26) State Change CS_INIT -> CS_ROUTING 2012-03-09 18:46:16.954363 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-09 18:46:16.954363 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:30 at 10.3.0.26) State INIT going to sleep 2012-03-09 18:46:16.954363 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:30 at 10.3.0.26) Running State Change CS_ROUTING 2012-03-09 18:46:16.954363 [DEBUG] switch_channel.c:1886 (sofia/internal/sip:30 at 10.3.0.26) Callstate Change DOWN -> RINGING send 1253 bytes to udp/[10.3.0.26]:5060 at 17:46:16.964763: ------------------------------------------------------------------------ INVITE sip:30 at 10.3.0.26 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK8acFFp1m4BjDc Max-Forwards: 69 From: "B " ;tag=1F3Sc0D3UNy6K To: Call-ID: 9cf1865a-e4b2-122f-cd91-00900b1be504 CSeq: 25321524 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 369 X-FS-Support: update_display,send_info Remote-Party-ID: "B " ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1331298730 1331298731 IN IP4 10.3.0.4 s=FreeSWITCH c=IN IP4 10.3.0.4 t=0 0 m=audio 16446 RTP/AVP 9 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 16438 RTP/AVP 98 99 100 34 31 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-2000/90000 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 ------------------------------------------------------------------------ 2012-03-09 18:46:16.954363 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:30 at 10.3.0.26) State ROUTING 2012-03-09 18:46:16.954363 [DEBUG] mod_sofia.c:148 sofia/internal/sip:30 at 10.3.0.26 SOFIA ROUTING 2012-03-09 18:46:16.954363 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:30 at 10.3.0.26) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-03-09 18:46:16.954363 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-09 18:46:16.954363 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:30 at 10.3.0.26) State ROUTING going to sleep 2012-03-09 18:46:16.954363 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:30 at 10.3.0.26) Running State Change CS_CONSUME_MEDIA 2012-03-09 18:46:16.954363 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:30 at 10.3.0.26) State CONSUME_MEDIA 2012-03-09 18:46:16.954363 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:30 at 10.3.0.26) State CONSUME_MEDIA going to sleep 2012-03-09 18:46:16.954363 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-09 18:46:16.954363 [DEBUG] sofia.c:5532 Channel sofia/internal/sip:30 at 10.3.0.26 entering state [calling][0] recv 411 bytes from udp/[10.3.0.26]:5060 at 17:46:16.976025: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK8acFFp1m4BjDc From: "B" ;tag=1F3Sc0D3UNy6K To: "C" ;tag=93CE8DA-22B4DC39 CSeq: 25321524 INVITE Call-ID: 9cf1865a-e4b2-122f-cd91-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 448 bytes from udp/[10.3.0.26]:5060 at 17:46:17.023373: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK8acFFp1m4BjDc From: "B" ;tag=1F3Sc0D3UNy6K To: "C" ;tag=93CE8DA-22B4DC39 CSeq: 25321524 INVITE Call-ID: 9cf1865a-e4b2-122f-cd91-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Allow-Events: talk,hold,conference Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:17.014010 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-09 18:46:17.014010 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-09 18:46:17.014010 [DEBUG] sofia.c:5532 Channel sofia/internal/sip:30 at 10.3.0.26 entering state [proceeding][180] 2012-03-09 18:46:17.014010 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/sip:30 at 10.3.0.26! 2012-03-09 18:46:17.014010 [NOTICE] mod_sofia.c:2514 Ring-Ready sofia/internal/20 at 10.3.0.4! send 799 bytes to udp/[10.3.0.25]:5060 at 17:46:17.034554: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bK924971a7A89CEC02 From: "B" ;tag=FAB3376C-15AEB62F To: ;tag=0690a5vZyc8Kr Call-ID: 12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 Remote-Party-ID: "Outbound Call" <30>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ 2012-03-09 18:46:17.014010 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/20 at 10.3.0.4 [BREAK] 2012-03-09 18:46:17.014010 [DEBUG] sofia.c:5532 Channel sofia/internal/20 at 10.3.0.4 entering state [early][180] 2012-03-09 18:46:17.014010 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/20 at 10.3.0.4 [BREAK] 2012-03-09 18:46:17.014010 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/internal/20 at 10.3.0.4! recv 617 bytes from udp/[10.3.0.25]:5060 at 17:46:17.752817: ------------------------------------------------------------------------ REFER sip:mod_sofia at 10.3.0.4:5060 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bK91e647a3A2636CDE From: "B" ;tag=C3930705-7AD5A1D0 To: "A" ;tag=ymQF7eUr4tUeH CSeq: 2 REFER Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Refer-To: Referred-By: Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:17.733940 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:17.772935 [DEBUG] sofia.c:6522 Process REFER to [830 at 10.3.0.4] 2012-03-09 18:46:17.772935 [DEBUG] sofia.c:6540 Replaces: [12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25] send 703 bytes to udp/[10.3.0.25]:5060 at 17:46:17.773525: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bK91e647a3A2636CDE From: "B" ;tag=C3930705-7AD5A1D0 To: "A" ;tag=ymQF7eUr4tUeH Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 CSeq: 2 REFER Contact: Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:17.772935 [NOTICE] sofia.c:6623 Attended Transfer on originating session c3c26f12-6a0f-11e1-8e6d-299021af10b6 2012-03-09 18:46:17.772935 [DEBUG] switch_ivr.c:1711 (sofia/internal/10 at 10.3.0.4) State Change CS_EXECUTE -> CS_ROUTING 2012-03-09 18:46:17.772935 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:17.772935 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:17.772935 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/10 at 10.3.0.4 to inline[endless_playback:local_stream://moh,park at default] send 851 bytes to udp/[10.3.0.25]:5060 at 17:46:17.775981: ------------------------------------------------------------------------ NOTIFY sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK9K57gHjr1m8ZQ Max-Forwards: 70 From: "A " ;tag=ymQF7eUr4tUeH To: ;tag=C3930705-7AD5A1D0 Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 CSeq: 25321521 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: refer;id=2 Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 16 SIP/2.0 200 OK ------------------------------------------------------------------------ send 638 bytes to udp/[10.3.0.25]:5060 at 17:46:17.776716: ------------------------------------------------------------------------ BYE sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKaXy0jc3UyXyjK Max-Forwards: 70 From: ;tag=0690a5vZyc8Kr To: "B" ;tag=FAB3376C-15AEB62F Call-ID: 12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25 CSeq: 25321524 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="normal_clearing" Content-Length: 0 ------------------------------------------------------------------------ send 653 bytes to udp/[10.3.0.25]:5060 at 17:46:17.777176: ------------------------------------------------------------------------ SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bK924971a7A89CEC02 From: "B" ;tag=FAB3376C-15AEB62F To: ;tag=0690a5vZyc8Kr Call-ID: 12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25 CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 420 bytes from udp/[10.3.0.25]:5060 at 17:46:17.788488: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK9K57gHjr1m8ZQ From: "A" ;tag=ymQF7eUr4tUeH To: "B" ;tag=C3930705-7AD5A1D0 CSeq: 25321521 NOTIFY Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:17.772935 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] recv 432 bytes from udp/[10.3.0.25]:5060 at 17:46:17.793916: ------------------------------------------------------------------------ BYE sip:mod_sofia at 10.3.0.4:5060 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bK6975a5b1C0C10CDC From: "B" ;tag=C3930705-7AD5A1D0 To: "A" ;tag=ymQF7eUr4tUeH CSeq: 3 BYE Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:17.794057 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:17.794057 [DEBUG] switch_ivr_play_say.c:1678 done playing file local_stream://moh 2012-03-09 18:46:17.794057 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/sip:20 at 10.3.0.25] 2012-03-09 18:46:17.794057 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:17.794057 [DEBUG] switch_channel.c:2848 (sofia/internal/sip:20 at 10.3.0.25) Callstate Change HELD -> HANGUP 2012-03-09 18:46:17.794057 [NOTICE] switch_ivr_bridge.c:669 Hangup sofia/internal/sip:20 at 10.3.0.25 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2012-03-09 18:46:17.794057 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/sip:20 at 10.3.0.25 [KILL] 2012-03-09 18:46:17.794057 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:420 (sofia/internal/sip:20 at 10.3.0.25) State EXCHANGE_MEDIA going to sleep 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:20 at 10.3.0.25) Running State Change CS_HANGUP 2012-03-09 18:46:17.794057 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:17.794057 [DEBUG] switch_ivr_bridge.c:329 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:17.794057 [DEBUG] switch_ivr_bridge.c:499 sofia/internal/sip:20 at 10.3.0.25 ending bridge by request from write function 2012-03-09 18:46:17.794057 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/10 at 10.3.0.4] 2012-03-09 18:46:17.794057 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:20 at 10.3.0.25) State HANGUP 2012-03-09 18:46:17.794057 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:20 at 10.3.0.25 hanging up, cause: NORMAL_CLEARING 2012-03-09 18:46:17.794057 [DEBUG] switch_ivr_bridge.c:1403 sofia/internal/sip:20 at 10.3.0.25 skip receive message [UNBRIDGE] (channel is hungup already) 2012-03-09 18:46:17.794057 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/10 at 10.3.0.4) State EXECUTE going to sleep 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_ROUTING recv 738 bytes from udp/[10.3.0.25]:5060 at 17:46:17.800685: ------------------------------------------------------------------------ CANCEL sip:830 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bK924971a7A89CEC02 From: "B" ;tag=FAB3376C-15AEB62F To: CSeq: 2 CANCEL Call-ID: 12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Proxy-Authorization: Digest username="20", realm="10.3.0.4", nonce="c3bdc00c-6a0f-11e1-8e6c-299021af10b6", qop=auth, cnonce="xa7J+697IQEBbl7", nc=00000002, uri="sip:830 at 10.3.0.4:5060;user=phone", response="6c89aa15f8428bb6f6daa6df020c448f", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:17.794057 [DEBUG] switch_channel.c:1886 (sofia/internal/10 at 10.3.0.4) Callstate Change ACTIVE -> RINGING send 266 bytes to udp/[10.3.0.25]:5060 at 17:46:17.801231: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bK924971a7A89CEC02 From: "B" ;tag=FAB3376C-15AEB62F To: ;tag=0690a5vZyc8Kr Call-ID: 12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25 CSeq: 2 CANCEL Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/10 at 10.3.0.4) State ROUTING 2012-03-09 18:46:17.794057 [DEBUG] mod_sofia.c:148 sofia/internal/10 at 10.3.0.4 SOFIA ROUTING 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:104 sofia/internal/10 at 10.3.0.4 Standard ROUTING 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/10 at 10.3.0.4) State Change CS_ROUTING -> CS_EXECUTE 2012-03-09 18:46:17.794057 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/10 at 10.3.0.4) State ROUTING going to sleep 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_EXECUTE 2012-03-09 18:46:17.794057 [DEBUG] switch_channel.c:1888 (sofia/internal/10 at 10.3.0.4) Callstate Change RINGING -> ACTIVE 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/10 at 10.3.0.4) State EXECUTE 2012-03-09 18:46:17.794057 [DEBUG] mod_sofia.c:241 sofia/internal/10 at 10.3.0.4 SOFIA EXECUTE 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:192 sofia/internal/10 at 10.3.0.4 Standard EXECUTE EXECUTE sofia/internal/10 at 10.3.0.4 endless_playback(local_stream://moh) 2012-03-09 18:46:17.794057 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/16000] 16000hz 2012-03-09 18:46:17.794057 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16 at 16000hz 1 channels 20ms 2012-03-09 18:46:17.794057 [DEBUG] mod_sofia.c:513 Sending BYE to sofia/internal/sip:20 at 10.3.0.25 send 622 bytes to udp/[10.3.0.25]:5060 at 17:46:17.808816: ------------------------------------------------------------------------ BYE sip:20 at 10.3.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKB6QSm7KZU6m5e Max-Forwards: 70 From: "A " ;tag=ymQF7eUr4tUeH To: ;tag=C3930705-7AD5A1D0 Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 CSeq: 25321522 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:20 at 10.3.0.25 Standard HANGUP, cause: NORMAL_CLEARING ------------------------------------------------------------------------ 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:20 at 10.3.0.25) State HANGUP going to sleep 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/sip:20 at 10.3.0.25) State Change CS_HANGUP -> CS_REPORTING 2012-03-09 18:46:17.794057 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:20 at 10.3.0.25) Running State Change CS_REPORTING 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:20 at 10.3.0.25) State REPORTING 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:20 at 10.3.0.25 Standard REPORTING, cause: NORMAL_CLEARING 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:20 at 10.3.0.25) State REPORTING going to sleep 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/sip:20 at 10.3.0.25) State Change CS_REPORTING -> CS_DESTROY 2012-03-09 18:46:17.794057 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:20 at 10.3.0.25 [BREAK] 2012-03-09 18:46:17.794057 [DEBUG] switch_core_session.c:1380 Session 182 (sofia/internal/sip:20 at 10.3.0.25) Locked, Waiting on external entities 2012-03-09 18:46:17.794057 [NOTICE] switch_core_session.c:1398 Session 182 (sofia/internal/sip:20 at 10.3.0.25) Ended 2012-03-09 18:46:17.794057 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/sip:20 at 10.3.0.25 [CS_DESTROY] 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/sip:20 at 10.3.0.25) Callstate Change HANGUP -> DOWN 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/sip:20 at 10.3.0.25) Running State Change CS_DESTROY 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:20 at 10.3.0.25) State DESTROY 2012-03-09 18:46:17.794057 [DEBUG] mod_sofia.c:374 sofia/internal/sip:20 at 10.3.0.25 SOFIA DESTROY 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:20 at 10.3.0.25 Standard DESTROY 2012-03-09 18:46:17.794057 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:20 at 10.3.0.25) State DESTROY going to sleep recv 403 bytes from udp/[10.3.0.25]:5060 at 17:46:17.828484: ------------------------------------------------------------------------ SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKaXy0jc3UyXyjK From: ;tag=0690a5vZyc8Kr To: "B" ;tag=FAB3376C-15AEB62F CSeq: 25321524 BYE Call-ID: 12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:17.794057 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/20 at 10.3.0.4 [BREAK] 2012-03-09 18:46:17.794057 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/20 at 10.3.0.4 [BREAK] 2012-03-09 18:46:17.794057 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/20 at 10.3.0.4 [BREAK] 2012-03-09 18:46:17.794057 [DEBUG] sofia.c:5532 Channel sofia/internal/20 at 10.3.0.4 entering state [terminated][481] 2012-03-09 18:46:17.794057 [DEBUG] switch_channel.c:2848 (sofia/internal/20 at 10.3.0.4) Callstate Change RINGING -> HANGUP 2012-03-09 18:46:17.794057 [NOTICE] sofia.c:6301 Hangup sofia/internal/20 at 10.3.0.4 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2012-03-09 18:46:17.794057 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/20 at 10.3.0.4 [KILL] 2012-03-09 18:46:17.794057 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/20 at 10.3.0.4 [BREAK] recv 536 bytes from udp/[10.3.0.25]:5060 at 17:46:17.833953: ------------------------------------------------------------------------ ACK sip:830 at 10.3.0.4:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bK924971a7A89CEC02 From: "B" ;tag=FAB3376C-15AEB62F To: ;tag=0690a5vZyc8Kr CSeq: 2 ACK Call-ID: 12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ recv 394 bytes from udp/[10.3.0.25]:5060 at 17:46:17.850039: ------------------------------------------------------------------------ SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKB6QSm7KZU6m5e From: "A" ;tag=ymQF7eUr4tUeH To: ;tag=C3930705-7AD5A1D0 CSeq: 25321522 BYE Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ send 262 bytes to udp/[10.3.0.25]:5060 at 17:46:17.852789: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.25;branch=z9hG4bK6975a5b1C0C10CDC From: "B" ;tag=C3930705-7AD5A1D0 To: "A" ;tag=ymQF7eUr4tUeH Call-ID: 98c5604c-e4b2-122f-cd91-00900b1be504 CSeq: 3 BYE Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:26.014188 [DEBUG] switch_channel.c:2598 (sofia/internal/10 at 10.3.0.4) State Change CS_EXECUTE -> CS_RESET 2012-03-09 18:46:26.014188 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:26.052925 [DEBUG] switch_ivr_play_say.c:1678 done playing file local_stream://moh 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/10 at 10.3.0.4) State EXECUTE going to sleep 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_RESET 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:413 (sofia/internal/10 at 10.3.0.4) State RESET 2012-03-09 18:46:26.052925 [DEBUG] mod_sofia.c:166 sofia/internal/10 at 10.3.0.4 SOFIA RESET 2012-03-09 18:46:26.052925 [DEBUG] switch_channel.c:2600 (sofia/internal/10 at 10.3.0.4) State Change CS_RESET -> CS_EXECUTE 2012-03-09 18:46:26.052925 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:26.052925 [DEBUG] switch_channel.c:2848 (sofia/internal/sip:30 at 10.3.0.26) Callstate Change RINGING -> HANGUP 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:413 (sofia/internal/10 at 10.3.0.4) State RESET going to sleep 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_EXECUTE 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/10 at 10.3.0.4) State EXECUTE 2012-03-09 18:46:26.052925 [DEBUG] mod_sofia.c:241 sofia/internal/10 at 10.3.0.4 SOFIA EXECUTE 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:192 sofia/internal/10 at 10.3.0.4 Standard EXECUTE EXECUTE sofia/internal/10 at 10.3.0.4 info() 2012-03-09 18:46:26.052925 [NOTICE] switch_ivr_originate.c:3075 Hangup sofia/internal/sip:30 at 10.3.0.26 [CS_CONSUME_MEDIA] [ATTENDED_TRANSFER] 2012-03-09 18:46:26.052925 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/sip:30 at 10.3.0.26 [KILL] 2012-03-09 18:46:26.052925 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-09 18:46:26.052925 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [ATTENDED_TRANSFER] 2012-03-09 18:46:26.052925 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 601 [ATTENDED_TRANSFER] 2012-03-09 18:46:26.052925 [INFO] mod_dptools.c:2922 Originate Failed. Cause: ATTENDED_TRANSFER 2012-03-09 18:46:26.052925 [DEBUG] switch_core_session.c:2285 sofia/internal/20 at 10.3.0.4 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/20 at 10.3.0.4) State EXECUTE going to sleep 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/20 at 10.3.0.4) Running State Change CS_HANGUP 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/20 at 10.3.0.4) State HANGUP 2012-03-09 18:46:26.052925 [DEBUG] mod_sofia.c:469 Channel sofia/internal/20 at 10.3.0.4 hanging up, cause: NORMAL_TEMPORARY_FAILURE 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:30 at 10.3.0.26) Running State Change CS_HANGUP 2012-03-09 18:46:26.052925 [INFO] mod_dptools.c:1439 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [ACTIVE] Channel-State-Number: [4] Channel-Name: [sofia/internal/10 at 10.3.0.4] Unique-ID: [bf969cf6-6a0f-11e1-8e5c-299021af10b6] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-HIT-Dialplan: [true] Channel-Presence-ID: [20 at 10.3.0.4] Channel-Call-UUID: [bf969cf6-6a0f-11e1-8e5c-299021af10b6] Answer-State: [answered] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Read-Codec-Bit-Rate: [64000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Channel-Write-Codec-Bit-Rate: [64000] Caller-Direction: [inbound] Caller-Username: [10] Caller-Dialplan: [inline] Caller-Caller-ID-Name: [A] Caller-Caller-ID-Number: [10] Caller-Callee-ID-Name: [Outbound Call] Caller-Callee-ID-Number: [20] Caller-Network-Addr: [10.3.0.22] Caller-ANI: [10] Caller-Destination-Number: [830] Caller-Unique-ID: [bf969cf6-6a0f-11e1-8e5c-299021af10b6] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/10 at 10.3.0.4] Caller-Profile-Index: [3] Caller-Profile-Created-Time: [1331315186012931] Caller-Channel-Created-Time: [1331315166913289] Caller-Channel-Answered-Time: [1331315171292944] Caller-Channel-Progress-Time: [1331315170032935] 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] variable_sip_local_sdp_str: [v=0 o=FreeSWITCH 1331298747 1331298748 IN IP4 10.3.0.4 s=FreeSWITCH c=IN IP4 10.3.0.4 t=0 0 m=audio 16424 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ] variable_local_media_ip: [10.3.0.4] variable_local_media_port: [16424] variable_advertised_media_ip: [10.3.0.4] variable_sip_use_pt: [9] variable_rtp_use_ssrc: [1271749377] variable_sip_2833_send_payload: [127] variable_sip_2833_recv_payload: [127] variable_last_bridge_to: [c1665710-6a0f-11e1-8e64-299021af10b6] variable_bridge_channel: [sofia/internal/sip:20 at 10.3.0.25] variable_bridge_uuid: [c1665710-6a0f-11e1-8e64-299021af10b6] variable_sip_to_tag: [XByp5KaN7H5UN] variable_sip_cseq: [2] variable_switch_m_sdp: [v=0 o=- 1167610359 1167610360 IN IP4 10.3.0.25 s=Polycom IP Phone c=IN IP4 0.0.0.0 t=0 0 m=audio 2230 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=sendonly m=video 0 RTP/AVP 98 99 100 34 31 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-2000/90000 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 ] variable_att_xfer_kill_uuid: [c3c26f12-6a0f-11e1-8e6d-299021af10b6] variable_transfer_history: [ARRAY::1331315177:c60f0154-6a0f-11e1-8e79-299021af10b6:bl_xfer:endless_playback:local_stream://moh,park/default/inline] variable_transfer_source: [1331315177:c60f0154-6a0f-11e1-8e79-299021af10b6:bl_xfer:endless_playback:local_stream://moh,park/default/inline] variable_bridge_hangup_cause: [NORMAL_CLEARING] variable_direction: [inbound] variable_uuid: [c3c26f12-6a0f-11e1-8e6d-299021af10b6] variable_session_id: [183] variable_sip_local_network_addr: [10.3.0.4] variable_sip_network_ip: [10.3.0.25] variable_sip_network_port: [5060] variable_sip_received_ip: [10.3.0.25] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_authorized: [true] variable_sip_number_alias: [20] variable_sip_auth_username: [20] variable_sip_auth_realm: [10.3.0.4] variable_number_alias: [20] variable_user_name: [20] variable_domain_name: [10.3.0.4] variable_record_stereo: [true] variable_default_gateway: [fonira] variable_default_areacode: [01] variable_transfer_fallback_extension: [operator] variable_toll_allow: [local,domestic,international,vas] variable_accountcode: [20] variable_user_context: [default] variable_effective_caller_id_name: [B ] variable_effective_caller_id_number: [20] variable_outbound_caller_id_name: [B ] variable_outbound_caller_id_number: [1997156020] variable_callgroup: [intercept] variable_sip_from_user: [20] variable_sip_from_uri: [20 at 10.3.0.4] variable_sip_from_host: [10.3.0.4] variable_sip_from_user_stripped: [20] variable_sip_from_tag: [FAB3376C-15AEB62F] variable_sofia_profile_name: [internal] variable_sip_full_via: [SIP/2.0/UDP 10.3.0.25;branch=z9hG4bK924971a7A89CEC02] variable_sip_from_display: [B] variable_sip_full_from: ["B" ;tag=FAB3376C-15AEB62F] variable_sip_full_to: [] variable_sip_req_params: [user=phone] variable_sip_req_user: [830] variable_sip_req_port: [5060] variable_sip_req_uri: [830 at 10.3.0.4:5060] variable_sip_req_host: [10.3.0.4] variable_sip_to_params: [user=phone] variable_sip_to_user: [830] variable_sip_to_uri: [830 at 10.3.0.4] variable_sip_to_host: [10.3.0.4] variable_sip_contact_user: [20] variable_sip_contact_uri: [20 at 10.3.0.25] variable_sip_contact_host: [10.3.0.25] variable_channel_name: [sofia/internal/20 at 10.3.0.4] variable_sip_call_id: [12bb6ca6-d165db9-ac5e8f24 at 10.3.0.25] variable_sip_user_agent: [PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933] variable_sip_via_host: [10.3.0.25] variable_max_forwards: [70] variable_presence_id: [20 at 10.3.0.4] variable_switch_r_sdp: [v=0 o=- 1167610362 1167610362 IN IP4 10.3.0.25 s=Polycom IP Phone c=IN IP4 10.3.0.25 t=0 0 a=sendrecv m=audio 2232 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ] variable_remote_media_ip: [10.3.0.25] variable_remote_media_port: [2232] variable_sip_audio_recv_pt: [9] variable_sip_use_codec_name: [G722] variable_sip_use_codec_rate: [8000] variable_sip_use_codec_ptime: [20] variable_read_codec: [G722] variable_read_rate: [16000] variable_write_codec: [G722] variable_write_rate: [16000] variable_dialed_user: [30] variable_dialed_domain: [10.3.0.4] variable_originate_disposition: [failure] variable_DIALSTATUS: [INVALIDARGS] variable_endpoint_disposition: [ATTENDED_TRANSFER] variable_sip_term_status: [481] variable_proto_specific_hangup_cause: [sip:481] variable_sip_term_cause: [41] variable_playback_seconds: [16] variable_playback_ms: [16600] variable_playback_samples: [132800] variable_call_uuid: [bf969cf6-6a0f-11e1-8e5c-299021af10b6] variable_current_application: [info] 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:30 at 10.3.0.26) State HANGUP 2012-03-09 18:46:26.052925 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:30 at 10.3.0.26 hanging up, cause: ATTENDED_TRANSFER EXECUTE sofia/internal/10 at 10.3.0.4 log(WARNING TEST4 for UUID c3c26f12-6a0f-11e1-8e6d-299021af10b6, src 10, dest 830) 2012-03-09 18:46:26.052925 [WARNING] mod_dptools.c:1420 TEST4 for UUID c3c26f12-6a0f-11e1-8e6d-299021af10b6, src 10, dest 830 EXECUTE sofia/internal/10 at 10.3.0.4 sleep(1000) 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:47 sofia/internal/20 at 10.3.0.4 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/20 at 10.3.0.4) State HANGUP going to sleep 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/20 at 10.3.0.4) State Change CS_HANGUP -> CS_REPORTING 2012-03-09 18:46:26.052925 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/20 at 10.3.0.4 [BREAK] 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/20 at 10.3.0.4) Running State Change CS_REPORTING 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/20 at 10.3.0.4) State REPORTING 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:79 sofia/internal/20 at 10.3.0.4 Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/20 at 10.3.0.4) State REPORTING going to sleep 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/20 at 10.3.0.4) State Change CS_REPORTING -> CS_DESTROY 2012-03-09 18:46:26.052925 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/20 at 10.3.0.4 [BREAK] 2012-03-09 18:46:26.052925 [DEBUG] switch_core_session.c:1380 Session 183 (sofia/internal/20 at 10.3.0.4) Locked, Waiting on external entities 2012-03-09 18:46:26.052925 [NOTICE] switch_core_session.c:1398 Session 183 (sofia/internal/20 at 10.3.0.4) Ended 2012-03-09 18:46:26.052925 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/20 at 10.3.0.4 [CS_DESTROY] 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/20 at 10.3.0.4) Callstate Change HANGUP -> DOWN 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/20 at 10.3.0.4) Running State Change CS_DESTROY 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/20 at 10.3.0.4) State DESTROY 2012-03-09 18:46:26.052925 [DEBUG] mod_sofia.c:374 sofia/internal/20 at 10.3.0.4 SOFIA DESTROY 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:86 sofia/internal/20 at 10.3.0.4 Standard DESTROY 2012-03-09 18:46:26.052925 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/20 at 10.3.0.4) State DESTROY going to sleep 2012-03-09 18:46:26.074508 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/sip:30 at 10.3.0.26 2012-03-09 18:46:26.074508 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:30 at 10.3.0.26 Standard HANGUP, cause: ATTENDED_TRANSFER 2012-03-09 18:46:26.074508 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:30 at 10.3.0.26) State HANGUP going to sleep 2012-03-09 18:46:26.074508 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/sip:30 at 10.3.0.26) State Change CS_HANGUP -> CS_REPORTING 2012-03-09 18:46:26.074508 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] 2012-03-09 18:46:26.074508 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:30 at 10.3.0.26) Running State Change CS_REPORTING 2012-03-09 18:46:26.074508 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:30 at 10.3.0.26) State REPORTING 2012-03-09 18:46:26.074508 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:30 at 10.3.0.26 Standard REPORTING, cause: ATTENDED_TRANSFER 2012-03-09 18:46:26.074508 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:30 at 10.3.0.26) State REPORTING going to sleep 2012-03-09 18:46:26.074508 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/sip:30 at 10.3.0.26) State Change CS_REPORTING -> CS_DESTROY send 334 bytes to udp/[10.3.0.26]:5060 at 17:46:26.077886: ------------------------------------------------------------------------ CANCEL sip:30 at 10.3.0.26 SIP/2.02012-03-09 18:46:26.074508 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:30 at 10.3.0.26 [BREAK] Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK8acFFp1m4BjDc Max-Forwards: 69 2012-03-09 18:46:26.074508 [DEBUG] switch_core_session.c:1380 Session 184 (sofia/internal/sip:30 at 10.3.0.26) Locked, Waiting on external entities From: "B " ;tag=1F3Sc0D3UNy6K To: Call-ID: 9cf1865a-e4b2-122f-cd91-00900b1be504 CSeq: 25321524 CANCEL2012-03-09 18:46:26.074508 [NOTICE] switch_core_session.c:1398 Session 184 (sofia/internal/sip:30 at 10.3.0.26) Ended Reason: FreeSWITCH;cause=601;text="ATTENDED_TRANSFER" Content-Length: 02012-03-09 18:46:26.074508 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/sip:30 at 10.3.0.26 [CS_DESTROY] ------------------------------------------------------------------------ 2012-03-09 18:46:26.074508 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/sip:30 at 10.3.0.26) Callstate Change HANGUP -> DOWN 2012-03-09 18:46:26.074508 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/sip:30 at 10.3.0.26) Running State Change CS_DESTROY 2012-03-09 18:46:26.074508 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:30 at 10.3.0.26) State DESTROY 2012-03-09 18:46:26.074508 [DEBUG] mod_sofia.c:374 sofia/internal/sip:30 at 10.3.0.26 SOFIA DESTROY 2012-03-09 18:46:26.074508 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:30 at 10.3.0.26 Standard DESTROY 2012-03-09 18:46:26.074508 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:30 at 10.3.0.26) State DESTROY going to sleep recv 386 bytes from udp/[10.3.0.26]:5060 at 17:46:26.085229: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK8acFFp1m4BjDc From: "B" ;tag=1F3Sc0D3UNy6K To: "C" CSeq: 25321524 CANCEL Call-ID: 9cf1865a-e4b2-122f-cd91-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ recv 422 bytes from udp/[10.3.0.26]:5060 at 17:46:26.091905: ------------------------------------------------------------------------ SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK8acFFp1m4BjDc From: "B" ;tag=1F3Sc0D3UNy6K To: "C" ;tag=93CE8DA-22B4DC39 CSeq: 25321524 INVITE Call-ID: 9cf1865a-e4b2-122f-cd91-00900b1be504 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ send 310 bytes to udp/[10.3.0.26]:5060 at 17:46:26.092411: ------------------------------------------------------------------------ ACK sip:30 at 10.3.0.26 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bK8acFFp1m4BjDc Max-Forwards: 69 From: "B " ;tag=1F3Sc0D3UNy6K To: "C" ;tag=93CE8DA-22B4DC39 Call-ID: 9cf1865a-e4b2-122f-cd91-00900b1be504 CSeq: 25321524 ACK Content-Length: 0 ------------------------------------------------------------------------ EXECUTE sofia/internal/10 at 10.3.0.4 log(WARNING TEST5 for UUID c3c26f12-6a0f-11e1-8e6d-299021af10b6, src 10, dest 830) 2012-03-09 18:46:27.072929 [WARNING] mod_dptools.c:1420 TEST5 for UUID c3c26f12-6a0f-11e1-8e6d-299021af10b6, src 10, dest 830 EXECUTE sofia/internal/10 at 10.3.0.4 sleep(1000) EXECUTE sofia/internal/10 at 10.3.0.4 log(WARNING TEST6 for UUID c3c26f12-6a0f-11e1-8e6d-299021af10b6, src 10, dest 830) 2012-03-09 18:46:28.092926 [WARNING] mod_dptools.c:1420 TEST6 for UUID c3c26f12-6a0f-11e1-8e6d-299021af10b6, src 10, dest 830 EXECUTE sofia/internal/10 at 10.3.0.4 sleep(1000) 2012-03-09 18:46:29.112925 [NOTICE] switch_core_state_machine.c:226 sofia/internal/10 at 10.3.0.4 has executed the last dialplan instruction, hanging up. 2012-03-09 18:46:29.112925 [DEBUG] switch_channel.c:2848 (sofia/internal/10 at 10.3.0.4) Callstate Change ACTIVE -> HANGUP 2012-03-09 18:46:29.112925 [NOTICE] switch_core_state_machine.c:228 Hangup sofia/internal/10 at 10.3.0.4 [CS_EXECUTE] [NORMAL_CLEARING] 2012-03-09 18:46:29.112925 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/10 at 10.3.0.4 [KILL] 2012-03-09 18:46:29.112925 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/10 at 10.3.0.4) State EXECUTE going to sleep 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_HANGUP 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/10 at 10.3.0.4) State HANGUP 2012-03-09 18:46:29.112925 [DEBUG] mod_sofia.c:469 Channel sofia/internal/10 at 10.3.0.4 hanging up, cause: NORMAL_CLEARING 2012-03-09 18:46:29.112925 [DEBUG] mod_sofia.c:513 Sending BYE to sofia/internal/10 at 10.3.0.4 send 640 bytes to udp/[10.3.0.22]:5060 at 17:46:29.121085: ------------------------------------------------------------------------ BYE sip:10 at 10.3.0.22 SIP/2.0 Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKcFHjp242rFBra Max-Forwards: 70 From: ;tag=XByp5KaN7H5UN To: "A" ;tag=A59BE1A5-8A2B1314 Call-ID: 73ec8749-ef90cb48-95fd5843 at 10.3.0.22 CSeq: 25321530 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b128198 2012-03-08 15-27-51 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:47 sofia/internal/10 at 10.3.0.4 Standard HANGUP, cause: NORMAL_CLEARING 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/10 at 10.3.0.4) State HANGUP going to sleep 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/10 at 10.3.0.4) State Change CS_HANGUP -> CS_REPORTING 2012-03-09 18:46:29.112925 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/10 at 10.3.0.4) Running State Change CS_REPORTING 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/10 at 10.3.0.4) State REPORTING 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:79 sofia/internal/10 at 10.3.0.4 Standard REPORTING, cause: NORMAL_CLEARING 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/10 at 10.3.0.4) State REPORTING going to sleep 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/10 at 10.3.0.4) State Change CS_REPORTING -> CS_DESTROY 2012-03-09 18:46:29.112925 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/10 at 10.3.0.4 [BREAK] 2012-03-09 18:46:29.112925 [DEBUG] switch_core_session.c:1380 Session 181 (sofia/internal/10 at 10.3.0.4) Locked, Waiting on external entities 2012-03-09 18:46:29.112925 [NOTICE] switch_core_session.c:1398 Session 181 (sofia/internal/10 at 10.3.0.4) Ended 2012-03-09 18:46:29.112925 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/10 at 10.3.0.4 [CS_DESTROY] 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/10 at 10.3.0.4) Callstate Change HANGUP -> DOWN 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/10 at 10.3.0.4) Running State Change CS_DESTROY 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/10 at 10.3.0.4) State DESTROY 2012-03-09 18:46:29.112925 [DEBUG] mod_sofia.c:374 sofia/internal/10 at 10.3.0.4 SOFIA DESTROY 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:86 sofia/internal/10 at 10.3.0.4 Standard DESTROY 2012-03-09 18:46:29.112925 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/10 at 10.3.0.4) State DESTROY going to sleep recv 401 bytes from udp/[10.3.0.22]:5060 at 17:46:29.128903: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.0.4;rport;branch=z9hG4bKcFHjp242rFBra From: ;tag=XByp5KaN7H5UN To: "A" ;tag=A59BE1A5-8A2B1314 CSeq: 25321530 BYE Call-ID: 73ec8749-ef90cb48-95fd5843 at 10.3.0.22 Contact: User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.3.1.0933 Accept-Language: de-de,de;q=0.9,en;q=0.8 Content-Length: 0 ------------------------------------------------------------------------ From Tim.Meade at Millicorp.com Fri Mar 9 20:59:57 2012 From: Tim.Meade at Millicorp.com (Tim Meade) Date: Fri, 9 Mar 2012 17:59:57 +0000 Subject: [Freeswitch-users] mod_xml_rpc and xmlapi not working Message-ID: <804D48104511D4468F0D60DF9D31003508C643AB@MAILBOX.millicorp.com> Per the wiki: http://wiki.freeswitch.org/wiki/Webapi#xmlapi Using xmlapi should return the results in mod_xml_rpc as xml. It appears that the source is returning the same table data as webapi with an xml header which is being rejected by my browser. I found an old email in the group from 2010 with the exact same issue. Am I missing something or is this a known bug? Thanks Tim From anthony.minessale at gmail.com Fri Mar 9 21:02:03 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 9 Mar 2012 12:02:03 -0600 Subject: [Freeswitch-users] uuid_kill In-Reply-To: References: Message-ID: uuid_kill is designed to hangup call legs from remote applications. issues belong on JIRA with a list of details, stuck calls id too vague. there needs to be the exact version of FS and all the other questions the jira asks. Make sure you are using GIT HEAD. Your problem can be dealt with swiftly if you supply the right info. If you have a reproducible problem, turn on the debugging log and reproduce it so you can post a trace. from FS console, console loglevel 7 or from remote start fs_cli with -l7 sofia global siptrace on On Fri, Mar 9, 2012 at 8:22 AM, Ken Rice wrote: > There was a recent bug where this happened... Update to current git, and > check it again if you are still getting these hung calls, please help us out > by using gcore on the still running freeswitch with the hung calls to get a > core dump, then open that with gdb and follow the bug reporting info at > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 > and open a Jira. > > Since I am throw this out on the mailing list for this bug please check to > make sure someone hasn?t already opened a bug on it before opening a new > one. Stop but #freeswitch on irc.freenode.net (our irc channel) if you have > a box in this state and we can get a core dump but you need help on it. > (note, we?ll have to crash it to get the right coredump so you don?t want to > gcore it on a production system with live calls) > > K > > > > On 3/8/12 11:47 PM, "Ast Coder" wrote: > > I am just learning FreeSwitch and I see a lot of stuck calls. > > Why is that? Does that really mean there are active channels to PSTN even > though my SIP end-point hanged up minutes ago??There shouldn't be any need > for uuid_kill. Is this common to FreeSwitch to see calls getting stuck? > > Thanks, > > On Fri, Mar 9, 2012 at 12:19 AM, ? wrote: > > Hi, > > sometimes I have some stuck calls in FS. ?show calls? is showing them to me. > If I try to kill the call with ?uuid_kill ? the call won?t be killed? > I have to restart freeswitch, but I suppose this is not the best way for > this problem. Deleting from database doesn?t help too? > > Are there any other ideas? > > regards, > Benjamin T. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ________________________________ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From kris at kriskinc.com Fri Mar 9 21:03:54 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 9 Mar 2012 13:03:54 -0500 Subject: [Freeswitch-users] Adding 1khz or high resolution timer to CentOS for FS virtualization In-Reply-To: <4F594878.9060003@puzzled.xs4all.nl> References: <333789DE5C38474EB3A478A538F4EBAB0A32C20D3C@prod-exch01.corp.vseinc.com> <4F594878.9060003@puzzled.xs4all.nl> Message-ID: If you're using CentOS 6 you could also try mod_timerfd. On Thu, Mar 8, 2012 at 7:02 PM, Patrick Lists wrote: > On 08-03-12 18:35, bedgar at vseinc.com wrote: >> We have kernel 2.6.18-194.8.1.el5 with FS 1.0.6 and are testing >> virtualizing FS on ESX5. I know there is constant hashing out in the >> community. Please help in locating any past postings in guiding us in >> this direction. > > For starters I would use git MASTER as 1.0.6 is ancient. Both CentOS 5 & > 6 have 1000 Hz timers which is what you want. > > Just search the mailinglist for "virtual", "virtualized" and > "virtualisation" and you'll get a lot of hits. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From anthony.minessale at gmail.com Fri Mar 9 21:08:07 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 9 Mar 2012 12:08:07 -0600 Subject: [Freeswitch-users] Adding 1khz or high resolution timer to CentOS for FS virtualization In-Reply-To: References: <333789DE5C38474EB3A478A538F4EBAB0A32C20D3C@prod-exch01.corp.vseinc.com> <4F594878.9060003@puzzled.xs4all.nl> Message-ID: on centos 6 timerfd will self-install itself into the core and be used by default On Fri, Mar 9, 2012 at 12:03 PM, Kristian Kielhofner wrote: > If you're using CentOS 6 you could also try mod_timerfd. > > On Thu, Mar 8, 2012 at 7:02 PM, Patrick Lists > wrote: >> On 08-03-12 18:35, bedgar at vseinc.com wrote: >>> We have kernel 2.6.18-194.8.1.el5 with FS 1.0.6 and are testing >>> virtualizing FS on ESX5. I know there is constant hashing out in the >>> community. Please help in locating any past postings in guiding us in >>> this direction. >> >> For starters I would use git MASTER as 1.0.6 is ancient. Both CentOS 5 & >> 6 have 1000 Hz timers which is what you want. >> >> Just search the mailinglist for "virtual", "virtualized" and >> "virtualisation" and you'll get a lot of hits. >> >> Regards, >> Patrick >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From bedgar at vseinc.com Fri Mar 9 21:46:11 2012 From: bedgar at vseinc.com (bedgar at vseinc.com) Date: Fri, 9 Mar 2012 13:46:11 -0500 Subject: [Freeswitch-users] Adding 1khz or high resolution timer to CentOS for FS virtualization In-Reply-To: References: <333789DE5C38474EB3A478A538F4EBAB0A32C20D3C@prod-exch01.corp.vseinc.com> <4F594878.9060003@puzzled.xs4all.nl> Message-ID: <333789DE5C38474EB3A478A538F4EBAB0A32C21114@prod-exch01.corp.vseinc.com> Thank you. I will install core on CentOS 6.2 and take a look at the time_test results in fs_cli -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, March 09, 2012 1:08 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Adding 1khz or high resolution timer to CentOS for FS virtualization on centos 6 timerfd will self-install itself into the core and be used by default On Fri, Mar 9, 2012 at 12:03 PM, Kristian Kielhofner wrote: > If you're using CentOS 6 you could also try mod_timerfd. > > On Thu, Mar 8, 2012 at 7:02 PM, Patrick Lists > wrote: >> On 08-03-12 18:35, bedgar at vseinc.com wrote: >>> We have kernel 2.6.18-194.8.1.el5 with FS 1.0.6 and are testing >>> virtualizing FS on ESX5. I know there is constant hashing out in the >>> community. Please help in locating any past postings in guiding us >>> in this direction. >> >> For starters I would use git MASTER as 1.0.6 is ancient. Both CentOS >> 5 & >> 6 have 1000 Hz timers which is what you want. >> >> Just search the mailinglist for "virtual", "virtualized" and >> "virtualisation" and you'll get a lot of hits. >> >> Regards, >> Patrick >> >> _____________________________________________________________________ >> ____ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> http://www.freeswitch.org > > > > -- > Kristian Kielhofner > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jmesquita at freeswitch.org Fri Mar 9 22:05:00 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Fri, 9 Mar 2012 16:05:00 -0300 Subject: [Freeswitch-users] session-api - How to continue with A-leg after bridge-timeout? In-Reply-To: References: <748938CE1BF2401E86D9253FAB16066D@freeswitch.org> Message-ID: <1BC9C75671C64AB3ADC2CAFDBA2442D7@freeswitch.org> I am sorry Christian, I would need to defer this to someone with more knowledge on the matter. -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday, March 9, 2012 at 2:51 PM, Christian Benke wrote: > > Hmm, what i don't understand - when i rebuild the same route-logic in > > XML, the initiating channel is not hung up but continues the steps in > > the extension it has been transfered to > > > > > Attached is a log of this scenario, i've only added an additional > "info" to the XML-dialplan posted before. > > The initial channel bf969cf6-6a0f-11e1-8e5c-299021af10b6 is still > there at the very end of the transfered call - there must be some way > to get the same result in my script-based routing and be able to > reroute this initial channel? > > Regards, > Christian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > Attachments: > - freeswitch_XML_DP_log.txt > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/df4724c3/attachment-0001.html From kris at kriskinc.com Fri Mar 9 22:15:10 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 9 Mar 2012 14:15:10 -0500 Subject: [Freeswitch-users] Adding 1khz or high resolution timer to CentOS for FS virtualization In-Reply-To: References: <333789DE5C38474EB3A478A538F4EBAB0A32C20D3C@prod-exch01.corp.vseinc.com> <4F594878.9060003@puzzled.xs4all.nl> Message-ID: AWESOME! On Fri, Mar 9, 2012 at 1:08 PM, Anthony Minessale wrote: > on centos 6 timerfd will self-install itself into the core and be used > by default > -- Kristian Kielhofner From kris at kriskinc.com Fri Mar 9 22:17:40 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 9 Mar 2012 14:17:40 -0500 Subject: [Freeswitch-users] Send PAI and RPID In-Reply-To: References: Message-ID: Many commercial implementations (Cisco, for example) send both by default. As long as you have the same data in both there's really no reason not to send both. On Mon, Mar 5, 2012 at 6:09 PM, Anto wrote: > Hi > > Thank you very much. So asked to send both, as some providers accept > PAI and other RPID :-S. > > Best regards > Anto > > 2012/3/5 Ken Rice : >> You can send both... They arent really mutually exclusive... Just some >> providers whine about it... Its really redundant... Also RPID was never a >> RFC, it was a draft that has stuck around and refuses to die... PAI is the >> only ratified RFC standard >> >> >> >> On 3/5/12 1:42 PM, "Michael Collins" wrote: >> >> FYI, >> >> A quick glance through the source suggests that PAI and RPID are mutually >> exclusive - i.e. you can set one or the other but not both. I'll defer to >> the experts on whether or not the SIP spec says you SHOULD or SHOULD NOT >> have both headers in a single message. >> >> -MC >> >> On Thu, Mar 1, 2012 at 4:20 AM, Anto wrote: >> >> Hello >> >> To my FreeSWITCH servers, they come PAI and RPID headers, sent to the >> carrier but only one (the one I have configured with > name="caller-id-type" value="pid"/>). Is there any way to send both >> headers?. Thanks >> >> Best regards >> Anto >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> ________________________________ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From vipkilla at gmail.com Fri Mar 9 22:36:37 2012 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 9 Mar 2012 14:36:37 -0500 Subject: [Freeswitch-users] State of GUIs In-Reply-To: References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> Message-ID: On Fri, Mar 9, 2012 at 11:04 AM, Andrew Cassidy wrote: > As with a number of other people, I'm writing one to suit my needs too. Mine will have everything you could possibly need :) From bdfoster at endigotech.com Fri Mar 9 22:58:52 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 9 Mar 2012 14:58:52 -0500 Subject: [Freeswitch-users] Unhold message Message-ID: Hola, I'd like to document the unhold variable found here: http://wiki.freeswitch.org/wiki/Mod_dptools#U The page for this variable doesn't exist. Does anyone know if this variable still exists? If so, can someone explain it to me so I can wikify it? Thanks! -BDF -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/3f4ef260/attachment.html From asteriskcoding at gmail.com Fri Mar 9 23:04:18 2012 From: asteriskcoding at gmail.com (Ast Coder) Date: Fri, 9 Mar 2012 15:04:18 -0500 Subject: [Freeswitch-users] State of GUIs In-Reply-To: References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> Message-ID: We would be interested to put development time towards a project that allows for multi-tenancy, billing, and all in one Web GUI. Currently testing and hiring people to build it for us. Best, On Fri, Mar 9, 2012 at 2:36 PM, Vik Killa wrote: > On Fri, Mar 9, 2012 at 11:04 AM, Andrew Cassidy > wrote: > > As with a number of other people, I'm writing one to suit my needs too. > > Mine will have everything you could possibly need :) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/9f5fac74/attachment.html From sip at inbox.com Fri Mar 9 23:30:09 2012 From: sip at inbox.com (Jimmy Godbout) Date: Fri, 9 Mar 2012 12:30:09 -0800 Subject: [Freeswitch-users] Getting SIP verbose message Message-ID: Hi, Is it possible to get the phrase associated with a SIP code in a variable ? If I receive an error saying "insufficient funds", is it possible to retrieve it inside a variable in FS ? Thanks, Sipster ____________________________________________________________ GET FREE SMILEYS FOR YOUR IM & EMAIL - Learn more at http://www.inbox.com/smileys Works with AIM?, MSN? Messenger, Yahoo!? Messenger, ICQ?, Google Talk? and most webmails From fernandojdk at gmail.com Fri Mar 9 18:16:17 2012 From: fernandojdk at gmail.com (Fernando - NextBilling IP Solutions) Date: Fri, 9 Mar 2012 12:16:17 -0300 (Hora oficial do Brasil) Subject: [Freeswitch-users] State of GUIs References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> Message-ID: <4F5A1EC1.000001.26984@FLIGHTPC> I'm ready to help in PHP. Let's go. ? Atenciosamente, Importante: Esta mensagem, incluindo todo seu conte?do, cont?m informa??es confidenciais legalmente protegidas e destinadas a indiv?duo e prop?sito espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o mais breve poss?vel. As informa??es contidas nesta mensagem e em seu conte?do s?o de responsabilidade de seu autor, n?o representando necessariamente id?ias, opini?es, pensamentos ou qualquer forma de posicionamento por parte da NextBilling IP Solutions. P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." -------Original Message------- From: Roger Castaldo Date: 09/03/2012 12:09:22 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] State of GUIs Jay am I to take your response as you are interested in helping with the development of the project and have some background in C#? On Thu, Mar 8, 2012 at 10:11 AM, Anthony Minessale wrote: We encourage people to make GUIs for FreeSWITCH and we have mixed results. Being controlled externally is actually baked into the design of FreeSWITCH. Many of the abstract concepts int the core that people are still unearthing are all about developing external apps. The exercise to build one was intentionally left to the end user. The point was to harness the hard part about telephony so you could focus on your business use case. Really, based on the requirements of the original poster, I think a comercial solution is in order. I also would not judge activity by release dates / commits etc. FreeSWITCH itself has not been released since 2009 yet the C code has grown by 35 megabits since then with commits nearly hourly. We just happen to be more open about our development than others may be comfortable with. Anyway, I know for a fact all of the above list in the original post are still up and running but not everyone has the resources to run an active community. As referred to earlier in this thread, we made the CudaTEL PBX at Barracuda Networks which is an appliance designed to do everything described above and that effort could not be done without a commercial incentive to develop it and a team of dedicated employees. Everything you ever wanted is typically hard to come by when dealing with Free GUIs so the alternative is to invest in one of them and get them to focus on the functionality you seek. On Thu, Mar 8, 2012 at 2:17 AM, Brett Wilson wrote: I checked out freepbx. Freepbx v3, the complete rewrite that was supposed to work with freeswitch, actually got spun off into the blue.box project. Seems that the blue.box project is basically dead. The last commit was November 2011. Fusionpbx looks to be the only game in town. I was wrong, it is still being developed. I just did a new install of it today, and allowed it to auto update. There are indeed some changes that I see from my install from about 4 months ago. So that is a positive thing, but its too bad that the feel is still lacking. I took a look at some code, and what I saw did not impress me. Display code mixed right in with the logic. I did not look for more than about 30 seconds, but IMO it is not ?proper? application design. I am a big proponent of MVC architecture, and what I saw of fusionpbx code does not look all that impressive. I give credit where credit is due. The platform is functioning to an extent, most things actually work. I do realize that developing something of that size requires tons of effort. My ideal solution would be something driven by ExtJS and PHP on the backend. ExtJS provides the most advanced and rich javascript controls i have seen. Plus its sister product, sencha touch, would enable great admin functionality from a smartphone or tablet device. Here I go rambling on? I wish someone would develop an awesome GUI for freeswitch to help get it into the limelight where it deserves to be. If I had the time I would love to create a comprehensive and great looking GUI. Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: Brett.Wilson at launch3.net AOL IM: Brett.Wilson at launch3.net www.Launch3.net www.Launch3telecom.com *************************** From: Andrew Cassidy [mailto:andrew at cassidywebservices.co.uk] Sent: Wednesday, March 07, 2012 6:46 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] State of GUIs Look at http://www.freepybx.org although I didn't manage to get it installed I do have fusion installed on a text box although personally I'm not keen on it. On 7 March 2012 08:35, Brett Wilson wrote: Hey guys, I was wondering if anyone had any info on the current state of FS guis? It seems that most of the gui projects (blue.box, fusionpbx) have been mostly abandoned in terms of development. I would like something simple for end users to self-administer if needed. The problem I have with fusionpbx is that config files are overwritten by whatever is in the database. If you hand-edit a config file, fusion will not parse the file and load those settings into the interface. I realize that takes much more coding to do than using a database to simply write config files. But I feel that the freeswitch interface could be improved anyway. Unfortunately it seems that project has been abandoned. There have not been any releases since mid-2011. Anything new out or around the corner? Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: Brett.Wilson at launch3.net AOL IM: Brett.Wilson at launch3.net www.Launch3.net www.Launch3telecom.com *************************** _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/jpeg Size: 1715 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/827b076c/attachment-0007.jpe From rgarrett at garrettnet.net Fri Mar 9 19:08:35 2012 From: rgarrett at garrettnet.net (Robbie A. Garrett) Date: Fri, 9 Mar 2012 16:08:35 +0000 Subject: [Freeswitch-users] Unable to call cisco from freeswitch/blubox Message-ID: Hi All, My home use lab uses cisco call manager 8.0. My old setup was cisco call manager to asterisk and then to google voice. Since asterisk and google voice are now broken; I'm looking to move in the direction of free switch. Currently My cisco call manager can call any extension on free switch and leave voicemail's. When I add google talk to frees witch I'm able to call from my cisco VoIP phone, through free switch, and to google. My issue is that free switch is unable to call my cisco call manager. Below is an output of the log of a call from extension 2002 to extension 4000. From what I can tell is that free-switch is not even trying to talk to 10.0.1.99 (my call manger). It's failing before it gets that far. 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[G7221:115:32000:20:48000] 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[G7221:107:16000:20:32000] 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[G722:9:8000:20:64000] 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[PCMU:0:8000:20:64000] 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:2721 Set Codec sofia/sipinterface_1/peter_gibbons at bluebox PCMU/8000 20 ms 160 samples 64000 bits 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4457 Set 2833 dtmf send/recv payload to 101 2012-03-09 08:05:13.418262 [DEBUG] sofia.c:4732 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_NEW -> CS_INIT 2012-03-09 08:05:13.418262 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_INIT 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:342 (sofia/sipinterface_1/peter_gibbons at bluebox) State INIT 2012-03-09 08:05:13.419735 [DEBUG] mod_sofia.c:83 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA INIT 2012-03-09 08:05:13.419735 [DEBUG] mod_sofia.c:123 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_INIT -> CS_ROUTING 2012-03-09 08:05:13.419735 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:342 (sofia/sipinterface_1/peter_gibbons at bluebox) State INIT going to sleep 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_ROUTING 2012-03-09 08:05:13.419735 [DEBUG] switch_channel.c:1615 (sofia/sipinterface_1/peter_gibbons at bluebox) Callstate Change DOWN -> RINGING 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:345 (sofia/sipinterface_1/peter_gibbons at bluebox) State ROUTING 2012-03-09 08:05:13.419735 [DEBUG] mod_sofia.c:146 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA ROUTING 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:77 sofia/sipinterface_1/peter_gibbons at bluebox Standard ROUTING 2012-03-09 08:05:13.419735 [INFO] mod_dialplan_xml.c:331 Processing peter_gibbons ->4000 in context context_1 Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->conditioning_callerid] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [conditioning_callerid] ${internal_caller_id_number}(2002) =~ /^.+$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_name=${internal_caller_id_name}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_number=${internal_caller_id_number}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->postroute_global] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Absolute Condition [postroute_global] Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->preanswer_gtalk] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [preanswer_gtalk] source(mod_sofia) =~ /^mod_dingaling$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_trunk_1_pattern_5] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [main_trunk_1_pattern_5] destination_number(4000) =~ /^(40[0-9]{2})$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(prepend=) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [main_trunk_1_pattern_5] ${outbound_caller_id_number}(5555552002) =~ /^.+$/ break=never Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action export(sip_cid_type=rpid) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [main_trunk_1_pattern_5] destination_number(4000) =~ /^(40[0-9]{2})$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(failure_causes=NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action bridge(sofia/gateway/trunk_1/${prepend}4000) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_6] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_6] destination_number(4000) =~ /^2006$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_5] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_5] destination_number(4000) =~ /^2005$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_4] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_4] destination_number(4000) =~ /^2004$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_3] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_3] destination_number(4000) =~ /^2003$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_2] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_2] destination_number(4000) =~ /^2002$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_1] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_1] destination_number(4000) =~ /^2001$/ break=on-false 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:119 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_ROUTING -> CS_EXECUTE 2012-03-09 08:05:13.419735 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:345 (sofia/sipinterface_1/peter_gibbons at bluebox) State ROUTING going to sleep 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_EXECUTE 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:352 (sofia/sipinterface_1/peter_gibbons at bluebox) State EXECUTE 2012-03-09 08:05:13.419735 [DEBUG] mod_sofia.c:239 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA EXECUTE 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:157 sofia/sipinterface_1/peter_gibbons at bluebox Standard EXECUTE EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_name=Peter Gibbons) 2012-03-09 08:05:13.419735 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_name]=[Peter Gibbons] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_number=2002) 2012-03-09 08:05:13.419735 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_number]=[2002] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox hash(insert/10.0.1.3-spymap/peter_gibbons/bc1115e3-b592-454c-8813-a6a7fb2754d4) EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox hash(insert/10.0.1.3-last_dial/peter_gibbons/4000) EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox hash(insert/10.0.1.3-last_dial/global/bc1115e3-b592-454c-8813-a6a7fb2754d4) EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(RFC2822_DATE=Fri, 09 Mar 2012 08:05:13 -0800) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [RFC2822_DATE]=[Fri, 09 Mar 2012 08:05:13 -0800] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(prepend=) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [prepend]=[UNDEF] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_name=Peter Gibbons) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_name]=[Peter Gibbons] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_number=5555552002) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_number]=[5555552002] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox export(sip_cid_type=rpid) 2012-03-09 08:05:13.421692 [DEBUG] switch_channel.c:933 EXPORT (export_vars) [sip_cid_type]=[rpid] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(failure_causes=NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [failure_causes]=[NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox bridge(sofia/gateway/trunk_1/4000) 2012-03-09 08:05:13.421692 [DEBUG] switch_channel.c:890 sofia/sipinterface_1/peter_gibbons at bluebox EXPORTING[export_vars] [sip_cid_type]=[rpid] to event 2012-03-09 08:05:13.421692 [ERR] mod_sofia.c:3738 Invalid Gateway 2012-03-09 08:05:13.421692 [NOTICE] mod_sofia.c:4060 Close Channel N/A [CS_NEW] 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:434 () Running State Change CS_DESTROY 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:444 (N/A) State DESTROY 2012-03-09 08:05:13.421692 [DEBUG] mod_sofia.c:361 N/A SOFIA DESTROY 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:444 (N/A) State DESTROY going to sleep 2012-03-09 08:05:13.421692 [ERR] switch_ivr_originate.c:2605 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2012-03-09 08:05:13.421692 [DEBUG] switch_ivr_originate.c:3413 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT] 2012-03-09 08:05:13.421692 [INFO] mod_dptools.c:2579 Originate Failed. Cause: INVALID_NUMBER_FORMAT 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:2610 Failure causes [NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH]: Cause: INVALID_NUMBER_FORMAT 2012-03-09 08:05:13.421692 [NOTICE] switch_core_state_machine.c:189 sofia/sipinterface_1/peter_gibbons at bluebox has executed the last dialplan instruction, hanging up. 2012-03-09 08:05:13.421692 [DEBUG] switch_channel.c:2457 (sofia/sipinterface_1/peter_gibbons at bluebox) Callstate Change RINGING -> HANGUP 2012-03-09 08:05:13.421692 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/sipinterface_1/peter_gibbons at bluebox [CS_EXECUTE] [NORMAL_CLEARING] 2012-03-09 08:05:13.421692 [DEBUG] switch_channel.c:2473 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [KILL] 2012-03-09 08:05:13.421692 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:352 (sofia/sipinterface_1/peter_gibbons at bluebox) State EXECUTE going to sleep 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_HANGUP 2012-03-09 08:05:13.423711 [DEBUG] switch_core_state_machine.c:539 (sofia/sipinterface_1/peter_gibbons at bluebox) State HANGUP 2012-03-09 08:05:13.423711 [DEBUG] mod_sofia.c:456 Channel sofia/sipinterface_1/peter_gibbons at bluebox hanging up, cause: NORMAL_CLEARING 2012-03-09 08:05:13.427318 [DEBUG] mod_sofia.c:518 Responding to INVITE with: 480 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:46 sofia/sipinterface_1/peter_gibbons at bluebox Standard HANGUP, cause: NORMAL_CLEARING 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:539 (sofia/sipinterface_1/peter_gibbons at bluebox) State HANGUP going to sleep 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:337 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_HANGUP -> CS_REPORTING 2012-03-09 08:05:13.427318 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_REPORTING 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:599 (sofia/sipinterface_1/peter_gibbons at bluebox) State REPORTING 2012-03-09 08:05:13.586710 [DEBUG] switch_core_state_machine.c:53 sofia/sipinterface_1/peter_gibbons at bluebox Standard REPORTING, cause: NORMAL_CLEARING 2012-03-09 08:05:13.586710 [DEBUG] switch_core_state_machine.c:599 (sofia/sipinterface_1/peter_gibbons at bluebox) State REPORTING going to sleep 2012-03-09 08:05:13.586710 [DEBUG] switch_core_state_machine.c:331 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_REPORTING -> CS_DESTROY 2012-03-09 08:05:13.586710 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.586710 [DEBUG] switch_core_session.c:1224 Session 66 (sofia/sipinterface_1/peter_gibbons at bluebox) Locked, Waiting on external entities 2012-03-09 08:05:13.586710 [NOTICE] switch_core_session.c:1242 Session 66 (sofia/sipinterface_1/peter_gibbons at bluebox) Ended 2012-03-09 08:05:13.586710 [NOTICE] switch_core_session.c:1244 Close Channel sofia/sipinterface_1/peter_gibbons at bluebox [CS_DESTROY] 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:431 (sofia/sipinterface_1/peter_gibbons at bluebox) Callstate Change HANGUP -> DOWN 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:434 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_DESTROY 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:444 (sofia/sipinterface_1/peter_gibbons at bluebox) State DESTROY 2012-03-09 08:05:13.588724 [DEBUG] mod_sofia.c:361 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA DESTROY 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:60 sofia/sipinterface_1/peter_gibbons at bluebox Standard DESTROY 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:444 (sofia/sipinterface_1/peter_gibbons at bluebox) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/1260f9ea/attachment-0001.html From freeswitch at earthspike.net Fri Mar 9 23:55:57 2012 From: freeswitch at earthspike.net (John) Date: Fri, 09 Mar 2012 20:55:57 +0000 Subject: [Freeswitch-users] session-api - How to continue with A-leg after bridge-timeout? In-Reply-To: References: <748938CE1BF2401E86D9253FAB16066D@freeswitch.org> Message-ID: <4F5A6E5D.3030306@earthspike.net> On 09/03/12 17:51, Christian Benke wrote: >> Hmm, what i don't understand - when i rebuild the same route-logic in >> XML, the initiating channel is not hung up but continues the steps in >> the extension it has been transfered to > Attached is a log of this scenario, i've only added an additional > "info" to the XML-dialplan posted before. > > The initial channel bf969cf6-6a0f-11e1-8e5c-299021af10b6 is still > there at the very end of the transfered call - there must be some way > to get the same result in my script-based routing and be able to > reroute this initial channel? > Christian, It sounds like something that would be better driven from the ESL rather than a script that effectively forms part of the dialplan. As Jo?o says, when you lose the leg you are executing inside, you lose control. Also, you are far better posting logs to pastebin.freeswitch.org. The URL stays in the original post whereas the attachment gets lost further down the post chain. And they are easier to read. John From freeswitch at earthspike.net Fri Mar 9 23:58:45 2012 From: freeswitch at earthspike.net (John) Date: Fri, 09 Mar 2012 20:58:45 +0000 Subject: [Freeswitch-users] Please help me would like to use PHP ESL In-Reply-To: References: Message-ID: <4F5A6F05.2010706@earthspike.net> On 09/03/12 10:34, Mohamed Alam wrote: > I am new to this. > I would like to use PHP ESL for text chat,voice chat. > How to create database. Where to add users. > How voice chat will be generated. > Please help me Mohamed, Your question is too vague. You need to tell us what you are trying to achieve (the end result), what you have tried and what doesn't work. Put source code and logs in pastebin.freeswitch.org and, if you think you have found a bug, make sure you are using the latest version (git HEAD). Then someone might be able to help. (I cannot as my PHP is too rusty.) John From aksrini at hotmail.com Sat Mar 10 00:01:04 2012 From: aksrini at hotmail.com (Srini K) Date: Fri, 9 Mar 2012 13:01:04 -0800 Subject: [Freeswitch-users] =?windows-1256?q?How_to_get_session_obj_from_u?= =?windows-1256?q?uid=FE?= Message-ID: Hi, How to get the Session object if I know the uuid using mod_managed. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/44d4c3d9/attachment.html From jmesquita at freeswitch.org Sat Mar 10 00:01:02 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Fri, 9 Mar 2012 18:01:02 -0300 Subject: [Freeswitch-users] State of GUIs In-Reply-To: <4F5A1EC1.000001.26984@FLIGHTPC> References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> <4F5A1EC1.000001.26984@FLIGHTPC> Message-ID: Love to see Brazilians take initiative! Way to go Fernando!!! :-) Abra?os, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday, March 9, 2012 at 12:16 PM, Fernando - NextBilling IP Solutions wrote: > I'm ready to help in PHP. > > Let's go. > > > > Atenciosamente, > > Importante: > Esta mensagem, incluindo todo seu conte?do, cont?m informa??es confidenciais, legalmente protegidas e destinadas a indiv?duo e prop?sito espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o mais breve poss?vel. > As informa??es contidas nesta mensagem e em seu conte?do s?o de responsabilidade de seu autor, n?o representando necessariamente id?ias, opini?es, pensamentos ou qualquer forma de posicionamento por parte da NextBilling IP Solutions. > > > P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." > > > -------Original Message------- > > From: Roger Castaldo (mailto:roger.castaldo at gmail.com) > Date: 09/03/2012 12:09:22 > To: FreeSWITCH Users Help (mailto:freeswitch-users at lists.freeswitch.org) > Subject: Re: [Freeswitch-users] State of GUIs > > > Jay am I to take your response as you are interested in helping with the development of the project and have some background in C#? > > On Thu, Mar 8, 2012 at 10:11 AM, Anthony Minessale wrote: > We encourage people to make GUIs for FreeSWITCH and we have mixed results. Being controlled externally is actually baked into the design of FreeSWITCH. Many of the abstract concepts int the core that people are still unearthing are all about developing external apps. The exercise to build one was intentionally left to the end user. The point was to harness the hard part about telephony so you could focus on your business use case. Really, based on the requirements of the original poster, I think a comercial solution is in order. I also would not judge activity by release dates / commits etc. FreeSWITCH itself has not been released since 2009 yet the C code has grown by 35 megabits since then with commits nearly hourly. We just happen to be more open about our development than others may be comfortable with. > > Anyway, I know for a fact all of the above list in the original post are still up and running but not everyone has the resources to run an active community. > > As referred to earlier in this thread, we made the CudaTEL PBX at Barracuda Networks which is an appliance designed to do everything described above and that effort could not be done without a commercial incentive to develop it and a team of dedicated employees. > > Everything you ever wanted is typically hard to come by when dealing with Free GUIs so the alternative is to invest in one of them and get them to focus on the functionality you seek. > > > > > > > > On Thu, Mar 8, 2012 at 2:17 AM, Brett Wilson wrote: > > I checked out freepbx. Freepbx v3, the complete rewrite that was supposed to work with freeswitch, actually got spun off into the blue.box project. Seems that the blue.box project is basically dead. The last commit was November 2011. Fusionpbx looks to be the only game in town. I was wrong, it is still being developed. I just did a new install of it today, and allowed it to auto update. There are indeed some changes that I see from my install from about 4 months ago. So that is a positive thing, but its too bad that the feel is still lacking. I took a look at some code, and what I saw did not impress me. Display code mixed right in with the logic. I did not look for more than about 30 seconds, but IMO it is not ?proper? application design. I am a big proponent of MVC architecture, and what I saw of fusionpbx code does not look all that impressive. I give credit where credit is due. The platform is functioning to an extent, most things actually work. I do realize that developing something of that size requires tons of effort. My ideal solution would be something driven by ExtJS and PHP on the backend. ExtJS provides the most advanced and rich javascript controls i have seen. Plus its sister product, sencha touch, would enable great admin functionality from a smartphone or tablet device. Here I go rambling on? I wish someone would develop an awesome GUI for freeswitch to help get it into the limelight where it deserves to be. If I had the time I would love to create a comprehensive and great looking GUI. > > > > > > Brett Wilson > > > IT Department > > > Launch 3 Ventures, LLC > > > 134 Myer Street > > > Hackensack, NJ 07601 > > > Phone: 877.878.9134 (tel:877.878.9134) > Fax: 646.536.3866 (tel:646.536.3866) > > > Email: Brett.Wilson at launch3.net (mailto:Brett.Wilson at launch3.net) > > > AOL IM: Brett.Wilson at launch3.net (mailto:Brett.Wilson at launch3.net) > > > www.Launch3.net > > > www.Launch3telecom.com (http://www.launch3telecom.com/) > > > *************************** > > > > > > > > > > From: Andrew Cassidy [mailto:andrew at cassidywebservices.co.uk] > Sent: Wednesday, March 07, 2012 6:46 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] State of GUIs > > > > > > Look at http://www.freepybx.org although I didn't manage to get it installed. I do have fusion installed on a text box although personally I'm not keen on it. > > > > > > > On 7 March 2012 08:35, Brett Wilson wrote: > > > Hey guys, > > > I was wondering if anyone had any info on the current state of FS guis? It seems that most of the gui projects (blue.box, fusionpbx) have been mostly abandoned in terms of development. I would like something simple for end users to self-administer if needed. The problem I have with fusionpbx is that config files are overwritten by whatever is in the database. If you hand-edit a config file, fusion will not parse the file and load those settings into the interface. I realize that takes much more coding to do than using a database to simply write config files. But I feel that the freeswitch interface could be improved anyway. Unfortunately it seems that project has been abandoned. There have not been any releases since mid-2011. > > > > > > Anything new out or around the corner? > > > > > > Brett Wilson > > > IT Department > > > Launch 3 Ventures, LLC > > > 134 Myer Street > > > Hackensack, NJ 07601 > > > Phone: 877.878.9134 (tel:877.878.9134) > Fax: 646.536.3866 (tel:646.536.3866) > > > Email: Brett.Wilson at launch3.net (mailto:Brett.Wilson at launch3.net) > > > AOL IM: Brett.Wilson at launch3.net (mailto:Brett.Wilson at launch3.net) > > > www.Launch3.net (http://www.Launch3.net) > > > www.Launch3telecom.com (http://www.launch3telecom.com/) > > > *************************** > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > > > > > > > -- > Andrew Cassidy BSc (Hons) MBCS > > > Managing Director; Cassidy Web Services Ltd > > > > T: 03300 100 960 F: 03300 100 961 > > > > E: andrew at cassidywebservices.co.uk (mailto:andrew at cassidywebservices.co.uk) > > > > W: www.cassidywebservices.co.uk (http://www.cassidywebservices.co.uk) > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com (mailto:MSN%3Aanthony_minessale at hotmail.com) > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:PAYPAL%3Aanthony.minessale at gmail.com) > IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org (mailto:sip%3A888 at conference.freeswitch.org) > googletalk:conf+888 at conference.freeswitch.org (mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org) > pstn:+19193869900 (tel:%2B19193869900) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image0032.jpg Type: image/jpeg Size: 1715 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/580ce598/attachment-0007.jpg From benkokakao at gmail.com Sat Mar 10 00:28:39 2012 From: benkokakao at gmail.com (Christian Benke) Date: Fri, 9 Mar 2012 22:28:39 +0100 Subject: [Freeswitch-users] session-api - How to continue with A-leg after bridge-timeout? In-Reply-To: <4F5A6E5D.3030306@earthspike.net> References: <748938CE1BF2401E86D9253FAB16066D@freeswitch.org> <4F5A6E5D.3030306@earthspike.net> Message-ID: On 9 March 2012 21:55, John wrote: > It sounds like something that would be better driven from the ESL rather > than a script that effectively forms part of the dialplan. ?As Jo?o > says, when you lose the leg you are executing inside, you lose control. Hmm, I think i found the cause of the problem - Polycom is handling "Attended Transfers" resp. "Consultation Transfers" somewhat unorthodox, where B does not have to wait for C to pickup - FS can't cope well with this situation. A proper Attended Transfer or a proper Blind Transfer handles these situations much better. Anthony pointed this out to me offlist on an other issue. In essence, i have the option to reconfigure my Polycoms to not allow to transfer unanswered Attended Transfers with the parameter "voIpProt.SIP.allowTransferOnProceeding" - and instruct my clients how to use a "real" Blind Transfer instead. With a proper Blind Transfer, A sends a invite to C and the call is reinitated properly from the start. This issue was really bugging me for 2 weeks now, but the solution appears simple and at a totally different place than i had looked. Who would have thought? :-) Sometimes you lose sight of the forest for the trees... > Also, you are far better posting logs to pastebin.freeswitch.org. ?The > URL stays in the original post whereas the attachment gets lost further > down the post chain. ?And they are easier to read. Ah, yeah, thanks. I was so distracted and cocerned that i forgot about that ;-) Cheers, Christian From benkokakao at gmail.com Sat Mar 10 00:32:27 2012 From: benkokakao at gmail.com (Christian Benke) Date: Fri, 9 Mar 2012 22:32:27 +0100 Subject: [Freeswitch-users] session-api - How to continue with A-leg after bridge-timeout? In-Reply-To: References: <748938CE1BF2401E86D9253FAB16066D@freeswitch.org> <4F5A6E5D.3030306@earthspike.net> Message-ID: > Hmm, I think i found the cause of the problem :%s/cause/solution/ From gabe at gundy.org Sat Mar 10 00:54:34 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Fri, 9 Mar 2012 14:54:34 -0700 Subject: [Freeswitch-users] State of GUIs In-Reply-To: References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> Message-ID: On Fri, Mar 9, 2012 at 9:04 AM, Andrew Cassidy wrote: > As with a number of other people, I'm writing one to suit my needs too. Mine > is being written in Python/Django and is aimed at being multi tenant in a > cloud setting or single user on a manged device setting sharing as much of > the code from both scenarios as possible. If you're doing Django (or any Python), check out Parseltone: https://parseltone.org/browser/trunk/parseltone/django/apps/freeswitch/ It's all under the MPL. Right now it's a bit of this and that, some real good stuff and a little cruft. But, feel free to look around and use what you like. When we can get some of our projects done, we'll revisit that and give it some love. All this is done in the release early and release often mindset :) Best, Gabe From spautz2 at telefaks.biz Sat Mar 10 01:29:33 2012 From: spautz2 at telefaks.biz (David Spautz) Date: Fri, 09 Mar 2012 23:29:33 +0100 Subject: [Freeswitch-users] Debian lenny + Freeswitch + mod_event_zmq + module load error In-Reply-To: References: Message-ID: <4F5A844D.9010500@telefaks.biz> Hi, when i compile the last freeswitch revision from git with mod_event_zmq enabled there are no errors. But if i run freeswitch from the command line, i get this error 2012-03-09 23:13:36.010451 [CRIT] switch_loadable_module.c:1290 Error Loading module /usr/local/freeswitch/mod/mod_event_zmq.so **/usr/local/freeswitch/mod/mod_event_zmq.so: undefined symbol: uuid_generate** I have installed and removed different versions of libzmq libzmq-dev, uuid, libuuid, libuuid-dev. Some from debian sit, other some from ubuntu( because mod_event_zmq run under ubuntu withot any errors.), and still others direct from the original sources. But nothing helped. How can i use mod_event_zmq under debian lenny? Which versions of libraries i need? Thanks for help. David From curriegrad2004 at gmail.com Sat Mar 10 02:45:59 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 9 Mar 2012 15:45:59 -0800 Subject: [Freeswitch-users] Unhold message In-Reply-To: References: Message-ID: According to mod_dptools.c: "Send a un-hold message" On Fri, Mar 9, 2012 at 11:58 AM, Brian Foster wrote: > Hola, > > I'd like to document the unhold variable found here: > > http://wiki.freeswitch.org/wiki/Mod_dptools#U > > The page for this variable doesn't exist. Does anyone know if this variable > still exists? If so, can someone explain it to me so I can wikify it? > > Thanks! > > -BDF > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kjoseph.us at gmail.com Sat Mar 10 02:47:33 2012 From: kjoseph.us at gmail.com (Joseph Khoury) Date: Fri, 9 Mar 2012 15:47:33 -0800 Subject: [Freeswitch-users] State of GUIs In-Reply-To: References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> Message-ID: We are working on one using Node.js, autoscaling distributed system. We welcome all help. We will publish the source on github when we reach alpha state. If you are interested fell free to contact me. Big thank you for Anthony and the team. Joseph ABpin On Fri, Mar 9, 2012 at 1:54 PM, Gabriel Gunderson wrote: > On Fri, Mar 9, 2012 at 9:04 AM, Andrew Cassidy > wrote: > > As with a number of other people, I'm writing one to suit my needs too. > Mine > > is being written in Python/Django and is aimed at being multi tenant in a > > cloud setting or single user on a manged device setting sharing as much > of > > the code from both scenarios as possible. > > If you're doing Django (or any Python), check out Parseltone: > > https://parseltone.org/browser/trunk/parseltone/django/apps/freeswitch/ > > It's all under the MPL. Right now it's a bit of this and that, some > real good stuff and a little cruft. But, feel free to look around and > use what you like. > > When we can get some of our projects done, we'll revisit that and give > it some love. All this is done in the release early and release often > mindset :) > > Best, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/40dac73a/attachment.html From curriegrad2004 at gmail.com Sat Mar 10 02:49:25 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 9 Mar 2012 15:49:25 -0800 Subject: [Freeswitch-users] Unhold message In-Reply-To: References: Message-ID: Also forgot to mention, yes this is still present... But... It's not a variable, it's an application. Been in FS since the svn days really On Fri, Mar 9, 2012 at 3:45 PM, curriegrad2004 wrote: > According to mod_dptools.c: > > "Send a un-hold message" > > On Fri, Mar 9, 2012 at 11:58 AM, Brian Foster wrote: >> Hola, >> >> I'd like to document the unhold variable found here: >> >> http://wiki.freeswitch.org/wiki/Mod_dptools#U >> >> The page for this variable doesn't exist. Does anyone know if this variable >> still exists? If so, can someone explain it to me so I can wikify it? >> >> Thanks! >> >> -BDF >> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> From kjoseph.us at gmail.com Sat Mar 10 03:35:15 2012 From: kjoseph.us at gmail.com (Joseph Khoury) Date: Fri, 9 Mar 2012 16:35:15 -0800 Subject: [Freeswitch-users] State of GUIs In-Reply-To: References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> Message-ID: feel* :) not fell On Fri, Mar 9, 2012 at 3:47 PM, Joseph Khoury wrote: > We are working on one using Node.js, autoscaling distributed system. We > welcome all help. We will publish the source on github when we reach alpha > state. > If you are interested fell free to contact me. > > Big thank you for Anthony and the team. > > Joseph > ABpin > > On Fri, Mar 9, 2012 at 1:54 PM, Gabriel Gunderson wrote: > >> On Fri, Mar 9, 2012 at 9:04 AM, Andrew Cassidy >> wrote: >> > As with a number of other people, I'm writing one to suit my needs too. >> Mine >> > is being written in Python/Django and is aimed at being multi tenant in >> a >> > cloud setting or single user on a manged device setting sharing as much >> of >> > the code from both scenarios as possible. >> >> If you're doing Django (or any Python), check out Parseltone: >> >> https://parseltone.org/browser/trunk/parseltone/django/apps/freeswitch/ >> >> It's all under the MPL. Right now it's a bit of this and that, some >> real good stuff and a little cruft. But, feel free to look around and >> use what you like. >> >> When we can get some of our projects done, we'll revisit that and give >> it some love. All this is done in the release early and release often >> mindset :) >> >> Best, >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/486f6ad0/attachment-0001.html From jmesquita at freeswitch.org Sat Mar 10 07:38:41 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Sat, 10 Mar 2012 01:38:41 -0300 Subject: [Freeswitch-users] State of GUIs In-Reply-To: References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> Message-ID: <07CA705635574091A0842F676EADB31D@freeswitch.org> You don't handle a dynamic dialplan on this? I am not very fond of Django and it has been a while since I last touched it so excuse me if I am mistaken. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday, March 9, 2012 at 6:54 PM, Gabriel Gunderson wrote: > On Fri, Mar 9, 2012 at 9:04 AM, Andrew Cassidy > wrote: > > As with a number of other people, I'm writing one to suit my needs too. Mine > > is being written in Python/Django and is aimed at being multi tenant in a > > cloud setting or single user on a manged device setting sharing as much of > > the code from both scenarios as possible. > > > > > If you're doing Django (or any Python), check out Parseltone: > > https://parseltone.org/browser/trunk/parseltone/django/apps/freeswitch/ > > It's all under the MPL. Right now it's a bit of this and that, some > real good stuff and a little cruft. But, feel free to look around and > use what you like. > > When we can get some of our projects done, we'll revisit that and give > it some love. All this is done in the release early and release often > mindset :) > > Best, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120310/cbe1d6af/attachment.html From gabe at gundy.org Sat Mar 10 10:47:45 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 10 Mar 2012 00:47:45 -0700 Subject: [Freeswitch-users] State of GUIs In-Reply-To: <07CA705635574091A0842F676EADB31D@freeswitch.org> References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> <07CA705635574091A0842F676EADB31D@freeswitch.org> Message-ID: 2012/3/9 Jo?o Mesquita : > You don't handle a dynamic dialplan on this? I am not very fond of Django > and it has been a while since I last touched it so excuse me if I am > mistaken. Sorry you're not a fan of Django. I really enjoy it :) It keeps us productive and helps us get things done. Anyway, most of the dynamic stuff built for mod_xml_curl is for configuration and CDRs. If you're familiar with Django, you'll see that Parseltone gives you some-pre built views that provide a little logic and create the context to render the templates in. You have the actual templates. You have some models that provide a data backend for some of the more obvious configs (IVRs, ACLs etc.) and the Admin configuration to make it all pretty. You can find most of that with the link I shared earlier. Again, at it's current state, it's just a helpful collections of Python modules that all have to do with some aspect of FreeSWITCH, nothing more. Besides Django stuff, there is a pretty nice Event Socket library: https://parseltone.org/browser/trunk/parseltone/eventsocket The Twisted stuff is further along than the Django stuff, but even so, there is so much more we want to do with it. One cool thing about the current EventSocket stuff is that it has 3 modes of operation... each one a little more advanced than the last: 1) Raw Event Socket: Just connect and handle the passing of the data. 2) Basic Event Socket: Adds parsing of responses, methods for api and bgapi and well... basic operation. 3) Event Socket This adds the neat feature of tying FreeSWITCH's bgapi and Twisted's deferreds together. One last think we're getting ready to add is a WSGI component that fits between any WSGI client/server (so you don't have to use it with Django, but because it's WSGI, you can). For example, a Flask user could put this between Apache's mod_wsgi and Flask. Anyway, it captures all of the POST data and any other information from mod_xml_curl, saves it and then passes it on to Django/Flask/etc. for processing. It then gets the response and saves it to a database so the it can be examined at a later time. It also can display that data using it's own web interface. Fun stuff. Well, I shouldn't get too off topic. Anyone who is interested should consider hitting me off-list. Gabe Hope someone finds something useful there :) We'll get back on it when we have time... heh. Gabe From sdevoy at bizfocused.com Sat Mar 10 00:22:46 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 9 Mar 2012 16:22:46 -0500 Subject: [Freeswitch-users] Unable to call cisco from freeswitch/blubox In-Reply-To: References: Message-ID: <0dcd01ccfe3a$c5ff4160$51fdc420$@com> Hi Robbie, I am no expert, but I can help move this along some. The first error is "2012-03-09 08:05:13.421692 [ERR] mod_sofia.c:3738 Invalid Gateway" which is from the dial plan statement "bridge(sofia/gateway/trunk_1/${prepend}4000)" FS doesn't seem to know what "trunk_1" is. I had this and resolved it on my FS (although it was to a VOIP provider gateway). In your SIP configuration (where you define things like port 5060) you must have a section that must contain a entity. Working out the params and variables for your gateway I cannot help as I don't know call manager at all. I had not defined my gateway WITHIN a sip interface, so FS didn't really know how to get to it. If you have it defined, you should see it in ./fs_cli with a "sofia status" command. Hope that helps some. Sean From: Robbie A. Garrett [mailto:rgarrett at garrettnet.net] Sent: Friday, March 09, 2012 11:09 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Unable to call cisco from freeswitch/blubox Hi All, My home use lab uses cisco call manager 8.0. My old setup was cisco call manager to asterisk and then to google voice. Since asterisk and google voice are now broken; I'm looking to move in the direction of free switch. Currently My cisco call manager can call any extension on free switch and leave voicemail's. When I add google talk to frees witch I'm able to call from my cisco VoIP phone, through free switch, and to google. My issue is that free switch is unable to call my cisco call manager. Below is an output of the log of a call from extension 2002 to extension 4000. >From what I can tell is that free-switch is not even trying to talk to 10.0.1.99 (my call manger). It's failing before it gets that far. 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[G7221:115:32000:20:48000] 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[G7221:107:16000:20:32000] 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[G722:9:8000:20:64000] 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[PCMU:0:8000:20:64000] 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:2721 Set Codec sofia/sipinterface_1/peter_gibbons at bluebox PCMU/8000 20 ms 160 samples 64000 bits 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4457 Set 2833 dtmf send/recv payload to 101 2012-03-09 08:05:13.418262 [DEBUG] sofia.c:4732 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_NEW -> CS_INIT 2012-03-09 08:05:13.418262 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_INIT 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:342 (sofia/sipinterface_1/peter_gibbons at bluebox) State INIT 2012-03-09 08:05:13.419735 [DEBUG] mod_sofia.c:83 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA INIT 2012-03-09 08:05:13.419735 [DEBUG] mod_sofia.c:123 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_INIT -> CS_ROUTING 2012-03-09 08:05:13.419735 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:342 (sofia/sipinterface_1/peter_gibbons at bluebox) State INIT going to sleep 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_ROUTING 2012-03-09 08:05:13.419735 [DEBUG] switch_channel.c:1615 (sofia/sipinterface_1/peter_gibbons at bluebox) Callstate Change DOWN -> RINGING 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:345 (sofia/sipinterface_1/peter_gibbons at bluebox) State ROUTING 2012-03-09 08:05:13.419735 [DEBUG] mod_sofia.c:146 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA ROUTING 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:77 sofia/sipinterface_1/peter_gibbons at bluebox Standard ROUTING 2012-03-09 08:05:13.419735 [INFO] mod_dialplan_xml.c:331 Processing peter_gibbons ->4000 in context context_1 Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->conditioning_callerid] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [conditioning_callerid] ${internal_caller_id_number}(2002) =~ /^.+$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_name=${internal_caller_id_name}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_number=${internal_caller_id_number}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->postroute_global] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Absolute Condition [postroute_global] Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe r}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->preanswer_gtalk] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [preanswer_gtalk] source(mod_sofia) =~ /^mod_dingaling$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_trunk_1_pattern_5] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [main_trunk_1_pattern_5] destination_number(4000) =~ /^(40[0-9]{2})$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(prepend=) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [main_trunk_1_pattern_5] ${outbound_caller_id_number}(5555552002) =~ /^.+$/ break=never Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action export(sip_cid_type=rpid) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [main_trunk_1_pattern_5] destination_number(4000) =~ /^(40[0-9]{2})$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(failure_causes=NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action bridge(sofia/gateway/trunk_1/${prepend}4000) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_6] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_6] destination_number(4000) =~ /^2006$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_5] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_5] destination_number(4000) =~ /^2005$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_4] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_4] destination_number(4000) =~ /^2004$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_3] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_3] destination_number(4000) =~ /^2003$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_2] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_2] destination_number(4000) =~ /^2002$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_1] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_1] destination_number(4000) =~ /^2001$/ break=on-false 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:119 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_ROUTING -> CS_EXECUTE 2012-03-09 08:05:13.419735 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:345 (sofia/sipinterface_1/peter_gibbons at bluebox) State ROUTING going to sleep 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_EXECUTE 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:352 (sofia/sipinterface_1/peter_gibbons at bluebox) State EXECUTE 2012-03-09 08:05:13.419735 [DEBUG] mod_sofia.c:239 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA EXECUTE 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:157 sofia/sipinterface_1/peter_gibbons at bluebox Standard EXECUTE EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_name=Peter Gibbons) 2012-03-09 08:05:13.419735 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_name]=[Peter Gibbons] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_number=2002) 2012-03-09 08:05:13.419735 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_number]=[2002] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox hash(insert/10.0.1.3-spymap/peter_gibbons/bc1115e3-b592-454c-8813-a6a7fb2754 d4) EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox hash(insert/10.0.1.3-last_dial/peter_gibbons/4000) EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox hash(insert/10.0.1.3-last_dial/global/bc1115e3-b592-454c-8813-a6a7fb2754d4) EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(RFC2822_DATE=Fri, 09 Mar 2012 08:05:13 -0800) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [RFC2822_DATE]=[Fri, 09 Mar 2012 08:05:13 -0800] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(prepend=) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [prepend]=[UNDEF] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_name=Peter Gibbons) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_name]=[Peter Gibbons] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_number=5555552002) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_number]=[5555552002] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox export(sip_cid_type=rpid) 2012-03-09 08:05:13.421692 [DEBUG] switch_channel.c:933 EXPORT (export_vars) [sip_cid_type]=[rpid] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(failure_causes=NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [failure_causes]=[NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox bridge(sofia/gateway/trunk_1/4000) 2012-03-09 08:05:13.421692 [DEBUG] switch_channel.c:890 sofia/sipinterface_1/peter_gibbons at bluebox EXPORTING[export_vars] [sip_cid_type]=[rpid] to event 2012-03-09 08:05:13.421692 [ERR] mod_sofia.c:3738 Invalid Gateway 2012-03-09 08:05:13.421692 [NOTICE] mod_sofia.c:4060 Close Channel N/A [CS_NEW] 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:434 () Running State Change CS_DESTROY 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:444 (N/A) State DESTROY 2012-03-09 08:05:13.421692 [DEBUG] mod_sofia.c:361 N/A SOFIA DESTROY 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:444 (N/A) State DESTROY going to sleep 2012-03-09 08:05:13.421692 [ERR] switch_ivr_originate.c:2605 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2012-03-09 08:05:13.421692 [DEBUG] switch_ivr_originate.c:3413 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT] 2012-03-09 08:05:13.421692 [INFO] mod_dptools.c:2579 Originate Failed. Cause: INVALID_NUMBER_FORMAT 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:2610 Failure causes [NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH]: Cause: INVALID_NUMBER_FORMAT 2012-03-09 08:05:13.421692 [NOTICE] switch_core_state_machine.c:189 sofia/sipinterface_1/peter_gibbons at bluebox has executed the last dialplan instruction, hanging up. 2012-03-09 08:05:13.421692 [DEBUG] switch_channel.c:2457 (sofia/sipinterface_1/peter_gibbons at bluebox) Callstate Change RINGING -> HANGUP 2012-03-09 08:05:13.421692 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/sipinterface_1/peter_gibbons at bluebox [CS_EXECUTE] [NORMAL_CLEARING] 2012-03-09 08:05:13.421692 [DEBUG] switch_channel.c:2473 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [KILL] 2012-03-09 08:05:13.421692 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:352 (sofia/sipinterface_1/peter_gibbons at bluebox) State EXECUTE going to sleep 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_HANGUP 2012-03-09 08:05:13.423711 [DEBUG] switch_core_state_machine.c:539 (sofia/sipinterface_1/peter_gibbons at bluebox) State HANGUP 2012-03-09 08:05:13.423711 [DEBUG] mod_sofia.c:456 Channel sofia/sipinterface_1/peter_gibbons at bluebox hanging up, cause: NORMAL_CLEARING 2012-03-09 08:05:13.427318 [DEBUG] mod_sofia.c:518 Responding to INVITE with: 480 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:46 sofia/sipinterface_1/peter_gibbons at bluebox Standard HANGUP, cause: NORMAL_CLEARING 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:539 (sofia/sipinterface_1/peter_gibbons at bluebox) State HANGUP going to sleep 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:337 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_HANGUP -> CS_REPORTING 2012-03-09 08:05:13.427318 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_REPORTING 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:599 (sofia/sipinterface_1/peter_gibbons at bluebox) State REPORTING 2012-03-09 08:05:13.586710 [DEBUG] switch_core_state_machine.c:53 sofia/sipinterface_1/peter_gibbons at bluebox Standard REPORTING, cause: NORMAL_CLEARING 2012-03-09 08:05:13.586710 [DEBUG] switch_core_state_machine.c:599 (sofia/sipinterface_1/peter_gibbons at bluebox) State REPORTING going to sleep 2012-03-09 08:05:13.586710 [DEBUG] switch_core_state_machine.c:331 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_REPORTING -> CS_DESTROY 2012-03-09 08:05:13.586710 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.586710 [DEBUG] switch_core_session.c:1224 Session 66 (sofia/sipinterface_1/peter_gibbons at bluebox) Locked, Waiting on external entities 2012-03-09 08:05:13.586710 [NOTICE] switch_core_session.c:1242 Session 66 (sofia/sipinterface_1/peter_gibbons at bluebox) Ended 2012-03-09 08:05:13.586710 [NOTICE] switch_core_session.c:1244 Close Channel sofia/sipinterface_1/peter_gibbons at bluebox [CS_DESTROY] 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:431 (sofia/sipinterface_1/peter_gibbons at bluebox) Callstate Change HANGUP -> DOWN 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:434 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_DESTROY 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:444 (sofia/sipinterface_1/peter_gibbons at bluebox) State DESTROY 2012-03-09 08:05:13.588724 [DEBUG] mod_sofia.c:361 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA DESTROY 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:60 sofia/sipinterface_1/peter_gibbons at bluebox Standard DESTROY 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:444 (sofia/sipinterface_1/peter_gibbons at bluebox) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120309/3ee57e48/attachment-0001.html From grsingh750 at gmail.com Sat Mar 10 13:38:23 2012 From: grsingh750 at gmail.com (guru singh) Date: Sat, 10 Mar 2012 16:08:23 +0530 Subject: [Freeswitch-users] Event Socket Library on different client machine Message-ID: Hi, How can I move the ESL lib to different client machines (where FS is not installed) ? I ran 'make pymod' in libs/esl. On this machine I can go into python prompt and 'import ESL' without a problem I then moved ESL.py and ESL.pyc to the virtualenv of my python project on a different machine (venv/python2.7/site-packages) On this box, import ESL fails. The ESL wiki page says that it has no dependencies on freeswitch and can be moved around. What am I doing wrong? Thanks guru From development.milos at gmail.com Sat Mar 10 12:01:47 2012 From: development.milos at gmail.com (Milos Jovanovic) Date: Sat, 10 Mar 2012 10:01:47 +0100 Subject: [Freeswitch-users] MOD_XML_CDR is sending invalid XML... Message-ID: Hello, I use freeswitch and php web interface for it. When I tried to set mod_xml_cdr to send the POST requests to the URL of my script I get the XML passed in "cdr" variable but the channel variables are not url encoded. Here is how the channel variable looks like: "56011588754" ;tag=as6b531a4d When I set double logging (http and disk), in the XML file stored on the hard drive the channel variables are properly encoded. I tried to update FreeSWITCH (make current) but still nothing. Any suggestions? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120310/2433e4fc/attachment.html From peter at onemetric.com Sat Mar 10 14:14:45 2012 From: peter at onemetric.com (Peter Blackford) Date: Sat, 10 Mar 2012 22:14:45 +1100 Subject: [Freeswitch-users] MOD_XML_CDR is sending invalid XML... In-Reply-To: References: Message-ID: How are you viewing the results? If you are looking at the xml through a web browser then the browser is most likely un-encoding it. I was playing with this the other day and had no issues. On 10 March 2012 20:01, Milos Jovanovic wrote: > Hello, > > I use freeswitch and php web interface for it. When I tried to set > mod_xml_cdr to send the POST requests to the URL of my script I get the XML > passed in "cdr" variable but the channel variables are not url encoded. > > Here is how the channel variable looks like: > "56011588754" >;tag=as6b531a4d > > When I set double logging (http and disk), in the XML file stored on the > hard drive the channel variables are properly encoded. > > I tried to update FreeSWITCH (make current) but still nothing. > > Any suggestions? Thanks in advance. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Peter Blackford* *Technical Director* *P: *02 4032 7940 *M:* 0412 699 176* * *E:* peter at onemetric.com.au *W:* www.onemetric.com.au -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120310/45e4327d/attachment.html From rgarrett at garrettnet.net Sat Mar 10 16:04:25 2012 From: rgarrett at garrettnet.net (Robbie A. Garrett) Date: Sat, 10 Mar 2012 13:04:25 +0000 Subject: [Freeswitch-users] Unable to call cisco from freeswitch/blubox In-Reply-To: <0dcd01ccfe3a$c5ff4160$51fdc420$@com> Message-ID: Thanks for the response Sean, My gateway was in the configuration but is under sip interface 3. I move it under sip interface 1 and though I no longer get the error in bold below.. I still can not make a phone ring on the call manager side. 2012-03-10 05:01:21.362045 [DEBUG] sofia.c:6288 IP 10.0.1.141 Rejected by acl "net_list_5". Falling back to Digest auth. 2012-03-10 05:01:21.366989 [DEBUG] sofia.c:6288 IP 10.0.1.141 Rejected by acl "net_list_5". Falling back to Digest auth. 2012-03-10 05:01:21.368779 [NOTICE] switch_channel.c:784 New Channel sofia/sipinterface_1/peter_gibbons at bluebox [af0c695f-5d4b-452a-8cc4-5c7ba8841593] 2012-03-10 05:01:21.368779 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_NEW 2012-03-10 05:01:21.368779 [DEBUG] switch_core_state_machine.c:324 (sofia/sipinterface_1/peter_gibbons at bluebox) State NEW 2012-03-10 05:01:21.376921 [DEBUG] sofia.c:4559 Channel sofia/sipinterface_1/peter_gibbons at bluebox entering state [received][100] 2012-03-10 05:01:21.376921 [DEBUG] sofia.c:4570 Remote SDP: v=0 o=iSoftPhone 1331384504 1 IN IP4 10.0.1.141 s=conversation c=IN IP4 10.0.1.141 t=0 0 m=audio 10500 RTP/AVP 0 8 3 97 98 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 G726-32/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 2012-03-10 05:01:21.376921 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[G7221:115:32000:20:48000] 2012-03-10 05:01:21.376921 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[G7221:107:16000:20:32000] 2012-03-10 05:01:21.376921 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[G722:9:8000:20:64000] 2012-03-10 05:01:21.376921 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[PCMU:0:8000:20:64000] 2012-03-10 05:01:21.376921 [DEBUG] sofia_glue.c:2721 Set Codec sofia/sipinterface_1/peter_gibbons at bluebox PCMU/8000 20 ms 160 samples 64000 bits 2012-03-10 05:01:21.376921 [DEBUG] sofia_glue.c:4457 Set 2833 dtmf send/recv payload to 101 2012-03-10 05:01:21.376921 [DEBUG] sofia.c:4732 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_NEW -> CS_INIT 2012-03-10 05:01:21.376921 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-10 05:01:21.378955 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_INIT 2012-03-10 05:01:21.378955 [DEBUG] switch_core_state_machine.c:342 (sofia/sipinterface_1/peter_gibbons at bluebox) State INIT 2012-03-10 05:01:21.378955 [DEBUG] mod_sofia.c:83 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA INIT 2012-03-10 05:01:21.378955 [DEBUG] mod_sofia.c:123 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_INIT -> CS_ROUTING 2012-03-10 05:01:21.378955 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-10 05:01:21.378955 [DEBUG] switch_core_state_machine.c:342 (sofia/sipinterface_1/peter_gibbons at bluebox) State INIT going to sleep 2012-03-10 05:01:21.378955 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_ROUTING 2012-03-10 05:01:21.380921 [DEBUG] switch_channel.c:1615 (sofia/sipinterface_1/peter_gibbons at bluebox) Callstate Change DOWN -> RINGING 2012-03-10 05:01:21.380921 [DEBUG] switch_core_state_machine.c:345 (sofia/sipinterface_1/peter_gibbons at bluebox) State ROUTING 2012-03-10 05:01:21.380921 [DEBUG] mod_sofia.c:146 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA ROUTING 2012-03-10 05:01:21.380921 [DEBUG] switch_core_state_machine.c:77 sofia/sipinterface_1/peter_gibbons at bluebox Standard ROUTING 2012-03-10 05:01:21.380921 [INFO] mod_dialplan_xml.c:331 Processing peter_gibbons ->4000 in context context_1 Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->dingaling_1_pattern_1] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [dingaling_1_pattern_1] destination_number(4000) =~ /^1{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->conditioning_callerid] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [conditioning_callerid] ${internal_caller_id_number}(2002) =~ /^.+$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_name=${internal_caller_id_name}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_number=${internal_caller_id_number}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->postroute_global] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Absolute Condition [postroute_global] Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->preanswer_gtalk] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [preanswer_gtalk] source(mod_sofia) =~ /^mod_dingaling$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_trunk_1_pattern_5] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [main_trunk_1_pattern_5] destination_number(4000) =~ /^(40[0-9]{2})$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(prepend=) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [main_trunk_1_pattern_5] ${outbound_caller_id_number}(5555552002) =~ /^.+$/ break=never Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action export(sip_cid_type=rpid) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [main_trunk_1_pattern_5] destination_number(4000) =~ /^(40[0-9]{2})$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(failure_causes=NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action bridge(sofia/gateway/trunk_1/${prepend}4000) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_6] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_6] destination_number(4000) =~ /^2006$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_5] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_5] destination_number(4000) =~ /^2005$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_4] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_4] destination_number(4000) =~ /^2004$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_3] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_3] destination_number(4000) =~ /^2003$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_2] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_2] destination_number(4000) =~ /^2002$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_1] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_1] destination_number(4000) =~ /^2001$/ break=on-false 2012-03-10 05:01:21.380921 [DEBUG] switch_core_state_machine.c:119 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_ROUTING -> CS_EXECUTE 2012-03-10 05:01:21.380921 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-10 05:01:21.380921 [DEBUG] switch_core_state_machine.c:345 (sofia/sipinterface_1/peter_gibbons at bluebox) State ROUTING going to sleep 2012-03-10 05:01:21.380921 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_EXECUTE 2012-03-10 05:01:21.380921 [DEBUG] switch_core_state_machine.c:352 (sofia/sipinterface_1/peter_gibbons at bluebox) State EXECUTE 2012-03-10 05:01:21.380921 [DEBUG] mod_sofia.c:239 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA EXECUTE 2012-03-10 05:01:21.380921 [DEBUG] switch_core_state_machine.c:157 sofia/sipinterface_1/peter_gibbons at bluebox Standard EXECUTE EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_name=Peter Gibbons) 2012-03-10 05:01:21.380921 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_name]=[Peter Gibbons] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_number=2002) 2012-03-10 05:01:21.382939 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_number]=[2002] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox hash(insert/10.0.1.3-spymap/peter_gibbons/af0c695f-5d4b-452a-8cc4-5c7ba8841593) EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox hash(insert/10.0.1.3-last_dial/peter_gibbons/4000) EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox hash(insert/10.0.1.3-last_dial/global/af0c695f-5d4b-452a-8cc4-5c7ba8841593) EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(RFC2822_DATE=Sat, 10 Mar 2012 05:01:21 -0800) 2012-03-10 05:01:21.382939 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [RFC2822_DATE]=[Sat, 10 Mar 2012 05:01:21 -0800] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(prepend=) 2012-03-10 05:01:21.382939 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [prepend]=[UNDEF] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_name=Peter Gibbons) 2012-03-10 05:01:21.382939 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_name]=[Peter Gibbons] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_number=5555552002) 2012-03-10 05:01:21.382939 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_number]=[5555552002] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox export(sip_cid_type=rpid) 2012-03-10 05:01:21.382939 [DEBUG] switch_channel.c:933 EXPORT (export_vars) [sip_cid_type]=[rpid] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(failure_causes=NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH) 2012-03-10 05:01:21.382939 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [failure_causes]=[NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox bridge(sofia/gateway/trunk_1/4000) 2012-03-10 05:01:21.384935 [DEBUG] switch_channel.c:890 sofia/sipinterface_1/peter_gibbons at bluebox EXPORTING[export_vars] [sip_cid_type]=[rpid] to event 2012-03-10 05:01:21.384935 [NOTICE] switch_channel.c:784 New Channel sofia/sipinterface_1/4000 [149cbebe-befd-4a6a-9fe6-3597c88ade68] 2012-03-10 05:01:21.386950 [DEBUG] mod_sofia.c:3968 (sofia/sipinterface_1/4000) State Change CS_NEW -> CS_INIT 2012-03-10 05:01:21.386950 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/4000 [BREAK] 2012-03-10 05:01:21.386950 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/4000) Running State Change CS_INIT 2012-03-10 05:01:21.388953 [DEBUG] switch_core_state_machine.c:342 (sofia/sipinterface_1/4000) State INIT 2012-03-10 05:01:21.388953 [DEBUG] mod_sofia.c:83 sofia/sipinterface_1/4000 SOFIA INIT 2012-03-10 05:01:21.388953 [DEBUG] mod_sofia.c:123 (sofia/sipinterface_1/4000) State Change CS_INIT -> CS_ROUTING 2012-03-10 05:01:21.388953 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/4000 [BREAK] 2012-03-10 05:01:21.388953 [DEBUG] sofia.c:4559 Channel sofia/sipinterface_1/4000 entering state [calling][0] 2012-03-10 05:01:21.388953 [DEBUG] switch_core_state_machine.c:342 (sofia/sipinterface_1/4000) State INIT going to sleep 2012-03-10 05:01:21.388953 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/4000) Running State Change CS_ROUTING 2012-03-10 05:01:21.388953 [DEBUG] switch_channel.c:1615 (sofia/sipinterface_1/4000) Callstate Change DOWN -> RINGING 2012-03-10 05:01:21.388953 [DEBUG] switch_core_state_machine.c:345 (sofia/sipinterface_1/4000) State ROUTING 2012-03-10 05:01:21.388953 [DEBUG] mod_sofia.c:146 sofia/sipinterface_1/4000 SOFIA ROUTING 2012-03-10 05:01:21.388953 [DEBUG] switch_ivr_originate.c:66 (sofia/sipinterface_1/4000) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-03-10 05:01:21.388953 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/4000 [BREAK] 2012-03-10 05:01:21.388953 [DEBUG] switch_core_state_machine.c:345 (sofia/sipinterface_1/4000) State ROUTING going to sleep 2012-03-10 05:01:21.388953 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/4000) Running State Change CS_CONSUME_MEDIA 2012-03-10 05:01:21.388953 [DEBUG] switch_core_state_machine.c:364 (sofia/sipinterface_1/4000) State CONSUME_MEDIA 2012-03-10 05:01:21.388953 [DEBUG] switch_core_state_machine.c:364 (sofia/sipinterface_1/4000) State CONSUME_MEDIA going to sleep 2012-03-10 05:01:21.390900 [DEBUG] sofia.c:4559 Channel sofia/sipinterface_1/4000 entering state [terminated][503] 2012-03-10 05:01:21.390900 [DEBUG] switch_channel.c:2457 (sofia/sipinterface_1/4000) Callstate Change RINGING -> HANGUP 2012-03-10 05:01:21.390900 [NOTICE] sofia.c:5190 Hangup sofia/sipinterface_1/4000 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2012-03-10 05:01:21.390900 [DEBUG] switch_channel.c:2473 Send signal sofia/sipinterface_1/4000 [KILL] 2012-03-10 05:01:21.390900 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/4000 [BREAK] 2012-03-10 05:01:21.390900 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/4000) Running State Change CS_HANGUP 2012-03-10 05:01:21.390900 [DEBUG] switch_core_state_machine.c:539 (sofia/sipinterface_1/4000) State HANGUP 2012-03-10 05:01:21.390900 [DEBUG] mod_sofia.c:450 sofia/sipinterface_1/4000 Overriding SIP cause 503 with 503 from the other leg 2012-03-10 05:01:21.390900 [DEBUG] mod_sofia.c:456 Channel sofia/sipinterface_1/4000 hanging up, cause: NORMAL_TEMPORARY_FAILURE 2012-03-10 05:01:21.392800 [DEBUG] switch_ivr_originate.c:3413 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] 2012-03-10 05:01:21.392800 [INFO] mod_dptools.c:2579 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2012-03-10 05:01:21.392800 [DEBUG] mod_dptools.c:2610 Failure causes [NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH]: Cause: NORMAL_TEMPORARY_FAILURE 2012-03-10 05:01:21.392800 [NOTICE] switch_core_state_machine.c:189 sofia/sipinterface_1/peter_gibbons at bluebox has executed the last dialplan instruction, hanging up. 2012-03-10 05:01:21.392800 [DEBUG] switch_channel.c:2457 (sofia/sipinterface_1/peter_gibbons at bluebox) Callstate Change RINGING -> HANGUP 2012-03-10 05:01:21.392800 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/sipinterface_1/peter_gibbons at bluebox [CS_EXECUTE] [NORMAL_CLEARING] 2012-03-10 05:01:21.392800 [DEBUG] switch_channel.c:2473 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [KILL] 2012-03-10 05:01:21.392800 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-10 05:01:21.392800 [DEBUG] switch_core_state_machine.c:352 (sofia/sipinterface_1/peter_gibbons at bluebox) State EXECUTE going to sleep 2012-03-10 05:01:21.392800 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_HANGUP 2012-03-10 05:01:21.392800 [DEBUG] switch_core_state_machine.c:539 (sofia/sipinterface_1/peter_gibbons at bluebox) State HANGUP 2012-03-10 05:01:21.392800 [DEBUG] mod_sofia.c:450 sofia/sipinterface_1/peter_gibbons at bluebox Overriding SIP cause 480 with 503 from the other leg 2012-03-10 05:01:21.392800 [DEBUG] mod_sofia.c:456 Channel sofia/sipinterface_1/peter_gibbons at bluebox hanging up, cause: NORMAL_CLEARING 2012-03-10 05:01:21.394771 [DEBUG] switch_core_state_machine.c:46 sofia/sipinterface_1/4000 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2012-03-10 05:01:21.394771 [DEBUG] switch_core_state_machine.c:539 (sofia/sipinterface_1/4000) State HANGUP going to sleep 2012-03-10 05:01:21.394771 [DEBUG] switch_core_state_machine.c:337 (sofia/sipinterface_1/4000) State Change CS_HANGUP -> CS_REPORTING 2012-03-10 05:01:21.394771 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/4000 [BREAK] 2012-03-10 05:01:21.394771 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/4000) Running State Change CS_REPORTING 2012-03-10 05:01:21.394771 [DEBUG] switch_core_state_machine.c:599 (sofia/sipinterface_1/4000) State REPORTING 2012-03-10 05:01:21.394771 [DEBUG] switch_core_state_machine.c:53 sofia/sipinterface_1/4000 Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2012-03-10 05:01:21.394771 [DEBUG] switch_core_state_machine.c:599 (sofia/sipinterface_1/4000) State REPORTING going to sleep 2012-03-10 05:01:21.394771 [DEBUG] switch_core_state_machine.c:331 (sofia/sipinterface_1/4000) State Change CS_REPORTING -> CS_DESTROY 2012-03-10 05:01:21.394771 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/4000 [BREAK] 2012-03-10 05:01:21.394771 [DEBUG] switch_core_session.c:1224 Session 180 (sofia/sipinterface_1/4000) Locked, Waiting on external entities 2012-03-10 05:01:21.394771 [NOTICE] switch_core_session.c:1242 Session 180 (sofia/sipinterface_1/4000) Ended 2012-03-10 05:01:21.394771 [NOTICE] switch_core_session.c:1244 Close Channel sofia/sipinterface_1/4000 [CS_DESTROY] 2012-03-10 05:01:21.394771 [DEBUG] switch_core_state_machine.c:431 (sofia/sipinterface_1/4000) Callstate Change HANGUP -> DOWN 2012-03-10 05:01:21.394771 [DEBUG] switch_core_state_machine.c:434 (sofia/sipinterface_1/4000) Running State Change CS_DESTROY 2012-03-10 05:01:21.394771 [DEBUG] switch_core_state_machine.c:444 (sofia/sipinterface_1/4000) State DESTROY 2012-03-10 05:01:21.394771 [DEBUG] mod_sofia.c:361 sofia/sipinterface_1/4000 SOFIA DESTROY 2012-03-10 05:01:21.394771 [DEBUG] switch_core_state_machine.c:60 sofia/sipinterface_1/4000 Standard DESTROY 2012-03-10 05:01:21.394771 [DEBUG] switch_core_state_machine.c:444 (sofia/sipinterface_1/4000) State DESTROY going to sleep 2012-03-10 05:01:21.398824 [DEBUG] mod_sofia.c:518 Responding to INVITE with: 503 2012-03-10 05:01:21.398824 [DEBUG] switch_core_state_machine.c:46 sofia/sipinterface_1/peter_gibbons at bluebox Standard HANGUP, cause: NORMAL_CLEARING 2012-03-10 05:01:21.398824 [DEBUG] switch_core_state_machine.c:539 (sofia/sipinterface_1/peter_gibbons at bluebox) State HANGUP going to sleep 2012-03-10 05:01:21.398824 [DEBUG] switch_core_state_machine.c:337 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_HANGUP -> CS_REPORTING 2012-03-10 05:01:21.398824 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-10 05:01:21.398824 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_REPORTING 2012-03-10 05:01:21.398824 [DEBUG] switch_core_state_machine.c:599 (sofia/sipinterface_1/peter_gibbons at bluebox) State REPORTING 2012-03-10 05:01:21.555960 [DEBUG] switch_core_state_machine.c:53 sofia/sipinterface_1/peter_gibbons at bluebox Standard REPORTING, cause: NORMAL_CLEARING 2012-03-10 05:01:21.555960 [DEBUG] switch_core_state_machine.c:599 (sofia/sipinterface_1/peter_gibbons at bluebox) State REPORTING going to sleep 2012-03-10 05:01:21.556972 [DEBUG] switch_core_state_machine.c:331 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_REPORTING -> CS_DESTROY 2012-03-10 05:01:21.556972 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-10 05:01:21.556972 [DEBUG] switch_core_session.c:1224 Session 179 (sofia/sipinterface_1/peter_gibbons at bluebox) Locked, Waiting on external entities 2012-03-10 05:01:21.556972 [NOTICE] switch_core_session.c:1242 Session 179 (sofia/sipinterface_1/peter_gibbons at bluebox) Ended 2012-03-10 05:01:21.556972 [NOTICE] switch_core_session.c:1244 Close Channel sofia/sipinterface_1/peter_gibbons at bluebox [CS_DESTROY] 2012-03-10 05:01:21.556972 [DEBUG] switch_core_state_machine.c:431 (sofia/sipinterface_1/peter_gibbons at bluebox) Callstate Change HANGUP -> DOWN 2012-03-10 05:01:21.556972 [DEBUG] switch_core_state_machine.c:434 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_DESTROY 2012-03-10 05:01:21.556972 [DEBUG] switch_core_state_machine.c:444 (sofia/sipinterface_1/peter_gibbons at bluebox) State DESTROY 2012-03-10 05:01:21.556972 [DEBUG] mod_sofia.c:361 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA DESTROY 2012-03-10 05:01:21.556972 [DEBUG] switch_core_state_machine.c:60 sofia/sipinterface_1/peter_gibbons at bluebox Standard DESTROY 2012-03-10 05:01:21.556972 [DEBUG] switch_core_state_machine.c:444 (sofia/sipinterface_1/peter_gibbons at bluebox) State DESTROY going to sleep From: Sean Devoy > Reply-To: FreeSWITCH Users Help > Date: Fri, 9 Mar 2012 16:22:46 -0500 To: 'FreeSWITCH Users Help' > Subject: Re: [Freeswitch-users] Unable to call cisco from freeswitch/blubox Hi Robbie, I am no expert, but I can help move this along some. The first error is ?2012-03-09 08:05:13.421692 [ERR] mod_sofia.c:3738 Invalid Gateway? which is from the dial plan statement ?bridge(sofia/gateway/trunk_1/${prepend}4000)? FS doesn?t seem to know what ?trunk_1? is. I had this and resolved it on my FS (although it was to a VOIP provider gateway). In your SIP configuration (where you define things like port 5060) you must have a section that must contain a entity. Working out the params and variables for your gateway I cannot help as I don?t know call manager at all. I had not defined my gateway WITHIN a sip interface, so FS didn?t really know how to get to it. If you have it defined, you should see it in ./fs_cli with a ?sofia status? command. Hope that helps some. Sean From: Robbie A. Garrett [mailto:rgarrett at garrettnet.net] Sent: Friday, March 09, 2012 11:09 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Unable to call cisco from freeswitch/blubox Hi All, My home use lab uses cisco call manager 8.0. My old setup was cisco call manager to asterisk and then to google voice. Since asterisk and google voice are now broken; I'm looking to move in the direction of free switch. Currently My cisco call manager can call any extension on free switch and leave voicemail's. When I add google talk to frees witch I'm able to call from my cisco VoIP phone, through free switch, and to google. My issue is that free switch is unable to call my cisco call manager. Below is an output of the log of a call from extension 2002 to extension 4000. From what I can tell is that free-switch is not even trying to talk to 10.0.1.99 (my call manger). It's failing before it gets that far. 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[G7221:115:32000:20:48000] 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[G7221:107:16000:20:32000] 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[G722:9:8000:20:64000] 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:0:64000]/[PCMU:0:8000:20:64000] 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:2721 Set Codec sofia/sipinterface_1/peter_gibbons at bluebox PCMU/8000 20 ms 160 samples 64000 bits 2012-03-09 08:05:13.418262 [DEBUG] sofia_glue.c:4457 Set 2833 dtmf send/recv payload to 101 2012-03-09 08:05:13.418262 [DEBUG] sofia.c:4732 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_NEW -> CS_INIT 2012-03-09 08:05:13.418262 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_INIT 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:342 (sofia/sipinterface_1/peter_gibbons at bluebox) State INIT 2012-03-09 08:05:13.419735 [DEBUG] mod_sofia.c:83 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA INIT 2012-03-09 08:05:13.419735 [DEBUG] mod_sofia.c:123 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_INIT -> CS_ROUTING 2012-03-09 08:05:13.419735 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:342 (sofia/sipinterface_1/peter_gibbons at bluebox) State INIT going to sleep 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_ROUTING 2012-03-09 08:05:13.419735 [DEBUG] switch_channel.c:1615 (sofia/sipinterface_1/peter_gibbons at bluebox) Callstate Change DOWN -> RINGING 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:345 (sofia/sipinterface_1/peter_gibbons at bluebox) State ROUTING 2012-03-09 08:05:13.419735 [DEBUG] mod_sofia.c:146 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA ROUTING 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:77 sofia/sipinterface_1/peter_gibbons at bluebox Standard ROUTING 2012-03-09 08:05:13.419735 [INFO] mod_dialplan_xml.c:331 Processing peter_gibbons ->4000 in context context_1 Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->conditioning_callerid] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [conditioning_callerid] ${internal_caller_id_number}(2002) =~ /^.+$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_name=${internal_caller_id_name}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_number=${internal_caller_id_number}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->postroute_global] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Absolute Condition [postroute_global] Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->preanswer_gtalk] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [preanswer_gtalk] source(mod_sofia) =~ /^mod_dingaling$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_trunk_1_pattern_5] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [main_trunk_1_pattern_5] destination_number(4000) =~ /^(40[0-9]{2})$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(prepend=) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [main_trunk_1_pattern_5] ${outbound_caller_id_number}(5555552002) =~ /^.+$/ break=never Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action export(sip_cid_type=rpid) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (PASS) [main_trunk_1_pattern_5] destination_number(4000) =~ /^(40[0-9]{2})$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action set(failure_causes=NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Action bridge(sofia/gateway/trunk_1/${prepend}4000) Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_6] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_6] destination_number(4000) =~ /^2006$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_5] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_5] destination_number(4000) =~ /^2005$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_4] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_4] destination_number(4000) =~ /^2004$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_3] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_3] destination_number(4000) =~ /^2003$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_2] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_2] destination_number(4000) =~ /^2002$/ break=on-false Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox parsing [context_1->main_number_1] continue=true Dialplan: sofia/sipinterface_1/peter_gibbons at bluebox Regex (FAIL) [main_number_1] destination_number(4000) =~ /^2001$/ break=on-false 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:119 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_ROUTING -> CS_EXECUTE 2012-03-09 08:05:13.419735 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:345 (sofia/sipinterface_1/peter_gibbons at bluebox) State ROUTING going to sleep 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_EXECUTE 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:352 (sofia/sipinterface_1/peter_gibbons at bluebox) State EXECUTE 2012-03-09 08:05:13.419735 [DEBUG] mod_sofia.c:239 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA EXECUTE 2012-03-09 08:05:13.419735 [DEBUG] switch_core_state_machine.c:157 sofia/sipinterface_1/peter_gibbons at bluebox Standard EXECUTE EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_name=Peter Gibbons) 2012-03-09 08:05:13.419735 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_name]=[Peter Gibbons] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_number=2002) 2012-03-09 08:05:13.419735 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_number]=[2002] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox hash(insert/10.0.1.3-spymap/peter_gibbons/bc1115e3-b592-454c-8813-a6a7fb2754d4) EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox hash(insert/10.0.1.3-last_dial/peter_gibbons/4000) EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox hash(insert/10.0.1.3-last_dial/global/bc1115e3-b592-454c-8813-a6a7fb2754d4) EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(RFC2822_DATE=Fri, 09 Mar 2012 08:05:13 -0800) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [RFC2822_DATE]=[Fri, 09 Mar 2012 08:05:13 -0800] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(prepend=) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [prepend]=[UNDEF] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_name=Peter Gibbons) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_name]=[Peter Gibbons] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(effective_caller_id_number=5555552002) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [effective_caller_id_number]=[5555552002] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox export(sip_cid_type=rpid) 2012-03-09 08:05:13.421692 [DEBUG] switch_channel.c:933 EXPORT (export_vars) [sip_cid_type]=[rpid] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox set(failure_causes=NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH) 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:1028 sofia/sipinterface_1/peter_gibbons at bluebox SET [failure_causes]=[NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH] EXECUTE sofia/sipinterface_1/peter_gibbons at bluebox bridge(sofia/gateway/trunk_1/4000) 2012-03-09 08:05:13.421692 [DEBUG] switch_channel.c:890 sofia/sipinterface_1/peter_gibbons at bluebox EXPORTING[export_vars] [sip_cid_type]=[rpid] to event 2012-03-09 08:05:13.421692 [ERR] mod_sofia.c:3738 Invalid Gateway 2012-03-09 08:05:13.421692 [NOTICE] mod_sofia.c:4060 Close Channel N/A [CS_NEW] 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:434 () Running State Change CS_DESTROY 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:444 (N/A) State DESTROY 2012-03-09 08:05:13.421692 [DEBUG] mod_sofia.c:361 N/A SOFIA DESTROY 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:444 (N/A) State DESTROY going to sleep 2012-03-09 08:05:13.421692 [ERR] switch_ivr_originate.c:2605 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2012-03-09 08:05:13.421692 [DEBUG] switch_ivr_originate.c:3413 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT] 2012-03-09 08:05:13.421692 [INFO] mod_dptools.c:2579 Originate Failed. Cause: INVALID_NUMBER_FORMAT 2012-03-09 08:05:13.421692 [DEBUG] mod_dptools.c:2610 Failure causes [NORMAL_CLEARING,ORIGINATOR_CANCEL,CRASH]: Cause: INVALID_NUMBER_FORMAT 2012-03-09 08:05:13.421692 [NOTICE] switch_core_state_machine.c:189 sofia/sipinterface_1/peter_gibbons at bluebox has executed the last dialplan instruction, hanging up. 2012-03-09 08:05:13.421692 [DEBUG] switch_channel.c:2457 (sofia/sipinterface_1/peter_gibbons at bluebox) Callstate Change RINGING -> HANGUP 2012-03-09 08:05:13.421692 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/sipinterface_1/peter_gibbons at bluebox [CS_EXECUTE] [NORMAL_CLEARING] 2012-03-09 08:05:13.421692 [DEBUG] switch_channel.c:2473 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [KILL] 2012-03-09 08:05:13.421692 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:352 (sofia/sipinterface_1/peter_gibbons at bluebox) State EXECUTE going to sleep 2012-03-09 08:05:13.421692 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_HANGUP 2012-03-09 08:05:13.423711 [DEBUG] switch_core_state_machine.c:539 (sofia/sipinterface_1/peter_gibbons at bluebox) State HANGUP 2012-03-09 08:05:13.423711 [DEBUG] mod_sofia.c:456 Channel sofia/sipinterface_1/peter_gibbons at bluebox hanging up, cause: NORMAL_CLEARING 2012-03-09 08:05:13.427318 [DEBUG] mod_sofia.c:518 Responding to INVITE with: 480 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:46 sofia/sipinterface_1/peter_gibbons at bluebox Standard HANGUP, cause: NORMAL_CLEARING 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:539 (sofia/sipinterface_1/peter_gibbons at bluebox) State HANGUP going to sleep 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:337 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_HANGUP -> CS_REPORTING 2012-03-09 08:05:13.427318 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_REPORTING 2012-03-09 08:05:13.427318 [DEBUG] switch_core_state_machine.c:599 (sofia/sipinterface_1/peter_gibbons at bluebox) State REPORTING 2012-03-09 08:05:13.586710 [DEBUG] switch_core_state_machine.c:53 sofia/sipinterface_1/peter_gibbons at bluebox Standard REPORTING, cause: NORMAL_CLEARING 2012-03-09 08:05:13.586710 [DEBUG] switch_core_state_machine.c:599 (sofia/sipinterface_1/peter_gibbons at bluebox) State REPORTING going to sleep 2012-03-09 08:05:13.586710 [DEBUG] switch_core_state_machine.c:331 (sofia/sipinterface_1/peter_gibbons at bluebox) State Change CS_REPORTING -> CS_DESTROY 2012-03-09 08:05:13.586710 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/peter_gibbons at bluebox [BREAK] 2012-03-09 08:05:13.586710 [DEBUG] switch_core_session.c:1224 Session 66 (sofia/sipinterface_1/peter_gibbons at bluebox) Locked, Waiting on external entities 2012-03-09 08:05:13.586710 [NOTICE] switch_core_session.c:1242 Session 66 (sofia/sipinterface_1/peter_gibbons at bluebox) Ended 2012-03-09 08:05:13.586710 [NOTICE] switch_core_session.c:1244 Close Channel sofia/sipinterface_1/peter_gibbons at bluebox [CS_DESTROY] 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:431 (sofia/sipinterface_1/peter_gibbons at bluebox) Callstate Change HANGUP -> DOWN 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:434 (sofia/sipinterface_1/peter_gibbons at bluebox) Running State Change CS_DESTROY 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:444 (sofia/sipinterface_1/peter_gibbons at bluebox) State DESTROY 2012-03-09 08:05:13.588724 [DEBUG] mod_sofia.c:361 sofia/sipinterface_1/peter_gibbons at bluebox SOFIA DESTROY 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:60 sofia/sipinterface_1/peter_gibbons at bluebox Standard DESTROY 2012-03-09 08:05:13.588724 [DEBUG] switch_core_state_machine.c:444 (sofia/sipinterface_1/peter_gibbons at bluebox) State DESTROY going to sleep _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120310/c6fd9177/attachment-0001.html From grsingh750 at gmail.com Sat Mar 10 17:56:54 2012 From: grsingh750 at gmail.com (guru singh) Date: Sat, 10 Mar 2012 20:26:54 +0530 Subject: [Freeswitch-users] Event Socket Library on different client machine In-Reply-To: References: Message-ID: Ok, So I hadn't moved the .so file. I did that and now i get the error 'Wrong ELF class: ELFCLASS64' . I think this is because I compiled on a 64bit system and moved it to a 32bit system. Is there a way I can chose target as 32bit while compiling the esl module on 64bit or do I have to install entire fs on 32bit box to compile ESL lib? thanks On Sat, Mar 10, 2012 at 4:08 PM, guru singh wrote: > Hi, > > How can I move the ESL lib to different client machines (where FS is > not installed) ? > I ran 'make pymod' in libs/esl. On this machine I can go into python > prompt and 'import ESL' without a problem > I then moved ESL.py and ESL.pyc to the virtualenv of my python project > on a different machine (venv/python2.7/site-packages) > On this box, import ESL fails. The ESL wiki page says that it has no > dependencies on freeswitch and can be moved around. > What am I doing wrong? > > Thanks > guru From peter.olsson at visionutveckling.se Sat Mar 10 18:07:24 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 10 Mar 2012 15:07:24 +0000 Subject: [Freeswitch-users] Event Socket Library on different client machine In-Reply-To: References: , Message-ID: <1FFF97C269757C458224B7C895F35F15050DAF@cantor.std.visionutv.se> I think this should help you: http://wiki.freeswitch.org/wiki/Installation_Guide#Compiling_32_bit_on_64_bit_Systems /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för guru singh [grsingh750 at gmail.com] Skickat: den 10 mars 2012 15:56 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Event Socket Library on different client machine Ok, So I hadn't moved the .so file. I did that and now i get the error 'Wrong ELF class: ELFCLASS64' . I think this is because I compiled on a 64bit system and moved it to a 32bit system. Is there a way I can chose target as 32bit while compiling the esl module on 64bit or do I have to install entire fs on 32bit box to compile ESL lib? thanks On Sat, Mar 10, 2012 at 4:08 PM, guru singh wrote: > Hi, > > How can I move the ESL lib to different client machines (where FS is > not installed) ? > I ran 'make pymod' in libs/esl. On this machine I can go into python > prompt and 'import ESL' without a problem > I then moved ESL.py and ESL.pyc to the virtualenv of my python project > on a different machine (venv/python2.7/site-packages) > On this box, import ESL fails. The ESL wiki page says that it has no > dependencies on freeswitch and can be moved around. > What am I doing wrong? > > Thanks > guru _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f5b6afa32761959820876! From grsingh750 at gmail.com Sat Mar 10 18:26:01 2012 From: grsingh750 at gmail.com (guru singh) Date: Sat, 10 Mar 2012 20:56:01 +0530 Subject: [Freeswitch-users] Event Socket Library on different client machine In-Reply-To: <1FFF97C269757C458224B7C895F35F15050DAF@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F15050DAF@cantor.std.visionutv.se> Message-ID: Hi Peter, I read that, but isn't that for compiling entire FS tree? Can I compile just the ESL lib as 32bit target? On Sat, Mar 10, 2012 at 8:37 PM, Peter Olsson wrote: > I think this should help you: > > http://wiki.freeswitch.org/wiki/Installation_Guide#Compiling_32_bit_on_64_bit_Systems > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för guru singh [grsingh750 at gmail.com] > Skickat: den 10 mars 2012 15:56 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Event Socket Library on different client ? machine > > Ok, > > So I hadn't moved the .so file. I did that and now i get the error > 'Wrong ELF class: ELFCLASS64' . I think this is because I compiled on > a 64bit system and moved it to a 32bit system. Is there a way I can > chose target as 32bit while compiling the esl module on 64bit or do I > have to install entire fs on 32bit box to compile ESL lib? > > thanks > > On Sat, Mar 10, 2012 at 4:08 PM, guru singh wrote: >> Hi, >> >> How can I move the ESL lib to different client machines (where FS is >> not installed) ? >> I ran 'make pymod' in libs/esl. On this machine I can go into python >> prompt and 'import ESL' without a problem >> I then moved ESL.py and ESL.pyc to the virtualenv of my python project >> on a different machine (venv/python2.7/site-packages) >> On this box, import ESL fails. The ESL wiki page says that it has no >> dependencies on freeswitch and can be moved around. >> What am I doing wrong? >> >> Thanks >> guru > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f5b6afa32761959820876! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From babak.freeswitch at gmail.com Sun Mar 11 00:59:08 2012 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Sun, 11 Mar 2012 01:29:08 +0330 Subject: [Freeswitch-users] reloading mod_java crashes freeswitch is it normal? Message-ID: Hi Recently I was trying to use mod_java and it is working fine, but whenever I issue reload mod_java command freeswitch crashes. Is it normal?. I'm using jre 1.7 u2. thanx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120311/f2ca0cd7/attachment.html From gabe at gundy.org Sun Mar 11 01:23:34 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 10 Mar 2012 15:23:34 -0700 Subject: [Freeswitch-users] blocking destination number In-Reply-To: <4F5A1591.8090703@softnet.si> References: <4F5A1591.8090703@softnet.si> Message-ID: On Fri, Mar 9, 2012 at 7:37 AM, Miha Zoubek wrote: > what is the bast way to block destination number for certain user. > Is it possible to do it in user/dir? Ultimately, it would be a dialplan configuration, but you could set the variable that you match in the directory. See the example of 'toll_allow' in the default FreeSWITCH configuration... This is set on a per-user basis in the DIRECTORY: But it's considered while evaluating the DIALPLAN: You could easily do something similar by having a list of numbers that they can't call listed in the directory (maybe something like 'restricted_numbers') and check to make sure the destination doesn't match it in the dialplan. Good luck! Gabe From gabe at gundy.org Sun Mar 11 01:33:12 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 10 Mar 2012 15:33:12 -0700 Subject: [Freeswitch-users] reloading mod_java crashes freeswitch is it normal? In-Reply-To: References: Message-ID: On Sat, Mar 10, 2012 at 2:59 PM, babak yakhchali wrote: > Recently I was trying to use mod_java and it is working fine, but whenever I > issue reload mod_java command freeswitch crashes. Is it normal?. I'm using > jre 1.7 u2. I don't use Java, but it's not normal for modules to crash FreeSWITCH when being loaded or unloaded. I would... 1) make you're running on the latest code. 2) make sure it's reproducible on a clean system and a clean build (CentOS would be best). 3) check to see if there is a bug already reported in Jira. If there is, add what helpful information you can. If there is not, file a bug and watch for it to be resolved there. Let us know once you have it resolved. Perhaps even a link to Jira. Good luck. Gabe From gabe at gundy.org Sun Mar 11 01:47:15 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 10 Mar 2012 15:47:15 -0700 Subject: [Freeswitch-users] Getting SIP verbose message In-Reply-To: References: Message-ID: On Fri, Mar 9, 2012 at 1:30 PM, Jimmy Godbout wrote: > Is it possible to get the phrase associated with a SIP code in a variable ? If I receive an error saying "insufficient funds", is it possible to retrieve it inside a variable in FS ? Most of the SIP related variables I've seen are to tweak the SIP messages. While I don't know for sure if there is a variable for your specific case, I can recommend that you do a uuid_dump and see what you have to work with. Good luck, Gabe From gabe at gundy.org Sun Mar 11 01:56:38 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 10 Mar 2012 15:56:38 -0700 Subject: [Freeswitch-users] SIP Invite Contact Header In-Reply-To: <49C5FCA19A8A114493EBAACA42FE5899104EB9A7@1AERDCEXCHMBX1.AER.AERCO.local> References: <49C5FCA19A8A114493EBAACA42FE5899104EB9A7@1AERDCEXCHMBX1.AER.AERCO.local> Message-ID: On Wed, Mar 7, 2012 at 8:58 AM, Rob Moore wrote: > I?ve tried setting sip_contact_user but that doesn?t seem to have any effect > on the contents of the contact header. Plus I would ?really prefer to set > this in the gateway profile so it only effects calls passing over a single > gateway. I see these additional variables: sip_contact_params sip_contact_user sip_contact_port sip_contact_uri sip_contact_host If you see this page on the wiki, it shows setting contact info in the GW: http://wiki.freeswitch.org/wiki/Clarification:gateways#conf.2Fsip_profiles.2Fexternal.2Fexample.xml Don't know if it works, but hope those two things get you going. Gabe From engelster at gmail.com Sat Mar 10 20:59:53 2012 From: engelster at gmail.com (Der Engel) Date: Sat, 10 Mar 2012 12:59:53 -0500 Subject: [Freeswitch-users] State of GUIs In-Reply-To: References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> <07CA705635574091A0842F676EADB31D@freeswitch.org> Message-ID: Why doesn't cudatel open sources their UI and receive testing, feedback and contribution from the community, for sure a lot a people are willing to contribute, they could do something like how redhat/fedora works, or switchbox/asterisknow. From bdfoster at endigotech.com Sun Mar 11 02:05:56 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 10 Mar 2012 18:05:56 -0500 Subject: [Freeswitch-users] State of GUIs In-Reply-To: References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> <07CA705635574091A0842F676EADB31D@freeswitch.org> Message-ID: That's part of how they make their money. They contribute a ton to this community already, I wouldn't ask them to open source their GUI. On Mar 10, 2012 5:57 PM, "Der Engel" wrote: > Why doesn't cudatel open sources their UI and receive testing, > feedback and contribution from the community, for sure a lot a people > are willing to contribute, they could do something like how > redhat/fedora works, or switchbox/asterisknow. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120310/1f8bc969/attachment.html From gabe at gundy.org Sun Mar 11 02:12:35 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 10 Mar 2012 16:12:35 -0700 Subject: [Freeswitch-users] execution ends after lcr command In-Reply-To: <20120302133153.142490@gmx.com> References: <20120302133153.142490@gmx.com> Message-ID: On Fri, Mar 2, 2012 at 6:31 AM, adam harris wrote: > The poision point where the lcr command is run in my application is not the last command, why does freeswitch think it has and hang up the call? Can we get a log from the console when this is happening? Maybe even a SIP trace if there is any SIP activity. Gabe From avlubimov at gmail.com Sun Mar 11 02:28:25 2012 From: avlubimov at gmail.com (Lubimov Alexey) Date: Sun, 11 Mar 2012 03:28:25 +0400 Subject: [Freeswitch-users] git freeswitch cysco 504g and sipnet.ru unable to call gateway Message-ID: <4F5BE399.3090704@gmail.com> any ideas, why freeswitch send bye after 501 Not Implemented from spa 504G? recv 491 bytes from udp/[192.168.26.20]:5061 at 23:20:09.663308: ------------------------------------------------------------------------ SUBSCRIBE sip:10 at sip.comp-house.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-77d49247 From: "avlubimov" ;tag=e45937ca160bbb95 To: "avlubimov" Call-ID: 3034efae-325ec501 at 192.168.26.20 CSeq: 49907 SUBSCRIBE Max-Forwards: 70 Contact: "avlubimov" Expires: 15 Event: line-seize User-Agent: Cisco/SPA504G-7.4.8a Call-Info: ;appearance-index=1 Content-Length: 0 ------------------------------------------------------------------------ send 733 bytes to udp/[192.168.26.20]:5061 at 23:20:09.665718: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-77d49247 From: "avlubimov" ;tag=e45937ca160bbb95 To: "avlubimov" ;tag=tguJ7zPUrCJ5 Call-ID: 3034efae-325ec501 at 192.168.26.20 CSeq: 49907 SUBSCRIBE Contact: Expires: 15 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=15 Content-Length: 0 ------------------------------------------------------------------------ send 850 bytes to udp/[192.168.26.20]:5061 at 23:20:09.666191: ------------------------------------------------------------------------ NOTIFY sip:10 at 192.168.26.20:5061 SIP/2.0 Via: SIP/2.0/UDP 85.30.248.51;rport;branch=z9hG4bK9eB52ag7rNKtg Max-Forwards: 70 From: "avlubimov" ;tag=6KmvQg3e5gvgS To: "avlubimov" ;tag=e45937ca160bbb95 Call-ID: 3034efae-325ec501 at 192.168.26.20 CSeq: 25374740 NOTIFY Contact: Expires: 15 Call-Info: ;appearance-index=1 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: line-seize Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=15 Content-Length: 0 ------------------------------------------------------------------------ send 992 bytes to udp/[192.168.26.20]:5061 at 23:20:09.667243: ------------------------------------------------------------------------ NOTIFY sip:10 at 192.168.26.20:5061 SIP/2.0 Via: SIP/2.0/UDP 85.30.248.51;rport;branch=z9hG4bKar4X450apy9cc Route: ;transport=udp Max-Forwards: 70 From: "avlubimov" ;tag=eRwWYuXZc6fo To: "avlubimov" ;tag=908912105ff8d4f8 Call-ID: 8acdc9f0-ea487e98 at 192.168.26.20 CSeq: 187271947 NOTIFY Contact: Call-Info: ;appearance-index=1;appearance-state=seized Call-Info: ;appearance-index=*;appearance-state=idle User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: call-info Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3352 Content-Length: 0 ------------------------------------------------------------------------ recv 320 bytes from udp/[192.168.26.20]:5061 at 23:20:09.718698: ------------------------------------------------------------------------ SIP/2.0 200 OK To: "avlubimov" ;tag=e45937ca160bbb95 From: "avlubimov" ;tag=6KmvQg3e5gvgS Call-ID: 3034efae-325ec501 at 192.168.26.20 CSeq: 25374740 NOTIFY Via: SIP/2.0/UDP 85.30.248.51;branch=z9hG4bK9eB52ag7rNKtg Server: Cisco/SPA504G-7.4.8a Content-Length: 0 ------------------------------------------------------------------------ recv 320 bytes from udp/[192.168.26.20]:5061 at 23:20:09.737243: ------------------------------------------------------------------------ SIP/2.0 200 OK To: "avlubimov" ;tag=908912105ff8d4f8 From: "avlubimov" ;tag=eRwWYuXZc6fo Call-ID: 8acdc9f0-ea487e98 at 192.168.26.20 CSeq: 187271947 NOTIFY Via: SIP/2.0/UDP 85.30.248.51;branch=z9hG4bKar4X450apy9cc Server: Cisco/SPA504G-7.4.8a Content-Length: 0 ------------------------------------------------------------------------ recv 507 bytes from udp/[192.168.26.20]:5061 at 23:20:13.562633: ------------------------------------------------------------------------ SUBSCRIBE sip:10 at 85.30.248.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-3f015cc3 From: "avlubimov" ;tag=e45937ca160bbb95 To: "avlubimov" ;tag=tguJ7zPUrCJ5 Call-ID: 3034efae-325ec501 at 192.168.26.20 CSeq: 49908 SUBSCRIBE Max-Forwards: 70 Contact: "avlubimov" Expires: 0 Event: line-seize User-Agent: Cisco/SPA504G-7.4.8a Call-Info: ;appearance-index=1 Content-Length: 0 ------------------------------------------------------------------------ send 743 bytes to udp/[192.168.26.20]:5061 at 23:20:13.564567: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-3f015cc3 From: "avlubimov" ;tag=e45937ca160bbb95 To: "avlubimov" ;tag=tguJ7zPUrCJ5 Call-ID: 3034efae-325ec501 at 192.168.26.20 CSeq: 49908 SUBSCRIBE Contact: Expires: 0 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Length: 0 ------------------------------------------------------------------------ send 917 bytes to udp/[192.168.26.20]:5061 at 23:20:13.565467: ------------------------------------------------------------------------ NOTIFY sip:10 at 192.168.26.20:5061 SIP/2.0 Via: SIP/2.0/UDP 85.30.248.51;rport;branch=z9hG4bKB1Xp60HeK7ZZQ Route: ;transport=udp Max-Forwards: 70 From: "avlubimov" ;tag=eRwWYuXZc6fo To: "avlubimov" ;tag=908912105ff8d4f8 Call-ID: 8acdc9f0-ea487e98 at 192.168.26.20 CSeq: 187271948 NOTIFY Contact: Call-Info: ;appearance-index=*;appearance-state=idle User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: call-info Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3348 Content-Length: 0 ------------------------------------------------------------------------ recv 320 bytes from udp/[192.168.26.20]:5061 at 23:20:13.601524: ------------------------------------------------------------------------ SIP/2.0 200 OK To: "avlubimov" ;tag=908912105ff8d4f8 From: "avlubimov" ;tag=eRwWYuXZc6fo Call-ID: 8acdc9f0-ea487e98 at 192.168.26.20 CSeq: 187271948 NOTIFY Via: SIP/2.0/UDP 85.30.248.51;branch=z9hG4bKB1Xp60HeK7ZZQ Server: Cisco/SPA504G-7.4.8a Content-Length: 0 ------------------------------------------------------------------------ recv 490 bytes from udp/[192.168.26.20]:5061 at 23:20:15.178280: ------------------------------------------------------------------------ SUBSCRIBE sip:10 at sip.comp-house.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-6b39d1c From: "avlubimov" ;tag=6517ea9082c439af To: "avlubimov" Call-ID: 51e5d0e4-932d676b at 192.168.26.20 CSeq: 46361 SUBSCRIBE Max-Forwards: 70 Contact: "avlubimov" Expires: 15 Event: line-seize User-Agent: Cisco/SPA504G-7.4.8a Call-Info: ;appearance-index=1 Content-Length: 0 ------------------------------------------------------------------------ send 732 bytes to udp/[192.168.26.20]:5061 at 23:20:15.180173: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-6b39d1c From: "avlubimov" ;tag=6517ea9082c439af To: "avlubimov" ;tag=UsgRjAPCH9cF Call-ID: 51e5d0e4-932d676b at 192.168.26.20 CSeq: 46361 SUBSCRIBE Contact: Expires: 15 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=15 Content-Length: 0 ------------------------------------------------------------------------ send 850 bytes to udp/[192.168.26.20]:5061 at 23:20:15.180411: ------------------------------------------------------------------------ NOTIFY sip:10 at 192.168.26.20:5061 SIP/2.0 Via: SIP/2.0/UDP 85.30.248.51;rport;branch=z9hG4bKcaQF8U2HggpjK Max-Forwards: 70 From: "avlubimov" ;tag=7vDNSBmj2Sj3m To: "avlubimov" ;tag=6517ea9082c439af Call-ID: 51e5d0e4-932d676b at 192.168.26.20 CSeq: 25374743 NOTIFY Contact: Expires: 15 Call-Info: ;appearance-index=1 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: line-seize Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=15 Content-Length: 0 ------------------------------------------------------------------------ send 992 bytes to udp/[192.168.26.20]:5061 at 23:20:15.181545: ------------------------------------------------------------------------ NOTIFY sip:10 at 192.168.26.20:5061 SIP/2.0 Via: SIP/2.0/UDP 85.30.248.51;rport;branch=z9hG4bKDKg89pKNDSc5e Route: ;transport=udp Max-Forwards: 70 From: "avlubimov" ;tag=eRwWYuXZc6fo To: "avlubimov" ;tag=908912105ff8d4f8 Call-ID: 8acdc9f0-ea487e98 at 192.168.26.20 CSeq: 187271949 NOTIFY Contact: Call-Info: ;appearance-index=1;appearance-state=seized Call-Info: ;appearance-index=*;appearance-state=idle User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: call-info Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3346 Content-Length: 0 ------------------------------------------------------------------------ recv 320 bytes from udp/[192.168.26.20]:5061 at 23:20:15.237027: ------------------------------------------------------------------------ SIP/2.0 200 OK To: "avlubimov" ;tag=6517ea9082c439af From: "avlubimov" ;tag=7vDNSBmj2Sj3m Call-ID: 51e5d0e4-932d676b at 192.168.26.20 CSeq: 25374743 NOTIFY Via: SIP/2.0/UDP 85.30.248.51;branch=z9hG4bKcaQF8U2HggpjK Server: Cisco/SPA504G-7.4.8a Content-Length: 0 ------------------------------------------------------------------------ recv 320 bytes from udp/[192.168.26.20]:5061 at 23:20:15.251656: ------------------------------------------------------------------------ SIP/2.0 200 OK To: "avlubimov" ;tag=908912105ff8d4f8 From: "avlubimov" ;tag=eRwWYuXZc6fo Call-ID: 8acdc9f0-ea487e98 at 192.168.26.20 CSeq: 187271949 NOTIFY Via: SIP/2.0/UDP 85.30.248.51;branch=z9hG4bKDKg89pKNDSc5e Server: Cisco/SPA504G-7.4.8a Content-Length: 0 ------------------------------------------------------------------------ recv 855 bytes from udp/[192.168.26.20]:5061 at 23:20:17.554852: ------------------------------------------------------------------------ INVITE sip:89168030237 at sip.comp-house.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-207e455d From: "avlubimov" ;tag=5697f670d4f0a03co1 To: Call-ID: a7634980-1f9ee86c at 192.168.26.20 CSeq: 101 INVITE Max-Forwards: 70 Contact: "avlubimov" Expires: 240 User-Agent: Cisco/SPA504G-7.4.8a Call-Info: ;appearance-index=1 Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 29037 29037 IN IP4 192.168.26.20 s=- c=IN IP4 192.168.26.20 t=0 0 m=audio 16500 RTP/AVP 18 0 8 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------------------------------------------------------ send 320 bytes to udp/[192.168.26.20]:5061 at 23:20:17.555281: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-207e455d From: "avlubimov" ;tag=5697f670d4f0a03co1 To: Call-ID: a7634980-1f9ee86c at 192.168.26.20 CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ send 808 bytes to udp/[192.168.26.20]:5061 at 23:20:17.557294: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-207e455d From: "avlubimov" ;tag=5697f670d4f0a03co1 To: ;tag=856DU64NZ28Ng Call-ID: a7634980-1f9ee86c at 192.168.26.20 CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="comp-house.ru", nonce="991c9820-6b07-11e1-b573-bdc32e505683", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 471 bytes from udp/[192.168.26.20]:5061 at 23:20:17.583006: ------------------------------------------------------------------------ ACK sip:89168030237 at sip.comp-house.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-207e455d From: "avlubimov" ;tag=5697f670d4f0a03co1 To: ;tag=856DU64NZ28Ng Call-ID: a7634980-1f9ee86c at 192.168.26.20 CSeq: 101 ACK Max-Forwards: 70 Contact: "avlubimov" User-Agent: Cisco/SPA504G-7.4.8a Call-Info: ;appearance-index=1 Content-Length: 0 ------------------------------------------------------------------------ recv 1102 bytes from udp/[192.168.26.20]:5061 at 23:20:17.588436: ------------------------------------------------------------------------ INVITE sip:89168030237 at sip.comp-house.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-c30ceb25 From: "avlubimov" ;tag=5697f670d4f0a03co1 To: Call-ID: a7634980-1f9ee86c at 192.168.26.20 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="10",realm="comp-house.ru",nonce="991c9820-6b07-11e1-b573-bdc32e505683",uri="sip:89168030237 at sip.comp-house.ru",algorithm=MD5,response="1a4163060aa44faacdd2f674ffb7f13b",qop=auth,nc=00000001,cnonce="526a5637" Contact: "avlubimov" Expires: 240 User-Agent: Cisco/SPA504G-7.4.8a Call-Info: ;appearance-index=1 Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 29037 29037 IN IP4 192.168.26.20 s=- c=IN IP4 192.168.26.20 t=0 0 m=audio 16500 RTP/AVP 18 0 8 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------------------------------------------------------ send 320 bytes to udp/[192.168.26.20]:5061 at 23:20:17.588834: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-c30ceb25 From: "avlubimov" ;tag=5697f670d4f0a03co1 To: Call-ID: a7634980-1f9ee86c at 192.168.26.20 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ send 885 bytes to udp/[192.168.26.20]:5061 at 23:20:21.262101: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-c30ceb25 From: "avlubimov" ;tag=5697f670d4f0a03co1 To: ;tag=9e06v1NSvBZ8B Call-ID: a7634980-1f9ee86c at 192.168.26.20 CSeq: 102 INVITE Contact: Call-Info: ;appearance-index=1 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ send 1089 bytes to udp/[192.168.26.20]:5061 at 23:20:27.649934: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-c30ceb25 From: "avlubimov" ;tag=5697f670d4f0a03co1 To: ;tag=9e06v1NSvBZ8B Call-ID: a7634980-1f9ee86c at 192.168.26.20 CSeq: 102 INVITE Contact: Call-Info: ;appearance-index=1 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 171 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1610198872 1610198873 IN IP4 85.30.248.51 s=FreeSWITCH c=IN IP4 85.30.248.51 t=0 0 m=audio 0 RTP/AVP 96 a=rtpmap:96 G729/8000 a=fmtp:96 annexb=no ------------------------------------------------------------------------ recv 732 bytes from udp/[192.168.26.20]:5061 at 23:20:27.679656: ------------------------------------------------------------------------ ACK sip:89168030237 at 85.30.248.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-d832aefe From: "avlubimov" ;tag=5697f670d4f0a03co1 To: ;tag=9e06v1NSvBZ8B Call-ID: a7634980-1f9ee86c at 192.168.26.20 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="10",realm="comp-house.ru",nonce="991c9820-6b07-11e1-b573-bdc32e505683",uri="sip:89168030237 at sip.comp-house.ru",algorithm=MD5,response="1a4163060aa44faacdd2f674ffb7f13b",qop=auth,nc=00000001,cnonce="526a5637" Contact: "avlubimov" User-Agent: Cisco/SPA504G-7.4.8a Call-Info: ;appearance-index=1 Content-Length: 0 ------------------------------------------------------------------------ send 909 bytes to udp/[192.168.26.20]:5061 at 23:20:27.681881: ------------------------------------------------------------------------ UPDATE sip:10 at 192.168.26.20:5061 SIP/2.0 Via: SIP/2.0/UDP 85.30.248.51;rport;branch=z9hG4bKev90Bj4ra22Qa Max-Forwards: 70 From: ;tag=9e06v1NSvBZ8B To: "avlubimov" ;tag=5697f670d4f0a03co1 Call-ID: a7634980-1f9ee86c at 192.168.26.20 CSeq: 25374749 UPDATE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 171 P-Asserted-Identity: "Outbound Call" v=0 o=FreeSWITCH 1610198872 1610198873 IN IP4 85.30.248.51 s=FreeSWITCH c=IN IP4 85.30.248.51 t=0 0 m=audio 0 RTP/AVP 96 a=rtpmap:96 G729/8000 a=fmtp:96 annexb=no ------------------------------------------------------------------------ recv 506 bytes from udp/[192.168.26.20]:5061 at 23:20:27.686466: ------------------------------------------------------------------------ SUBSCRIBE sip:10 at 85.30.248.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-e8394fc From: "avlubimov" ;tag=6517ea9082c439af To: "avlubimov" ;tag=UsgRjAPCH9cF Call-ID: 51e5d0e4-932d676b at 192.168.26.20 CSeq: 46362 SUBSCRIBE Max-Forwards: 70 Contact: "avlubimov" Expires: 0 Event: line-seize User-Agent: Cisco/SPA504G-7.4.8a Call-Info: ;appearance-index=1 Content-Length: 0 ------------------------------------------------------------------------ send 742 bytes to udp/[192.168.26.20]:5061 at 23:20:27.688327: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.26.20:5061;branch=z9hG4bK-e8394fc From: "avlubimov" ;tag=6517ea9082c439af To: "avlubimov" ;tag=UsgRjAPCH9cF Call-ID: 51e5d0e4-932d676b at 192.168.26.20 CSeq: 46362 SUBSCRIBE Contact: Expires: 0 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Length: 0 ------------------------------------------------------------------------ send 917 bytes to udp/[192.168.26.20]:5061 at 23:20:27.689850: ------------------------------------------------------------------------ NOTIFY sip:10 at 192.168.26.20:5061 SIP/2.0 Via: SIP/2.0/UDP 85.30.248.51;rport;branch=z9hG4bKF52SDDNv7aSap Route: ;transport=udp Max-Forwards: 70 From: "avlubimov" ;tag=eRwWYuXZc6fo To: "avlubimov" ;tag=908912105ff8d4f8 Call-ID: 8acdc9f0-ea487e98 at 192.168.26.20 CSeq: 187271950 NOTIFY Contact: Call-Info: ;appearance-index=*;appearance-state=idle User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: call-info Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3334 Content-Length: 0 ------------------------------------------------------------------------ recv 332 bytes from udp/[192.168.26.20]:5061 at 23:20:27.714474: ------------------------------------------------------------------------ SIP/2.0 501 Not Implemented To: "avlubimov" ;tag=5697f670d4f0a03co1 From: ;tag=9e06v1NSvBZ8B Call-ID: a7634980-1f9ee86c at 192.168.26.20 CSeq: 25374749 UPDATE Via: SIP/2.0/UDP 85.30.248.51;branch=z9hG4bKev90Bj4ra22Qa Server: Cisco/SPA504G-7.4.8a Content-Length: 0 ------------------------------------------------------------------------ send 599 bytes to udp/[192.168.26.20]:5061 at 23:20:27.715049: ------------------------------------------------------------------------ BYE sip:10 at 192.168.26.20:5061 SIP/2.0 Via: SIP/2.0/UDP 85.30.248.51;rport;branch=z9hG4bKgevjF85Z4KFXH Max-Forwards: 70 From: ;tag=9e06v1NSvBZ8B To: "avlubimov" ;tag=5697f670d4f0a03co1 Call-ID: a7634980-1f9ee86c at 192.168.26.20 CSeq: 25374750 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 320 bytes from udp/[192.168.26.20]:5061 at 23:20:27.731308: ------------------------------------------------------------------------ SIP/2.0 200 OK To: "avlubimov" ;tag=908912105ff8d4f8 From: "avlubimov" ;tag=eRwWYuXZc6fo Call-ID: 8acdc9f0-ea487e98 at 192.168.26.20 CSeq: 187271950 NOTIFY Via: SIP/2.0/UDP 85.30.248.51;branch=z9hG4bKF52SDDNv7aSap Server: Cisco/SPA504G-7.4.8a Content-Length: 0 ------------------------------------------------------------------------ recv 332 bytes from udp/[192.168.26.20]:5061 at 23:20:27.736262: ------------------------------------------------------------------------ SIP/2.0 405 Method Not Allowed To: "avlubimov" ;tag=5697f670d4f0a03co1 From: ;tag=9e06v1NSvBZ8B Call-ID: a7634980-1f9ee86c at 192.168.26.20 CSeq: 25374750 BYE Via: SIP/2.0/UDP 85.30.248.51;branch=z9hG4bKgevjF85Z4KFXH Server: Cisco/SPA504G-7.4.8a Content-Length: 0 ------------------------------------------------------------------------ From tang.du at hotmail.com Sun Mar 11 07:22:40 2012 From: tang.du at hotmail.com (tangdu) Date: Sat, 10 Mar 2012 20:22:40 -0800 (PST) Subject: [Freeswitch-users] Skypopen problem on centos5.8 Message-ID: <1331439760882-7362245.post@n2.nabble.com> First, Giovanni, Thank You for the new skypopen installer? With mod skypopen wiki?I'm trying to install skypopen on my centOS server witout success. when run /usr/local/freeswitch/skypopen/skype-clients-startup-dir/start_skype_clients.sh? error message? ERROR: Module snd_pcm_oss does not exist in /proc/modules ERROR: Module snd_mixer_oss does not exist in /proc/modules ERROR: Module snd_seq_oss does not exist in /proc/modules mknod: `/dev/dsp': File exists insmod: error inserting '/usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko': -1 File exists how could i solved this problem? Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Skypopen-problem-on-centos5-8-tp7362245p7362245.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Sun Mar 11 11:01:12 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 11 Mar 2012 08:01:12 +0000 Subject: [Freeswitch-users] Event Socket Library on different client machine In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15050DAF@cantor.std.visionutv.se>, Message-ID: <1FFF97C269757C458224B7C895F35F15050F04@cantor.std.visionutv.se> I don't know really - I guess you could edit the Makefile manually, or just build the entire FS source tree, it will just take some time.. :) /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för guru singh [grsingh750 at gmail.com] Skickat: den 10 mars 2012 16:26 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Event Socket Library on different client machine Hi Peter, I read that, but isn't that for compiling entire FS tree? Can I compile just the ESL lib as 32bit target? On Sat, Mar 10, 2012 at 8:37 PM, Peter Olsson wrote: > I think this should help you: > > http://wiki.freeswitch.org/wiki/Installation_Guide#Compiling_32_bit_on_64_bit_Systems > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för guru singh [grsingh750 at gmail.com] > Skickat: den 10 mars 2012 15:56 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Event Socket Library on different client machine > > Ok, > > So I hadn't moved the .so file. I did that and now i get the error > 'Wrong ELF class: ELFCLASS64' . I think this is because I compiled on > a 64bit system and moved it to a 32bit system. Is there a way I can > chose target as 32bit while compiling the esl module on 64bit or do I > have to install entire fs on 32bit box to compile ESL lib? > > thanks > > On Sat, Mar 10, 2012 at 4:08 PM, guru singh wrote: >> Hi, >> >> How can I move the ESL lib to different client machines (where FS is >> not installed) ? >> I ran 'make pymod' in libs/esl. On this machine I can go into python >> prompt and 'import ESL' without a problem >> I then moved ESL.py and ESL.pyc to the virtualenv of my python project >> on a different machine (venv/python2.7/site-packages) >> On this box, import ESL fails. The ESL wiki page says that it has no >> dependencies on freeswitch and can be moved around. >> What am I doing wrong? >> >> Thanks >> guru > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f5b71b232766041427376! From all.eforums at gmail.com Sun Mar 11 12:03:02 2012 From: all.eforums at gmail.com (A E [Gmail]) Date: Sun, 11 Mar 2012 05:03:02 -0400 Subject: [Freeswitch-users] Skypopen problem on centos5.8 In-Reply-To: <1331439760882-7362245.post@n2.nabble.com> References: <1331439760882-7362245.post@n2.nabble.com> Message-ID: On Mar 10, 2012 11:23 PM, "tangdu" wrote: > First, Giovanni, Thank You for the new skypopen installer? > With mod skypopen wiki?I'm trying to install skypopen on my centOS server > witout success. > when run > > /usr/local/freeswitch/skypopen/skype-clients-startup-dir/start_skype_clients.sh? > error message? > ERROR: Module snd_pcm_oss does not exist in /proc/modules > ERROR: Module snd_mixer_oss does not exist in /proc/modules > ERROR: Module snd_seq_oss does not exist in /proc/modules > mknod: `/dev/dsp': File exists > insmod: error inserting > '/usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko': -1 > File exists > how could i solved this problem? > Thanks > > Hi, If you read this messages properly you'll realise that all's well with your startup. Because of the way the instructions are written it has already made you create the dsp device and load the skypopen module which is why it says that it's already created or already loaded. Everything else is not important. It only checks to see if any of those modules i.e. snd_pcm etc exist and it tries to delete/stop/unload them before starting. If you do a "ps -ef | grep skyp" you should see the Skype client running. Also confirm you have Xvfb running. HTH aeg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120311/b687e590/attachment.html From virbhati at gmail.com Sun Mar 11 13:22:10 2012 From: virbhati at gmail.com (virendra bhati) Date: Sun, 11 Mar 2012 15:52:10 +0530 Subject: [Freeswitch-users] How to get and store values from mysql database to Lua script Message-ID: Hi List, Below is my code by which I have connected Lua and mysql DB and displayed into FreeSwitch CLI. local dbh = freeswitch.Dbh("Billing","Billing","Tzru9unbcfuW6PV") if assert(dbh:connected()) == true then freeswitch.consoleLog("info", "connected"); assert(dbh:query("SELECT id, profile_name FROM sofia_conf", function(row) for key, val in pairs(row) do row[key] = val -- freeswitch.consoleLog("info", " row.id"); -- freeswitch.consoleLog("info", " profile_name"); freeswitch.consoleLog("info", "row[key]"); -- freeswitch.consoleLog("info", " profile_name"); end end)) -- stream:write(string.format("%5s : %s\n", row.id, row.profile_name)) -- end) else freeswitch.consoleLog("info", "not connected"); end How to get and store values from mysql database to Lua variables ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati at gmail.com Skype id:- virbhati2 Hyderabad(India) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120311/e48b233e/attachment.html From freeswitch-list at puzzled.xs4all.nl Sun Mar 11 15:14:58 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Sun, 11 Mar 2012 13:14:58 +0100 Subject: [Freeswitch-users] How to get and store values from mysql database to Lua script In-Reply-To: References: Message-ID: <4F5C9742.5050404@puzzled.xs4all.nl> On 11-03-12 11:22, virendra bhati wrote: > How to get and store values from mysql database to Lua variables ? http://www.lua.org/docs.html http://www.amazon.com/Programming-Second-Edition-Roberto-Ierusalimschy/dp/8590379825/ref=sr_1_1?ie=UTF8&qid=1331467993&sr=8-1 Regards, Patrick From potxoka at gmail.com Sun Mar 11 22:47:46 2012 From: potxoka at gmail.com (Anto) Date: Sun, 11 Mar 2012 20:47:46 +0100 Subject: [Freeswitch-users] Send PAI and RPID In-Reply-To: References: Message-ID: That I want to do, send the two, but do not know if can do in FreeSWITCH. Regards Anto 2012/3/9 Kristian Kielhofner : > Many commercial implementations (Cisco, for example) send both by > default. ?As long as you have the same data in both there's really no > reason not to send both. > > On Mon, Mar 5, 2012 at 6:09 PM, Anto wrote: >> Hi >> >> Thank you very much. So asked to send both, as some providers accept >> PAI and other RPID :-S. >> >> Best regards >> Anto >> >> 2012/3/5 Ken Rice : >>> You can send both... They arent really mutually exclusive... Just some >>> providers whine about it... Its really redundant... Also RPID was never a >>> RFC, it was a draft that has stuck around and refuses to die... PAI is the >>> only ratified RFC standard >>> >>> >>> >>> On 3/5/12 1:42 PM, "Michael Collins" wrote: >>> >>> FYI, >>> >>> A quick glance through the source suggests that PAI and RPID are mutually >>> exclusive - i.e. you can set one or the other but not both. I'll defer to >>> the experts on whether or not the SIP spec says you SHOULD or SHOULD NOT >>> have both headers in a single message. >>> >>> -MC >>> >>> On Thu, Mar 1, 2012 at 4:20 AM, Anto wrote: >>> >>> Hello >>> >>> To my FreeSWITCH servers, they come PAI and RPID headers, sent to the >>> carrier but only one (the one I have configured with >> name="caller-id-type" value="pid"/>). Is there any way to send both >>> headers?. Thanks >>> >>> Best regards >>> Anto >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> ________________________________ >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From paul at cupis.co.uk Mon Mar 12 00:39:30 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Sun, 11 Mar 2012 21:39:30 +0000 Subject: [Freeswitch-users] Send PAI and RPID In-Reply-To: References: Message-ID: <4F5D1B92.8000803@cupis.co.uk> On 11/03/12 19:47, Anto wrote: > That I want to do, send the two, but do not know if can do in FreeSWITCH. If you really need to do this, why not set FreeSWITCH to send PAID and then in your dialplan manually construct and add an RPID header to outbound INVITEs for the relevant gateways? From potxoka at gmail.com Mon Mar 12 00:42:50 2012 From: potxoka at gmail.com (Anto) Date: Sun, 11 Mar 2012 22:42:50 +0100 Subject: [Freeswitch-users] Codec negotiation with carriers In-Reply-To: References: Message-ID: Hi I still do not find the solution and not really understanding, because it works:-S regards anto 2012/3/7 Anto : > Hello > > Attached file, with the traces of the different tests (with different > configurations). > > http://pastebin.freeswitch.org/18599 > > I have read the url that you mentioned, the initial guide FreeSWITCH, > that of mod_sofia, applications, etc.. I believe that most of the wiki > (maybe when do not give the solution, read as much documentation is > worse idea :-S, lock me more). > > I made a configuration that works (I have not tested the audio), but > earlier (before I started "touch" the profiles) if I could talk to a > physical phone (several times). The problem is that I can not > understand why it works and sometimes not, and I would like to learn > :-). Not only do and forget, so I would like to learn and less > disturbing to the mail list and (maybe in the future) to help other > newbies like me :-). Thanks ! > > Best regards > Anto > > 2012/3/7 Michael Collins : >> You may want to read up on codec negotiation: >> http://wiki.freeswitch.org/wiki/Codec_negotiation >> >> There are different ways to handle codecs depending on your needs. I'd read >> that page first and then try out some of the suggestions. If you're still >> having trouble then I'd recommend getting SIP traces of the traffic and >> putting them on pastebin.freeswitch.org. The gang here is pretty good at >> looking over logs and helping with diagnosing problems. :) >> >> -MC >> >> On Tue, Mar 6, 2012 at 2:30 PM, Anto wrote: >>> >>> Hi >>> >>> I am following the steps in this direction >>> "http://wiki.freeswitch.org/wiki/SBC_Setup" and >>> "http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice", >>> I reread the whole entire wiki (or so I lack), but do not quite >>> assimilate or finding the right formula to operate the bridge :-S. >>> >>> I captured traffic with ngrep, I enabled sip-trace, console logconsole >>> 8, etc., but unless the transcoding error (only two of the hundreds of >>> combinations of settings that I have), I have not seen anything >>> "weird" :-S >>> >>> I have 3 suppliers, each with this codec: >>> >>> 1) ? ? ? ? ? 2) ? ? ? ? ? ? ?3) >>> G729 ? ? ? ?G729 ? ? ? ?G729 >>> G711u ? ? ?G711A ? ? ?G711A >>> G711A ? ? G711u ? ? ? G711u >>> ? ? ? ? ? ? ? ?G723 ? ? ? ? G723 >>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?G722 >>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?GSM >>> >>> I think I understand that when making an outside call, FreeSWITCH >>> follow these steps: >>> >>> USER -> ( ? Dialplan -> profile (internal) -> bridge (external) -> >>> profile (external) ? ) -> PROVIDER >>> >>> PROVIDER -> ( ? Dialplan -> profile (external) -> bridge (internal) -> >>> profile (internal) ?) -> USER >>> >>> right? >>> >>> Internal and external I set as follows (and not many changes have >>> done, and not remember it, because I've been testing days). If >>> outbound (outbound-codec-prefs) all codecs specified system does not >>> handle the call, I have to specify these by hand. If active >>> inbound-proxy-media, not the caller. Some of the time I worked, but >>> gave me an error that it can do transcoding G729 codec (I do >>> passthrough), but the proxy does not work half. >>> >>> If the outbound property (outbound-codec-prefs) all codecs specified >>> system does not handle the call, I have to specify these by hand. If >>> active inbound-proxy-media, not the caller. Some of the time I worked, >>> but gave me an error that it can do transcoding G729 codec (I want to >>> make passthrough), but the "proxy media" does not work. >>> >>> Basically, what I do is that local users can use all the codecs >>> allowed (iLBC, GSM, ...) and make an outside call, use the carrier >>> that will indicate the priority but the free codec. >>> >>> With this configuration works for me, but I would like to understand >>> why so if it works and otherwise no. Coming to understand how to >>> configure properly and so as not to disturb the mail list ;-). Thanks >>> ! >>> >>> Best regards >>> Anto >>> >>> vars.xml >>> >>> >> >>> data="global_codec_prefs=iLBC,G7221,speex,PCMU,PCMA,BV16,G726-32,GSM,G729,G723,AMR"/> >>> >> >>> data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,iLBC,G7221,speex,BV16,G726-32,GSM"/> >>> >>> internal.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> external.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> dialplan/outbound.xml >>> >>> >>> ? ? ? ? >>> ? ? ? ? ? ? ? ? >>> ? ? ? ? ? ? ? ? ?>> expression="^(\d+)$"> >>> ? ? ? ? ? ? ? ? ? ? ? ? >>> ? ? ? ? ? ? ? ? ? ? ? ? >>> ? ? ? ? ? ? ? ? ? ? ? ? >>> ? ? ? ? ? ? ? ? ? ? ? ?>> data="sofia/gateway/provider-2/$1"/> >>> ? ? ? ? ? ? ? ? ? >>> ? ? ? ? ? ? ? ? >>> ? ? ? ? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> From shane at longwhitecloud.com Sun Mar 11 11:12:52 2012 From: shane at longwhitecloud.com (Shane Harrison) Date: Sun, 11 Mar 2012 21:12:52 +1300 Subject: [Freeswitch-users] Transferring calls from analog extension Message-ID: Spent most of th day trying to get an attended transfer working on an analog extension. Ideally I would like a 2 -way call to be "transfered" when the user hangs up ie. incoming call answered by analog extension (a-leg?), hookflash, dial 3rd party (b-leg?), talk to them and hang up to bridge a-leg and b-leg. However I couldn't find any hints on how to do this. What I did find is information on using att_txfer application in the dialplan as setup in the standard features.xml. This all works OK, I push *4 on the analog extension, the a-leg gets MOH, analog extension gets the played dialtone but pushing keys on the analog extension to enter the new destination does nothing. It appears as if DTMF is ignored. Analog extension users FreeTDM. Has context as default in freetdm.conf. Any pointers appreciated. Below is the console debug. It starts after the call is setup from a SIP (308) extension to the analog extension freetdm/2:1. 2012-03-11 19:38:50.393576 [DEBUG] ftdm_io.c:3530 [s2c1][1:4] Queuing DTMF * (debug = 0) 2012-03-11 19:38:50.393576 [DEBUG] mod_freetdm.c:799 Queuing DTMF [*] in channel FreeTDM/2:1/ device 2:1 2012-03-11 19:38:50.393576 [DEBUG] switch_ivr_bridge.c:391 Send signal sofia/internal/308 at 192.168.7.49 [BREAK] 2012-03-11 19:38:51.073573 [DEBUG] ftdm_io.c:3530 [s2c1][1:4] Queuing DTMF 4 (debug = 0) 2012-03-11 19:38:51.073573 [DEBUG] mod_freetdm.c:799 Queuing DTMF [4] in channel FreeTDM/2:1/ device 2:1 2012-03-11 19:38:51.073573 [DEBUG] switch_ivr_async.c:3058 sofia/internal/ 308 at 192.168.7.49 Processing meta digit '4' [execute_extension::att_xfer XML features] 2012-03-11 19:38:51.073573 [DEBUG] switch_core_session.c:1014 Send signal FreeTDM/2:1/ [BREAK] 2012-03-11 19:38:51.073573 [DEBUG] switch_ivr_bridge.c:391 Send signal sofia/internal/308 at 192.168.7.49 [BREAK] 2012-03-11 19:38:51.093575 [DEBUG] switch_core_session.c:731 Send signal FreeTDM/2:1/ [BREAK] 2012-03-11 19:38:51.173589 [DEBUG] switch_core_session.c:1014 Send signal sofia/internal/308 at 192.168.7.49 [BREAK] 2012-03-11 19:38:51.193585 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/308 at 192.168.7.49 [BREAK] 2012-03-11 19:38:51.373573 [DEBUG] switch_ivr.c:591 sofia/internal/ 308 at 192.168.7.49 Command Execute playback(local_stream://moh) EXECUTE sofia/internal/308 at 192.168.7.49 playback(local_stream://moh) 2012-03-11 19:38:51.373573 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz 2012-03-11 19:38:51.373573 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16 at 8000hz 1 channels 20ms 2012-03-11 19:38:51.373573 [DEBUG] switch_ivr.c:591 FreeTDM/2:1/ Command Execute execute_extension(att_xfer XML features) EXECUTE FreeTDM/2:1/ execute_extension(att_xfer XML features) 2012-03-11 19:38:51.373573 [INFO] mod_dialplan_xml.c:485 Processing Shane PC <308>->att_xfer in context features Dialplan: FreeTDM/2:1/ parsing [features->dx] continue=false Dialplan: FreeTDM/2:1/ Regex (FAIL) [dx] destination_number(att_xfer) =~ /^dx$/ break=on-false Dialplan: FreeTDM/2:1/ parsing [features->att_xfer] continue=false Dialplan: FreeTDM/2:1/ Regex (PASS) [att_xfer] destination_number(att_xfer) =~ /^att_xfer$/ break=on-false Dialplan: FreeTDM/2:1/ Action read(3 4 'tone_stream://%(10000,0,350,440)' digits 30000 #) Dialplan: FreeTDM/2:1/ Action set(origination_cancel_key=#) Dialplan: FreeTDM/2:1/ Action att_xfer(user/${digits}@xxxx.xxx.nz) 2012-03-11 19:38:51.373573 [NOTICE] switch_core_session.c:2386 Execute read(3 4 'tone_stream://%(10000,0,350,440)' digits 30000 #) EXECUTE FreeTDM/2:1/ read(3 4 'tone_stream://%(10000,0,350,440)' digits 30000 #) 2012-03-11 19:38:51.373573 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16 at 8000hz 1 channels 20ms 2012-03-11 19:39:01.410613 [DEBUG] switch_ivr_play_say.c:1678 done playing file tone_stream://%(10000,0,350,440) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120311/d8cfc842/attachment.html From lazyvirus at gmx.com Sun Mar 11 19:01:28 2012 From: lazyvirus at gmx.com (Bzzz) Date: Sun, 11 Mar 2012 17:01:28 +0100 Subject: [Freeswitch-users] fs + fusionpbx registered but not Message-ID: <20120311170128.274057c3@anubis.defcon1> Hi list, I'm trying to use FS w/ fusionpbx: users & extensions created. My 2 softphones says they're successfully registered, fusionpbx says that too, but when I'm trying to make a call between them (whatever direction), FS log says: 2012-03-11 15:59:18.925845 [DEBUG] switch_event.c:1522 Parsing variable [sip_invite_domain]=[192.168.1.25] 2012-03-11 15:59:18.925845 [DEBUG] switch_event.c:1522 Parsing variable [presence_id]=[jy sur anubis at 192.168.1.25] 2012-03-11 15:59:18.925845 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2012-03-11 15:59:18.925845 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2012-03-11 15:59:18.925845 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2012-03-11 15:59:18.925845 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2012-03-11 15:59:18.925845 [INFO] mod_dptools.c:2922 Originate Failed. Cause: USER_NOT_REGISTERED Where am I wrong? JY -- The Supreme Court does it with all deliberate speed. From lazyvirus at gmx.com Sun Mar 11 19:11:11 2012 From: lazyvirus at gmx.com (Bzzz) Date: Sun, 11 Mar 2012 17:11:11 +0100 Subject: [Freeswitch-users] connexion attempts behing a firewall (wtf?) Message-ID: <20120311171111.1257067b@anubis.defcon1> Hi, I just setup an FS svr w/ fusionpbx (only ext.+users created), and I found THAT in the FS log: 2012-03-11 16:51:30.795812 [DEBUG] sofia.c:7567 IP 72.55.156.56 Rejected by acl "domains". Falling back to Digest auth. 2012-03-11 16:51:30.795812 [WARNING] sofia_reg.c:1422 SIP auth challenge (INVITE) on sofia profile 'internal' for [88775950945170 at 86.68.18.226] from ip 72.55.156.56 2012-03-11 16:51:31.115813 [DEBUG] sofia.c:7567 IP 72.55.156.56 Rejected by acl "domains". Falling back to Digest auth. 2012-03-11 16:51:31.115813 [WARNING] sofia_reg.c:1422 SIP auth challenge (INVITE) on sofia profile 'internal' for [011441212790583 at 86.68.18.226] from ip 72.55.156.56 2012-03-11 16:51:31.985813 [DEBUG] sofia.c:7567 IP 72.55.156.56 Rejected by acl "domains". Falling back to Digest auth. 2012-03-11 16:51:31.985813 [WARNING] sofia_reg.c:1422 SIP auth challenge (INVITE) on sofia profile 'internal' for [00441212790587 at 86.68.18.226] from ip 72.55.156.56 2012-03-11 16:51:33.015828 [DEBUG] sofia.c:7567 IP 72.55.156.56 Rejected by acl "domains". Falling back to Digest auth. 2012-03-11 16:51:33.015828 [WARNING] sofia_reg.c:1422 SIP auth challenge (INVITE) on sofia profile 'internal' for [000441212790581 at 86.68.18.226] from ip 72.55.156.56 How can this freak reach my svr as I'm in my LAN and my modem box integrates a firewall. Oook (but not): just checked my box and found that there are 4 uPNP new rules: 0 UDP 5060 192.168.1.25 5060 1 TCP 5060 192.168.1.25 5060 2 UDP 5080 192.168.1.25 5080 3 TCP 5080 192.168.1.25 5080 I understand FS is opening these ports to be reached by external subscribers, but where can I stop it to do so until I made my internal tests, change these ports and limited VoIP to TLS only? JY -- masturbation, n.: Coming unscrewed. From bdfoster at endigotech.com Mon Mar 12 02:21:01 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 11 Mar 2012 19:21:01 -0400 Subject: [Freeswitch-users] connexion attempts behing a firewall (wtf?) In-Reply-To: <20120311171111.1257067b@anubis.defcon1> References: <20120311171111.1257067b@anubis.defcon1> Message-ID: This happens all the time. You need fail2ban set up properly. There are articles on both the freeswitch and fusionpbx wikis. I run a public server and I get people doing scans all the time. Does it worry me? No. Of course not. That's what fail2ban and other measures are for. Don't need to get all tinfoil hat about this. On Mar 11, 2012 7:09 PM, "Bzzz" wrote: > Hi, > > I just setup an FS svr w/ fusionpbx (only ext.+users created), and I > found THAT in the FS log: > > 2012-03-11 16:51:30.795812 [DEBUG] sofia.c:7567 IP 72.55.156.56 Rejected > by acl "domains". Falling back to Digest auth. > 2012-03-11 16:51:30.795812 [WARNING] sofia_reg.c:1422 SIP auth challenge > (INVITE) on sofia profile 'internal' for [88775950945170 at 86.68.18.226] > from ip 72.55.156.56 > 2012-03-11 16:51:31.115813 [DEBUG] sofia.c:7567 IP 72.55.156.56 Rejected > by acl "domains". Falling back to Digest auth. > 2012-03-11 16:51:31.115813 [WARNING] sofia_reg.c:1422 SIP auth challenge > (INVITE) on sofia profile 'internal' for [011441212790583 at 86.68.18.226] > from ip 72.55.156.56 > 2012-03-11 16:51:31.985813 [DEBUG] sofia.c:7567 IP 72.55.156.56 Rejected > by acl "domains". Falling back to Digest auth. > 2012-03-11 16:51:31.985813 [WARNING] sofia_reg.c:1422 SIP auth challenge > (INVITE) on sofia profile 'internal' for [00441212790587 at 86.68.18.226] > from ip 72.55.156.56 > 2012-03-11 16:51:33.015828 [DEBUG] sofia.c:7567 IP 72.55.156.56 Rejected > by acl "domains". Falling back to Digest auth. > 2012-03-11 16:51:33.015828 [WARNING] sofia_reg.c:1422 SIP auth challenge > (INVITE) on sofia profile 'internal' for [000441212790581 at 86.68.18.226] > from ip 72.55.156.56 > > How can this freak reach my svr as I'm in my LAN and my modem box > integrates a firewall. > > Oook (but not): just checked my box and found that there are 4 > uPNP new rules: > 0 UDP 5060 192.168.1.25 5060 > 1 TCP 5060 192.168.1.25 5060 > 2 UDP 5080 192.168.1.25 5080 > 3 TCP 5080 192.168.1.25 5080 > > I understand FS is opening these ports to be reached by external > subscribers, but where can I stop it to do so until I made my > internal tests, change these ports and limited VoIP to TLS only? > > JY > -- > masturbation, n.: > Coming unscrewed. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120311/d7bd5759/attachment-0001.html From freeswitch-list at puzzled.xs4all.nl Mon Mar 12 03:25:35 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Mon, 12 Mar 2012 01:25:35 +0100 Subject: [Freeswitch-users] connexion attempts behing a firewall (wtf?) In-Reply-To: <20120311171111.1257067b@anubis.defcon1> References: <20120311171111.1257067b@anubis.defcon1> Message-ID: <4F5D427F.3000908@puzzled.xs4all.nl> On 11-03-12 17:11, Bzzz wrote: [snip] > I understand FS is opening these ports to be reached by external > subscribers, but where can I stop it to do so until I made my > internal tests, change these ports and limited VoIP to TLS only? http://wiki.freeswitch.org/wiki/Auto_NAT Regards, Patrick From lazyvirus at gmx.com Mon Mar 12 02:38:44 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 12 Mar 2012 00:38:44 +0100 Subject: [Freeswitch-users] connexion attempts behing a firewall (wtf?) In-Reply-To: References: <20120311171111.1257067b@anubis.defcon1> Message-ID: <20120312003844.25ce725e@anubis.defcon1> On Sun, 11 Mar 2012 19:21:01 -0400 Brian Foster wrote: > This happens all the time. You need fail2ban set up properly. There are > articles on both the freeswitch and fusionpbx wikis. It is, and tested as the wiki says:) but attacker isn't banned. This show me this is also linked to my previous post: softphones & fusionpbx say they're registered, but when I try to make a call between them FS throw me USER_NOT_REGISTERED errors, this is why the attacker's not jettisoned: FS don't seem to answer to a correct registration:( > I run a public server > and I get people doing scans all the time. Does it worry me? No. Of course > not. That's what fail2ban and other measures are for. Interesting, could you describe "other measures", PLS? -- Draft beer, not boys! From lazyvirus at gmx.com Mon Mar 12 03:36:42 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 12 Mar 2012 01:36:42 +0100 Subject: [Freeswitch-users] connexion attempts behing a firewall (wtf?) In-Reply-To: <4F5D427F.3000908@puzzled.xs4all.nl> References: <20120311171111.1257067b@anubis.defcon1> <4F5D427F.3000908@puzzled.xs4all.nl> Message-ID: <20120312013642.45140da0@anubis.defcon1> On Mon, 12 Mar 2012 01:25:35 +0100 Patrick Lists wrote: > > http://wiki.freeswitch.org/wiki/Auto_NAT Ah, ok, thanks Patrick (this will be very interesting in a near future:) -- Q: What is the difference between snow-men and snow-women? A: Snowballs! From packetandy at gmail.com Mon Mar 12 04:44:08 2012 From: packetandy at gmail.com (andy) Date: Sun, 11 Mar 2012 18:44:08 -0700 Subject: [Freeswitch-users] seg fault on latest git Message-ID: <4F5D54E8.4090808@gmail.com> Hi all, I just got segmentation faults with a build from git (yesterday). After launching FS with mod_xml_curl, everything seems normal. If I unload mod_xml_curl from fs_cli, the next call crashes the system with a seg fault. I was able to repeat this behavior several times in a row. cheers From lazyvirus at gmx.com Mon Mar 12 06:10:28 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 12 Mar 2012 04:10:28 +0100 Subject: [Freeswitch-users] bootstrap Message-ID: <20120312041028.5e70cb57@anubis.defcon1> Hi, is it necessary to re-bootstrap @ each update, or just once for the first compilation? -- BOFH excuse #404: Sysadmin accidentally destroyed pager with a large hammer. From singhai.piyush at gmail.com Mon Mar 12 10:40:42 2012 From: singhai.piyush at gmail.com (piyush singhai) Date: Mon, 12 Mar 2012 13:10:42 +0530 Subject: [Freeswitch-users] Check for freeswitch session exists Message-ID: Hello, I am using ESL lib for making connection with freeswitch from the Application. Some time we miss the hangup event in application and instead make another execute request on the same session, lets say for playback(). Then we receive CHANNEL_EXECUTE_COMPLETE for application 'park' instead of playback(). Is this an indication that the session has already ended (in this case due to hangup by the user)? Otherwise how can we check before executing any new function that the session is still alive at fresswitch through ESL? --Piyush -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120312/92725ae0/attachment.html From peter.olsson at visionutveckling.se Mon Mar 12 10:56:11 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 12 Mar 2012 07:56:11 +0000 Subject: [Freeswitch-users] seg fault on latest git Message-ID: <1FFF97C269757C458224B7C895F35F15051135@cantor.std.visionutv.se> Please report to Jira, and supply all needed information.Also make sure to search open issues first, so it doesn't exist already. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r andy Skickat: den 12 mars 2012 02:44 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] seg fault on latest git Hi all, I just got segmentation faults with a build from git (yesterday). After launching FS with mod_xml_curl, everything seems normal. If I unload mod_xml_curl from fs_cli, the next call crashes the system with a seg fault. I was able to repeat this behavior several times in a row. cheers _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f5da5de32761177412446! From peter.olsson at visionutveckling.se Mon Mar 12 10:59:20 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 12 Mar 2012 07:59:20 +0000 Subject: [Freeswitch-users] Check for freeswitch session exists Message-ID: <1FFF97C269757C458224B7C895F35F15051141@cantor.std.visionutv.se> Since ESL is event driven, there is no way to be 100% sure of this. However, the hangup event won't be missed, I guess it will just show up a little bit later (if you continue to read a few more events). So you will just have to handle the "errors" if you try to playback when the channel is hung up already, and then when you receive the hangup event, you know for sure that the call has ended. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r piyush singhai Skickat: den 12 mars 2012 08:41 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Check for freeswitch session exists Hello, I am using ESL lib for making connection with freeswitch from the Application. Some time we miss the hangup event in application and instead make another execute request on the same session, lets say for playback(). Then we receive CHANNEL_EXECUTE_COMPLETE for application 'park' instead of playback(). Is this an indication that the session has already ended (in this case due to hangup by the user)? Otherwise how can we check before executing any new function that the session is still alive at fresswitch through ESL? --Piyush !DSPAM:4f5da6e632765021882877! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120312/76fb6031/attachment.html From fdelawarde at wirelessmundi.com Mon Mar 12 12:17:43 2012 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Mon, 12 Mar 2012 10:17:43 +0100 Subject: [Freeswitch-users] removing caller_id_name In-Reply-To: <1330605100.12944.132.camel@luna.madrid.commsmundi.com> References: <1330527551.12944.47.camel@luna.madrid.commsmundi.com> <2AB1E3C03738764094BDD615FEE595AD037468A5@AMSPRD0402MB111.eurprd04.prod.outlook.com> <1330529520.12944.64.camel@luna.madrid.commsmundi.com> <1330531312.12944.65.camel@luna.madrid.commsmundi.com> <1330532922.12944.74.camel@luna.madrid.commsmundi.com> <1330605100.12944.132.camel@luna.madrid.commsmundi.com> Message-ID: <1331543863.11617.178.camel@luna.madrid.commsmundi.com> http://jira.freeswitch.org/browse/FS-3984 Fran?ois. On Thu, 2012-03-01 at 13:31 +0100, Fran?ois Delawarde wrote: > Thanks for the help, did you find anything? > > If there is no way, I'll try to find time to do some patch that would > take "_undef_" into account for PAI and RPID headers (right now it only > works for the From header). > > Fran?ois. > > On Wed, 2012-02-29 at 11:32 -0500, Brian Foster wrote: > > Hmm... let me do some research. > > > > > > -BDF > > > > On Wed, Feb 29, 2012 at 11:28 AM, Fran?ois Delawarde > > wrote: > > Can't do that, "caller_id_name" is read-only (part of caller > > profile). > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#caller_id_name > > > > Please keep firing ideas, I'm sure it must be possible > > somehow! > > > > Fran?ois. > > > > > > On Wed, 2012-02-29 at 11:10 -0500, Brian Foster wrote: > > > There's probably a couple of different ways to do it. It's > > whatever > > > works. I don't like setting things to an empty string just > > as a > > > personal preference (even though it accomplishes the same > > result). > > > > > > > > > > > > > > > > > > -BDF > > > > > > On Wed, Feb 29, 2012 at 11:01 AM, Fran?ois Delawarde > > > wrote: > > > Nice try but no, it would unset > > "effective_caller_id_name", > > > and not > > > "caller_id_name". > > > > > > I would need to be able to set it to an empty > > string, but for > > > FS, empty > > > string is like an "unset". > > > > > > Fran?ois. > > > > > > > > > On Wed, 2012-02-29 at 10:42 -0500, Brian Foster > > wrote: > > > > What about doing: > > > > > > > > > > data="effective_caller_id_name"/> > > > > > > > > ..? > > > > > > > > -BDF > > > > > > > > On Feb 29, 2012 10:32 AM, "Fran?ois Delawarde" > > > > wrote: > > > > Yeah, my analog phone has that function to > > be able > > > to add the > > > > last CID > > > > to its directory, but it uses the > > caller_id_name as > > > contact > > > > name and > > > > caller_id_number as contact number. > > > > > > > > Each time I want to add someone that > > called me to > > > the phone's > > > > directory, > > > > I have to delete the preset contact name > > and add > > > what I want. > > > > > > > > It's a pain for lazy people like me. > > > > > > > > Fran?ois. > > > > > > > > > > > > On Wed, 2012-02-29 at 07:23 -0800, Mitch > > Capper > > > wrote: > > > > > Any reason not to set it to the phone > > number? > > > > > > > > > > ~Mitch > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > Professional FreeSWITCH Consulting > > Services: > > > > > consulting at freeswitch.org > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > FreeSWITCH-powered IP PBX: The CudaTel > > > Communication Server > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > http://www.freeswitch.org > > > > > http://wiki.freeswitch.org > > > > > http://www.cluecon.com > > > > > > > > > > FreeSWITCH-users mailing list > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting > > Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > FreeSWITCH-powered IP PBX: The CudaTel > > Communication > > > Server > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > FreeSWITCH-powered IP PBX: The CudaTel > > Communication Server > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > > Server > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > -- > > > Brian D. Foster > > > Endigo Computer LLC > > > Email: bdfoster at endigotech.com > > > Phone: 317-429-1069 > > > Indianapolis, Indiana, USA > > > > > > This message contains confidential information and is > > intended for > > > those listed in the "To:", "CC:", and/or "BCC:" fields of > > the message > > > header.If you are not the intended recipient you are > > notified that > > > disclosing, copying, distributing or taking any action in > > reliance on > > > the contents of this information is strictly prohibited. > > E-mail > > > transmission cannot be guaranteed to be secure or error-free > > as > > > information could be intercepted, corrupted, lost, > > destroyed, arrive > > > late or incomplete, or contain viruses. The sender therefore > > does not > > > accept liability for any errors or omissions in the contents > > of this > > > message, which arise as a result of e-mail transmission. If > > > verification is required please request a hard-copy version. > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Brian D. Foster > > Endigo Computer LLC > > Email: bdfoster at endigotech.com > > Phone: 317-429-1069 > > Indianapolis, Indiana, USA > > > > This message contains confidential information and is intended for > > those listed in the "To:", "CC:", and/or "BCC:" fields of the message > > header.If you are not the intended recipient you are notified that > > disclosing, copying, distributing or taking any action in reliance on > > the contents of this information is strictly prohibited. E-mail > > transmission cannot be guaranteed to be secure or error-free as > > information could be intercepted, corrupted, lost, destroyed, arrive > > late or incomplete, or contain viruses. The sender therefore does not > > accept liability for any errors or omissions in the contents of this > > message, which arise as a result of e-mail transmission. If > > verification is required please request a hard-copy version. > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From miha at softnet.si Mon Mar 12 12:44:59 2012 From: miha at softnet.si (Miha Zoubek) Date: Mon, 12 Mar 2012 10:44:59 +0100 Subject: [Freeswitch-users] blocking destination number In-Reply-To: References: <4F5A1591.8090703@softnet.si> Message-ID: <4F5DC59B.10205@softnet.si> On 03/10/2012 11:23 PM, Gabriel Gunderson wrote: > On Fri, Mar 9, 2012 at 7:37 AM, Miha Zoubek wrote: >> what is the bast way to block destination number for certain user. >> Is it possible to do it in user/dir? > Ultimately, it would be a dialplan configuration, but you could set > the variable that you match in the directory. See the example of > 'toll_allow' in the default FreeSWITCH configuration... > > > This is set on a per-user basis in the DIRECTORY: > > > > > > But it's considered while evaluating the DIALPLAN: > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_id_name}"/> > data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> > > > > > > You could easily do something similar by having a list of numbers that > they can't call listed in the directory (maybe something like > 'restricted_numbers') and check to make sure the destination doesn't > match it in the dialplan. > > Good luck! > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Thank you! Regards, Miha From vfclists at gmail.com Mon Mar 12 12:47:41 2012 From: vfclists at gmail.com (Frank Church) Date: Mon, 12 Mar 2012 09:47:41 +0000 Subject: [Freeswitch-users] How to avoid changing of internal profile binding when VPN comes up? Message-ID: I am running Freeswitch on a VM which uses 10.0.2.x subnet for NAT, and bridges to the hosts subnet on 192.168.x.x network. When I start the server the internal and external profiles automatically bind to the 10.0.2.x subnet. I don't need to edit any of the IPs in vars.xml or internal.xml or external.xml. But when the VPN comes up Freeswitch automatically binds to the VPNs subnet. Are there some rules which determine what internal and external profiles bind themselves to by default? Is it the internet gateway with lowest metric that they are bound to? -- Frank Church ======================= http://devblog.brahmancreations.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120312/7b8fd78f/attachment.html From singhai.piyush at gmail.com Mon Mar 12 13:13:03 2012 From: singhai.piyush at gmail.com (piyush singhai) Date: Mon, 12 Mar 2012 15:43:03 +0530 Subject: [Freeswitch-users] how does eval command works Message-ID: I want to know the channel state corresponding to chan_uuid. I tried eval but it is not working can anybody tell me how it will work. I followed the wiki http://wiki.freeswitch.org/wiki/Mod_commands --Piyush -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120312/fc859d66/attachment.html From b2m at a-cti.com Mon Mar 12 14:12:15 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Mon, 12 Mar 2012 16:42:15 +0530 Subject: [Freeswitch-users] How to disable Re-invite? Message-ID: 2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:2925 Already using PCMU 2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:4988 Set 2833 dtmf send payload to 101 2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:3213 Audio params are unchanged for sofia/external/+919884730340. 2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:3223 sofia/external/+919884730340 Setting audio receive payload in *Re-INVITE to 0* * * Thanks, Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120312/f4350943/attachment.html From dgarcia at anew.com.ve Mon Mar 12 16:19:48 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Mon, 12 Mar 2012 08:49:48 -0430 Subject: [Freeswitch-users] State of GUIs In-Reply-To: References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net> <03f301ccfd03$d7777080$86665180$@launch3.net> <4F58B19E.7070101@puzzled.xs4all.nl> Message-ID: <4F5DF7F4.2030606@anew.com.ve> Hi Roger Could you share your work? I would like to try it? On 3/8/2012 9:16 AM, Roger Castaldo wrote: > I wanted to avoid putting myself out there but there is another gui > project that I have avoided releasing as open source on account of the > fact that I am having issues finding developers to help. It is > written in C#, built to run on mono or .net. Runs on windows, debian, > slitaz and centos can be designed to run perfectly on others. It > contains ability to configure full server stuff as well as controls > the call flow of freeswitch through the sockets library. It is > designed to be web 2.0 compliant as its all done through javascript > and a single web page using ajax calls with json encoding for > efficiency. There is still some things needing to be done such as > developing a billling module for it, ensuring that mobile browsers can > operate it successfully and adding new functionality through modules > as well as completing one or two modules. The few people who I have > had take a look and try it out have been impressed and slowly, as I > have time i have been cleaning up code, making it more efficient and > trying to improve functionality. It takes time though when its one > person and they are not getting paid to do it, its more of a hobby. > > On Thu, Mar 8, 2012 at 8:18 AM, Patrick Lists > > wrote: > > On 08-03-12 09:17, Brett Wilson wrote: > > I checked out freepbx. Freepbx v3, the complete rewrite that was > > supposed to work with freeswitch, actually got spun off into the > > blue.box project. Seems that the blue.box project is basically > dead. The > > last commit was November 2011. Fusionpbx looks to be the only > game in > > town. I was wrong, it is still being developed. I just did a new > install > > of it today, and allowed it to auto update. There are indeed some > > changes that I see from my install from about 4 months ago. So > that is a > > positive thing, but its too bad that the feel is still lacking. > I took a > > look at some code, and what I saw did not impress me. Display > code mixed > > right in with the logic. I did not look for more than about 30 > seconds, > > but IMO it is not ?proper? application design. I am a big > proponent of > > MVC architecture, and what I saw of fusionpbx code does not look all > > that impressive. I give credit where credit is due. The platform is > > functioning to an extent, most things actually work. I do > realize that > > developing something of that size requires tons of effort. My ideal > > solution would be something driven by ExtJS and PHP on the backend. > > ExtJS provides the most advanced and rich javascript controls i have > > seen. Plus its sister product, sencha touch, would enable great > admin > > functionality from a smartphone or tablet device. Here I go > rambling on? > > I wish someone would develop an awesome GUI for freeswitch to > help get > > it into the limelight where it deserves to be. If I had the time > I would > > love to create a comprehensive and great looking GUI. > > I you like blue.box more than fusionpbx but feel things are missing in > blue.box then why not talk to the 2600hz developers and hire them > to add > that functionality? > > If you are looking for a slick GUI then have a look at the Cudatel: > / > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1913 / Virus Database: 2114/4866 - Release Date: 03/12/12 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120312/5c7b3333/attachment-0001.html From peter.olsson at visionutveckling.se Mon Mar 12 16:47:38 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 12 Mar 2012 13:47:38 +0000 Subject: [Freeswitch-users] how does eval command works Message-ID: <1FFF97C269757C458224B7C895F35F150514DB@cantor.std.visionutv.se> I just tried using eval as described on the wiki, and it worked for me... This is what I tried; eval uuid:09aeb781-4d31-40c4-b2be-b1bb4ee63508 ${channel-state} /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r piyush singhai Skickat: den 12 mars 2012 11:13 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] how does eval command works I want to know the channel state corresponding to chan_uuid. I tried eval but it is not working can anybody tell me how it will work. I followed the wiki http://wiki.freeswitch.org/wiki/Mod_commands --Piyush !DSPAM:4f5dcaf432769251570223! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120312/60d41712/attachment.html From peter.olsson at visionutveckling.se Mon Mar 12 16:49:58 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 12 Mar 2012 13:49:58 +0000 Subject: [Freeswitch-users] bootstrap Message-ID: <1FFF97C269757C458224B7C895F35F150514E5@cantor.std.visionutv.se> Usually just once. Sometimes a bigger change might be commited to git which requires a re-bootstrap, but that is very rare. Usually you just need "make current", and it will do everything for you. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Bzzz Skickat: den 12 mars 2012 04:10 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] bootstrap Hi, is it necessary to re-bootstrap @ each update, or just once for the first compilation? -- BOFH excuse #404: Sysadmin accidentally destroyed pager with a large hammer. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f5da5dd32767885415725! From peter.olsson at visionutveckling.se Mon Mar 12 16:52:54 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 12 Mar 2012 13:52:54 +0000 Subject: [Freeswitch-users] connexion attempts behing a firewall (wtf?) Message-ID: <1FFF97C269757C458224B7C895F35F150514F0@cantor.std.visionutv.se> If you get USER_NOT_REGISTERED, that means that the user is not registered. Please do a "show registrations", and you will see who is registered to your server. Also - if you want to disable NAT for now (and don't let FS punch holes in your fw) you can also start FS using the "-nonat" switch. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Bzzz Skickat: den 12 mars 2012 00:39 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] connexion attempts behing a firewall (wtf?) On Sun, 11 Mar 2012 19:21:01 -0400 Brian Foster wrote: > This happens all the time. You need fail2ban set up properly. There > are articles on both the freeswitch and fusionpbx wikis. It is, and tested as the wiki says:) but attacker isn't banned. This show me this is also linked to my previous post: softphones & fusionpbx say they're registered, but when I try to make a call between them FS throw me USER_NOT_REGISTERED errors, this is why the attacker's not jettisoned: FS don't seem to answer to a correct registration:( > I run a public server > and I get people doing scans all the time. Does it worry me? No. Of > course not. That's what fail2ban and other measures are for. Interesting, could you describe "other measures", PLS? -- Draft beer, not boys! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f5da5e232761141321256! From ira at connectmevoice.com Mon Mar 12 17:44:22 2012 From: ira at connectmevoice.com (Ira Tessler) Date: Mon, 12 Mar 2012 10:44:22 -0400 Subject: [Freeswitch-users] Question on Multiple Registrations with Polycom 501 Message-ID: <07d9ccbebe3f1bbcd5c0e705b5299706@mail.gmail.com> I have a Polycom 501 that is going in and out of registration and creating multiple registrations. Anyone have an ideas why? Thanks! Registrations: ================================================================================================= Call-ID: 6a66a4f8-215d8f26-cf1e7593 at 192.168.4.115 User: 101 at 19036.cmvtele.com Contact: "user" Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134 Status: Registered(UDP)(unknown) EXP(2012-03-12 10:41:31) EXPSECS(14) Host: fs2.cmvvoip.com IP: 4.59.18.226 Port: 5060 Auth-User: 101 Auth-Realm: 19036.cmvtele.com MWI-Account: 101 at 19036.cmvtele.com Call-ID: c388378d-c3835e0b-c4deaab0 at 192.168.4.115 User: 101 at 19036.cmvtele.com Contact: "user" Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134 Status: Registered(UDP)(unknown) EXP(2012-03-12 10:51:24) EXPSECS(607) Host: fs2.cmvvoip.com IP: 4.59.18.226 Port: 5060 Auth-User: 101 Auth-Realm: 19036.cmvtele.com MWI-Account: 101 at 19036.cmvtele.com Call-ID: 5d0c1f2e-69c026bc-dc40a9b9 at 192.168.4.115 User: 101 at 19036.cmvtele.com Contact: "user" Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134 Status: Registered(UDP)(unknown) EXP(2012-03-12 10:52:06) EXPSECS(649) Host: fs2.cmvvoip.com IP: 4.59.18.226 Port: 5060 Auth-User: 101 Auth-Realm: 19036.cmvtele.com MWI-Account: 101 at 19036.cmvtele.com Total items returned: 3 Ira Tessler ConnectMe (732) 490-9007 x2 www.connectmevoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120312/e0fa4d90/attachment.html From lists at telefaks.de Mon Mar 12 18:27:18 2012 From: lists at telefaks.de (Peter Steinbach) Date: Mon, 12 Mar 2012 16:27:18 +0100 Subject: [Freeswitch-users] Audio problems with unimrcp and Vestec ASR In-Reply-To: <4F53A59E.9070001@telefaks.de> References: <4F5374C3.800@telefaks.de> <1FFF97C269757C458224B7C895F35F1504CD2F@cantor.std.visionutv.se> <4F53A59E.9070001@telefaks.de> Message-ID: <4F5E15D6.1080900@telefaks.de> Just for the records... after extensive testing with Vestec support, we were still at the point that Freeswitch was sending empty RTP packets to the MRCP server. After updating Freeswitch today to newest GIT (our previous GIT version was from 29-Jan-2012), recognition via MRCP worked successfully. Thanks to all who supported on this issue. Best regards Peter Am 04.03.2012 18:25, schrieb Peter Steinbach: > Hello Peter, > > thanks for the hint. It was using L16 codec, so I changed the > avaliable codecs for the mrcp profile, so that it only accepts PCMA. > Now it negociates PCMA, see FS log: > > 2012-03-04 18:12:40.266537 [INFO] rtsp_client.c:929 () Receive RTSP > Stream 192.168.178.221:54797 <-> 192.168.178.180:1554 [324 bytes] > RTSP/1.0 200 OK > CSeq: 1 > Transport: RTP/AVP;unicast;client_port=4008-4009;server_port=5022-5023 > Session: dafbbc3e7efb4474 > Content-Type: application/sdp > Content-Length: 145 > v=0 > o=UniMRCPServer 0 0 IN IP4 192.168.178.180 > s=- > c=IN IP4 192.168.178.180 > t=0 0 > m=audio 5022 RTP/AVP 8 > a=rtpmap:8 PCMA/8000 > a=recvonly > > But the behaviour is the same. The RTP stream I grepped on the network > seems always to be empty (silence). > All UDP RTP Packets sent from FS look like this (hex values): > 80:08:00:08:00:00:05:00:f9:53:b5:e6:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5 > > > > Best regards > Peter > > > Am 04.03.2012 15:17, schrieb Peter Olsson: >> What codec is used on the call leg? I tried some ASR (UniMRCP to >> Nuance) once, and I noticed that the audio passed to Nuance seemed to >> fail when using HD audio, when I did the same test with standard PCMA >> it worked. I never had the time to investigate any further though, >> since this was onle for a test, and nothing I needed to implement yet. >> >> /Peter >> >> ________________________________ >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org >> [freeswitch-users-bounces at lists.freeswitch.org] f?r Peter Steinbach >> [lists at telefaks.de] >> Skickat: den 4 mars 2012 14:57 >> Till: FreeSWITCH Users Help >> ?mne: [Freeswitch-users] Audio problems with unimrcp and Vestec ASR >> >> I have an Audio problem with unimrcp and Vestec ASR with mrcp v1. >> >> This seems to be a similar problem to thread: "ASR from Freeswitch to >> MS Speech Server [using MRCP Connector] - Audio Problem" >> >> What does work: >> >> * MRCP command are passed successsfull in both ways on port 1554 >> (vestec mrcp server) >> * Grammar is accepted and recognition is started >> * I can see RTP stream (wireshark) from FS port 400x to the mrcp >> server >> * Result (002 no-input-timeout) is sent back to Freeeswicth and >> is processed by FS successfully. >> >> What does NOT work: >> >> * RTP stream is always empty (silence), i grepped this on the >> network with ngrep, RTP data is empty, there is no change in the >> audio data when I speak. >> * so voice recognition will always time out (002 no-input-timeout) >> >> Here are my configs: >> unimrcp.conf.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> unimrcpserver-mrcp-v1.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> See logs here >> http://pastebin.freeswitch.org/18570 >> >> Any help is appreciated. >> >> Best egards >> Peter >> >> -- >> With kind regards >> Peter Steinbach >> >> Telefaks Services GmbH >> mailto:lists (att) telefaks.de >> Internet: www.telefaks.de >> >> >> >> !DSPAM:4f5373f932761937219840! >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From anthony.minessale at gmail.com Mon Mar 12 19:41:59 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 12 Mar 2012 11:41:59 -0500 Subject: [Freeswitch-users] how does eval command works In-Reply-To: <1FFF97C269757C458224B7C895F35F150514DB@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F150514DB@cantor.std.visionutv.se> Message-ID: eval basically does nothing allowing expansions via API variables and substitution to occur with no actual app or function directly executed. On Mon, Mar 12, 2012 at 8:47 AM, Peter Olsson wrote: > I just tried using eval as described on the wiki, and it worked for me... > This is what I tried; > > > > eval uuid:09aeb781-4d31-40c4-b2be-b1bb4ee63508 ${channel-state} > > > > /Peter > > > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r piyush singhai > Skickat: den 12 mars 2012 11:13 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] how does eval command works > > > > I want to know the channel state corresponding to chan_uuid. I tried eval > but it is not working can anybody tell me how it will work. > > I followed the wiki > > http://wiki.freeswitch.org/wiki/Mod_commands > > --Piyush > !DSPAM:4f5dcaf432769251570223! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From kris at kriskinc.com Mon Mar 12 20:12:34 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 12 Mar 2012 13:12:34 -0400 Subject: [Freeswitch-users] How to disable Re-invite? In-Reply-To: References: Message-ID: Bala, Can you give any more information? It doesn't look like this re-INVITE is changing the session but I can't be sure what's going on without a full console log and SIP trace. Also, the client sends the re-INVITEs. If you want to disable them you should look there first. On Mon, Mar 12, 2012 at 7:12 AM, Balamurugan Mahendran wrote: > > 2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:2925 Already using PCMU > 2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:4988 Set 2833 dtmf send > payload to 101 > 2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:3213 Audio params are > unchanged for sofia/external/+919884730340. > 2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:3223 > sofia/external/+919884730340 Setting audio receive payload in Re-INVITE to 0 > > Thanks, > Bala > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From kris at kriskinc.com Mon Mar 12 20:13:15 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 12 Mar 2012 13:13:15 -0400 Subject: [Freeswitch-users] How to avoid changing of internal profile binding when VPN comes up? In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Mod_sofia#Forcing_SIP_profile_to_use_a_static_IP_address On Mon, Mar 12, 2012 at 5:47 AM, Frank Church wrote: > > I am running Freeswitch on a VM which uses 10.0.2.x subnet for NAT, and > bridges to the hosts subnet on 192.168.x.x network. When I start the server > the internal and external profiles automatically bind to the 10.0.2.x > subnet. I don't need to edit any of the IPs in vars.xml or internal.xml or > external.xml. But when the VPN comes up Freeswitch automatically binds to > the VPNs subnet. > > Are there some rules which determine what internal and external profiles > bind themselves to by default? Is it? the internet gateway with lowest > metric that they are bound to? > > -- > Frank Church > > ======================= > http://devblog.brahmancreations.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From kris at kriskinc.com Mon Mar 12 20:14:50 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 12 Mar 2012 13:14:50 -0400 Subject: [Freeswitch-users] Send PAI and RPID In-Reply-To: <4F5D1B92.8000803@cupis.co.uk> References: <4F5D1B92.8000803@cupis.co.uk> Message-ID: This is much more difficult than it seems - you need to set the screening type, etc. Ideally FreeSWITCH would include a PAI+RPID option. Perhaps the OP should create a bounty? On Sun, Mar 11, 2012 at 5:39 PM, Paul Cupis wrote: > On 11/03/12 19:47, Anto wrote: >> That I want to do, send the two, but do not know if can do in FreeSWITCH. > > If you really need to do this, why not set FreeSWITCH to send PAID and > then in your dialplan manually construct and add an RPID header to > outbound INVITEs for the relevant gateways? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From krice at freeswitch.org Mon Mar 12 22:19:03 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 12 Mar 2012 13:19:03 -0600 Subject: [Freeswitch-users] FreeSWITCH 1.2 is on the horizon. Message-ID: The FreeSWITCH Development Team is happy to announce that 1.2 is officially on the horizon. Starting Wed March 14th 2012, the Development branch of FreeSWITCH will reach a Feature Freeze. What does this mean for you the user? It means we will have a stable known feature set heading into the Release Candidate Cycle. Only patches that Fix Bugs will be accepted. If you have a feature you would like to see included, get us a Jira with a full patch set so we can evaluate its inclusion with 1.2. What we need from the community: Testing, Testing, more Testing, and Documentation Updates. The freeze will last 2 to 4 weeks as we spool up testing and everything else we need to get the Release Candidates ready for prime time. If you have outstanding bugs on Jira, please help us help you during this time by making sure all information on them is up to date. Grab the latest GIT Head and see if your bugs have been resolved and someone forgot to close your Jira. If you want to help or need some help diagnosing and issue visit us on IRC via irc.freenode.net/#freeswitch any time. The FreeSWITCH Dev Team -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120312/f72a4bb2/attachment.html From anthony.minessale at gmail.com Mon Mar 12 22:19:31 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 12 Mar 2012 14:19:31 -0500 Subject: [Freeswitch-users] How to avoid changing of internal profile binding when VPN comes up? In-Reply-To: References: Message-ID: also consider hard coding local_ip_v4 to the desired IP which will propagate to most of the default configs. On Mon, Mar 12, 2012 at 12:13 PM, Kristian Kielhofner wrote: > http://wiki.freeswitch.org/wiki/Mod_sofia#Forcing_SIP_profile_to_use_a_static_IP_address > > On Mon, Mar 12, 2012 at 5:47 AM, Frank Church wrote: >> >> I am running Freeswitch on a VM which uses 10.0.2.x subnet for NAT, and >> bridges to the hosts subnet on 192.168.x.x network. When I start the server >> the internal and external profiles automatically bind to the 10.0.2.x >> subnet. I don't need to edit any of the IPs in vars.xml or internal.xml or >> external.xml. But when the VPN comes up Freeswitch automatically binds to >> the VPNs subnet. >> >> Are there some rules which determine what internal and external profiles >> bind themselves to by default? Is it? the internet gateway with lowest >> metric that they are bound to? >> >> -- >> Frank Church >> >> ======================= >> http://devblog.brahmancreations.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From ivan.ilves at gmail.com Mon Mar 12 15:20:50 2012 From: ivan.ilves at gmail.com (Ivan Ilves) Date: Mon, 12 Mar 2012 14:20:50 +0200 Subject: [Freeswitch-users] Check SRTP encryption on Leg B Message-ID: Dear FreeSWITCH pros, I need to check SRTP encryption on both call legs. Checking SRTP encryption on leg A was easy enough: --- CUT HERE --- --- CUT HERE --- But how do I check SRTP encryption on leg B? I need this to disallow improperly configured devices to receive unencrypted calls. Could you please help me with this? Thanks in advance!!! From lazyvirus at gmx.com Mon Mar 12 17:37:54 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 12 Mar 2012 15:37:54 +0100 Subject: [Freeswitch-users] connexion attempts behing a firewall (wtf?) In-Reply-To: <1FFF97C269757C458224B7C895F35F150514F0@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F150514F0@cantor.std.visionutv.se> Message-ID: <20120312153754.5799fa92@anubis.defcon1> On Mon, 12 Mar 2012 13:52:54 +0000 Peter Olsson wrote: Hi Peter, > If you get USER_NOT_REGISTERED, that means that the user is not > registered. Please do a "show registrations", and you will see who > is registered to your server. Thanks, that's exactly the command I was looking after! (but I missed the final 's'). show registrations reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname 01,192.168.1.25,yeveyyihyxqvepr at anubis.defcon1,sofia/internal/sip:01 at 192.168.1.25:5000,1331565571,192.168.1.25,5000,udp,anubis 02,192.168.1.25,tufsyqgvkshuonx at osiris.defcon1,sofia/internal/sip:02 at 192.168.1.50:5100,1331565589,192.168.1.50,5100,udp,anubis 2 total. So, it seems that my 2 softphone are correctly registered, which is in accordance w/ what fusionpbx and the phones says; I guess the strange string before machines' names is normal. Raaaaaaaahhhhhhhh! > Also - if you want to disable NAT for now (and don't let FS punch > holes in your fw) you can also start FS using the "-nonat" switch. Yeah, Patrick sent me a useful link about Auto_NAT. As a matter of fact my PB is coming from the box itself: once some uPNP forwards have been set, I can't get rid about them, even disabling uPNP don't work; I'm obliged to disable uPNP AND reboot the box to do that, thanks to sagem:(( (after disabling, it tells that Nb of rules is 0, though) So, this is my solution: keep ADSL box non-uPNP until FS conf is cooked "aux petits oignons":) But the most "interesting" thing is those forwards lead to someone in Brazil trying to authenticate, but fail2ban failed to ban him because there was no "auth failure" answer from FS - However, when I tested it from my LAN, I was forbidden access after 3 unsuccessful registration attempts. I'm a bit lost at this point: both phone registered and still no communication possible (forgot to say: I fall directly to the VM message when trying that.) Jean-Yves -- Q: What's meaner than a pit bull with AIDS? A: The guy that gave it to him. From lazyvirus at gmx.com Mon Mar 12 17:38:36 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 12 Mar 2012 15:38:36 +0100 Subject: [Freeswitch-users] bootstrap In-Reply-To: <1FFF97C269757C458224B7C895F35F150514E5@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F150514E5@cantor.std.visionutv.se> Message-ID: <20120312153836.4d619b71@anubis.defcon1> On Mon, 12 Mar 2012 13:49:58 +0000 Peter Olsson wrote: > Usually just once. Sometimes a bigger change might be commited to git which requires a re-bootstrap, but that is very rare. > > Usually you just need "make current", and it will do everything for you. Thanks Peter. -- From lazyvirus at gmx.com Mon Mar 12 23:43:08 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 12 Mar 2012 21:43:08 +0100 Subject: [Freeswitch-users] [OT] Re: connexion attempts behing a firewall (wtf?) In-Reply-To: <20120312153754.5799fa92@anubis.defcon1> References: <1FFF97C269757C458224B7C895F35F150514F0@cantor.std.visionutv.se> <20120312153754.5799fa92@anubis.defcon1> Message-ID: <20120312214308.2f686104@anubis.defcon1> On Mon, 12 Mar 2012 15:37:54 +0100 Bzzz wrote: Sorry for the off-topic, but I experience something strange: I sent this one @ 15:37 local (+0100) and just received it from the mailing list - does anybody have the same delay, or is it gmx.com that play it slooow? JY -- Losing your faith is a lot like losing your virginity you don't realise how irritating it was 'til it's gone. From singhai.piyush at gmail.com Tue Mar 13 05:09:58 2012 From: singhai.piyush at gmail.com (piyush singhai) Date: Tue, 13 Mar 2012 07:39:58 +0530 Subject: [Freeswitch-users] how does eval command works In-Reply-To: References: <1FFF97C269757C458224B7C895F35F150514DB@cantor.std.visionutv.se> Message-ID: Yes Correct it is also working for me eval uuid:09aeb781-4d31-40c4-b2be-b1bb4ee63508 ${channel-state} What my expectation with this command is when i run this command with while true loop it should return for that channel CS_NEW--->CS_INIT--->CS_ROUTING--->CS_EXECUTE--->CS_HANGUP--->CS_REPORTING-->CS_DESTROY But i see only CS_EXECUTE. After CS_EXECUTE this command returns "no reply" for that channel. --Piyush On Mon, Mar 12, 2012 at 10:11 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > eval basically does nothing allowing expansions via API variables and > substitution to occur with no actual app or function directly > executed. > > > On Mon, Mar 12, 2012 at 8:47 AM, Peter Olsson > wrote: > > I just tried using eval as described on the wiki, and it worked for me... > > This is what I tried; > > > > > > > > eval uuid:09aeb781-4d31-40c4-b2be-b1bb4ee63508 ${channel-state} > > > > > > > > /Peter > > > > > > > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r piyush > singhai > > Skickat: den 12 mars 2012 11:13 > > Till: FreeSWITCH Users Help > > ?mne: [Freeswitch-users] how does eval command works > > > > > > > > I want to know the channel state corresponding to chan_uuid. I tried eval > > but it is not working can anybody tell me how it will work. > > > > I followed the wiki > > > > http://wiki.freeswitch.org/wiki/Mod_commands > > > > --Piyush > > !DSPAM:4f5dcaf432769251570223! > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/fe8a2129/attachment.html From anton.jugatsu at gmail.com Tue Mar 13 08:01:51 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Tue, 13 Mar 2012 09:01:51 +0400 Subject: [Freeswitch-users] FreeSWITCH 1.2 is on the horizon. In-Reply-To: References: Message-ID: This is great news! 12 ????? 2012 ?. 23:19 ???????????? Ken Rice ???????: > The FreeSWITCH Development Team is happy to announce that 1.2 is > officially on the horizon. > Starting Wed March 14th 2012, the Development branch of FreeSWITCH will > reach a Feature Freeze. > > What does this mean for you the user? It means we will have a stable known > feature set heading into the Release Candidate Cycle. Only patches that Fix > Bugs will be accepted. If you have a feature you would like to see > included, get us a Jira with a full patch set so we can evaluate its > inclusion with 1.2. > > What we need from the community: Testing, Testing, more Testing, and > Documentation Updates. > > The freeze will last 2 to 4 weeks as we spool up testing and everything > else we need to get the Release Candidates ready for prime time. > > If you have outstanding bugs on Jira, please help us help you during this > time by making sure all information on them is up to date. Grab the latest > GIT Head and see if your bugs have been resolved and someone forgot to > close your Jira. > > If you want to help or need some help diagnosing and issue visit us on IRC > via irc.freenode.net/#freeswitch any time. > > The FreeSWITCH Dev Team > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/dbe1ba71/attachment.html From bob.mccarthy at experient.com Tue Mar 13 10:38:19 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Tue, 13 Mar 2012 01:38:19 -0600 Subject: [Freeswitch-users] Caller-Callee-ID-Name(please help) In-Reply-To: <4F509B49.7060703@softnet.si> References: <4F5076E4.5080904@softnet.si> <4F509B49.7060703@softnet.si> Message-ID: <040301cd00ec$44a83a90$cdf8afb0$@mccarthy@experient.com> How did you figure it out ? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Miha Zoubek Sent: Friday, March 02, 2012 3:05 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Caller-Callee-ID-Name(please help) On 03/02/2012 08:29 AM, Miha Zoubek wrote: > Hi, > > How can I set this variable on a leg(Caller-Callee-ID-Name)? I have try > to put in user/dir but with no luck. > > Usr dir: > > > > > > > > > > log from cdr on b leg: > > > > Caller-Direction: [outbound] > Caller-Username: [018108500.fs_kabelvoip1] > Caller-Dialplan: [XML] > Caller-Caller-ID-Name: [Outbound Call] > Caller-Caller-ID-Number: [38651357952] > > > > a leg: > Caller-Caller-ID-Name: [018108500] > Caller-Caller-ID-Number: [018108500.fs_kabelvoip1] > Caller-Callee-ID-Name: [Outbound Call] > Caller-Callee-ID-Number: [38651357952] > > Thanks! > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > I figure it out! Thanks guys! miha _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From miha at softnet.si Tue Mar 13 10:39:49 2012 From: miha at softnet.si (Miha Zoubek) Date: Tue, 13 Mar 2012 08:39:49 +0100 Subject: [Freeswitch-users] stip variable In-Reply-To: <73D50A0D-B8C3-4EB2-91A5-6B2C76C82907@freeswitch.org> References: <4F588FE5.3090402@softnet.si> <73D50A0D-B8C3-4EB2-91A5-6B2C76C82907@freeswitch.org> Message-ID: <4F5EF9C5.6020202@softnet.si> Hi @Brian, I was looking at Enum.conf but having problem with modifying it. I can remove sofia/internal/ and Enum will return just 18005551212 at tf.voipmich.com. What is the best why to get rid of @tf.voipmich.com as I need only number? I was trying to trim it like exp. {18005551212 at tf.voipmich.com:10:10} but this is not working properly as length of number change. Thanks! Regards, Miha On 03/08/2012 04:10 PM, Brian West wrote: > or modify enum.conf to only give you the numer? > > /b > > On Mar 8, 2012, at 4:54 AM, Miha Zoubek wrote: > >> Hi, >> >> how can I strip thissofia/internal/18005551212 at tf.voipmich.com, so >> that I would have only number 18005551212? >> >> This return enum. >> >> >> Thanks! >> Miha > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/3a6d35d9/attachment-0001.html From miha at softnet.si Tue Mar 13 11:29:15 2012 From: miha at softnet.si (Miha Zoubek) Date: Tue, 13 Mar 2012 09:29:15 +0100 Subject: [Freeswitch-users] Caller-Callee-ID-Name(please help) In-Reply-To: <040301cd00ec$44a83a90$cdf8afb0$@mccarthy@experient.com> References: <4F5076E4.5080904@softnet.si> <4F509B49.7060703@softnet.si> <040301cd00ec$44a83a90$cdf8afb0$@mccarthy@experient.com> Message-ID: <4F5F055B.304@softnet.si> On 03/13/2012 08:38 AM, Bob McCarthy wrote: > How did you figure it out ? > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Miha > Zoubek > Sent: Friday, March 02, 2012 3:05 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Caller-Callee-ID-Name(please help) > > On 03/02/2012 08:29 AM, Miha Zoubek wrote: >> Hi, >> >> How can I set this variable on a leg(Caller-Callee-ID-Name)? I have try >> to put in user/dir but with no luck. >> >> Usr dir: >> >> >> >> >> >> >> >> >> >> log from cdr on b leg: >> >> >> >> Caller-Direction: [outbound] >> Caller-Username: [018108500.fs_kabelvoip1] >> Caller-Dialplan: [XML] >> Caller-Caller-ID-Name: [Outbound Call] >> Caller-Caller-ID-Number: [38651357952] >> >> >> >> a leg: >> Caller-Caller-ID-Name: [018108500] >> Caller-Caller-ID-Number: [018108500.fs_kabelvoip1] >> Caller-Callee-ID-Name: [Outbound Call] >> Caller-Callee-ID-Number: [38651357952] >> >> Thanks! >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > I figure it out! > > Thanks guys! > > miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > HI, I think I did it like this: data="{origination_callee_id_name='${sip_req_user}'}sofia/external... Regards, Miha From b2m at a-cti.com Tue Mar 13 12:00:45 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Tue, 13 Mar 2012 14:30:45 +0530 Subject: [Freeswitch-users] Choppy Audio Recordings Message-ID: MP3 recording for voicemail is bad with yesterdays build, I tried to change it as wav looks better. Not sure its only me or for many. Thanks, Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/da666a35/attachment.html From xing2kin at yahoo.com Tue Mar 13 12:46:27 2012 From: xing2kin at yahoo.com (king2kin) Date: Tue, 13 Mar 2012 02:46:27 -0700 (PDT) Subject: [Freeswitch-users] Failed to pass arguments to lua script of FreeSwitch ('arg' and 'arg[i]' are nil values) Message-ID: <1331631987.12002.YahooMailNeo@web162802.mail.bf1.yahoo.com> Hi all, ? It's said we could access the arguments passed to lua script by variable 'arg[i]' where i > 0. ? for example,? { lua dummy.lua? aa 100 } then, we will have as follows: arg[1] -> aa arg[2] -> 100 ? However, I always get 'nil' values for 'arg[i]' or 'arg' while running the lua script. ? How can I pass and retrieve arguments or parameters to a lua script of FreeSwitch? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/002d908a/attachment.html From asilva at wirelessmundi.com Tue Mar 13 13:12:50 2012 From: asilva at wirelessmundi.com (Antonio) Date: Tue, 13 Mar 2012 11:12:50 +0100 Subject: [Freeswitch-users] Failed to pass arguments to lua script of FreeSwitch ('arg' and 'arg[i]' are nil values) In-Reply-To: <1331631987.12002.YahooMailNeo@web162802.mail.bf1.yahoo.com> References: <1331631987.12002.YahooMailNeo@web162802.mail.bf1.yahoo.com> Message-ID: <1331633570.21039.35.camel@marces.madrid.commsmundi.com> Hi, Your missing an "V" in the arg name, "argv". You can try do the following: in diaplan: in the lua: var1 = argv[1] or "default value to avoid errors" var2 = argv[1] or "default value var2" freeswitch.consoleLog("info", tostring(var1) .. "\n") freeswitch.consoleLog("info", tostring(var2) .. "\n") you can also call the lua script directly from the cli: fs:> luarun yourcript.lua var1 var2 Regards, Ant?nio On Tue, 2012-03-13 at 02:46 -0700, king2kin wrote: > Hi all, > > It's said we could access the arguments passed to lua script by > variable 'arg[i]' where i > 0. > > for example, { lua dummy.lua aa 100 } > then, we will have as follows: > arg[1] -> aa > arg[2] -> 100 > > However, I always get 'nil' values for 'arg[i]' or 'arg' while running > the lua script. > > How can I pass and retrieve arguments or parameters to a lua script of > FreeSwitch? > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Un cordial saludo / Best regards, _________________________ Ant?nio Silva E-mail:asilva at wirelessmundi.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/5eb00e2a/attachment.html From B.Tietz at pinguin.ag Tue Mar 13 13:39:01 2012 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Tue, 13 Mar 2012 11:39:01 +0100 Subject: [Freeswitch-users] Stuck Calls Message-ID: <07BF4904977CC645B485E970424193AD0FF1109F8B@localhost> Hi, I have some stuck calls in the list when I issue "show detailed_calls" Trying to kill them with "uuid_kill " or "fsctl hupall normal_clearing dialed_ext 1234" doesn't help. FS is configured as HA-node with odbc in the core and sofia. Any other suggestions?! Benjamin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/84901fba/attachment.html From peter.olsson at visionutveckling.se Tue Mar 13 13:46:41 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 13 Mar 2012 10:46:41 +0000 Subject: [Freeswitch-users] Stuck Calls Message-ID: <1FFF97C269757C458224B7C895F35F15051C7C@cantor.std.visionutv.se> Are you on latest git? There was a fix made for this about a week ago. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r B.Tietz at pinguin.ag Skickat: den 13 mars 2012 11:39 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] Stuck Calls Hi, I have some stuck calls in the list when I issue "show detailed_calls" Trying to kill them with "uuid_kill " or "fsctl hupall normal_clearing dialed_ext 1234" doesn't help. FS is configured as HA-node with odbc in the core and sofia. Any other suggestions?! Benjamin !DSPAM:4f5f22d932768997615399! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/e1aa9eb0/attachment-0001.html From Vladislav.Grishin at vts24.ru Tue Mar 13 13:48:24 2012 From: Vladislav.Grishin at vts24.ru (=?koi8-r?B?98zBxMnTzMHXIOfSydvJzg==?=) Date: Tue, 13 Mar 2012 14:48:24 +0400 Subject: [Freeswitch-users] make[1]: *** [mod_shout-install] Error 1 Message-ID: I have FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 17-48-30 -0600) on Centos 5.5 >From src directory I run command first time [root at FS-A freeswitch]# make mod_shout-install : --2012-03-11 13:59:40-- http://files.freeswitch.org/downloads/libs/libshout-2.2.2.tar.gz Resolving files.freeswitch.org... 67.201.31.224 Connecting to files.freeswitch.org|67.201.31.224|:80... connected. HTTP request sent, awaiting response... 200 OK Length: 478582 (467K) [application/x-gzip] Saving to: `libshout-2.2.2.tar.gz' 100%[======================================================================= ===================================================>] 478,582 969K/s in 0.5s 2012-03-11 13:59:40 (969 KB/s) - `libshout-2.2.2.tar.gz' saved [478582/478582] /bin/sh: patch: command not found make[3]: *** [/usr/local/src/freeswitch/libs/libshout-2.2.2] Error 127 make[2]: *** [install] Error 1 make[1]: *** [mod_shout-install] Error 1 make: *** [mod_shout-install] Error 2 [root at FS-A freeswitch]# second time [root at FS-A freeswitch]# make mod_shout-install : Making all in doc Compiling /usr/local/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c... mkdir .libs Compiling /usr/local/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c ... Creating mod_shout.so... installing mod_shout.so [root at FS-A freeswitch]# All ok Vladislav Grishin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/0341a6a2/attachment.html From B.Tietz at pinguin.ag Tue Mar 13 14:03:14 2012 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Tue, 13 Mar 2012 12:03:14 +0100 Subject: [Freeswitch-users] Stuck Calls In-Reply-To: <1FFF97C269757C458224B7C895F35F15051C7C@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F15051C7C@cantor.std.visionutv.se> Message-ID: <07BF4904977CC645B485E970424193AD0FF1109FDE@localhost> Not yet... It's a productive system and I would like to check every version... but thanks fort hat hint... I'll try the latest GIT VG, Benjamin T. Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Peter Olsson Gesendet: Dienstag, 13. M?rz 2012 11:47 An: 'FreeSWITCH Users Help' Betreff: Re: [Freeswitch-users] Stuck Calls Are you on latest git? There was a fix made for this about a week ago. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r B.Tietz at pinguin.ag Skickat: den 13 mars 2012 11:39 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] Stuck Calls Hi, I have some stuck calls in the list when I issue "show detailed_calls" Trying to kill them with "uuid_kill " or "fsctl hupall normal_clearing dialed_ext 1234" doesn't help. FS is configured as HA-node with odbc in the core and sofia. Any other suggestions?! Benjamin !DSPAM:4f5f22d932768997615399! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/bf164a88/attachment.html From b2m at a-cti.com Tue Mar 13 14:09:42 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Tue, 13 Mar 2012 16:39:42 +0530 Subject: [Freeswitch-users] make[1]: *** [mod_shout-install] Error 1 In-Reply-To: References: Message-ID: I guess either the download or extract was interrupted, try deleting libshout-2.2.2.tar.gz and extracted content. Then go for make and make install. Thanks, Bala 2012/3/13 ????????? ?????? > I have FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 17-48-30 -0600) > on Centos 5.5**** > > ** ** > > From src directory I run command first time**** > > ** ** > > [root at FS-A freeswitch]# make mod_shout-install**** > > ...**** > > --2012-03-11 13:59:40-- > http://files.freeswitch.org/downloads/libs/libshout-2.2.2.tar.gz**** > > Resolving files.freeswitch.org... 67.201.31.224**** > > Connecting to files.freeswitch.org|67.201.31.224|:80... connected.**** > > HTTP request sent, awaiting response... 200 OK**** > > Length: 478582 (467K) [application/x-gzip]**** > > Saving to: `libshout-2.2.2.tar.gz'**** > > ** ** > > 100%[==========================================================================================================================>] > 478,582 969K/s in 0.5s**** > > ** ** > > 2012-03-11 13:59:40 (969 KB/s) - `libshout-2.2.2.tar.gz' saved > [478582/478582]**** > > ** ** > > /bin/sh: patch: command not found**** > > make[3]: *** [/usr/local/src/freeswitch/libs/libshout-2.2.2] Error 127**** > > make[2]: *** [install] Error 1**** > > make[1]: *** [mod_shout-install] Error 1**** > > make: *** [mod_shout-install] Error 2**** > > [root at FS-A freeswitch]#**** > > ** ** > > second time **** > > ** ** > > [root at FS-A freeswitch]# make mod_shout-install**** > > ...**** > > Making all in doc**** > > Compiling > /usr/local/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c...**** > > mkdir .libs**** > > Compiling /usr/local/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c > ...**** > > Creating mod_shout.so...**** > > installing mod_shout.so**** > > [root at FS-A freeswitch]# **** > > ** ** > > All ok**** > > ** ** > > Vladislav Grishin**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/a2853921/attachment-0001.html From cstomi.levlist at gmail.com Tue Mar 13 17:04:43 2012 From: cstomi.levlist at gmail.com (Tamas.Cseke ) Date: Tue, 13 Mar 2012 15:04:43 +0100 Subject: [Freeswitch-users] att_xfer uuid_bridge result Message-ID: <4F5F53FB.3080504@gmail.com> Hello, We'd like to know if the att_xfer was successful. Let me explain. User calls a client and transfers it to another user with att_xfer Users talk to each other, then one of them hangs up. Here we need the information if the client is up (because the user has to make administrative work after the call) If the hangup is made by the user and the client is alive, the other user can do the administation. But if the client isn't up, when the user channel tries to uuid bridge the client the user channel goes down. We are thinking about we might need a channel variable containing the result of the uuid bridge (with the client if it happend) so we could check this in the CHANNEL_HANGUP_COMPLETE event and decide wether the user has to do the administration or the other user. We would like to ask your opinion and advices about this before doing anything. Thanks, Tamas From anita.hall at simmortel.com Tue Mar 13 18:09:55 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Tue, 13 Mar 2012 20:39:55 +0530 Subject: [Freeswitch-users] Check for freeswitch session exists In-Reply-To: References: Message-ID: Does this work ? http://wiki.freeswitch.org/wiki/Event_List#CHANNEL_STATE regards, Anita On Mon, Mar 12, 2012 at 1:10 PM, piyush singhai wrote: > Hello, > > I am using ESL lib for making connection with freeswitch from the > Application. Some time we miss the hangup event in application and instead > make another execute request on the same session, lets say for playback(). > Then we receive CHANNEL_EXECUTE_COMPLETE for application 'park' instead of > playback(). Is this an indication that the session has already ended (in > this case due to hangup by the user)? Otherwise how can we check before > executing any new function that the session is still alive at fresswitch > through ESL? > > > --Piyush > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/9c6ceb20/attachment.html From msc at freeswitch.org Tue Mar 13 18:50:39 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 13 Mar 2012 08:50:39 -0700 Subject: [Freeswitch-users] make[1]: *** [mod_shout-install] Error 1 In-Reply-To: References: Message-ID: It looks like the patch utility wasn't found or isn't installed on your system. Try doing a yum search on "patch" and installing it. -MC 2012/3/13 ????????? ?????? > I have FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 17-48-30 > -0600) on Centos 5.5**** > > ** ** > > From src directory I run command first time**** > > ** ** > > [root at FS-A freeswitch]# make mod_shout-install**** > > ...**** > > --2012-03-11 13:59:40-- > http://files.freeswitch.org/downloads/libs/libshout-2.2.2.tar.gz**** > > Resolving files.freeswitch.org... 67.201.31.224**** > > Connecting to files.freeswitch.org|67.201.31.224|:80... connected.**** > > HTTP request sent, awaiting response... 200 OK**** > > Length: 478582 (467K) [application/x-gzip]**** > > Saving to: `libshout-2.2.2.tar.gz'**** > > ** ** > > 100%[==========================================================================================================================>] > 478,582 969K/s in 0.5s**** > > ** ** > > 2012-03-11 13:59:40 (969 KB/s) - `libshout-2.2.2.tar.gz' saved > [478582/478582]**** > > ** ** > > /bin/sh: patch: command not found**** > > make[3]: *** [/usr/local/src/freeswitch/libs/libshout-2.2.2] Error 127**** > > make[2]: *** [install] Error 1**** > > make[1]: *** [mod_shout-install] Error 1**** > > make: *** [mod_shout-install] Error 2**** > > [root at FS-A freeswitch]#**** > > ** ** > > second time **** > > ** ** > > [root at FS-A freeswitch]# make mod_shout-install**** > > ...**** > > Making all in doc**** > > Compiling > /usr/local/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c...**** > > mkdir .libs**** > > Compiling /usr/local/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c > ...**** > > Creating mod_shout.so...**** > > installing mod_shout.so**** > > [root at FS-A freeswitch]# **** > > ** ** > > All ok**** > > ** ** > > Vladislav Grishin**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/48cba703/attachment.html From msc at freeswitch.org Tue Mar 13 19:17:13 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 13 Mar 2012 09:17:13 -0700 Subject: [Freeswitch-users] user dial-string getting variables In-Reply-To: <1331061712.3526.55.camel@marces.madrid.commsmundi.com> References: <1331061712.3526.55.camel@marces.madrid.commsmundi.com> Message-ID: Sorry for the late reply... Did you figure this one out yet? I think you can set the uuid of the b-leg to a known value and then do a uuid_dump (or whatever) to check stuff on that uuid: You will have ${new_uuid} on the A leg which contains the B leg uuid. I suppose you could also skip the create_uuid/origination_uuid thing and just use the value in ${signal_bond}. However, personally, I like creating the uuid beforehand and knowing what the B leg's uuid will be. In any case, you can do a uuid_dump XXXX where XXXX is whatever is in ${new_uuid} (or in ${signal_bond}) Let us know what happens. -MC On Tue, Mar 6, 2012 at 11:21 AM, Antonio wrote: > ** > Hi, > > Is it possible to access a variable that is passed per channel, in some > script called in the directory.xml, parameter dial-string? > > I need to be able to set the var1 dynamically from the dialplan, so i can > parse it from my lua script. > > Using the set command directly it will set the variables for both channels. > > > For example, I dialed two users: > > > > > > > > then in my directory.xml, i have: > > > > > So when the return dial-string for the user 100 has it's own var1, and > when dial the user 102 has another value. > > I already try to uuid_dump in dial.lua, but i can't see the variable > var1... > > Using this method, [var1=value], it will set the variable var1 for the B > channel, but can i access it from the A channel? Or is there another way > to set this variables before processing the dial-string... > > Thanks, > Ant?nio > > > -- > > Un cordial saludo / Best regards, > > _________________________ > > Ant?nio Silva > > E-mail:asilva at wirelessmundi.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/e1bb5de6/attachment-0001.html From adam.kelloway at newpace.ca Tue Mar 13 19:30:35 2012 From: adam.kelloway at newpace.ca (Adam Kelloway) Date: Tue, 13 Mar 2012 13:30:35 -0300 Subject: [Freeswitch-users] Dialplan bridge results in choppy audio Message-ID: <4F5F762B.9000402@newpace.ca> Hi there, Using the dialplan tool, I am trying to bridge a call on one freeswitch instance to another freeswitch instance on the same network. The call is bridged as expected, but the audio is very choppy. The first few seconds is basically inaudible, and after that the quality is better but still quite poor. In searching previous threads, I could only find audio problems related to using loopback, but nothing for the bridge dialplan tool. Also note that the ptime is 20 on both legs. Am I missing something here? Is there another way I should be transferring a call to another freeswitch instance? Here is what the bridge looks like: EXECUTE sofia/profile_1_external/userA at hostA.com bridge(sofia/internal/userB at hostB) Both instances are running 1.0.head-git-72bb196 2012-03-12 16-07-56 -0700 Thanks, Adam From peter.olsson at visionutveckling.se Tue Mar 13 19:38:14 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 13 Mar 2012 16:38:14 +0000 Subject: [Freeswitch-users] Dialplan bridge results in choppy audio Message-ID: <1FFF97C269757C458224B7C895F35F150522CE@cantor.std.visionutv.se> Are you running on real hardware, or virtual machines? /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Adam Kelloway Skickat: den 13 mars 2012 17:31 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Dialplan bridge results in choppy audio Hi there, Using the dialplan tool, I am trying to bridge a call on one freeswitch instance to another freeswitch instance on the same network. The call is bridged as expected, but the audio is very choppy. The first few seconds is basically inaudible, and after that the quality is better but still quite poor. In searching previous threads, I could only find audio problems related to using loopback, but nothing for the bridge dialplan tool. Also note that the ptime is 20 on both legs. Am I missing something here? Is there another way I should be transferring a call to another freeswitch instance? Here is what the bridge looks like: EXECUTE sofia/profile_1_external/userA at hostA.com bridge(sofia/internal/userB at hostB) Both instances are running 1.0.head-git-72bb196 2012-03-12 16-07-56 -0700 Thanks, Adam _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f5f749732761982117328! From Rob.Moore at Aeriandi.com Tue Mar 13 19:43:14 2012 From: Rob.Moore at Aeriandi.com (Rob Moore) Date: Tue, 13 Mar 2012 16:43:14 +0000 Subject: [Freeswitch-users] SIP Invite Contact Header In-Reply-To: References: <49C5FCA19A8A114493EBAACA42FE5899104EB9A7@1AERDCEXCHMBX1.AER.AERCO.local> Message-ID: <49C5FCA19A8A114493EBAACA42FE5899104EF0D9@1AERDCEXCHMBX1.AER.AERCO.local> Thanks Gabe, It turns out we had some previous configuration that was overriding it and simply setting was enough to change both from and contact. Thanks again for your help, got me looking in the right direction -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel Gunderson Sent: 10 March 2012 22:57 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP Invite Contact Header On Wed, Mar 7, 2012 at 8:58 AM, Rob Moore wrote: > I've tried setting sip_contact_user but that doesn't seem to have any > effect on the contents of the contact header. Plus I would ?really > prefer to set this in the gateway profile so it only effects calls > passing over a single gateway. I see these additional variables: sip_contact_params sip_contact_user sip_contact_port sip_contact_uri sip_contact_host If you see this page on the wiki, it shows setting contact info in the GW: http://wiki.freeswitch.org/wiki/Clarification:gateways#conf.2Fsip_profiles.2Fexternal.2Fexample.xml Don't know if it works, but hope those two things get you going. Gabe _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From Rob.Moore at Aeriandi.com Tue Mar 13 19:57:22 2012 From: Rob.Moore at Aeriandi.com (Rob Moore) Date: Tue, 13 Mar 2012 16:57:22 +0000 Subject: [Freeswitch-users] Remove INFO and MESSAGE from SIP Invite Header Message-ID: <49C5FCA19A8A114493EBAACA42FE5899104EF0F4@1AERDCEXCHMBX1.AER.AERCO.local> Hi Guys, I'm looking for a way to remove INFO and MESSAGE from the SIP Invite message freeswitch generates when I attempt to use a certain gateway. The system used by one of our providers to provide a TLS solution doesn't accept these methods and is rejecting a call. I've tried to remove INFO by setting within the gateway configuration and attempted to disable message presence ( ) to remove the MESSAGE method request but neither seem to have any effect. Does anyone have any suggestions on how I can remove these? Thanks Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/d4bbc5e0/attachment.html From xing2kin at yahoo.com Tue Mar 13 20:11:36 2012 From: xing2kin at yahoo.com (king2kin) Date: Tue, 13 Mar 2012 10:11:36 -0700 (PDT) Subject: [Freeswitch-users] Failed to pass arguments to lua script of FreeSwitch ('arg' and 'arg[i]' are nil values) In-Reply-To: <1331633570.21039.35.camel@marces.madrid.commsmundi.com> References: <1331631987.12002.YahooMailNeo@web162802.mail.bf1.yahoo.com> <1331633570.21039.35.camel@marces.madrid.commsmundi.com> Message-ID: <1331658696.43848.YahooMailNeo@web162805.mail.bf1.yahoo.com> Thank you very much for your help! it now works.????? From: Antonio To: FreeSWITCH Users Help Sent: Tuesday, March 13, 2012 6:12 PM Subject: Re: [Freeswitch-users] Failed to pass arguments to lua script of FreeSwitch ('arg' and 'arg[i]' are nil values) Hi,Your? missing an "V" in the arg name, "argv".You can try do the following:in diaplan:in the lua:var1 = argv[1] or "default value to avoid errors"var2 = argv[1] or "default value var2"freeswitch.consoleLog("info", tostring(var1) .. "\n")freeswitch.consoleLog("info", tostring(var2) .. "\n")you can also call the lua script directly from the cli:fs:> luarun yourcript.lua var1 var2Regards,Ant?nioOn Tue, 2012-03-13 at 02:46 -0700, king2kin wrote: Hi all, >? >It's said we could access the arguments passed to lua script by variable 'arg[i]' where i > 0. >? >for example,? { lua dummy.lua? aa 100 } >then, we will have as follows: >arg[1] -> aa >arg[2] -> 100 >? >However, I always get 'nil' values for 'arg[i]' or 'arg' while running the lua script. >? >How can I pass and retrieve arguments or parameters to a lua script of FreeSwitch? >? >? >? >? >? >? >? >? >_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Un cordial saludo / Best regards, ?_________________________ Ant?nio Silva E-mail:asilva at wirelessmundi.com _________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120313/2a50bfcf/attachment.html From lazyvirus at gmx.com Tue Mar 13 22:40:28 2012 From: lazyvirus at gmx.com (Bzzz) Date: Tue, 13 Mar 2012 20:40:28 +0100 Subject: [Freeswitch-users] [SOLVED] Re: connexion attempts behing a firewall (wtf?) In-Reply-To: <20120312153754.5799fa92@anubis.defcon1> References: <1FFF97C269757C458224B7C895F35F150514F0@cantor.std.visionutv.se> <20120312153754.5799fa92@anubis.defcon1> Message-ID: <20120313204028.39d2633e@anubis.defcon1> On Mon, 12 Mar 2012 15:37:54 +0100 Bzzz wrote: The chair/keyboard interface had a flaw: some FusionPBX xmls were missing as the installation was manual. JY -- I'd like to meet the man who invented sex and see what he's working on now. From faisal.rehman22 at hotmail.com Wed Mar 14 09:40:04 2012 From: faisal.rehman22 at hotmail.com (Faisal Rehman) Date: Wed, 14 Mar 2012 11:40:04 +0500 Subject: [Freeswitch-users] Freeswitch replies with dynamic payload Message-ID: Hi Everyone, We are working on scenario where Freeswitch is terminating calls a vendor with g729 passthru mode. But when we get the call terminated , termination side replies with static payload but freeswitch replies with Dynamic payload. Our customer wants to be replied with only static payload. How can we solve this? Situation: Originator -> freeswitch -> termination In session progress : Termination side sends, m=audio 64034 RTP/AVP 18 101^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M Freeswitch sends following to origination. m=audio 0 RTP/AVP 96 101 a=rtpmap:96 G729/8000 a=fmtp:96 annexb=no Configuration File external.xml: for presence. --> Regards, Faisal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/c23b73e2/attachment.html From danishmoosa at gmail.com Wed Mar 14 10:13:32 2012 From: danishmoosa at gmail.com (Muhammad Danish Moosa) Date: Wed, 14 Mar 2012 12:13:32 +0500 Subject: [Freeswitch-users] Freeswitch replies with dynamic payload In-Reply-To: References: Message-ID: Hi This was interesting,my understanding was that SDP is(or SHOULD be) never touched in case of disabled transcoding and enabled bypass media. It might be something related to negotiation settings. On Wed, Mar 14, 2012 at 11:40 AM, Faisal Rehman wrote: > *Hi Everyone,* > > > We are working on scenario where Freeswitch is terminating calls a vendor > with g729 passthru mode. But when we get the call terminated , termination > side replies with static payload but freeswitch replies with *Dynamic*payload. Our customer wants to be replied with only static payload. How can > we solve this? > > *Situation:* > > Originator -> freeswitch -> termination > > *In session progress :* > > Termination side sends, > > m=audio 64034 RTP/AVP 18 101^M > a=rtpmap:18 G729/8000^M > a=fmtp:18 annexb=no^M > > Freeswitch sends following to origination. > > m=audio 0 RTP/AVP 96 101 > a=rtpmap:96 G729/8000 > a=fmtp:96 annexb=no > > > *Configuration File* > * > * > * > * > *external.xml:* > > > > > > > > > > > > > > > > > > > > > > > > > > > > > for presence. > --> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Regards, > > *Faisal * > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/ca9c0cf8/attachment-0001.html From Vladislav.Grishin at vts24.ru Wed Mar 14 10:35:39 2012 From: Vladislav.Grishin at vts24.ru (=?koi8-r?B?98zBxMnTzMHXIOfSydvJzg==?=) Date: Wed, 14 Mar 2012 11:35:39 +0400 Subject: [Freeswitch-users] make[1]: *** [mod_shout-install] Error 1 In-Reply-To: References: Message-ID: <1A96149B6DD14AF7BEA7A85F0B52F149@mservice.local> After I installed the "patch", successfully works from the first. Vladislav Grishin _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, March 13, 2012 7:51 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] make[1]: *** [mod_shout-install] Error 1 After I installed the program team successfully works from the first. It looks like the patch utility wasn't found or isn't installed on your system. Try doing a yum search on "patch" and installing it. -MC 2012/3/13 ????????? ?????? I have FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 17-48-30 -0600) on Centos 5.5 >From src directory I run command first time [root at FS-A freeswitch]# make mod_shout-install : --2012-03-11 13:59:40-- http://files.freeswitch.org/downloads/libs/libshout-2.2.2.tar.gz Resolving files.freeswitch.org... 67.201.31.224 Connecting to files.freeswitch.org|67.201.31.224|:80... connected. HTTP request sent, awaiting response... 200 OK Length: 478582 (467K) [application/x-gzip] Saving to: `libshout-2.2.2.tar.gz' 100%[======================================================================= ===================================================>] 478,582 969K/s in 0.5s 2012-03-11 13:59:40 (969 KB/s) - `libshout-2.2.2.tar.gz' saved [478582/478582] /bin/sh: patch: command not found make[3]: *** [/usr/local/src/freeswitch/libs/libshout-2.2.2] Error 127 make[2]: *** [install] Error 1 make[1]: *** [mod_shout-install] Error 1 make: *** [mod_shout-install] Error 2 [root at FS-A freeswitch]# second time [root at FS-A freeswitch]# make mod_shout-install : Making all in doc Compiling /usr/local/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c... mkdir .libs Compiling /usr/local/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c ... Creating mod_shout.so... installing mod_shout.so [root at FS-A freeswitch]# All ok Vladislav Grishin _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/1f7f3b1c/attachment.html From msc at freeswitch.org Wed Mar 14 10:45:28 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Mar 2012 00:45:28 -0700 Subject: [Freeswitch-users] stip variable In-Reply-To: <4F5EF9C5.6020202@softnet.si> References: <4F588FE5.3090402@softnet.si> <73D50A0D-B8C3-4EB2-91A5-6B2C76C82907@freeswitch.org> <4F5EF9C5.6020202@softnet.si> Message-ID: Try this: Also, you may need 'inline="true"' depending on when you need the value that ends up in ${my_number}. -MC On Tue, Mar 13, 2012 at 12:39 AM, Miha Zoubek wrote: > Hi @Brian, > > I was looking at Enum.conf but having problem with modifying it. > > > > I can remove sofia/internal/ and Enum will return just > 18005551212 at tf.voipmich.com. > > What is the best why to get rid of @tf.voipmich.com as I need only number? > > I was trying to trim it like exp. {18005551212 at tf.voipmich.com:10:10} but > this is not working properly as length of number change. > > Thanks! > > Regards, > Miha > > > > > > > On 03/08/2012 04:10 PM, Brian West wrote: > > or modify enum.conf to only give you the numer? > > /b > > On Mar 8, 2012, at 4:54 AM, Miha Zoubek wrote: > > Hi, > > how can I strip this sofia/internal/18005551212 at tf.voipmich.com, so that > I would have only number 18005551212? > > This return enum. > > > Thanks! > Miha > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/5136702f/attachment-0001.html From miha at softnet.si Wed Mar 14 11:31:47 2012 From: miha at softnet.si (Miha) Date: Wed, 14 Mar 2012 09:31:47 +0100 Subject: [Freeswitch-users] stip variable In-Reply-To: References: <4F588FE5.3090402@softnet.si> <73D50A0D-B8C3-4EB2-91A5-6B2C76C82907@freeswitch.org> <4F5EF9C5.6020202@softnet.si> Message-ID: <4F605773.8030202@softnet.si> On 03/14/2012 08:45 AM, Michael Collins wrote: > Try this: > > > > Also, you may need 'inline="true"' depending on when you need the > value that ends up in ${my_number}. > > -MC > > On Tue, Mar 13, 2012 at 12:39 AM, Miha Zoubek > wrote: > > Hi @Brian, > > I was looking at Enum.conf but having problem with modifying it. > > replace="sofia/internal/$1"/> > > I can remove sofia/internal/ and Enum will return just > 18005551212 at tf.voipmich.com . > > What is the best why to get rid of @tf.voipmich.com > as I need only number? > > I was trying to trim it like exp. > {18005551212 at tf.voipmich.com:10:10 > } but this is not > working properly as length of number change. > > Thanks! > > Regards, > Miha > > > > > > > On 03/08/2012 04:10 PM, Brian West wrote: >> or modify enum.conf to only give you the numer? >> >> /b >> >> On Mar 8, 2012, at 4:54 AM, Miha Zoubek wrote: >> >>> Hi, >>> >>> how can I strip thissofia/internal/18005551212 at tf.voipmich.com >>> , so that I >>> would have only number 18005551212 ? >>> >>> This return enum. >>> >>> >>> Thanks! >>> Miha >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org @MC, thanks! IT works:) Regards, Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/cfd541ec/attachment.html From msc at freeswitch.org Wed Mar 14 15:24:11 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Mar 2012 05:24:11 -0700 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Message-ID: Hello all, Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_14 Ken Rice will be discussing the feature freeze and versioning. Please join the discussion to learn more about how you can help the project get up to version 1.2 as quickly as possible! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/a9e8d576/attachment.html From msc at freeswitch.org Wed Mar 14 15:37:24 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Mar 2012 05:37:24 -0700 Subject: [Freeswitch-users] Freeswitch replies with dynamic payload In-Reply-To: References: Message-ID: On Wed, Mar 14, 2012 at 12:13 AM, Muhammad Danish Moosa < danishmoosa at gmail.com> wrote: > Hi > > This was interesting,my understanding was that SDP is(or SHOULD be) never > touched in case of disabled transcoding and enabled bypass media. It might > be something related to negotiation settings. > That was my understanding as well. I recommend that you get a SIP trace of the traffic to/from the destination and see if the destination is what is doing the dynamic payload thing. Usually FreeSWITCH just passes through what it receives since it's a B2BUA... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/04e65ac3/attachment.html From adam.kelloway at newpace.ca Wed Mar 14 15:42:18 2012 From: adam.kelloway at newpace.ca (Adam Kelloway) Date: Wed, 14 Mar 2012 09:42:18 -0300 Subject: [Freeswitch-users] Dialplan bridge results in choppy audio In-Reply-To: <1FFF97C269757C458224B7C895F35F150522CE@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F150522CE@cantor.std.visionutv.se> Message-ID: <4F60922A.8080804@newpace.ca> These are virtual machines. Would this have a negative effect? On 3:59 PM, Peter Olsson wrote: > Are you running on real hardware, or virtual machines? > > /Peter > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Adam Kelloway > Skickat: den 13 mars 2012 17:31 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] Dialplan bridge results in choppy audio > > Hi there, > > Using the dialplan tool, I am trying to bridge a call on one freeswitch instance to another freeswitch instance on the same network. The call is bridged as expected, but the audio is very choppy. The first few seconds is basically inaudible, and after that the quality is better but still quite poor. In searching previous threads, I could only find audio problems related to using loopback, but nothing for the bridge dialplan tool. Also note that the ptime is 20 on both legs. > > Am I missing something here? Is there another way I should be transferring a call to another freeswitch instance? > > Here is what the bridge looks like: > > EXECUTE sofia/profile_1_external/userA at hostA.com > bridge(sofia/internal/userB at hostB) > > Both instances are running 1.0.head-git-72bb196 2012-03-12 16-07-56 -0700 > > Thanks, > > Adam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f5f749732761982117328! > > > -- Adam -- NewPace Logo Adam Kelloway Software Engineer, NewPace phone +1 (902) 406--8375 x1031 email Adam.Kelloway at NewPace.com aim /msn Adam.Kelloway @NewPace.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/c800e330/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Newpace_50x50.png Type: image/png Size: 4620 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/c800e330/attachment-0001.png From avi at avimarcus.net Wed Mar 14 15:48:08 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 14 Mar 2012 14:48:08 +0200 Subject: [Freeswitch-users] Dialplan bridge results in choppy audio In-Reply-To: <4F60922A.8080804@newpace.ca> References: <1FFF97C269757C458224B7C895F35F150522CE@cantor.std.visionutv.se> <4F60922A.8080804@newpace.ca> Message-ID: Yep. When you are virtualized, both the timer that handles the media is affected, and having the processing power to handle the media it at the mercy of the other processes. OpenVZ supposedly provides the best timing.. or in xen 1000 mhz kernals.. but still... if you have a performance issue with something like this, it's probably due to the virtualization. Try it bare-metal and see if you have the same problem. Is there much load on this box? -Avi On Wed, Mar 14, 2012 at 2:42 PM, Adam Kelloway wrote: > ** > These are virtual machines. Would this have a negative effect? > > On 3:59 PM, Peter Olsson wrote: > > Are you running on real hardware, or virtual machines? > > /Peter > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] F?r Adam Kelloway > Skickat: den 13 mars 2012 17:31 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] Dialplan bridge results in choppy audio > > Hi there, > > Using the dialplan tool, I am trying to bridge a call on one freeswitch instance to another freeswitch instance on the same network. The call is bridged as expected, but the audio is very choppy. The first few seconds is basically inaudible, and after that the quality is better but still quite poor. In searching previous threads, I could only find audio problems related to using loopback, but nothing for the bridge dialplan tool. Also note that the ptime is 20 on both legs. > > Am I missing something here? Is there another way I should be transferring a call to another freeswitch instance? > > Here is what the bridge looks like: > > EXECUTE sofia/profile_1_external/userA at hostA.com > bridge(sofia/internal/userB at hostB) > > Both instances are running 1.0.head-git-72bb196 2012-03-12 16-07-56 -0700 > > Thanks, > > Adam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > !DSPAM:4f5f749732761982117328! > > > > > > -- > Adam > -- > [image: NewPace Logo] > > > Adam Kelloway > Software Engineer, NewPace phone +1 (902) 406?8375 x1031 email > Adam.Kelloway at NewPace.com aim/msn > Adam.Kelloway at NewPace.ca > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/c074e930/attachment.html From peter.olsson at visionutveckling.se Wed Mar 14 15:56:22 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 14 Mar 2012 12:56:22 +0000 Subject: [Freeswitch-users] Dialplan bridge results in choppy audio Message-ID: <1FFF97C269757C458224B7C895F35F15052A5F@cantor.std.visionutv.se> Yeah - that's the reason for bad audio. If you want to be sure - never use virtual stuff, at least not for production systems. The main issue is that timing in virtualized environments are not good enough. If you Goggle it, you will find loads of information about this. Much related to Asterisk, but it's the same problem for all VoIP systems. /Peter Fr?n: Adam Kelloway [mailto:adam.kelloway at newpace.ca] Skickat: den 14 mars 2012 13:42 Till: FreeSWITCH Users Help Kopia: Peter Olsson ?mne: Re: Re: [Freeswitch-users] Dialplan bridge results in choppy audio These are virtual machines. Would this have a negative effect? On 3:59 PM, Peter Olsson wrote: Are you running on real hardware, or virtual machines? /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Adam Kelloway Skickat: den 13 mars 2012 17:31 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Dialplan bridge results in choppy audio Hi there, Using the dialplan tool, I am trying to bridge a call on one freeswitch instance to another freeswitch instance on the same network. The call is bridged as expected, but the audio is very choppy. The first few seconds is basically inaudible, and after that the quality is better but still quite poor. In searching previous threads, I could only find audio problems related to using loopback, but nothing for the bridge dialplan tool. Also note that the ptime is 20 on both legs. Am I missing something here? Is there another way I should be transferring a call to another freeswitch instance? Here is what the bridge looks like: EXECUTE sofia/profile_1_external/userA at hostA.com bridge(sofia/internal/userB at hostB) Both instances are running 1.0.head-git-72bb196 2012-03-12 16-07-56 -0700 Thanks, Adam _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Adam -- [NewPace Logo] Adam Kelloway Software Engineer, NewPace phone +1 (902) 406-8375 x1031 email Adam.Kelloway at NewPace.com aim/msn Adam.Kelloway@NewPace.ca !DSPAM:4f608f8332762193319106! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/45fef835/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 4620 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/45fef835/attachment-0001.png From lists at telefaks.de Wed Mar 14 16:56:20 2012 From: lists at telefaks.de (Peter Steinbach) Date: Wed, 14 Mar 2012 14:56:20 +0100 Subject: [Freeswitch-users] Stuck Calls In-Reply-To: <07BF4904977CC645B485E970424193AD0FF1109F8B@localhost> References: <07BF4904977CC645B485E970424193AD0FF1109F8B@localhost> Message-ID: <4F60A384.7070003@telefaks.de> On our system, some channels were still present in the channel table of the core database after uuid_kill. After deleting those uuid entries in the channel table, all was fine. Best regards Peter Am 13.03.2012 11:39, schrieb B.Tietz at pinguin.ag: > > Hi, > > I have some stuck calls in the list when I issue "show detailed_calls" > > Trying to kill them with "uuid_kill " or "fsctl hupall > normal_clearing dialed_ext 1234" doesn't help. > > FS is configured as HA-node with odbc in the core and sofia. > > Any other suggestions?! > > Benjamin > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/6f497666/attachment.html From lazyvirus at gmx.com Wed Mar 14 16:58:36 2012 From: lazyvirus at gmx.com (Bzzz) Date: Wed, 14 Mar 2012 14:58:36 +0100 Subject: [Freeswitch-users] SIPS & SRTP questions Message-ID: <20120314145836.45391515@anubis.defcon1> Hi list, I'm running a FS server @home to create a small network of people that need confidentiality upon their calls/conferences. I'm reading the SIPS/SRTP FS wiki page at this moment and I've got some questions: * I've a dyndns WAN name for my adsl box (that I loop on my DNS to the server LAN's name: nslookup extname.org => 192.168.1.50); so I suppose I must have the WAN name as the 'cn' in my cert/key for external clients being able to connect? (but what about internal ones?) * If I'm not using auto-nat, is forwarding port 5081 from the WAN to the server sufficient for making and receiving external calls? * BTW, SIP 5060 is supposed to be internal, so why is it also part of the uPNP forwards? * How can I force FS to only work in SSLv23 + SRTP modes? (how?) * Is there a possibility for each user to have its own certificate, so I would be able to revoke permissions atomically if needed? Jean-Yves -- We are Pentium of Borg. Division is futile. You will be approximated. (seen in someone's .signature) From peter.olsson at visionutveckling.se Wed Mar 14 17:05:01 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 14 Mar 2012 14:05:01 +0000 Subject: [Freeswitch-users] Stuck Calls Message-ID: <1FFF97C269757C458224B7C895F35F15052B69@cantor.std.visionutv.se> Are you running latest git? /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Peter Steinbach Skickat: den 14 mars 2012 14:56 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Stuck Calls On our system, some channels were still present in the channel table of the core database after uuid_kill. After deleting those uuid entries in the channel table, all was fine. Best regards Peter Am 13.03.2012 11:39, schrieb B.Tietz at pinguin.ag: Hi, I have some stuck calls in the list when I issue "show detailed_calls" Trying to kill them with "uuid_kill " or "fsctl hupall normal_clearing dialed_ext 1234" doesn't help. FS is configured as HA-node with odbc in the core and sofia. Any other suggestions?! Benjamin _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de !DSPAM:4f60a29232761811354587! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/f4c91c57/attachment.html From adam.kelloway at newpace.ca Wed Mar 14 17:14:33 2012 From: adam.kelloway at newpace.ca (Adam Kelloway) Date: Wed, 14 Mar 2012 11:14:33 -0300 Subject: [Freeswitch-users] Dialplan bridge results in choppy audio In-Reply-To: References: <1FFF97C269757C458224B7C895F35F150522CE@cantor.std.visionutv.se> <4F60922A.8080804@newpace.ca> Message-ID: <4F60A7C9.50600@newpace.ca> Thank you both for the clarification. Both of the VMs I have freeswitch on are run from the same box, which also runs a number of other VMs with other applications. This is not a production environment, however, so there is virtually no 'call load'. I will see if I can test it out on real hardware, thanks very much. Adam On 3:59 PM, Avi Marcus wrote: > Yep. When you are virtualized, both the timer that handles the media > is affected, and having the processing power to handle the media it at > the mercy of the other processes. > OpenVZ supposedly provides the best timing.. or in xen 1000 mhz > kernals.. but still... if you have a performance issue with something > like this, it's probably due to the virtualization. > > Try it bare-metal and see if you have the same problem. > > Is there much load on this box? > > -Avi > > > > On Wed, Mar 14, 2012 at 2:42 PM, Adam Kelloway > > wrote: > > These are virtual machines. Would this have a negative effect? > > On 3:59 PM, Peter Olsson wrote: >> Are you running on real hardware, or virtual machines? >> >> /Peter >> >> -----Ursprungligt meddelande----- >> Fr?n:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Adam Kelloway >> Skickat: den 13 mars 2012 17:31 >> Till: FreeSWITCH Users Help >> ?mne: [Freeswitch-users] Dialplan bridge results in choppy audio >> >> Hi there, >> >> Using the dialplan tool, I am trying to bridge a call on one freeswitch instance to another freeswitch instance on the same network. The call is bridged as expected, but the audio is very choppy. The first few seconds is basically inaudible, and after that the quality is better but still quite poor. In searching previous threads, I could only find audio problems related to using loopback, but nothing for the bridge dialplan tool. Also note that the ptime is 20 on both legs. >> >> Am I missing something here? Is there another way I should be transferring a call to another freeswitch instance? >> >> Here is what the bridge looks like: >> >> EXECUTEsofia/profile_1_external/userA at hostA.com >> bridge(sofia/internal/userB at hostB) >> >> Both instances are running 1.0.head-git-72bb196 2012-03-12 16-07-56 -0700 >> >> Thanks, >> >> Adam >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> !DSPAM:4f5f749732761982117328! >> >> >> > > -- > Adam > -- > NewPace Logo > > > Adam Kelloway > > Software Engineer, NewPace > phone +1 (902) 406--8375 x1031 > > email Adam.Kelloway at NewPace.com > aim/msn Adam.Kelloway at NewPace.ca > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Adam -- NewPace Logo Adam Kelloway Software Engineer, NewPace phone +1 (902) 406--8375 x1031 email Adam.Kelloway at NewPace.com aim /msn Adam.Kelloway @NewPace.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/f2500821/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Newpace_50x50.png Type: image/png Size: 4620 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/f2500821/attachment-0001.png From kris at kriskinc.com Wed Mar 14 17:15:25 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 14 Mar 2012 10:15:25 -0400 Subject: [Freeswitch-users] Freeswitch replies with dynamic payload In-Reply-To: References: Message-ID: Please post a full SIP trace with console output. It looks like you have bypass_media turned on. More than likely that broken reply SDP is coming from another device, not FreeSWITCH. On Wed, Mar 14, 2012 at 2:40 AM, Faisal Rehman wrote: > Hi Everyone, > > > We are working on scenario where Freeswitch is terminating calls a vendor > with g729 passthru mode. But when we get the call terminated , termination > side replies with static payload but freeswitch replies with Dynamic > payload. Our customer wants to be replied with only static payload. How can > we solve this? > > Situation: > > Originator? -> freeswitch -> termination > > In session progress : > > Termination side sends, > > m=audio 64034 RTP/AVP 18 101^M > a=rtpmap:18 G729/8000^M > a=fmtp:18 annexb=no^M > > Freeswitch sends following to origination. > > m=audio 0 RTP/AVP 96 101 > a=rtpmap:96 G729/8000 > a=fmtp:96 annexb=no > > > Configuration File > > > external.xml: > > > ? > ??? > ??????? > ??????? > ??? > ??? > ??? > ??? > ??? > ??? > > > ? > ???????? > > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ?????? for presence. > ??? --> > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ??? > ? > > > > Regards, > > Faisal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From mitch.capper at gmail.com Wed Mar 14 18:24:39 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Wed, 14 Mar 2012 08:24:39 -0700 Subject: [Freeswitch-users] SIPS & SRTP questions In-Reply-To: <20120314145836.45391515@anubis.defcon1> References: <20120314145836.45391515@anubis.defcon1> Message-ID: Hello Jean-Yves, well common-name matching is not required by all sip clients but may be by some of yours. Yes you would want to set this set as the cn name. Assuming your FS box is run on your adsl box (or the important ports are forwarded to your FS box) you shouldn't run into an issue as you already handle the natted client problem by faking the dns entry for your cn name. FS should be able to handle both external and internal clients without an issue. While you can certainly issue certificates for each client (Infact its encouraged!) see http://wiki.freeswitch.org/wiki/SIP_TLS#Step_4_Client_Configuration for generating individual client certs easily (although make sure you are running HEAD), there may be an issue with revocation. I am not sure if revocation is currently enabled(or supported) in the libsofia stack. Test it and see, I would bet it isn't. As for getting it enabled first would be to verify its supported by libsofia first, checking the documentation at http://sofia-sip.sourceforge.net/ specifically http://sofia-sip.sourceforge.net/refdocs/tport/tport__tag_8h.html may be a good place to start. If it supports revocation but we don't expose it then it may be an easy change, if it doesn't its going to be a bit more of an uphill battle as you will have to patch it and our sofia to add the option for revocation checks. If you post back with your results I may be willing to help with the work to add this additional security feature (assuming it doesn't already work). Normally I would guess that you would actually have everyone connecting on 5061 (or whatever your tls port is) that are authed users. 5060 is meant for users who auth against the server vs 5080 being more of the public /outbound side of the server is somewhat of how I have always looked at it. As for forcing SSLv23 and SRTP thats pretty straightforward: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#TLS documents the sofia options including: tls-only which will prevent sofia from even listening for un-encrypted connections. Settings tls-version to the sslv23 takes care of what version to use, the final is just how to ensure all calls are encrypted. As your clients can only connect encrypted with tls-only it takes care of ensuring the signalling channel is encrypted, to ensure SRTP just add sip_secure_media=true to the channel vars. ~Mitch From lazyvirus at gmx.com Wed Mar 14 19:06:52 2012 From: lazyvirus at gmx.com (Bzzz) Date: Wed, 14 Mar 2012 17:06:52 +0100 Subject: [Freeswitch-users] SIPS & SRTP questions In-Reply-To: References: <20120314145836.45391515@anubis.defcon1> Message-ID: <20120314170652.3d631175@anubis.defcon1> On Wed, 14 Mar 2012 08:24:39 -0700 Mitch Capper wrote: Hi Mitch, > > and internal clients without an issue. While you can certainly issue > certificates for each client (Infact its encouraged!) see > http://wiki.freeswitch.org/wiki/SIP_TLS#Step_4_Client_Configuration Oops, I sent my email before reaching this text. > for generating individual client certs easily (although make sure you > are running HEAD), there may be an issue with revocation. I am not > sure if revocation is currently enabled(or supported) in the libsofia > stack. Test it and see, I would bet it isn't. As for getting it > enabled first would be to verify its supported by libsofia first, > checking the documentation at http://sofia-sip.sourceforge.net/ > specifically http://sofia-sip.sourceforge.net/refdocs/tport/tport__tag_8h.html Thanks for this links; unfortunately I didn't find any primitive nor comment that address this issue, and a: grep -Ri revo * in the source tree didn't returned any results. This is weird as it force to re-generate another couple crt/key and distribute it to non-revoked users :( > > As for forcing SSLv23 and SRTP thats pretty straightforward: > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#TLS Thank for the link. > documents the sofia options including: > tls-only which will prevent sofia from even listening for un-encrypted > connections. Settings tls-version to the sslv23 takes care of what > version to use, the final is just how to ensure all calls are > encrypted. As your clients can only connect encrypted with tls-only > it takes care of ensuring the signalling channel is encrypted, > to > ensure SRTP just add sip_secure_media=true to the channel vars. That's exactly what I need:) JY -- From mitch.capper at gmail.com Wed Mar 14 19:34:15 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Wed, 14 Mar 2012 09:34:15 -0700 Subject: [Freeswitch-users] SIPS & SRTP questions In-Reply-To: <20120314170652.3d631175@anubis.defcon1> References: <20120314145836.45391515@anubis.defcon1> <20120314170652.3d631175@anubis.defcon1> Message-ID: JY, Keep in mind it may not quite be that simple as libsofia doesn't actually implement the ssl features it uses openssl, so the question is how does one enable revocation in openssl and is libsofia doing it:) It could be something like setting a CRL or the index or what have you so just grepping for revocation might not do it:) One other option is you could instead of using the CA cert have all the client certs catted together then if you want to revoke one of them you pull it out. Its clumsy but works right now:) ~Mitch From lazyvirus at gmx.com Wed Mar 14 20:14:19 2012 From: lazyvirus at gmx.com (Bzzz) Date: Wed, 14 Mar 2012 18:14:19 +0100 Subject: [Freeswitch-users] SIPS & SRTP questions In-Reply-To: References: <20120314145836.45391515@anubis.defcon1> <20120314170652.3d631175@anubis.defcon1> Message-ID: <20120314181419.181bc26b@anubis.defcon1> On Wed, 14 Mar 2012 09:34:15 -0700 Mitch Capper wrote: > One other > option is you could instead of using the CA cert have all the client > certs catted together then if you want to revoke one of them you pull > it out. Its clumsy but works right now:) This is a bit obscur to me Mitch, so I try to reformulate: Do you mean getting rid of the CA (the *whole* conf/ssl/CA?) and keeping only the clients certs in conf/ssl ? Without changing anything to the regular SIPS/SRTP conf? JY -- From mstockton at harqen.com Wed Mar 14 20:26:39 2012 From: mstockton at harqen.com (Matt Stockton) Date: Wed, 14 Mar 2012 10:26:39 -0700 Subject: [Freeswitch-users] Getting real-time amplitude of a channel Message-ID: Hi all, I am interested in providing a visualization of a channel in a conference. I am able to process the start talking and stop talking events from the conference, but I'm wondering, has anyone done anything that can get the real-time amplitude of a channel? I didn't see any built-in API commands to do this, but wondering if there's any modules or something else that I'm overlooking? Or maybe now is the time for me to dig into the API documentation? Just curious if anyone has done something like this before, and if so, if you could guide me in the right direction. Thanks! Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/3f8852fa/attachment.html From mitch.capper at gmail.com Wed Mar 14 21:36:14 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Wed, 14 Mar 2012 11:36:14 -0700 Subject: [Freeswitch-users] SIPS & SRTP questions In-Reply-To: <20120314181419.181bc26b@anubis.defcon1> References: <20120314145836.45391515@anubis.defcon1> <20120314170652.3d631175@anubis.defcon1> <20120314181419.181bc26b@anubis.defcon1> Message-ID: I believe if you replace cafile.pem with the contents of all the clients certs it will ensure that a connecting client cert matches a cert in that file. ~Mitch From lazyvirus at gmx.com Wed Mar 14 21:41:45 2012 From: lazyvirus at gmx.com (Bzzz) Date: Wed, 14 Mar 2012 19:41:45 +0100 Subject: [Freeswitch-users] SIPS & SRTP questions In-Reply-To: References: <20120314145836.45391515@anubis.defcon1> <20120314170652.3d631175@anubis.defcon1> <20120314181419.181bc26b@anubis.defcon1> Message-ID: <20120314194145.318c3811@anubis.defcon1> On Wed, 14 Mar 2012 11:36:14 -0700 Mitch Capper wrote: > I believe if you replace cafile.pem with the contents of all the > clients certs it will ensure that a connecting client cert matches a > cert in that file. Ha, ok, thanks Mitch I'll try that ASA my conf will be ok with regular way. -- Like winter snow on summer lawn, time past is time gone. From msc at freeswitch.org Wed Mar 14 22:30:35 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Mar 2012 12:30:35 -0700 Subject: [Freeswitch-users] Getting real-time amplitude of a channel In-Reply-To: References: Message-ID: AFAIK there isn't anything pre-rolled that you can use. However, I know that Tony's auto gain control (agc) logic checks the "volume" level coming in on a channel. You might be able to look in mod_conference.c and see where the agc calculates the incoming volume level and write an API that displays that value. I'd start by looking at the function check_agc_levels() and for references to member->agc_volume_in_level. Hopefully that will give you a place to start. -MC On Wed, Mar 14, 2012 at 10:26 AM, Matt Stockton wrote: > Hi all, > > I am interested in providing a visualization of a channel in a conference. > I am able to process the start talking and stop talking events from the > conference, but I'm wondering, has anyone done anything that can get the > real-time amplitude of a channel? I didn't see any built-in API commands to > do this, but wondering if there's any modules or something else that I'm > overlooking? Or maybe now is the time for me to dig into the API > documentation? > > Just curious if anyone has done something like this before, and if so, if > you could guide me in the right direction. > > Thanks! > Matt > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/a3dfda42/attachment.html From manieq at wp.eu Thu Mar 15 00:07:41 2012 From: manieq at wp.eu (Mariusz Czulada) Date: Wed, 14 Mar 2012 22:07:41 +0100 Subject: [Freeswitch-users] Playback images as video Message-ID: <4f61089d26d500.51184566@wp.pl> Hi all, I wonder if it is possible now to "playback" a still image (jpeg/png/tiff) in a video stream? While in IVR a new user could also view the same or similar text as transmited as voice. Regards, Mariusz From potxoka at gmail.com Thu Mar 15 00:18:21 2012 From: potxoka at gmail.com (Anto) Date: Wed, 14 Mar 2012 22:18:21 +0100 Subject: [Freeswitch-users] Codec negotiation with carriers In-Reply-To: References: Message-ID: Hello I have searched previous messages in the list, I consulted the book of FreeSWITCH (which I bought over a year), wiki and so on. I still do not understand how and why in some cases I work. Also I downloaded frontend to consult your code if there was something about this, but still the same. I have several weeks with this question and I can not find it. In the end I decided to spend the gateway to Asterisk, and you at least understand its operation. Thank you very much to all :-) Best regards Anto 2012/3/11 Anto : > Hi > > I still do not find the solution and not really understanding, because > it works:-S > > regards > anto > > 2012/3/7 Anto : >> Hello >> >> Attached file, with the traces of the different tests (with different >> configurations). >> >> http://pastebin.freeswitch.org/18599 >> >> I have read the url that you mentioned, the initial guide FreeSWITCH, >> that of mod_sofia, applications, etc.. I believe that most of the wiki >> (maybe when do not give the solution, read as much documentation is >> worse idea :-S, lock me more). >> >> I made a configuration that works (I have not tested the audio), but >> earlier (before I started "touch" the profiles) if I could talk to a >> physical phone (several times). The problem is that I can not >> understand why it works and sometimes not, and I would like to learn >> :-). Not only do and forget, so I would like to learn and less >> disturbing to the mail list and (maybe in the future) to help other >> newbies like me :-). Thanks ! >> >> Best regards >> Anto >> >> 2012/3/7 Michael Collins : >>> You may want to read up on codec negotiation: >>> http://wiki.freeswitch.org/wiki/Codec_negotiation >>> >>> There are different ways to handle codecs depending on your needs. I'd read >>> that page first and then try out some of the suggestions. If you're still >>> having trouble then I'd recommend getting SIP traces of the traffic and >>> putting them on pastebin.freeswitch.org. The gang here is pretty good at >>> looking over logs and helping with diagnosing problems. :) >>> >>> -MC >>> >>> On Tue, Mar 6, 2012 at 2:30 PM, Anto wrote: >>>> >>>> Hi >>>> >>>> I am following the steps in this direction >>>> "http://wiki.freeswitch.org/wiki/SBC_Setup" and >>>> "http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice", >>>> I reread the whole entire wiki (or so I lack), but do not quite >>>> assimilate or finding the right formula to operate the bridge :-S. >>>> >>>> I captured traffic with ngrep, I enabled sip-trace, console logconsole >>>> 8, etc., but unless the transcoding error (only two of the hundreds of >>>> combinations of settings that I have), I have not seen anything >>>> "weird" :-S >>>> >>>> I have 3 suppliers, each with this codec: >>>> >>>> 1) ? ? ? ? ? 2) ? ? ? ? ? ? ?3) >>>> G729 ? ? ? ?G729 ? ? ? ?G729 >>>> G711u ? ? ?G711A ? ? ?G711A >>>> G711A ? ? G711u ? ? ? G711u >>>> ? ? ? ? ? ? ? ?G723 ? ? ? ? G723 >>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?G722 >>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?GSM >>>> >>>> I think I understand that when making an outside call, FreeSWITCH >>>> follow these steps: >>>> >>>> USER -> ( ? Dialplan -> profile (internal) -> bridge (external) -> >>>> profile (external) ? ) -> PROVIDER >>>> >>>> PROVIDER -> ( ? Dialplan -> profile (external) -> bridge (internal) -> >>>> profile (internal) ?) -> USER >>>> >>>> right? >>>> >>>> Internal and external I set as follows (and not many changes have >>>> done, and not remember it, because I've been testing days). If >>>> outbound (outbound-codec-prefs) all codecs specified system does not >>>> handle the call, I have to specify these by hand. If active >>>> inbound-proxy-media, not the caller. Some of the time I worked, but >>>> gave me an error that it can do transcoding G729 codec (I do >>>> passthrough), but the proxy does not work half. >>>> >>>> If the outbound property (outbound-codec-prefs) all codecs specified >>>> system does not handle the call, I have to specify these by hand. If >>>> active inbound-proxy-media, not the caller. Some of the time I worked, >>>> but gave me an error that it can do transcoding G729 codec (I want to >>>> make passthrough), but the "proxy media" does not work. >>>> >>>> Basically, what I do is that local users can use all the codecs >>>> allowed (iLBC, GSM, ...) and make an outside call, use the carrier >>>> that will indicate the priority but the free codec. >>>> >>>> With this configuration works for me, but I would like to understand >>>> why so if it works and otherwise no. Coming to understand how to >>>> configure properly and so as not to disturb the mail list ;-). Thanks >>>> ! >>>> >>>> Best regards >>>> Anto >>>> >>>> vars.xml >>>> >>>> >>> >>>> data="global_codec_prefs=iLBC,G7221,speex,PCMU,PCMA,BV16,G726-32,GSM,G729,G723,AMR"/> >>>> >>> >>>> data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,iLBC,G7221,speex,BV16,G726-32,GSM"/> >>>> >>>> internal.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> external.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> dialplan/outbound.xml >>>> >>>> >>>> ? ? ? ? >>>> ? ? ? ? ? ? ? ? >>>> ? ? ? ? ? ? ? ? ?>>> expression="^(\d+)$"> >>>> ? ? ? ? ? ? ? ? ? ? ? ? >>>> ? ? ? ? ? ? ? ? ? ? ? ? >>>> ? ? ? ? ? ? ? ? ? ? ? ? >>>> ? ? ? ? ? ? ? ? ? ? ? ?>>> data="sofia/gateway/provider-2/$1"/> >>>> ? ? ? ? ? ? ? ? ? >>>> ? ? ? ? ? ? ? ? >>>> ? ? ? ? >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> From msc at freeswitch.org Thu Mar 15 00:24:43 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Mar 2012 14:24:43 -0700 Subject: [Freeswitch-users] Codec negotiation with carriers In-Reply-To: References: Message-ID: Did you get sip traces and logs of working vs. non-working calls and put them on pastebin? Most likely there is an explanation but it will take some time and effort to figure it out. -MC On Wed, Mar 14, 2012 at 2:18 PM, Anto wrote: > Hello > > I have searched previous messages in the list, I consulted the book of > FreeSWITCH (which I bought over a year), wiki and so on. I still do > not understand how and why in some cases I work. Also I downloaded > frontend to consult your code if there was something about this, but > still the same. I have several weeks with this question and I can not > find it. In the end I decided to spend the gateway to Asterisk, and > you at least understand its operation. Thank you very much to all :-) > > Best regards > Anto > > 2012/3/11 Anto : > > Hi > > > > I still do not find the solution and not really understanding, because > > it works:-S > > > > regards > > anto > > > > 2012/3/7 Anto : > >> Hello > >> > >> Attached file, with the traces of the different tests (with different > >> configurations). > >> > >> http://pastebin.freeswitch.org/18599 > >> > >> I have read the url that you mentioned, the initial guide FreeSWITCH, > >> that of mod_sofia, applications, etc.. I believe that most of the wiki > >> (maybe when do not give the solution, read as much documentation is > >> worse idea :-S, lock me more). > >> > >> I made a configuration that works (I have not tested the audio), but > >> earlier (before I started "touch" the profiles) if I could talk to a > >> physical phone (several times). The problem is that I can not > >> understand why it works and sometimes not, and I would like to learn > >> :-). Not only do and forget, so I would like to learn and less > >> disturbing to the mail list and (maybe in the future) to help other > >> newbies like me :-). Thanks ! > >> > >> Best regards > >> Anto > >> > >> 2012/3/7 Michael Collins : > >>> You may want to read up on codec negotiation: > >>> http://wiki.freeswitch.org/wiki/Codec_negotiation > >>> > >>> There are different ways to handle codecs depending on your needs. I'd > read > >>> that page first and then try out some of the suggestions. If you're > still > >>> having trouble then I'd recommend getting SIP traces of the traffic and > >>> putting them on pastebin.freeswitch.org. The gang here is pretty good > at > >>> looking over logs and helping with diagnosing problems. :) > >>> > >>> -MC > >>> > >>> On Tue, Mar 6, 2012 at 2:30 PM, Anto wrote: > >>>> > >>>> Hi > >>>> > >>>> I am following the steps in this direction > >>>> "http://wiki.freeswitch.org/wiki/SBC_Setup" and > >>>> "http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice", > >>>> I reread the whole entire wiki (or so I lack), but do not quite > >>>> assimilate or finding the right formula to operate the bridge :-S. > >>>> > >>>> I captured traffic with ngrep, I enabled sip-trace, console logconsole > >>>> 8, etc., but unless the transcoding error (only two of the hundreds of > >>>> combinations of settings that I have), I have not seen anything > >>>> "weird" :-S > >>>> > >>>> I have 3 suppliers, each with this codec: > >>>> > >>>> 1) 2) 3) > >>>> G729 G729 G729 > >>>> G711u G711A G711A > >>>> G711A G711u G711u > >>>> G723 G723 > >>>> G722 > >>>> GSM > >>>> > >>>> I think I understand that when making an outside call, FreeSWITCH > >>>> follow these steps: > >>>> > >>>> USER -> ( Dialplan -> profile (internal) -> bridge (external) -> > >>>> profile (external) ) -> PROVIDER > >>>> > >>>> PROVIDER -> ( Dialplan -> profile (external) -> bridge (internal) -> > >>>> profile (internal) ) -> USER > >>>> > >>>> right? > >>>> > >>>> Internal and external I set as follows (and not many changes have > >>>> done, and not remember it, because I've been testing days). If > >>>> outbound (outbound-codec-prefs) all codecs specified system does not > >>>> handle the call, I have to specify these by hand. If active > >>>> inbound-proxy-media, not the caller. Some of the time I worked, but > >>>> gave me an error that it can do transcoding G729 codec (I do > >>>> passthrough), but the proxy does not work half. > >>>> > >>>> If the outbound property (outbound-codec-prefs) all codecs specified > >>>> system does not handle the call, I have to specify these by hand. If > >>>> active inbound-proxy-media, not the caller. Some of the time I worked, > >>>> but gave me an error that it can do transcoding G729 codec (I want to > >>>> make passthrough), but the "proxy media" does not work. > >>>> > >>>> Basically, what I do is that local users can use all the codecs > >>>> allowed (iLBC, GSM, ...) and make an outside call, use the carrier > >>>> that will indicate the priority but the free codec. > >>>> > >>>> With this configuration works for me, but I would like to understand > >>>> why so if it works and otherwise no. Coming to understand how to > >>>> configure properly and so as not to disturb the mail list ;-). Thanks > >>>> ! > >>>> > >>>> Best regards > >>>> Anto > >>>> > >>>> vars.xml > >>>> > >>>> >>>> > >>>> > data="global_codec_prefs=iLBC,G7221,speex,PCMU,PCMA,BV16,G726-32,GSM,G729,G723,AMR"/> > >>>> >>>> > >>>> > data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,iLBC,G7221,speex,BV16,G726-32,GSM"/> > >>>> > >>>> internal.xml > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> external.xml > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> dialplan/outbound.xml > >>>> > >>>> > >>>> > >>>> > >>>> >>>> expression="^(\d+)$"> > >>>> > >>>> > >>>> > >>>> >>>> data="sofia/gateway/provider-2/$1"/> > >>>> > >>>> > >>>> > >>>> > >>>> > _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> http://www.cluecon.com > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120314/1548716e/attachment-0001.html From david.villasmil.work at gmail.com Thu Mar 15 02:58:14 2012 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 15 Mar 2012 00:58:14 +0100 Subject: [Freeswitch-users] Send calls through registered gateway Message-ID: Hello guys, I seem to be unable to send calls through a registered gateway... is there any sample dialplan available? thanks David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/a2ac1e64/attachment.html From gabe at gundy.org Thu Mar 15 03:23:37 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 14 Mar 2012 18:23:37 -0600 Subject: [Freeswitch-users] Send calls through registered gateway In-Reply-To: References: Message-ID: On Wed, Mar 14, 2012 at 5:58 PM, David Villasmil wrote: > I seem to be unable to send calls through a registered gateway... is there > any sample dialplan available? http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways Best, Gabe From david.villasmil.work at gmail.com Thu Mar 15 03:32:20 2012 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 15 Mar 2012 01:32:20 +0100 Subject: [Freeswitch-users] Send calls through registered gateway In-Reply-To: References: Message-ID: Hello, Thanks for the reply. Not what I'm looking for though. I register an asterisk with with, say with username 1001. I receive a call from user 2 and want to terminate what 1002 sends with the gateway (not a gateway per se, but a registered user). in the dialplan I have: But all asterisk gets is a call to extension "s" in the INVITE, even though the "To" header is OK.... how do I set it correctly?? INVITE sip:s at 5.6.7.8 SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgmBy3gy8BSKZB Max-Forwards: 69 From: "David Villasmil" ;tag=1ep74eBrB2Fcj To: Call-ID: 3393725d-e8d7-122f-6395-0013725ca38a CSeq: 25549284 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 482 X-FS-Support: update_display Remote-Party-ID: "David Villasmil" ;party=calling;screen=yes;privacy=off On Thu, Mar 15, 2012 at 1:23 AM, Gabriel Gunderson wrote: > On Wed, Mar 14, 2012 at 5:58 PM, David Villasmil > wrote: > > I seem to be unable to send calls through a registered gateway... is > there > > any sample dialplan available? > > > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways > > Best, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/61f55e56/attachment.html From gabe at gundy.org Thu Mar 15 05:22:45 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 14 Mar 2012 20:22:45 -0600 Subject: [Freeswitch-users] Send calls through registered gateway In-Reply-To: References: Message-ID: On Wed, Mar 14, 2012 at 6:32 PM, David Villasmil wrote: > I register an asterisk with with, say with username 1001. I receive a call > from user 2 and want to terminate what 1002 sends with the gateway (not a > gateway ?per se, but a registered user). I think you may have it backwards. If you want your Asterisk server to act as the GW to FreeSWITCH (you send calls from FS to Asterisk and it hit's its own dialplan), then you would setup a gateway in FreeSWITCH (see the sofia profie). You don't *need* FreeSWITCH to register to its GWs, but you can if you like. Then from FS's perspective, you would dial it like so: sofia/gateway/NAME_OF_GW/NUMBER_TO_DIAL I'm sorry if I misunderstood what you're doing, but I hope this helps. Best, Gabe From lists at telefaks.de Thu Mar 15 12:51:04 2012 From: lists at telefaks.de (Peter Steinbach) Date: Thu, 15 Mar 2012 10:51:04 +0100 Subject: [Freeswitch-users] Stuck Calls In-Reply-To: <1FFF97C269757C458224B7C895F35F15052B69@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F15052B69@cantor.std.visionutv.se> Message-ID: <4F61BB88.3040706@telefaks.de> We had that situation on a Freeswitch dated 29-Jan-2012 and we solved it by the described way. Best regards Peter Am 14.03.2012 15:05, schrieb Peter Olsson: > > Are you running latest git? > > /Peter > > *Fr?n:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *F?r *Peter > Steinbach > *Skickat:* den 14 mars 2012 14:56 > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] Stuck Calls > > On our system, some channels were still present in the channel table > of the core database after uuid_kill. After deleting those uuid > entries in the channel table, all was fine. > > Best regards > Peter > > Am 13.03.2012 11:39, schrieb B.Tietz at pinguin.ag: > > > Hi, > > I have some stuck calls in the list when I issue "show detailed_calls" > > Trying to kill them with "uuid_kill " or "fsctl hupall > normal_clearing dialed_ext 1234" doesn't help. > > FS is configured as HA-node with odbc in the core and sofia. > > Any other suggestions?! > > Benjamin > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet:www.telefaks.de > > > !DSPAM:4f60a29232761811354587! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/bf1e49c2/attachment-0001.html From muelbuesch at as-infodienste.de Thu Mar 15 17:56:04 2012 From: muelbuesch at as-infodienste.de (=?ISO-8859-15?Q?Marcus_M=FClb=FCsch?=) Date: Thu, 15 Mar 2012 15:56:04 +0100 Subject: [Freeswitch-users] I do not get any ringtones? Message-ID: <4F620304.6090406@as-infodienste.de> Hello all, I thought that just setting or something similar would make freeswicth giving a ringtone when it opens a SIP-connection. Alas, it doesn't. I freely admit that I feel lost in all the documents available; so a "RTFM" is an completely acceptable answer; as long as you point me to the correct place. Thank you. From cesar.bermudez at gmail.com Thu Mar 15 18:03:56 2012 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Thu, 15 Mar 2012 12:03:56 -0300 Subject: [Freeswitch-users] I do not get any ringtones? In-Reply-To: <4F620304.6090406@as-infodienste.de> References: <4F620304.6090406@as-infodienste.de> Message-ID: this its working for me: another working sample: Best regards. On Thu, Mar 15, 2012 at 11:56 AM, Marcus M?lb?sch < muelbuesch at as-infodienste.de> wrote: > Hello all, > > I thought that just setting data="tone_stream://%(400,200,400,450);%(400,2200,400,450)"/> or > something similar would make freeswicth giving a ringtone when it opens > a SIP-connection. > > Alas, it doesn't. > > I freely admit that I feel lost in all the documents available; so a > "RTFM" is an completely acceptable answer; as long as you point me to > the correct place. > > Thank you. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/94ccc111/attachment.html From lazyvirus at gmx.com Thu Mar 15 18:35:32 2012 From: lazyvirus at gmx.com (Bzzz) Date: Thu, 15 Mar 2012 16:35:32 +0100 Subject: [Freeswitch-users] sips, tls, srtp, etc Message-ID: <20120315163532.7507ef73@anubis.defcon1> FS latest FusionPBX ===================== Hi list, I read a lot from the wiki, but I'm a bit lost about securing calls. From what I read elsewhere, ssl+srtp seems to be the best solution, however the wiki only talks about tls+srtp. I modified conf/vars.xml as of the wiki & enabled xxx_ssl_enable, but put 'sslv23' instead of 'tls'; I also modified conf/directory/default.xml (from 'tls' to 'sslv23') as: is this right & sufficient? If not, what do I miss or what would be the best solution to make sure all calls will be secured both signaling & conversations? At this time, I make tests with the jitsi softphone (ex sip-commu?nicator) which seems to use SRTP once the SAS has been accepted from each side: that's what wireshark shows, but it also show that SIP is use instead of SIPS, is it because I move 'tls' to 'sslv23'? JY -- From bote_radio at botecomm.com Thu Mar 15 18:46:39 2012 From: bote_radio at botecomm.com (Bote Man) Date: Thu, 15 Mar 2012 11:46:39 -0400 Subject: [Freeswitch-users] I do not get any ringtones? In-Reply-To: <4F620304.6090406@as-infodienste.de> References: <4F620304.6090406@as-infodienste.de> Message-ID: <006f01cd02c2$d10cf250$7326d6f0$@com> I tried this yesterday, so your question got to me before I forget it :-) http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pre_answer Pre-answer sends a SIP status code 183 and apparently opens up early media so that they can hear whatever tone stream you see fit to play to them. I also copied the example from the dialplan and padded some sleep time on either side of it to provide a cushion. I abandoned my effort at injecting a short 2600 Hz "wink" because it was too loud and hurt the ear. Perhaps an amplitude parameter might be in order for tone_stream? HTH Bote > -----Original Message----- > From: Marcus M?lb?sch > Sent: Thursday, 15 March, 2012 10:56 > > Hello all, > > I thought that just setting data="tone_stream://%(400,200,400,450);%(400,2200,400,450)"/> or > something similar would make freeswicth giving a ringtone when it > opens > a SIP-connection. > > Alas, it doesn't. > > I freely admit that I feel lost in all the documents available; so > a > "RTFM" is an completely acceptable answer; as long as you point me to > the correct place. > > Thank you. > > From lazyvirus at gmx.com Thu Mar 15 18:56:13 2012 From: lazyvirus at gmx.com (Bzzz) Date: Thu, 15 Mar 2012 16:56:13 +0100 Subject: [Freeswitch-users] I do not get any ringtones? In-Reply-To: <006f01cd02c2$d10cf250$7326d6f0$@com> References: <4F620304.6090406@as-infodienste.de> <006f01cd02c2$d10cf250$7326d6f0$@com> Message-ID: <20120315165613.36a9758d@anubis.defcon1> On Thu, 15 Mar 2012 11:46:39 -0400 "Bote Man" wrote: > > I abandoned my effort at injecting a short 2600 Hz "wink" because it was > too loud and hurt the ear. Perhaps an amplitude parameter might be in > order for tone_stream? May be you could cheat by editing the 2600Hz sound file and lower its level to, say 33% or 25% (or even less) from what it is; it takes 10s with audacity to do so. -- From freeswitch-users at digitaldan.com Thu Mar 15 18:59:51 2012 From: freeswitch-users at digitaldan.com (Dan) Date: Thu, 15 Mar 2012 09:59:51 -0600 (MDT) Subject: [Freeswitch-users] Problem getting TALK and NOTALK events In-Reply-To: Message-ID: <2ceba846-8644-4c1d-846d-10376e062622@radio> Hi, I'm having some issues getting TALK / NOTALK events to fire on an incoming stream to Freeswitch. In my ESL application I am subscribing to RECORD_START RECORD_STOP TALK NOTALK, Below is the dial plan I am using: I can see that VAD is enabled: 2012-03-15 09:49:25.523832 [DEBUG] switch_rtp.c:4130 Activate VAD codec PCMU 20ms 2012-03-15 09:49:25.523832 [DEBUG] sofia_glue.c:3353 AUDIO RTP Engage VAD for sofia/external/3035551212 at 10.10.10.1 ( in out ) In my ESL app I get the RECORD_START and RECORD_STOP but not the talk events. I'm on git version "2c52f23 2012-02-18 08:37:47 -0600", Any ideas? Thanks. From bote_radio at botecomm.com Thu Mar 15 19:22:57 2012 From: bote_radio at botecomm.com (Bote Man) Date: Thu, 15 Mar 2012 12:22:57 -0400 Subject: [Freeswitch-users] I do not get any ringtones? In-Reply-To: <20120315165613.36a9758d@anubis.defcon1> References: <4F620304.6090406@as-infodienste.de> <006f01cd02c2$d10cf250$7326d6f0$@com> <20120315165613.36a9758d@anubis.defcon1> Message-ID: <007301cd02c7$e32f0770$a98d1650$@com> A tone_stream is not a sound file, it is a generated tone sequence. Bote > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Bzzz > Sent: Thursday, 15 March, 2012 11:56 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] I do not get any ringtones? > > On Thu, 15 Mar 2012 11:46:39 -0400 > "Bote Man" wrote: > > > > > I abandoned my effort at injecting a short 2600 Hz "wink" because it > was > > too loud and hurt the ear. Perhaps an amplitude parameter might be > in > > order for tone_stream? > > May be you could cheat by editing the 2600Hz sound file and lower its > level to, say 33% or 25% (or even less) from what it is; it takes > 10s with audacity to do so. > > -- > > ______________________________________________________________________ > ___ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From lazyvirus at gmx.com Thu Mar 15 19:36:10 2012 From: lazyvirus at gmx.com (Bzzz) Date: Thu, 15 Mar 2012 17:36:10 +0100 Subject: [Freeswitch-users] I do not get any ringtones? In-Reply-To: <007301cd02c7$e32f0770$a98d1650$@com> References: <4F620304.6090406@as-infodienste.de> <006f01cd02c2$d10cf250$7326d6f0$@com> <20120315165613.36a9758d@anubis.defcon1> <007301cd02c7$e32f0770$a98d1650$@com> Message-ID: <20120315173610.65cc21a1@anubis.defcon1> On Thu, 15 Mar 2012 12:22:57 -0400 "Bote Man" wrote: > A tone_stream is not a sound file, it is a generated tone sequence. > > Bote Oops, pwnd :) -- From mitch.capper at gmail.com Thu Mar 15 20:23:00 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Thu, 15 Mar 2012 10:23:00 -0700 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: <20120315163532.7507ef73@anubis.defcon1> References: <20120315163532.7507ef73@anubis.defcon1> Message-ID: Actually the wiki primarily recommend sslv23 and documents how to do it at: http://wiki.freeswitch.org/wiki/SIP_TLS#Configuration ~Mitch From lazyvirus at gmx.com Thu Mar 15 20:50:12 2012 From: lazyvirus at gmx.com (Bzzz) Date: Thu, 15 Mar 2012 18:50:12 +0100 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: References: <20120315163532.7507ef73@anubis.defcon1> Message-ID: <20120315185012.25862a47@anubis.defcon1> On Thu, 15 Mar 2012 10:23:00 -0700 Mitch Capper wrote: > Actually the wiki primarily recommend sslv23 and documents how to do > it at: http://wiki.freeswitch.org/wiki/SIP_TLS#Configuration That's what I followed & did, Mitch - at this time I just trying to have SIPS working; but I don't see 5061 nor 5081 ports opened:( JY -- From daveh at beachdognet.com Thu Mar 15 20:55:51 2012 From: daveh at beachdognet.com (Dave Horton) Date: Thu, 15 Mar 2012 13:55:51 -0400 Subject: [Freeswitch-users] ICMP port unreachable for RTCP messages? In-Reply-To: <994352FB-0ECA-4EC6-8F9C-44F0C36F7C95@dchorton.com> References: <994352FB-0ECA-4EC6-8F9C-44F0C36F7C95@dchorton.com> Message-ID: <2A33F2B6-7709-4A15-95D6-813DA9759524@beachdognet.com> I've noticed when taking a wireshark trace for other reasons that my freeswitch server is sending ICMP port unreachable messages in response to incoming RTCP Sender Reports. I was assuming freeswitch supported RTCP, certainly at least from the receiving side of things. Is this not the cas? From mitch.capper at gmail.com Thu Mar 15 21:39:45 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Thu, 15 Mar 2012 11:39:45 -0700 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: <20120315185012.25862a47@anubis.defcon1> References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> Message-ID: If you could paste the freeswitch log on startup to pb. Assuming tls is set to true on the sofia profile and the tls-cert-dir has the cacert.pem and agent.pem it should be starting up. You can also from a running freeswitch just turn debug level on (F8) run fsctl send_sighup (rotates to a clean log file) then type reload mod_sofia then fsctl send_sighup again once its finished loading and PB that log. ~Mitch From lazyvirus at gmx.com Thu Mar 15 22:21:39 2012 From: lazyvirus at gmx.com (Bzzz) Date: Thu, 15 Mar 2012 20:21:39 +0100 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> Message-ID: <20120315202139.6a994dd2@anubis.defcon1> On Thu, 15 Mar 2012 11:39:45 -0700 Mitch Capper wrote: > If you could paste the freeswitch log on startup to pb. Assuming tls > is set to true on the sofia profile ? I modified vars.xml to authorize ssl. > and the tls-cert-dir has the > cacert.pem and agent.pem it should be starting up. I hope so, I have: conf/ssl => agent.pem & cafile.pem conf/ssl/CA => cacert.pem, cacert.srl, cakey.pem & config.tpl > You can also from > a running freeswitch just turn debug level on (F8) run fsctl Huu, I don't have fsctl (nor into my source directory); but I see what you want; I'll cut the log file to recover just that. > send_sighup (rotates to a clean log file) then type reload mod_sofia > then fsctl send_sighup again once its finished loading and PB that > log. PB? JY -- BOFH excuse #429: Temporal anomaly -------------- next part -------------- A non-text attachment was scrubbed... Name: FS.LOG Type: text/x-log Size: 20772 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/295069dc/attachment.bin From paul at iamfine.com Thu Mar 15 23:16:08 2012 From: paul at iamfine.com (Paul) Date: Thu, 15 Mar 2012 13:16:08 -0700 (PDT) Subject: [Freeswitch-users] Mod_flite wont load Message-ID: <1331842568647-7376926.post@n2.nabble.com> March 15 Mac OSX 10.7.3 Problem loading mod_flite when starting freeswitch Error message is 2012-03-15 13:06:22.460110 [CRIT] switch_loadable_module.c:1295 Error Loading module /usr/local/freeswitch/mod/mod_flite.so **dlopen(/usr/local/freeswitch/mod/mod_flite.so, 6): Symbol not found: _register_cmu_us_kal16 Referenced from: /usr/local/freeswitch/mod/mod_flite.so Expected in: flat namespace in /usr/local/freeswitch/mod/mod_flite.so** Did a make current this morning to FreeSWITCH Version 1.0.head (git-fcab3de 2012-03-15 18-57-19 +0000) edited modules.conf to make it available commented out all cepestral edited conf/autoload_configs/modules.conf.xml to ensure it loads Fails on load -- I then tried to load from fs_CLI with > load mod_flite I get the same error I went back and did > sudo make mod_flite seems to be successful -- Here is the output making all mod_flite making in ... making in include ... making in src ... making in src/audio ... making in src/utils ... making in src/regex ... making in src/hrg ... making in src/stats ... making in src/speech ... making in src/lexicon ... making in src/synth ... making in src/wavesynth ... making in src/cg ... making in lang ... making in lang/cmulex ... making in lang/usenglish ... making in lang/cmu_us_kal ... making in lang/cmu_time_awb ... making in lang/cmu_us_kal16 ... making in lang/cmu_us_awb ... making in lang/cmu_us_rms ... making in lang/cmu_us_slt ... making in doc ... making in tools ... making in main ... Compiling /usr/local/src/freeswitch/src/mod/asr_tts/mod_flite/mod_flite.c... quiet_libtool: compile: gcc -I/usr/local/src/freeswitch/libs/flite-1.5.4-current/include -I/usr/local/src/freeswitch/libs/flite-1.5.4-current/include -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/asr_tts/mod_flite/mod_flite.c -fno-common -DPIC -o .libs/mod_flite.o quiet_libtool: compile: gcc -I/usr/local/src/freeswitch/libs/flite-1.5.4-current/include -I/usr/local/src/freeswitch/libs/flite-1.5.4-current/include -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/asr_tts/mod_flite/mod_flite.c -o mod_flite.o >/dev/null 2>&1 Creating mod_flite.la... Restarted FS and still same load error _______________________________________________ Struggling at this point to figure out whats preventing it from loading Any thoughts would be greatly appreciated Paul -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Mod-flite-wont-load-tp7376926p7376926.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gozdal at gmail.com Thu Mar 15 18:48:08 2012 From: gozdal at gmail.com (Marcin Gozdalik) Date: Thu, 15 Mar 2012 16:48:08 +0100 Subject: [Freeswitch-users] Event socket outbound for attended atransfer Message-ID: Hi I am using outbound event socket connection in dialplan using . The scenario is like this: 1. Call from PSTN to FS (leg A) with socket application in dialplan as above 2. FS bridges the call to endpoint (leg B) 3. Endpoint picks up the call (there is a call no 1 established with legs A-B) 4. Endpoint holds call no 1 5. Endpoint invites somebody via FS (leg C) with socket application in dialplan as above 6. FS invites PSTN gateway (leg D) 7. D picks up the call (not there is a call no 2 established with legs C-D) 8. Endpoint makes attended transfer between calls no 1 and no 2 9. A speaks with D, call no. 2 is ended, call no. 1 is modified to include legs A and D My event receiver (esl:8022) receives two connections - one for each call. However the second socket connection ends as soon as Endpoint does a transfer (point no 8). This doesn't happen when Endpoint does a blind transfer - both event connections last until A and D are disconnected (in blind transfer scenario there is no leg C, Endpoint sends REFER to D in point 5 but it creates another socket connection). Is there any method to keep the event connection alive until A and D are connected with attended transfer? I have tried "myevents", "events all plain" and "linger" but it doesn't help - second event connection is closed as soon as connection in transferred. -- Marcin Gozdalik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/65eb7e67/attachment.html From moshe3t at gmail.com Thu Mar 15 22:27:08 2012 From: moshe3t at gmail.com (moshe3) Date: Thu, 15 Mar 2012 15:27:08 -0400 Subject: [Freeswitch-users] Distributed Parking Lot Message-ID: <006301cd02e1$9d70b980$d8522c80$@com> HI I would like to check if there is a way to maintain a single parking lot scheme over a distributed (multiple FreeSWITCH) instances, We are implementing a multi tenant ?using multi server setup and I?m wondering if call parking will be able to work across multiple server, does it share a ODBC, if it does not what would be the best work around ? Or what would involve to get it work in a distributed environment Moshe3 From lazyvirus at gmx.com Thu Mar 15 23:35:25 2012 From: lazyvirus at gmx.com (Bzzz) Date: Thu, 15 Mar 2012 21:35:25 +0100 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> Message-ID: <20120315213525.0430d40b@anubis.defcon1> On Thu, 15 Mar 2012 11:39:45 -0700 Mitch Capper wrote: Ok, I found why I had "tls [false]": I had the same block commented below with false for tls (so, xml isn't able to comment several lines in one comment block, just another reason to hate it...) Now reloading mod_sofia show that tls is enabled - good; but I lost 5060 & 5080 ports and still no 5061 nor 5081 - weird. JY -- 40 isn't old. If you're a tree. From jcasale at activenetwerx.com Thu Mar 15 23:44:35 2012 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 15 Mar 2012 20:44:35 +0000 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: <20120315213525.0430d40b@anubis.defcon1> References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> , <20120315213525.0430d40b@anubis.defcon1> Message-ID: > (so, xml isn't able to comment several > lines in one comment block, just another reason to hate it...) Huh, i'd be inclined to say you wrote the comment wrong... As for the second comment, it makes perfect sense from a programmatic sense but I digress, I am sure... From potxoka at gmail.com Fri Mar 16 00:22:09 2012 From: potxoka at gmail.com (Anto) Date: Thu, 15 Mar 2012 22:22:09 +0100 Subject: [Freeswitch-users] Codec negotiation with carriers In-Reply-To: References: Message-ID: Hi If, upload a file to trace and explanation to this address http://pastebin.freeswitch.org/18599 I do not want disturb watching this ;-), I prefer to use a system to understand, this scenario and for future projects. With everything I've read do not really understand what the codecs :-S , but if I had been good to understand the rest of the operation of FreeSWITCH (or so I think). Thanks ! Regards Anto 2012/3/14 Michael Collins : > Did you get sip traces and logs of working vs. non-working calls and put > them on pastebin? Most likely there is an explanation but it will take some > time and effort to figure it out. > > -MC > > > On Wed, Mar 14, 2012 at 2:18 PM, Anto wrote: >> >> Hello >> >> I have searched previous messages in the list, I consulted the book of >> FreeSWITCH (which I bought over a year), wiki and so on. I still do >> not understand how and why in some cases I work. Also I downloaded >> frontend to consult your code if there was something about this, but >> still the same. I have several weeks with this question and I can not >> find it. In the end I decided to spend the gateway to Asterisk, and >> you at least understand its operation. Thank you very much to all :-) >> >> Best regards >> Anto >> >> 2012/3/11 Anto : >> > Hi >> > >> > I still do not find the solution and not really understanding, because >> > it works:-S >> > >> > regards >> > anto >> > >> > 2012/3/7 Anto : >> >> Hello >> >> >> >> Attached file, with the traces of the different tests (with different >> >> configurations). >> >> >> >> http://pastebin.freeswitch.org/18599 >> >> >> >> I have read the url that you mentioned, the initial guide FreeSWITCH, >> >> that of mod_sofia, applications, etc.. I believe that most of the wiki >> >> (maybe when do not give the solution, read as much documentation is >> >> worse idea :-S, lock me more). >> >> >> >> I made a configuration that works (I have not tested the audio), but >> >> earlier (before I started "touch" the profiles) if I could talk to a >> >> physical phone (several times). The problem is that I can not >> >> understand why it works and sometimes not, and I would like to learn >> >> :-). Not only do and forget, so I would like to learn and less >> >> disturbing to the mail list and (maybe in the future) to help other >> >> newbies like me :-). Thanks ! >> >> >> >> Best regards >> >> Anto >> >> >> >> 2012/3/7 Michael Collins : >> >>> You may want to read up on codec negotiation: >> >>> http://wiki.freeswitch.org/wiki/Codec_negotiation >> >>> >> >>> There are different ways to handle codecs depending on your needs. I'd >> >>> read >> >>> that page first and then try out some of the suggestions. If you're >> >>> still >> >>> having trouble then I'd recommend getting SIP traces of the traffic >> >>> and >> >>> putting them on pastebin.freeswitch.org. The gang here is pretty good >> >>> at >> >>> looking over logs and helping with diagnosing problems. :) >> >>> >> >>> -MC >> >>> >> >>> On Tue, Mar 6, 2012 at 2:30 PM, Anto wrote: >> >>>> >> >>>> Hi >> >>>> >> >>>> I am following the steps in this direction >> >>>> "http://wiki.freeswitch.org/wiki/SBC_Setup" and >> >>>> "http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice", >> >>>> I reread the whole entire wiki (or so I lack), but do not quite >> >>>> assimilate or finding the right formula to operate the bridge :-S. >> >>>> >> >>>> I captured traffic with ngrep, I enabled sip-trace, console >> >>>> logconsole >> >>>> 8, etc., but unless the transcoding error (only two of the hundreds >> >>>> of >> >>>> combinations of settings that I have), I have not seen anything >> >>>> "weird" :-S >> >>>> >> >>>> I have 3 suppliers, each with this codec: >> >>>> >> >>>> 1) ? ? ? ? ? 2) ? ? ? ? ? ? ?3) >> >>>> G729 ? ? ? ?G729 ? ? ? ?G729 >> >>>> G711u ? ? ?G711A ? ? ?G711A >> >>>> G711A ? ? G711u ? ? ? G711u >> >>>> ? ? ? ? ? ? ? ?G723 ? ? ? ? G723 >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?G722 >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?GSM >> >>>> >> >>>> I think I understand that when making an outside call, FreeSWITCH >> >>>> follow these steps: >> >>>> >> >>>> USER -> ( ? Dialplan -> profile (internal) -> bridge (external) -> >> >>>> profile (external) ? ) -> PROVIDER >> >>>> >> >>>> PROVIDER -> ( ? Dialplan -> profile (external) -> bridge (internal) >> >>>> -> >> >>>> profile (internal) ?) -> USER >> >>>> >> >>>> right? >> >>>> >> >>>> Internal and external I set as follows (and not many changes have >> >>>> done, and not remember it, because I've been testing days). If >> >>>> outbound (outbound-codec-prefs) all codecs specified system does not >> >>>> handle the call, I have to specify these by hand. If active >> >>>> inbound-proxy-media, not the caller. Some of the time I worked, but >> >>>> gave me an error that it can do transcoding G729 codec (I do >> >>>> passthrough), but the proxy does not work half. >> >>>> >> >>>> If the outbound property (outbound-codec-prefs) all codecs specified >> >>>> system does not handle the call, I have to specify these by hand. If >> >>>> active inbound-proxy-media, not the caller. Some of the time I >> >>>> worked, >> >>>> but gave me an error that it can do transcoding G729 codec (I want to >> >>>> make passthrough), but the "proxy media" does not work. >> >>>> >> >>>> Basically, what I do is that local users can use all the codecs >> >>>> allowed (iLBC, GSM, ...) and make an outside call, use the carrier >> >>>> that will indicate the priority but the free codec. >> >>>> >> >>>> With this configuration works for me, but I would like to understand >> >>>> why so if it works and otherwise no. Coming to understand how to >> >>>> configure properly and so as not to disturb the mail list ;-). Thanks >> >>>> ! >> >>>> >> >>>> Best regards >> >>>> Anto >> >>>> >> >>>> vars.xml >> >>>> >> >>>> > >>>> >> >>>> >> >>>> data="global_codec_prefs=iLBC,G7221,speex,PCMU,PCMA,BV16,G726-32,GSM,G729,G723,AMR"/> >> >>>> > >>>> >> >>>> >> >>>> data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,iLBC,G7221,speex,BV16,G726-32,GSM"/> >> >>>> >> >>>> internal.xml >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> external.xml >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> dialplan/outbound.xml >> >>>> >> >>>> >> >>>> ? ? ? ? >> >>>> ? ? ? ? ? ? ? ? >> >>>> ? ? ? ? ? ? ? ? ?> >>>> expression="^(\d+)$"> >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? >> >>>> ? ? ? ? ? ? ? ? ? ? ? ?> >>>> data="sofia/gateway/provider-2/$1"/> >> >>>> ? ? ? ? ? ? ? ? ? >> >>>> ? ? ? ? ? ? ? ? >> >>>> ? ? ? ? >> >>>> >> >>>> >> >>>> _________________________________________________________________________ >> >>>> Professional FreeSWITCH Consulting Services: >> >>>> consulting at freeswitch.org >> >>>> http://www.freeswitchsolutions.com >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> Official FreeSWITCH Sites >> >>>> http://www.freeswitch.org >> >>>> http://wiki.freeswitch.org >> >>>> http://www.cluecon.com >> >>>> >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>> http://www.freeswitch.org >> >>> >> >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From joe.jflemmings at gmail.com Fri Mar 16 00:57:38 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Thu, 15 Mar 2012 14:57:38 -0700 Subject: [Freeswitch-users] Record Session Message-ID: How can i record a call in two seperate directories at the same time the following does not work Joe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/fb4536f4/attachment-0001.html From msc at freeswitch.org Fri Mar 16 02:29:21 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 15 Mar 2012 16:29:21 -0700 Subject: [Freeswitch-users] Problem getting TALK and NOTALK events In-Reply-To: <2ceba846-8644-4c1d-846d-10376e062622@radio> References: <2ceba846-8644-4c1d-846d-10376e062622@radio> Message-ID: Can you confirm if the TALK/NOTALK events are even firing? Try listening to all events and then sifting through the barrage of data to see if those events are present. If they are then you know you just have a filtering issue. If they are not present then you know there's an issue with VAD or something like that. -MC On Thu, Mar 15, 2012 at 8:59 AM, Dan wrote: > Hi, I'm having some issues getting TALK / NOTALK events to fire on an > incoming stream to Freeswitch. In my ESL application I am subscribing to > RECORD_START RECORD_STOP TALK NOTALK, Below is the dial plan I am using: > > > > > > > > > > > > > I can see that VAD is enabled: > > 2012-03-15 09:49:25.523832 [DEBUG] switch_rtp.c:4130 Activate VAD codec > PCMU 20ms > 2012-03-15 09:49:25.523832 [DEBUG] sofia_glue.c:3353 AUDIO RTP Engage VAD > for sofia/external/3035551212 at 10.10.10.1 ( in out ) > > > In my ESL app I get the RECORD_START and RECORD_STOP but not the talk > events. I'm on git version "2c52f23 2012-02-18 08:37:47 -0600", Any > ideas? Thanks. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/69d6fe9b/attachment.html From msc at freeswitch.org Fri Mar 16 02:33:26 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 15 Mar 2012 16:33:26 -0700 Subject: [Freeswitch-users] I do not get any ringtones? In-Reply-To: <006f01cd02c2$d10cf250$7326d6f0$@com> References: <4F620304.6090406@as-infodienste.de> <006f01cd02c2$d10cf250$7326d6f0$@com> Message-ID: > I abandoned my effort at injecting a short 2600 Hz "wink" because it was > too loud and hurt the ear. Perhaps an amplitude parameter might be in > order for tone_stream? > Ask and ye shall receive! You just needed to RTFM in the right place: http://wiki.freeswitch.org/wiki/TGML Note the "v" param. An example of using it is down in the Perl example of the bong tone. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/3755f5b4/attachment.html From msc at freeswitch.org Fri Mar 16 02:35:19 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 15 Mar 2012 16:35:19 -0700 Subject: [Freeswitch-users] Record Session In-Reply-To: References: Message-ID: What possible advantage is there to recording to two separate directories at the same time? Would it not be better just to copy the recording once the call is over? -MC On Thu, Mar 15, 2012 at 2:57 PM, Joe Flemmings wrote: > How can i record a call in two seperate directories at the same time > > the following does not work > > data="$${base_dir}/recordings/admin/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > data="$${base_dir}/recordings/users/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > > Joe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/9c6af276/attachment.html From msc at freeswitch.org Fri Mar 16 02:44:13 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 15 Mar 2012 16:44:13 -0700 Subject: [Freeswitch-users] Codec negotiation with carriers In-Reply-To: References: Message-ID: Well, this is a little better, however you don't have proper freeswitch logs on all these calls. For example, only the first call has freeswitch debug output. The other calls have sip traces, but not the first call. One call has what appears to be info-level output, but not debug-level output. I'd recommend that if you have this much information it might be good to put each call example in its own pastebin. Also, be sure to give a detailed description of what kind of call you are documenting. Some of your traces/debugs have no information explaining what the call is doing. Whether you are reporting a working or failed call, be sure to mention what kind of call it is. In the case of a failed call, be sure to mention what it is you are trying to do and what call result you expected to see. Thanks! -MC On Thu, Mar 15, 2012 at 2:22 PM, Anto wrote: > Hi > > If, upload a file to trace and explanation to this address > http://pastebin.freeswitch.org/18599 > > I do not want disturb watching this ;-), I prefer to use a system to > understand, this scenario and for future projects. > > With everything I've read do not really understand what the codecs :-S > , but if I had been good to understand the rest of the operation of > FreeSWITCH (or so I think). Thanks ! > > Regards > Anto > > 2012/3/14 Michael Collins : > > Did you get sip traces and logs of working vs. non-working calls and put > > them on pastebin? Most likely there is an explanation but it will take > some > > time and effort to figure it out. > > > > -MC > > > > > > On Wed, Mar 14, 2012 at 2:18 PM, Anto wrote: > >> > >> Hello > >> > >> I have searched previous messages in the list, I consulted the book of > >> FreeSWITCH (which I bought over a year), wiki and so on. I still do > >> not understand how and why in some cases I work. Also I downloaded > >> frontend to consult your code if there was something about this, but > >> still the same. I have several weeks with this question and I can not > >> find it. In the end I decided to spend the gateway to Asterisk, and > >> you at least understand its operation. Thank you very much to all :-) > >> > >> Best regards > >> Anto > >> > >> 2012/3/11 Anto : > >> > Hi > >> > > >> > I still do not find the solution and not really understanding, because > >> > it works:-S > >> > > >> > regards > >> > anto > >> > > >> > 2012/3/7 Anto : > >> >> Hello > >> >> > >> >> Attached file, with the traces of the different tests (with different > >> >> configurations). > >> >> > >> >> http://pastebin.freeswitch.org/18599 > >> >> > >> >> I have read the url that you mentioned, the initial guide FreeSWITCH, > >> >> that of mod_sofia, applications, etc.. I believe that most of the > wiki > >> >> (maybe when do not give the solution, read as much documentation is > >> >> worse idea :-S, lock me more). > >> >> > >> >> I made a configuration that works (I have not tested the audio), but > >> >> earlier (before I started "touch" the profiles) if I could talk to a > >> >> physical phone (several times). The problem is that I can not > >> >> understand why it works and sometimes not, and I would like to learn > >> >> :-). Not only do and forget, so I would like to learn and less > >> >> disturbing to the mail list and (maybe in the future) to help other > >> >> newbies like me :-). Thanks ! > >> >> > >> >> Best regards > >> >> Anto > >> >> > >> >> 2012/3/7 Michael Collins : > >> >>> You may want to read up on codec negotiation: > >> >>> http://wiki.freeswitch.org/wiki/Codec_negotiation > >> >>> > >> >>> There are different ways to handle codecs depending on your needs. > I'd > >> >>> read > >> >>> that page first and then try out some of the suggestions. If you're > >> >>> still > >> >>> having trouble then I'd recommend getting SIP traces of the traffic > >> >>> and > >> >>> putting them on pastebin.freeswitch.org. The gang here is pretty > good > >> >>> at > >> >>> looking over logs and helping with diagnosing problems. :) > >> >>> > >> >>> -MC > >> >>> > >> >>> On Tue, Mar 6, 2012 at 2:30 PM, Anto wrote: > >> >>>> > >> >>>> Hi > >> >>>> > >> >>>> I am following the steps in this direction > >> >>>> "http://wiki.freeswitch.org/wiki/SBC_Setup" and > >> >>>> " > http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice", > >> >>>> I reread the whole entire wiki (or so I lack), but do not quite > >> >>>> assimilate or finding the right formula to operate the bridge :-S. > >> >>>> > >> >>>> I captured traffic with ngrep, I enabled sip-trace, console > >> >>>> logconsole > >> >>>> 8, etc., but unless the transcoding error (only two of the hundreds > >> >>>> of > >> >>>> combinations of settings that I have), I have not seen anything > >> >>>> "weird" :-S > >> >>>> > >> >>>> I have 3 suppliers, each with this codec: > >> >>>> > >> >>>> 1) 2) 3) > >> >>>> G729 G729 G729 > >> >>>> G711u G711A G711A > >> >>>> G711A G711u G711u > >> >>>> G723 G723 > >> >>>> G722 > >> >>>> GSM > >> >>>> > >> >>>> I think I understand that when making an outside call, FreeSWITCH > >> >>>> follow these steps: > >> >>>> > >> >>>> USER -> ( Dialplan -> profile (internal) -> bridge (external) -> > >> >>>> profile (external) ) -> PROVIDER > >> >>>> > >> >>>> PROVIDER -> ( Dialplan -> profile (external) -> bridge (internal) > >> >>>> -> > >> >>>> profile (internal) ) -> USER > >> >>>> > >> >>>> right? > >> >>>> > >> >>>> Internal and external I set as follows (and not many changes have > >> >>>> done, and not remember it, because I've been testing days). If > >> >>>> outbound (outbound-codec-prefs) all codecs specified system does > not > >> >>>> handle the call, I have to specify these by hand. If active > >> >>>> inbound-proxy-media, not the caller. Some of the time I worked, but > >> >>>> gave me an error that it can do transcoding G729 codec (I do > >> >>>> passthrough), but the proxy does not work half. > >> >>>> > >> >>>> If the outbound property (outbound-codec-prefs) all codecs > specified > >> >>>> system does not handle the call, I have to specify these by hand. > If > >> >>>> active inbound-proxy-media, not the caller. Some of the time I > >> >>>> worked, > >> >>>> but gave me an error that it can do transcoding G729 codec (I want > to > >> >>>> make passthrough), but the "proxy media" does not work. > >> >>>> > >> >>>> Basically, what I do is that local users can use all the codecs > >> >>>> allowed (iLBC, GSM, ...) and make an outside call, use the carrier > >> >>>> that will indicate the priority but the free codec. > >> >>>> > >> >>>> With this configuration works for me, but I would like to > understand > >> >>>> why so if it works and otherwise no. Coming to understand how to > >> >>>> configure properly and so as not to disturb the mail list ;-). > Thanks > >> >>>> ! > >> >>>> > >> >>>> Best regards > >> >>>> Anto > >> >>>> > >> >>>> vars.xml > >> >>>> > >> >>>> >> >>>> > >> >>>> > >> >>>> > data="global_codec_prefs=iLBC,G7221,speex,PCMU,PCMA,BV16,G726-32,GSM,G729,G723,AMR"/> > >> >>>> >> >>>> > >> >>>> > >> >>>> > data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,iLBC,G7221,speex,BV16,G726-32,GSM"/> > >> >>>> > >> >>>> internal.xml > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> external.xml > >> >>>> > >> >>>> > >> >>>> value="$${carriers_codec_prefs}"/> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> dialplan/outbound.xml > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> >> >>>> expression="^(\d+)$"> > >> >>>> > >> >>>> > >> >>>> > >> >>>> >> >>>> data="sofia/gateway/provider-2/$1"/> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> > _________________________________________________________________________ > >> >>>> Professional FreeSWITCH Consulting Services: > >> >>>> consulting at freeswitch.org > >> >>>> http://www.freeswitchsolutions.com > >> >>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> Official FreeSWITCH Sites > >> >>>> http://www.freeswitch.org > >> >>>> http://wiki.freeswitch.org > >> >>>> http://www.cluecon.com > >> >>>> > >> >>>> FreeSWITCH-users mailing list > >> >>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>> > >> >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>> http://www.freeswitch.org > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > _________________________________________________________________________ > >> >>> Professional FreeSWITCH Consulting Services: > >> >>> consulting at freeswitch.org > >> >>> http://www.freeswitchsolutions.com > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> Official FreeSWITCH Sites > >> >>> http://www.freeswitch.org > >> >>> http://wiki.freeswitch.org > >> >>> http://www.cluecon.com > >> >>> > >> >>> FreeSWITCH-users mailing list > >> >>> FreeSWITCH-users at lists.freeswitch.org > >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>> > >> >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>> http://www.freeswitch.org > >> >>> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/be12a872/attachment-0001.html From lazyvirus at gmx.com Fri Mar 16 02:54:37 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 16 Mar 2012 00:54:37 +0100 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> Message-ID: <20120316005437.478d0a74@anubis.defcon1> On Thu, 15 Mar 2012 11:39:45 -0700 Mitch Capper wrote: There's definitely something wrong in my conf: I restarted from the last working one (w/o tls nor srtp) to make sure my base was clear. I followed the wiki (http://wiki.freeswitch.org/wiki/SIP_TLS): * generated certs: CA & svr, * modified conf/vars (enabling port 5081 only), * restart FS netstat -pan|grep swit tcp 0 0 0.0.0.0:8787 0.0.0.0:* LISTEN 10576/freeswitch tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 10576/freeswitch tcp 0 0 192.168.1.25:5060 0.0.0.0:* LISTEN 10576/freeswitch tcp 0 0 127.0.0.1:8021 127.0.0.1:57654 ESTABLISHED 10576/freeswitch tcp6 0 0 ::1:5060 :::* LISTEN 10576/freeswitch udp 0 0 192.168.1.25:5060 0.0.0.0:* 10576/freeswitch udp 0 0 192.168.1.25:53226 192.168.1.1:5351 ESTABLISHED 10576/freeswitch udp6 0 0 ::1:5060 :::* 10576/freeswitch no more 5080 port, and of course no 5081. A "reload mod_sofia" agrees w/ my conf: tls false for 5061 & true for 5081. Could it be linked to the fact I use fusionpbx, or what else (apart nescaf?:)? JY -- The first guy that rats gets a belly-full of slugs in the head. Understand? -- Joey Glimco, trade unionist From mitch.capper at gmail.com Fri Mar 16 03:25:19 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Thu, 15 Mar 2012 17:25:19 -0700 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: <20120316005437.478d0a74@anubis.defcon1> References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> Message-ID: Post your sofia profile xml file and the log to pb we can take a look from there. ~Mitch From lazyvirus at gmx.com Fri Mar 16 04:00:11 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 16 Mar 2012 02:00:11 +0100 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> Message-ID: <20120316020011.00b15d5f@anubis.defcon1> On Thu, 15 Mar 2012 17:25:19 -0700 Mitch Capper wrote: > Post your sofia profile xml file I'm not really sure of what you want nor where to get it, so I attached conf/autoload_configs/sofia.conf.xml (?) I also attached the only file I see connected to one of my accounts: conf/directory/default/v_01.xml > and the log to pb we can take a look > from there. I don't know what "log to pb" means, so I attached the result of reload mod_sofia. JY -- -------------- next part -------------- A non-text attachment was scrubbed... Name: sofia.conf.xml Type: application/xml Size: 596 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/fa7383e1/attachment-0002.rdf -------------- next part -------------- A non-text attachment was scrubbed... Name: v_01.xml Type: application/xml Size: 994 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/fa7383e1/attachment-0003.rdf -------------- next part -------------- A non-text attachment was scrubbed... Name: FS.LOG Type: text/x-log Size: 21971 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/fa7383e1/attachment-0001.bin From bdfoster at endigotech.com Fri Mar 16 04:10:02 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 15 Mar 2012 21:10:02 -0400 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: <20120316020011.00b15d5f@anubis.defcon1> References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> Message-ID: We need to see the profile configuration file. See /usr/local/freeswitch/conf/sip_profiles/internal.xml Also 'pb' is the pastebin, so pastebin everything to http://pastebin.freeswitch.org. pb everything separately and give links and the description in your reply so we can decode what we are seeing. On Mar 15, 2012 9:01 PM, "Bzzz" wrote: > On Thu, 15 Mar 2012 17:25:19 -0700 > Mitch Capper wrote: > > > Post your sofia profile xml file > > I'm not really sure of what you want nor where to get it, so I > attached conf/autoload_configs/sofia.conf.xml (?) > > I also attached the only file I see connected to one of my accounts: > conf/directory/default/v_01.xml > > > and the log to pb we can take a look > > from there. > > I don't know what "log to pb" means, so I attached the result of > reload mod_sofia. > > JY > -- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/9f03f272/attachment.html From joe.jflemmings at gmail.com Fri Mar 16 04:26:09 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Thu, 15 Mar 2012 18:26:09 -0700 Subject: [Freeswitch-users] Record Session In-Reply-To: References: Message-ID: Comment sense dictates that I should have thought of that before asking the question but the short answer is that this should only occur in some recordings (NOT ALL) and is decided during routing. On Thu, Mar 15, 2012 at 4:35 PM, Michael Collins wrote: > What possible advantage is there to recording to two separate directories > at the same time? Would it not be better just to copy the recording once > the call is over? > > -MC > > On Thu, Mar 15, 2012 at 2:57 PM, Joe Flemmings wrote: > >> How can i record a call in two seperate directories at the same time >> >> the following does not work >> >> > data="$${base_dir}/recordings/admin/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> >> > data="$${base_dir}/recordings/users/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> >> >> Joe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/a390c7ff/attachment.html From lazyvirus at gmx.com Fri Mar 16 04:33:31 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 16 Mar 2012 02:33:31 +0100 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> Message-ID: <20120316023331.63dd8e9e@anubis.defcon1> On Thu, 15 Mar 2012 21:10:02 -0400 Brian Foster wrote: > We need to see the profile configuration file. See > /usr/local/freeswitch/conf/sip_profiles/internal.xml > > Also 'pb' is the pastebin, so pastebin everything to > http://pastebin.freeswitch.org. pb everything separately and give links and > the description in your reply so we can decode what we are seeing. Ok, this is clear. Log part corresponding to "reload mod_sofia": http://pastebin.freeswitch.org/18650 conf/sip_profiles/internal.xml http://pastebin.freeswitch.org/18651 conf/vars.xml http://pastebin.freeswitch.org/18652 conf/directory/default.xml http://pastebin.freeswitch.org/18655 conf/directory/default/v_01.xml http://pastebin.freeswitch.org/18654 JY -- From mitch.capper at gmail.com Fri Mar 16 04:33:17 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Thu, 15 Mar 2012 18:33:17 -0700 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: <20120316020011.00b15d5f@anubis.defcon1> References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> Message-ID: Brian is right on his comments, and the internal profile was missing but looking at the log I did notice a few things: It looks like TLS is enabled on your public profile and should be listening on 5081 and disabled on the internal profile. Normally you would want these reversed tls is generally used for users registering to your FS box rather than your public profile. With that said if you could run "sofia status" and then the "netstat -nlp | grep frees". Thanks ~Mitch From avi at avimarcus.net Fri Mar 16 04:33:56 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 16 Mar 2012 03:33:56 +0200 Subject: [Freeswitch-users] Record Session In-Reply-To: References: Message-ID: A workaround would be setting a hangup hook for a script to copy (or hardlink) the files when the call is over. Or have your CDR handler find a channel variable set and do it after the call. -Avi On Fri, Mar 16, 2012 at 3:26 AM, Joe Flemmings wrote: > > Comment sense dictates that I should have thought of that before asking > the question but the short answer is that this should only occur in some > recordings (NOT ALL) and is decided during routing. > > > On Thu, Mar 15, 2012 at 4:35 PM, Michael Collins wrote: > >> What possible advantage is there to recording to two separate directories >> at the same time? Would it not be better just to copy the recording once >> the call is over? >> >> -MC >> >> On Thu, Mar 15, 2012 at 2:57 PM, Joe Flemmings wrote: >> >>> How can i record a call in two seperate directories at the same time >>> >>> the following does not work >>> >>> >> data="$${base_dir}/recordings/admin/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> >>> >> data="$${base_dir}/recordings/users/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> >>> >>> Joe >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/5fc713af/attachment-0001.html From mitch.capper at gmail.com Fri Mar 16 04:35:05 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Thu, 15 Mar 2012 18:35:05 -0700 Subject: [Freeswitch-users] Record Session In-Reply-To: References: Message-ID: It may be better to set a variable during routing for if the recording should be duplicated or not and then copy after the fact. The other option may be to symlink rather than create two recordings, this could be done upfront and has the nice side effect of saving space. ~mitch From mitch.capper at gmail.com Fri Mar 16 04:41:44 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Thu, 15 Mar 2012 18:41:44 -0700 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: <20120316023331.63dd8e9e@anubis.defcon1> References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> <20120316023331.63dd8e9e@anubis.defcon1> Message-ID: To confirm are you trying to enable tls on your public profile (5080/5081) or internal profile (5060/5061) Also in looking at the vars.xml you cannot comment out X-PRE-PROCESS lines. You must break the X-PRE-PROCESS tag up, as its actually handled BEFORE xml comments are taken into effect. I think you were noticing this oddness before and so the best way is to ensure you alter the tagname itself to fix that. ~Mitch From lazyvirus at gmx.com Fri Mar 16 04:46:55 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 16 Mar 2012 02:46:55 +0100 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> Message-ID: <20120316024655.05f32b43@anubis.defcon1> On Thu, 15 Mar 2012 18:33:17 -0700 Mitch Capper wrote: > Brian is right on his comments, and the internal profile was missing don't forget I'm a beginner to FS. > but looking at the log I did notice a few things: > It looks like TLS is enabled on your public profile and should be > listening on 5081 and disabled on the internal profile. Yep, as it is a test, I changed the first one in vars.xml > Normally you > would want these reversed tls is generally used for users registering > to your FS box rather than your public profile. With that said if you > could run "sofia status" and then the "netstat -nlp | grep frees". I read that external is generally used to connect to a VoIP provider, but I thought I could use it for friends elsewhere connecting to my svr by the Internet (?) Is it a mistake to separate LAN & WAN registrations, and shall I leave external alone except if some day I connect to a VoIP provider? JY -- A man who cannot seduce men cannot save them either. -- Soren Kierkegaard From lazyvirus at gmx.com Fri Mar 16 04:57:34 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 16 Mar 2012 02:57:34 +0100 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> <20120316023331.63dd8e9e@anubis.defcon1> Message-ID: <20120316025734.4d45769b@anubis.defcon1> On Thu, 15 Mar 2012 18:41:44 -0700 Mitch Capper wrote: > To confirm are you trying to enable tls on your public profile > (5080/5081) or internal profile (5060/5061) Well, it depends on your answer to my last post. > Also in looking at the vars.xml you cannot comment out X-PRE-PROCESS lines. > > You must break the X-PRE-PROCESS tag up, as its actually handled > BEFORE xml comments are taken into effect. I think you were noticing > this oddness before and so the best way is to ensure you alter the > tagname itself to fix that. This one's really weird! (I pasted your post into the file to make sure I won't ever forget that.) But it don't change anything as the commented line is located before tls activation, so FS first read a false, then a true (except if you tell me it only read once a parm, which would be VERY weird.) -- * Overfiend prefers girls who have developed, and DON'T FUCKING **GIGGLE** From joe.jflemmings at gmail.com Fri Mar 16 05:26:47 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Thu, 15 Mar 2012 19:26:47 -0700 Subject: [Freeswitch-users] Record Session In-Reply-To: References: Message-ID: Thanks guys for the input. On Thu, Mar 15, 2012 at 6:35 PM, Mitch Capper wrote: > It may be better to set a variable during routing for if the recording > should be duplicated or not and then copy after the fact. The other > option may be to symlink rather than create two recordings, this could > be done upfront and has the nice side effect of saving space. > > ~mitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/2a21eade/attachment.html From cmrienzo at gmail.com Fri Mar 16 05:55:10 2012 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Thu, 15 Mar 2012 22:55:10 -0400 Subject: [Freeswitch-users] Record Session In-Reply-To: References: Message-ID: This seems broken to me if it doesn't work. I'm pretty sure I've enabled multiple recordings before and I looked at the code again to be sure... a separate media bug gets created for each recording. What error do you see in the logs? On Thu, Mar 15, 2012 at 10:26 PM, Joe Flemmings wrote: > Thanks guys for the input. > > > On Thu, Mar 15, 2012 at 6:35 PM, Mitch Capper wrote: > >> It may be better to set a variable during routing for if the recording >> should be duplicated or not and then copy after the fact. The other >> option may be to symlink rather than create two recordings, this could >> be done upfront and has the nice side effect of saving space. >> >> ~mitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/7f7f5041/attachment.html From mitch.capper at gmail.com Fri Mar 16 06:34:15 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Thu, 15 Mar 2012 20:34:15 -0700 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: <20120316025734.4d45769b@anubis.defcon1> References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> <20120316023331.63dd8e9e@anubis.defcon1> <20120316025734.4d45769b@anubis.defcon1> Message-ID: No you double set most of the x-pre-process items so I didn't see any issues but its a very easy way to make a mistake by trying to comment them out:) It gets weirder if you try and comment out an include x-pre-process header as it will still include everything. Public certainly can be for remote users but generally public means not-authed (and generally less reason to be encrypted) but as you are just trying to test lets leave it as it is. I still however need: It looks like TLS is enabled on your public profile and should be listening on 5081 and disabled on the internal profile. With that said if you could run "sofia status" at the freeswitch console and then the "netstat -nlp | grep frees" and pb the results. ~Mitch From joe.jflemmings at gmail.com Fri Mar 16 07:56:34 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Thu, 15 Mar 2012 21:56:34 -0700 Subject: [Freeswitch-users] Record Session In-Reply-To: References: Message-ID: Actually you're right, what i should have said is that it does work when you set in "execute_on_answer" eg in a lua script This Works and records in both places --------------------------------------------------------- session:execute("record_session", "$${base_dir}/recordings/admin/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") session:execute("record_session", "$${base_dir}/recordings/user/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") This only records the second option and not the first ------------------------------------------------------------------------------- session:execute("export", "nolocal:execute_on_answer=record_session $${base_dir}/recordings/admin/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") session:execute("export", "nolocal:execute_on_answer=record_session $${base_dir}/recordings/user/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") On Thu, Mar 15, 2012 at 7:55 PM, Christopher Rienzo wrote: > This seems broken to me if it doesn't work. I'm pretty sure I've enabled > multiple recordings before and I looked at the code again to be sure... a > separate media bug gets created for each recording. What error do you see > in the logs? > > > On Thu, Mar 15, 2012 at 10:26 PM, Joe Flemmings wrote: > >> Thanks guys for the input. >> >> >> On Thu, Mar 15, 2012 at 6:35 PM, Mitch Capper wrote: >> >>> It may be better to set a variable during routing for if the recording >>> should be duplicated or not and then copy after the fact. The other >>> option may be to symlink rather than create two recordings, this could >>> be done upfront and has the nice side effect of saving space. >>> >>> ~mitch >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120315/e02f1356/attachment-0001.html From virbhati at gmail.com Fri Mar 16 08:07:14 2012 From: virbhati at gmail.com (virendra bhati) Date: Fri, 16 Mar 2012 10:37:14 +0530 Subject: [Freeswitch-users] Any help me to installed Luac compiler Message-ID: Hi list, I don't find any link on which luac compiler is mention. only one link was found on which description about Luac was mention. http://www.lua.org/manual/4.0/luac.html Anyone know how to installed Luac compiler in Centos ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati at gmail.com Skype id:- virbhati2 Hyderabad(India) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/e8b10c7d/attachment.html From avi at avimarcus.net Fri Mar 16 12:19:28 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 16 Mar 2012 11:19:28 +0200 Subject: [Freeswitch-users] Record Session In-Reply-To: References: Message-ID: That makes more sense. The first is running an application, so it creates the media bug. The second sets a variable... so the second time you set it, it overwrites it. You can set multiple execute_on by adding _n to it -- see: http://wiki.freeswitch.org/wiki/Channel_Variables#The_execute_on_family -Avi On Fri, Mar 16, 2012 at 6:56 AM, Joe Flemmings wrote: > Actually you're right, what i should have said is that it does work when > you set in "execute_on_answer" eg in a lua script > > This Works and records in both places > --------------------------------------------------------- > session:execute("record_session", > "$${base_dir}/recordings/admin/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") > session:execute("record_session", > "$${base_dir}/recordings/user/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") > > This only records the second option and not the first > > ------------------------------------------------------------------------------- > session:execute("export", "nolocal:execute_on_answer=record_session > $${base_dir}/recordings/admin/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") > session:execute("export", "nolocal:execute_on_answer=record_session > $${base_dir}/recordings/user/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") > > > > > > On Thu, Mar 15, 2012 at 7:55 PM, Christopher Rienzo wrote: > >> This seems broken to me if it doesn't work. I'm pretty sure I've enabled >> multiple recordings before and I looked at the code again to be sure... a >> separate media bug gets created for each recording. What error do you see >> in the logs? >> >> >> On Thu, Mar 15, 2012 at 10:26 PM, Joe Flemmings > > wrote: >> >>> Thanks guys for the input. >>> >>> >>> On Thu, Mar 15, 2012 at 6:35 PM, Mitch Capper wrote: >>> >>>> It may be better to set a variable during routing for if the recording >>>> should be duplicated or not and then copy after the fact. The other >>>> option may be to symlink rather than create two recordings, this could >>>> be done upfront and has the nice side effect of saving space. >>>> >>>> ~mitch >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/c7c6b84d/attachment.html From miha at softnet.si Fri Mar 16 12:20:08 2012 From: miha at softnet.si (Miha) Date: Fri, 16 Mar 2012 10:20:08 +0100 Subject: [Freeswitch-users] blocking destination number In-Reply-To: References: <4F5A1591.8090703@softnet.si> Message-ID: <4F6305C8.1070509@softnet.si> Hi @Gabriel, I was looking on wiki how to define new xml or document where I will have my restricted numbers. Where are domestic,international,local defined? Thanks! Miha On 3/10/2012 11:23 PM, Gabriel Gunderson wrote: > On Fri, Mar 9, 2012 at 7:37 AM, Miha Zoubek wrote: >> what is the bast way to block destination number for certain user. >> Is it possible to do it in user/dir? > Ultimately, it would be a dialplan configuration, but you could set > the variable that you match in the directory. See the example of > 'toll_allow' in the default FreeSWITCH configuration... > > > This is set on a per-user basis in the DIRECTORY: > > > > > > But it's considered while evaluating the DIALPLAN: > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_id_name}"/> > data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> > > > > > > You could easily do something similar by having a list of numbers that > they can't call listed in the directory (maybe something like > 'restricted_numbers') and check to make sure the destination doesn't > match it in the dialplan. > > Good luck! > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From lazyvirus at gmx.com Fri Mar 16 15:01:02 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 16 Mar 2012 13:01:02 +0100 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> <20120316023331.63dd8e9e@anubis.defcon1> <20120316025734.4d45769b@anubis.defcon1> Message-ID: <20120316130102.07b63e85@anubis.defcon1> On Thu, 15 Mar 2012 20:34:15 -0700 Mitch Capper wrote: > No you double set most of the x-pre-process items so I didn't see any > issues but its a very easy way to make a mistake by trying to comment > them out:) It gets weirder if you try and comment out an include > x-pre-process header as it will still include everything. According to a precedent post of yours, my test shows that breaking the x-pre-process header in 2 pieces works. > Public certainly can be for remote users but generally public means > not-authed (and generally less reason to be encrypted) but as you are > just trying to test lets leave it as it is. Hmm, this one will be entirely private (may be there will be a PSTN exhaust, but here through an ATA, not from a provider); it'll also be a way to make very private conferences. > I still however need: > It looks like TLS is enabled on your public profile and should be > listening on 5081 and disabled on the internal profile. Yep, this is what I want I just modified 1 line in vars.xml to do so. BTW, whether conf/directory/default.xml is modified or not don't have any impact: ASA vars.xml is modified, it disable the tls authorized branch. > With that said if you > could run "sofia status" at the freeswitch console and then the > "netstat -nlp | grep frees" and pb the results. done sofia status: http://pastebin.freeswitch.org/18665 I PB both results, just in case: netstat -pan|grep swi: http://pastebin.freeswitch.org/18666 netstat -pln|grep swi: http://pastebin.freeswitch.org/18668 JY -- The best way to avoid responsibility is to say, "I've got responsibilities." From lazyvirus at gmx.com Fri Mar 16 15:12:32 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 16 Mar 2012 13:12:32 +0100 Subject: [Freeswitch-users] Any help me to installed Luac compiler In-Reply-To: References: Message-ID: <20120316131232.79b66933@anubis.defcon1> On Fri, 16 Mar 2012 10:37:14 +0530 virendra bhati wrote: > I don't find any link on which luac compiler is mention. only one link was > found on which description about Luac was mention. > http://www.lua.org/manual/4.0/luac.html > > Anyone know how to installed Luac compiler in Centos ? On Debian, it is named "luajit" (w/ libluajit-xxx), as RH use is to pack all together, may be it is already installed; but you really don't need it because LUA is a very fast interpreter by construction. Unless you have big & complicated code, you won't notice any difference between interpreted & compiled - not to mention it is much easier to make a modification on a readable code. JY -- [Babe] Ruth made a big mistake when he gave up pitching. -- Tris Speaker, 1921 From cmrienzo at gmail.com Fri Mar 16 15:22:00 2012 From: cmrienzo at gmail.com (cmrienzo at gmail.com) Date: Fri, 16 Mar 2012 08:22:00 -0400 Subject: [Freeswitch-users] Record Session In-Reply-To: References: Message-ID: <26B5A7D4-618F-44C6-97A0-28025A7F1DC9@gmail.com> Yes, that makes sense. Execute_on_answer is a channel variable and will retain the last value set to it. You could execute_on_answer a script instead which starts both recordings. On Mar 16, 2012, at 0:56, Joe Flemmings wrote: > Actually you're right, what i should have said is that it does work when you set in "execute_on_answer" eg in a lua script > > This Works and records in both places > --------------------------------------------------------- > session:execute("record_session", "$${base_dir}/recordings/admin/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") > session:execute("record_session", "$${base_dir}/recordings/user/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") > > This only records the second option and not the first > ------------------------------------------------------------------------------- > session:execute("export", "nolocal:execute_on_answer=record_session $${base_dir}/recordings/admin/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") > session:execute("export", "nolocal:execute_on_answer=record_session $${base_dir}/recordings/user/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") > > > > > On Thu, Mar 15, 2012 at 7:55 PM, Christopher Rienzo wrote: > This seems broken to me if it doesn't work. I'm pretty sure I've enabled multiple recordings before and I looked at the code again to be sure... a separate media bug gets created for each recording. What error do you see in the logs? > > > On Thu, Mar 15, 2012 at 10:26 PM, Joe Flemmings wrote: > Thanks guys for the input. > > > On Thu, Mar 15, 2012 at 6:35 PM, Mitch Capper wrote: > It may be better to set a variable during routing for if the recording > should be duplicated or not and then copy after the fact. The other > option may be to symlink rather than create two recordings, this could > be done upfront and has the nice side effect of saving space. > > ~mitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/33c6cea4/attachment.html From avi at avimarcus.net Fri Mar 16 15:29:25 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 16 Mar 2012 14:29:25 +0200 Subject: [Freeswitch-users] Record Session In-Reply-To: <26B5A7D4-618F-44C6-97A0-28025A7F1DC9@gmail.com> References: <26B5A7D4-618F-44C6-97A0-28025A7F1DC9@gmail.com> Message-ID: or execute_on_answer_1 and execute_on_answer_2 as I linked to.. -Avi On Fri, Mar 16, 2012 at 2:22 PM, wrote: > Yes, that makes sense. Execute_on_answer is a channel variable and will > retain the last value set to it. You could execute_on_answer a script > instead which starts both recordings. > > > > On Mar 16, 2012, at 0:56, Joe Flemmings wrote: > > Actually you're right, what i should have said is that it does work when > you set in "execute_on_answer" eg in a lua script > > This Works and records in both places > --------------------------------------------------------- > session:execute("record_session", > "$${base_dir}/recordings/admin/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") > session:execute("record_session", > "$${base_dir}/recordings/user/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") > > This only records the second option and not the first > > ------------------------------------------------------------------------------- > session:execute("export", "nolocal:execute_on_answer=record_session > $${base_dir}/recordings/admin/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") > session:execute("export", "nolocal:execute_on_answer=record_session > $${base_dir}/recordings/user/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") > > > > > On Thu, Mar 15, 2012 at 7:55 PM, Christopher Rienzo wrote: > >> This seems broken to me if it doesn't work. I'm pretty sure I've enabled >> multiple recordings before and I looked at the code again to be sure... a >> separate media bug gets created for each recording. What error do you see >> in the logs? >> >> >> On Thu, Mar 15, 2012 at 10:26 PM, Joe Flemmings > > wrote: >> >>> Thanks guys for the input. >>> >>> >>> On Thu, Mar 15, 2012 at 6:35 PM, Mitch Capper wrote: >>> >>>> It may be better to set a variable during routing for if the recording >>>> should be duplicated or not and then copy after the fact. The other >>>> option may be to symlink rather than create two recordings, this could >>>> be done upfront and has the nice side effect of saving space. >>>> >>>> ~mitch >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/9a4ffb18/attachment-0001.html From freeswitch-users at digitaldan.com Fri Mar 16 16:25:36 2012 From: freeswitch-users at digitaldan.com (Dan) Date: Fri, 16 Mar 2012 07:25:36 -0600 (MDT) Subject: [Freeswitch-users] Problem getting TALK and NOTALK events In-Reply-To: <2494bf95-f5f6-4829-8155-cc762f8d9913@radio> Message-ID: <64256728-687c-408c-9c98-b3a045864462@radio> Thanks for the response, I went ahead and subscribed to all events, but am still not getting these events. I have also enabled in my sip profile, not sure if this is redundant or not with the dial plan directives. If VAD is indeed enabled but not working, is there something else I should try? Thanks. ----- Original Message ----- From: "Michael Collins" To: "FreeSWITCH Users Help" Sent: Thursday, March 15, 2012 5:29:21 PM Subject: Re: [Freeswitch-users] Problem getting TALK and NOTALK events Can you confirm if the TALK/NOTALK events are even firing? Try listening to all events and then sifting through the barrage of data to see if those events are present. If they are then you know you just have a filtering issue. If they are not present then you know there's an issue with VAD or something like that. -MC On Thu, Mar 15, 2012 at 8:59 AM, Dan < freeswitch-users at digitaldan.com > wrote: Hi, I'm having some issues getting TALK / NOTALK events to fire on an incoming stream to Freeswitch. In my ESL application I am subscribing to RECORD_START RECORD_STOP TALK NOTALK, Below is the dial plan I am using: I can see that VAD is enabled: 2012-03-15 09:49:25.523832 [DEBUG] switch_rtp.c:4130 Activate VAD codec PCMU 20ms 2012-03-15 09:49:25.523832 [DEBUG] sofia_glue.c:3353 AUDIO RTP Engage VAD for sofia/external/ 3035551212 at 10.10.10.1 ( in out ) In my ESL app I get the RECORD_START and RECORD_STOP but not the talk events. I'm on git version "2c52f23 2012-02-18 08:37:47 -0600", Any ideas? Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/47e4caa5/attachment.html From mitch.capper at gmail.com Fri Mar 16 17:35:17 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Fri, 16 Mar 2012 07:35:17 -0700 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: <20120316130102.07b63e85@anubis.defcon1> References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> <20120316023331.63dd8e9e@anubis.defcon1> <20120316025734.4d45769b@anubis.defcon1> <20120316130102.07b63e85@anubis.defcon1> Message-ID: Well something is certainly going wrong s sofia isn't showing it listening. Shutdown freeswitch and run: export SOFIA_DEBUG=9 export NUA_DEBUG=9 export NTA_DEBUG=9 export TPORT_DEBUG=9 Then start freeswitch enable console debug (F8), then do reload mod_sofia and PB the log of what happens. From lazyvirus at gmx.com Fri Mar 16 17:59:11 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 16 Mar 2012 15:59:11 +0100 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> <20120316023331.63dd8e9e@anubis.defcon1> <20120316025734.4d45769b@anubis.defcon1> <20120316130102.07b63e85@anubis.defcon1> Message-ID: <20120316155911.61361c42@anubis.defcon1> On Fri, 16 Mar 2012 07:35:17 -0700 Mitch Capper wrote: > Well something is certainly going wrong s sofia isn't showing it > listening. Shutdown freeswitch and run: > export SOFIA_DEBUG=9 > export NUA_DEBUG=9 > export NTA_DEBUG=9 > export TPORT_DEBUG=9 > > Then start freeswitch enable console debug (F8), then do reload In fact, when I hit F8 it says log level's 7; but issuing log level 9 worked - log's a bit hard to recover as it almost continuously scroll: is there a way to temporarily redirect the console output to a file? > mod_sofia and PB the log of what happens. but here we go: http://pastebin.freeswitch.org/18671 -- It is more rational to sacrifice one life than six. -- Spock, "The Galileo Seven", stardate 2822.3 From mitch.capper at gmail.com Fri Mar 16 17:59:39 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Fri, 16 Mar 2012 07:59:39 -0700 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> <20120316023331.63dd8e9e@anubis.defcon1> <20120316025734.4d45769b@anubis.defcon1> <20120316130102.07b63e85@anubis.defcon1> Message-ID: saw your PB and your issue. FS will drop priv by default and I am guessing the freeswitch user can't read the keys you generated, apply proper permissions to them and you should be in good shape: tport_tls_init_master(0x92a3f78): tls key = /usr/local/freeswitch/conf/ssl/agent.pem tls_init_context: invalid local certificate: /usr/local/freeswitch/conf/ssl/agent.pem tls_init_context: 0200100d:system library:fopen:Permission denied tls_init_context: 20074002:BIO routines:FILE_CTRL:system lib tls_init_context: 140ad002:SSL routines:SSL_CTX_use_certificate_file:system lib tls_init_context: invalid private key: /usr/local/freeswitch/conf/ssl/agent.pem tls_init_context(key): 0200100d:system library:fopen:Permission denied ~Mitch From mitch.capper at gmail.com Fri Mar 16 18:10:11 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Fri, 16 Mar 2012 08:10:11 -0700 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: <20120316155911.61361c42@anubis.defcon1> References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> <20120316023331.63dd8e9e@anubis.defcon1> <20120316025734.4d45769b@anubis.defcon1> <20120316130102.07b63e85@anubis.defcon1> <20120316155911.61361c42@anubis.defcon1> Message-ID: There is http://wiki.freeswitch.org/wiki/Fs_logger.pl which can capture debug output and post it to PB or a file. It currently requires you to apply a patch from http://jira.freeswitch.org/browse/FS-3188 but that may change soon. ~Mitch From lazyvirus at gmx.com Fri Mar 16 18:17:47 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 16 Mar 2012 16:17:47 +0100 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> <20120316023331.63dd8e9e@anubis.defcon1> <20120316025734.4d45769b@anubis.defcon1> <20120316130102.07b63e85@anubis.defcon1> Message-ID: <20120316161747.6224c7b3@anubis.defcon1> On Fri, 16 Mar 2012 07:59:39 -0700 Mitch Capper wrote: > saw your PB and your issue. FS will drop priv by default and I am > guessing the freeswitch user can't read the keys you generated, apply > proper permissions to them and you should be in good shape: > tport_tls_init_master(0x92a3f78): tls key = > /usr/local/freeswitch/conf/ssl/agent.pem > tls_init_context: invalid local certificate: > /usr/local/freeswitch/conf/ssl/agent.pem > tls_init_context: 0200100d:system library:fopen:Permission denied > tls_init_context: 20074002:BIO routines:FILE_CTRL:system lib > tls_init_context: 140ad002:SSL routines:SSL_CTX_use_certificate_file:system lib > tls_init_context: invalid private key: /usr/local/freeswitch/conf/ssl/agent.pem > tls_init_context(key): 0200100d:system library:fopen:Permission denied Yep, I also saw it. My mistake was to think FS was running under www-data:www-data (this because of fusionpbx) when it is running under www-data:nogroup, and as conf/ssl was root:www-data (perms 42740), it was impossible for it to read anything. I chown -R www-data ssl/, restarted and... its aliiiveee! Thanks a lot for debugging me:) BTW, log level 9 mean an almost continuous flow of data and it took me 4 times to correctly catch the reload logs; so, is there a way to redirect temporarily the console output to a file? JY -- QOTD: "The only real difference between men and women is that men are crabby all month long." From lazyvirus at gmx.com Fri Mar 16 18:31:34 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 16 Mar 2012 16:31:34 +0100 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> <20120316023331.63dd8e9e@anubis.defcon1> <20120316025734.4d45769b@anubis.defcon1> <20120316130102.07b63e85@anubis.defcon1> <20120316155911.61361c42@anubis.defcon1> Message-ID: <20120316163134.74fc0d0e@anubis.defcon1> On Fri, 16 Mar 2012 08:10:11 -0700 Mitch Capper wrote: > There is http://wiki.freeswitch.org/wiki/Fs_logger.pl which can > capture debug output and post it to PB or a file. It currently > requires you to apply a patch from > http://jira.freeswitch.org/browse/FS-3188 but that may change soon. Ok, I'll give it a try, thanks. JY -- Your canceled check is your receipt. From lazyvirus at gmx.com Fri Mar 16 18:44:14 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 16 Mar 2012 16:44:14 +0100 Subject: [Freeswitch-users] sips, tls, srtp, etc In-Reply-To: References: <20120315163532.7507ef73@anubis.defcon1> <20120315185012.25862a47@anubis.defcon1> <20120316005437.478d0a74@anubis.defcon1> <20120316020011.00b15d5f@anubis.defcon1> <20120316023331.63dd8e9e@anubis.defcon1> <20120316025734.4d45769b@anubis.defcon1> <20120316130102.07b63e85@anubis.defcon1> <20120316155911.61361c42@anubis.defcon1> Message-ID: <20120316164414.5189a5ba@anubis.defcon1> On Fri, 16 Mar 2012 08:10:11 -0700 Mitch Capper wrote: BTW, this has nothing to do with tls, but I found some thing interesting I don't really fully understand (because one is C++ and the other Java): after installing jitsi (ex sip-communicator), Twinkle can now use ZRTP !? JY -- Fidelity, n.: A virtue peculiar to those who are about to be betrayed. From joe.jflemmings at gmail.com Fri Mar 16 19:19:30 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Fri, 16 Mar 2012 09:19:30 -0700 Subject: [Freeswitch-users] Record Session In-Reply-To: References: <26B5A7D4-618F-44C6-97A0-28025A7F1DC9@gmail.com> Message-ID: Thanks Avi, that's in-fact what i was looking for. On Fri, Mar 16, 2012 at 5:29 AM, Avi Marcus wrote: > or execute_on_answer_1 and execute_on_answer_2 as I linked to.. > -Avi > > > On Fri, Mar 16, 2012 at 2:22 PM, wrote: > >> Yes, that makes sense. Execute_on_answer is a channel variable and will >> retain the last value set to it. You could execute_on_answer a script >> instead which starts both recordings. >> >> >> >> On Mar 16, 2012, at 0:56, Joe Flemmings wrote: >> >> Actually you're right, what i should have said is that it does work when >> you set in "execute_on_answer" eg in a lua script >> >> This Works and records in both places >> --------------------------------------------------------- >> session:execute("record_session", >> "$${base_dir}/recordings/admin/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") >> session:execute("record_session", >> "$${base_dir}/recordings/user/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") >> >> This only records the second option and not the first >> >> ------------------------------------------------------------------------------- >> session:execute("export", "nolocal:execute_on_answer=record_session >> $${base_dir}/recordings/admin/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") >> session:execute("export", "nolocal:execute_on_answer=record_session >> $${base_dir}/recordings/user/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav") >> >> >> >> >> On Thu, Mar 15, 2012 at 7:55 PM, Christopher Rienzo wrote: >> >>> This seems broken to me if it doesn't work. I'm pretty sure I've >>> enabled multiple recordings before and I looked at the code again to be >>> sure... a separate media bug gets created for each recording. What error >>> do you see in the logs? >>> >>> >>> On Thu, Mar 15, 2012 at 10:26 PM, Joe Flemmings < >>> joe.jflemmings at gmail.com> wrote: >>> >>>> Thanks guys for the input. >>>> >>>> >>>> On Thu, Mar 15, 2012 at 6:35 PM, Mitch Capper wrote: >>>> >>>>> It may be better to set a variable during routing for if the recording >>>>> should be duplicated or not and then copy after the fact. The other >>>>> option may be to symlink rather than create two recordings, this could >>>>> be done upfront and has the nice side effect of saving space. >>>>> >>>>> ~mitch >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/aae9deff/attachment.html From mstockton at harqen.com Fri Mar 16 19:59:58 2012 From: mstockton at harqen.com (Matt Stockton) Date: Fri, 16 Mar 2012 09:59:58 -0700 Subject: [Freeswitch-users] Getting real-time amplitude of a channel In-Reply-To: References: Message-ID: Hi Michael, Thanks for the info!! Is there information I can find on how the bounties work? I'm interesting in getting this feature, but it would probably take me much longer to implement than someone who knows the code well. I see a bounty wiki entry...is that active? or is there a better place to post such things? Thanks, Matt On Wed, Mar 14, 2012 at 12:30 PM, Michael Collins wrote: > AFAIK there isn't anything pre-rolled that you can use. However, I know > that Tony's auto gain control (agc) logic checks the "volume" level coming > in on a channel. You might be able to look in mod_conference.c and see > where the agc calculates the incoming volume level and write an API that > displays that value. > > I'd start by looking at the function check_agc_levels() and for references > to member->agc_volume_in_level. Hopefully that will give you a place to > start. > > -MC > > On Wed, Mar 14, 2012 at 10:26 AM, Matt Stockton wrote: > >> Hi all, >> >> I am interested in providing a visualization of a channel in a >> conference. I am able to process the start talking and stop talking events >> from the conference, but I'm wondering, has anyone done anything that can >> get the real-time amplitude of a channel? I didn't see any built-in API >> commands to do this, but wondering if there's any modules or something else >> that I'm overlooking? Or maybe now is the time for me to dig into the API >> documentation? >> >> Just curious if anyone has done something like this before, and if so, if >> you could guide me in the right direction. >> >> Thanks! >> Matt >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/01a885f3/attachment-0001.html From bdfoster at endigotech.com Fri Mar 16 19:58:27 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 16 Mar 2012 12:58:27 -0400 Subject: [Freeswitch-users] blocking destination number In-Reply-To: <4F6305C8.1070509@softnet.si> References: <4F5A1591.8090703@softnet.si> <4F6305C8.1070509@softnet.si> Message-ID: The toll_allow variable.would be defined in the actual user xml file located in /usr/local/freeswitch/conf/directory. Then you would evaluate that variable in the dialplan that would end up making the bridge to your gateway. -BDF On Mar 16, 2012 5:21 AM, "Miha" wrote: > Hi @Gabriel, > > I was looking on wiki how to define new xml or document where I will > have my restricted numbers. > Where are domestic,international,local defined? > > Thanks! > Miha > > On 3/10/2012 11:23 PM, Gabriel Gunderson wrote: > > On Fri, Mar 9, 2012 at 7:37 AM, Miha Zoubek wrote: > >> what is the bast way to block destination number for certain user. > >> Is it possible to do it in user/dir? > > Ultimately, it would be a dialplan configuration, but you could set > > the variable that you match in the directory. See the example of > > 'toll_allow' in the default FreeSWITCH configuration... > > > > > > This is set on a per-user basis in the DIRECTORY: > > > > > > > > > > > > But it's considered while evaluating the DIALPLAN: > > > > > > > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > > > data="effective_caller_id_name=${outbound_caller_id_name}"/> > > > data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> > > > > > > > > > > > > You could easily do something similar by having a list of numbers that > > they can't call listed in the directory (maybe something like > > 'restricted_numbers') and check to make sure the destination doesn't > > match it in the dialplan. > > > > Good luck! > > > > > > Gabe > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/0d3fbdca/attachment.html From jeff at jefflenk.com Fri Mar 16 20:46:10 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Fri, 16 Mar 2012 10:46:10 -0700 (PDT) Subject: [Freeswitch-users] Mod_flite wont load In-Reply-To: <1331842568647-7376926.post@n2.nabble.com> References: <1331842568647-7376926.post@n2.nabble.com> Message-ID: <1331919970183-7379705.post@n2.nabble.com> Please try git head and let me know if that works. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Mod-flite-wont-load-tp7376926p7379705.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch-list at puzzled.xs4all.nl Fri Mar 16 20:54:19 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Fri, 16 Mar 2012 18:54:19 +0100 Subject: [Freeswitch-users] Any help me to installed Luac compiler In-Reply-To: References: Message-ID: <4F637E4B.80201@puzzled.xs4all.nl> On 16-03-12 06:07, virendra bhati wrote: > Hi list, > > I don't find any link on which luac compiler is mention. only one link > was found on which description about Luac was mention. > http://www.lua.org/manual/4.0/luac.html > > Anyone know how to installed Luac compiler in Centos ? Can't you try a little harder to figure out these basic questions yourself? Did you even try to Google for "luac CentOS"? That would have given you a hint. Since your question is not a FreeSWITCH question but it is a lua question why not ask on the lua mailinglist or on the lua irc channel if they have one. Regards, Patrick From b2m at a-cti.com Fri Mar 16 21:04:23 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Fri, 16 Mar 2012 23:34:23 +0530 Subject: [Freeswitch-users] mod_shout issue Message-ID: Hi all, 2012-03-16 18:02:32.957534 [ERR] mod_shout.c:859 Error: MPG123 Error at /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:659. 2012-03-16 18:02:32.957534 [ERR] mod_shout.c:862 Error from mpg123: Invalid mpg123 handle. (code 10) Please let me know what Iam missing here. Thanks, Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/1aaabf36/attachment.html From erin.omeara at salmonbaytechnology.com Fri Mar 16 19:58:51 2012 From: erin.omeara at salmonbaytechnology.com (Erin O'Meara) Date: Fri, 16 Mar 2012 09:58:51 -0700 Subject: [Freeswitch-users] Strange VLAN problem with freeswitch Message-ID: I have moved offices down the street and extended my office network via vlan (old office was using VLAN also). I have two networks one internal and one wireless that I sell wireless service with. FusionPBX (In the cloud) -> Cable Modem -> PFSense -> VLAN -> Phone. my phone works fine for a few days on either network then calls stop working, if I switch the phone VLAN to the other VLAN it starts working again. I noticed this issue from my office that is connect of a Ubiquiti wireless link and VLAN extended to my new office. What happens is the calls ring and I can pickup but no audio in either direction. here is the pastebin of a call failing then I plugged the same phone into the other vlan and reboot and made a successful call. http://pastebin.freeswitch.org/18675 Regards, 206.905.9520 http://salmonbaytechnology.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/83591c05/attachment-0001.html From msc at freeswitch.org Fri Mar 16 23:39:55 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 16 Mar 2012 13:39:55 -0700 Subject: [Freeswitch-users] Getting real-time amplitude of a channel In-Reply-To: References: Message-ID: Post the bounty on the wiki and then shoot an email to the -dev list letting them know there's a new bounty available... -MC On Fri, Mar 16, 2012 at 9:59 AM, Matt Stockton wrote: > Hi Michael, > > Thanks for the info!! Is there information I can find on how the bounties > work? I'm interesting in getting this feature, but it would probably take > me much longer to implement than someone who knows the code well. I see a > bounty wiki entry...is that active? or is there a better place to post such > things? > > Thanks, > Matt > > > On Wed, Mar 14, 2012 at 12:30 PM, Michael Collins wrote: > >> AFAIK there isn't anything pre-rolled that you can use. However, I know >> that Tony's auto gain control (agc) logic checks the "volume" level coming >> in on a channel. You might be able to look in mod_conference.c and see >> where the agc calculates the incoming volume level and write an API that >> displays that value. >> >> I'd start by looking at the function check_agc_levels() and for >> references to member->agc_volume_in_level. Hopefully that will give you a >> place to start. >> >> -MC >> >> On Wed, Mar 14, 2012 at 10:26 AM, Matt Stockton wrote: >> >>> Hi all, >>> >>> I am interested in providing a visualization of a channel in a >>> conference. I am able to process the start talking and stop talking events >>> from the conference, but I'm wondering, has anyone done anything that can >>> get the real-time amplitude of a channel? I didn't see any built-in API >>> commands to do this, but wondering if there's any modules or something else >>> that I'm overlooking? Or maybe now is the time for me to dig into the API >>> documentation? >>> >>> Just curious if anyone has done something like this before, and if so, >>> if you could guide me in the right direction. >>> >>> Thanks! >>> Matt >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/f03e7063/attachment-0001.html From lists at telefaks.de Sat Mar 17 00:52:39 2012 From: lists at telefaks.de (Peter Steinbach) Date: Fri, 16 Mar 2012 22:52:39 +0100 Subject: [Freeswitch-users] Advice how to start with H.323 Message-ID: <4F63B627.5070206@telefaks.de> I want to create a H323 Gateway in order to receive calls from an external H323 Server and transfer them to another Freeswitch server. I started on my Ubuntu 10.04 with mod_opal and spent some hours with trying the following 2 ways: * Using buildopal.sh, I had to adapt the script slightly and had to do some symlinks in order to progress. At the end I got stuck with a compile error "class PSTUNClient' has no member named 'InvalidateCache'" during compilation of opal manager * I then installed the libpt and libopal packes from apt and started to compile mod_opal. I then got stuck with a compile error "errror: no matching function for call to 'OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, void*&, unsigned int&, OpalConnection::StringOptions*&)'" So this did not seem to be the right way as regards to future ability to compile it with system updates. Therefore I would like to ask you for some advice: * what is the best way to go: mod_opal or mod_h323? I think the feature set will be somehow the same for my purposes (I only need voice, PCMA), so I am asking, which one is more stable and compiles better. * what is best preferred platform (CentOS/Debian/??) for compiling this? * are there any dependencies for the opal and libpt versions? * Who had recently made running this solution and can give me some advice? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet:www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/03e8596f/attachment.html From nbhatti at gmail.com Sat Mar 17 01:01:21 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Sat, 17 Mar 2012 01:01:21 +0300 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 Message-ID: Hello community, I feel glad to announce the new release of *vBilling*, version 0.1.3. The first open source billing platform for FreeSWITCH. This versions has some major enhancements and supports many new features and bug fixes. Some of the highlights are: - *Complete reseller support* - Manage your resellers upto 3 levels. Once created by admin, a reseller can add their sub resellers depending on their permissions - *Mass updates of rates* - Edit your rates in bulk using the mass rate update module - *Built-in Help system* - vBilling now comes with built-in help system. Every element on each form has their own description and a tooltip. Just hover the mouse on the desired element and you will get a tooltip explaining what it means. - *Language localization* - Translate vBilling in your own language. You don't have to edit the bulky HTML/PHP templates. Just edit one language file and you have vBilling talk to you in your own language :) - *Customization of forms/output text* - Looking to customize messages on each screen? This can be done easily by editing one single file with your new messages. - *Carrier Management* - New and improved carrier management module - *Call detail records* - More enhanced CDR views - *New Module for Billing and Invoicing* - Enhanced module for billing and invoicing. But this is not all what we have. Take a look at our website, http://www.vbilling.org/ to explore some more serious features. Upgrading an existing install can never be so easier. Just browse to http://vbilling.org/get-started/ and download the install file, run it and relax. We will take care of the rest. If you have not installed yet, please do visit our website to get started with vBilling. If you face any issues, or do you have any questions, feel free to visit our forum @ http://forum.vbilling.org/ or send an email to support at vbilling.org. Our dedicated staff works round the clock and will try to help you as much as we can. Thanks and regards, Muhammad Naseer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/99d0038d/attachment.html From ZAlex at webley.com Sat Mar 17 01:57:47 2012 From: ZAlex at webley.com (Alex Zarubin) Date: Fri, 16 Mar 2012 17:57:47 -0500 Subject: [Freeswitch-users] Bridging after sending dtmf Message-ID: <74D0BA2985A4B04E8651FA01C8A58F39051F5A6D@vs-win-ex01.corp.parusi.com> Hello, Here are two dialplan examples of the simple thing I'm failing to do - bridge inbound (external profile) and outbound (internal profile) sip calls after sending some dtmf sequence to the outbound leg. In both cases (and several more I've tried) calls are bridged right after outbound leg picks up and dtmf are sent after the bridge. I would greatly appreciate your help/hints. Thank you. Alex ---------------------------------------------------------------------- Example 1. ---------------------------------------------------------------------- Example 2. Script lua_async.lua : dtmfseq = argv[1]; phone = argv[2]; destip = argv[3]; leg1uuid = argv[4]; cid = argv[5]; ands = ") and ("; logline = "Lua called with (" .. dtmfseq .. ands .. phone .. ands .. destip .. ands .. leg1uuid .. ands .. cid .. ")\n"; freeswitch.console_log("info", logline); api = freeswitch.API(); sofia_destination = "[origination_caller_id_number=" .. cid .. "]sofia/internal/" .. phone .. "@" .. destip; new_session = freeswitch.Session(sofia_destination); dispo = "None"; while (new_session:ready() and dispo ~= "ANSWER") do dispo = new_session:getVariable("endpoint_disposition") freeswitch.consoleLog("INFO", "disposition is '" .. dispo .. "'\n") os.execute("sleep 1") end -- while leg2uuid = new_session:get_uuid(); logline = "Lua leg2uuid (" .. leg2uuid .. ")\n"; freeswitch.console_log("info", logline); new_session:setAutoHangup(false); exestr_cmd = "uuid_send_dtmf " .. leg2uuid .. " w#w" .. dtmfseq; logline = "Lua calling api:executeString(" .. exestr_cmd .. ")\n"; freeswitch.console_log("info", logline); api:executeString(exestr_cmd); freeswitch.console_log("info", "Lua after api:executeString(...)\n"); exestr_cmd = "uuid_bridge " .. leg1uuid .. " " .. leg2uuid; logline = "Lua calling api:executeString(" .. exestr_cmd .. ")\n"; freeswitch.console_log("info", logline); api:executeString(exestr_cmd); freeswitch.console_log("info", "Lua after api:executeString(...)\n"); This message and any attachments to it are intended only for the addressee(s) identified above and may contain CONFIDENTIAL information. It is not intended for transmission to, or receipt by, any unauthorized persons. If you are not an intended recipient or an agent responsible for delivering it to an intended recipient, you have received this e-mail in error and any dissemination, distribution, or copying of this message or any attachment to it is strictly prohibited. If you have received this email in error, please (i) do not read it, (ii) reply to the sender that you received the message in error, and (iii) erase or destroy the message from your system. From lazyvirus at gmx.com Sat Mar 17 03:17:48 2012 From: lazyvirus at gmx.com (Bzzz) Date: Sat, 17 Mar 2012 01:17:48 +0100 Subject: [Freeswitch-users] acl.conf.xml don't understand Message-ID: <20120317011748.4e777bfd@anubis.defcon1> Hi list, There's something I don't understand: if I add my LAN cidr (allow) in the second block of acl.conf.xml (domains), I can't call anymore. From what I understood of the wiki, this should work because the block denies everyone but my line allows for LAN cidr. I guess I misunderstood something. JY -- From david.villasmil.work at gmail.com Sat Mar 17 03:38:06 2012 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 17 Mar 2012 01:38:06 +0100 Subject: [Freeswitch-users] Send calls through registered gateway In-Reply-To: References: Message-ID: HEllo, Thanks for the reply. Tried that but doesn't work. I register asterisk on FS. FS sends an "s at whatever" in the invite, with the actual number on the "Contact" header. but I set this on asterisk dialplan and now it works: exten => s,1,Goto(CONTEXT,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) thanks! On Thu, Mar 15, 2012 at 3:22 AM, Gabriel Gunderson wrote: > On Wed, Mar 14, 2012 at 6:32 PM, David Villasmil > wrote: > > I register an asterisk with with, say with username 1001. I receive a > call > > from user 2 and want to terminate what 1002 sends with the gateway (not a > > gateway per se, but a registered user). > > I think you may have it backwards. If you want your Asterisk server to > act as the GW to FreeSWITCH (you send calls from FS to Asterisk and it > hit's its own dialplan), then you would setup a gateway in FreeSWITCH > (see the sofia profie). You don't *need* FreeSWITCH to register to its > GWs, but you can if you like. > > Then from FS's perspective, you would dial it like so: > sofia/gateway/NAME_OF_GW/NUMBER_TO_DIAL > > I'm sorry if I misunderstood what you're doing, but I hope this helps. > > > Best, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/31378883/attachment.html From bdfoster at endigotech.com Sat Mar 17 05:30:45 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 16 Mar 2012 22:30:45 -0400 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: <4F63B627.5070206@telefaks.de> References: <4F63B627.5070206@telefaks.de> Message-ID: FYI there is a patch for the build script for mod_opal. I can't for the life of me remember what the ticket number is in JIRA, and I'm mobile so I can't really check it right now. It hasn't been committed to the code yet, so if someone would be willing to test on the latest revision let me know off list and I'll try and figure out the ticket number. As far as your predicament sir, a few things: On Mar 16, 2012 5:54 PM, "Peter Steinbach" wrote: > > I want to create a H323 Gateway in order to receive calls from an external H323 Server and transfer them to another Freeswitch server. > > I started on my Ubuntu 10.04 with mod_opal and spent some hours with trying the following 2 ways: > Using buildopal.sh, I had to adapt the script slightly and had to do some symlinks in order to progress. At the end I got stuck with a compile error "class PSTUNClient? has no member named ?InvalidateCache?" during compilation of opal manager > I then installed the libpt and libopal packes from apt and started to compile mod_opal. I then got stuck with a compile error "errror: no matching function for call to ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, void*&, unsigned int&, OpalConnection::StringOptions*&)?" > So this did not seem to be the right way as regards to future ability to compile it with system updates. > > Therefore I would like to ask you for some advice: > what is the best way to go: mod_opal or mod_h323? I think the feature set will be somehow the same for my purposes (I only need voice, PCMA), so I am asking, which one is more stable and compiles better. I'm actually not sure. I patched the build script for mod_Opal so that I could use IAX2. Fwiw Opal will not compile without a patch on the build script. The current build script gets the latest from svn or got, but it is indeed broken. > what is best preferred platform (CentOS/Debian/??) for compiling this? It boils down to what makes you comfortable. I personally use debian for everything, including freeswitch installs. Some wouldn't touch it with a 10 foot pole though. > are there any dependencies for the opal and libpt versions? Those should be taken care of by the build script (if there are any) > Who had recently made running this solution and can give me some advice? Can't help you there, sorry. :-) > -- > > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120316/98066bd9/attachment-0001.html From virbhati at gmail.com Sat Mar 17 10:26:44 2012 From: virbhati at gmail.com (virendra bhati) Date: Sat, 17 Mar 2012 12:56:44 +0530 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 69, Issue 127 In-Reply-To: References: Message-ID: Hi Patrick , I tryed a lot effort on google but no help i got and even i sent e-mail to lua mail list as well. but no one replied on it. Sorry but if you reply to me with details on luac then i will be a big help for me... On Sat, Mar 17, 2012 at 2:10 AM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: blocking destination number (Brian Foster) > 2. Re: Mod_flite wont load (Jeff Lenk) > 3. Re: Any help me to installed Luac compiler (Patrick Lists) > 4. mod_shout issue (Balamurugan Mahendran) > 5. Strange VLAN problem with freeswitch (Erin O'Meara) > 6. Re: Getting real-time amplitude of a channel (Michael Collins) > > > ---------- Forwarded message ---------- > From: Brian Foster > To: FreeSWITCH Users Help > Cc: > Date: Fri, 16 Mar 2012 12:58:27 -0400 > Subject: Re: [Freeswitch-users] blocking destination number > > The toll_allow variable.would be defined in the actual user xml file > located in /usr/local/freeswitch/conf/directory. Then you would evaluate > that variable in the dialplan that would end up making the bridge to your > gateway. > > -BDF > On Mar 16, 2012 5:21 AM, "Miha" wrote: > >> Hi @Gabriel, >> >> I was looking on wiki how to define new xml or document where I will >> have my restricted numbers. >> Where are domestic,international,local defined? >> >> Thanks! >> Miha >> >> On 3/10/2012 11:23 PM, Gabriel Gunderson wrote: >> > On Fri, Mar 9, 2012 at 7:37 AM, Miha Zoubek wrote: >> >> what is the bast way to block destination number for certain user. >> >> Is it possible to do it in user/dir? >> > Ultimately, it would be a dialplan configuration, but you could set >> > the variable that you match in the directory. See the example of >> > 'toll_allow' in the default FreeSWITCH configuration... >> > >> > >> > This is set on a per-user basis in the DIRECTORY: >> > >> > >> > >> > >> > >> > But it's considered while evaluating the DIALPLAN: >> > >> > >> > >> > >> > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> >> > > > data="effective_caller_id_name=${outbound_caller_id_name}"/> >> > > > data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> >> > >> > >> > >> > >> > >> > You could easily do something similar by having a list of numbers that >> > they can't call listed in the directory (maybe something like >> > 'restricted_numbers') and check to make sure the destination doesn't >> > match it in the dialplan. >> > >> > Good luck! >> > >> > >> > Gabe >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > ---------- Forwarded message ---------- > From: Jeff Lenk > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Fri, 16 Mar 2012 10:46:10 -0700 (PDT) > Subject: Re: [Freeswitch-users] Mod_flite wont load > Please try git head and let me know if that works. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Mod-flite-wont-load-tp7376926p7379705.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > ---------- Forwarded message ---------- > From: Patrick Lists > To: FreeSWITCH Users Help > Cc: > Date: Fri, 16 Mar 2012 18:54:19 +0100 > Subject: Re: [Freeswitch-users] Any help me to installed Luac compiler > On 16-03-12 06:07, virendra bhati wrote: > >> Hi list, >> >> I don't find any link on which luac compiler is mention. only one link >> was found on which description about Luac was mention. >> http://www.lua.org/manual/4.0/**luac.html >> >> Anyone know how to installed Luac compiler in Centos ? >> > > Can't you try a little harder to figure out these basic questions > yourself? Did you even try to Google for "luac CentOS"? That would have > given you a hint. > > Since your question is not a FreeSWITCH question but it is a lua question > why not ask on the lua mailinglist or on the lua irc channel if they have > one. > > Regards, > Patrick > > > > > ---------- Forwarded message ---------- > From: Balamurugan Mahendran > To: FreeSWITCH Users Help > Cc: > Date: Fri, 16 Mar 2012 23:34:23 +0530 > Subject: [Freeswitch-users] mod_shout issue > Hi all, > > 2012-03-16 18:02:32.957534 [ERR] mod_shout.c:859 Error: MPG123 Error at > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:659. > 2012-03-16 18:02:32.957534 [ERR] mod_shout.c:862 Error from mpg123: > Invalid mpg123 handle. (code 10) > > Please let me know what Iam missing here. > > Thanks, > Bala > > > ---------- Forwarded message ---------- > From: "Erin O'Meara" > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Fri, 16 Mar 2012 09:58:51 -0700 > Subject: [Freeswitch-users] Strange VLAN problem with freeswitch > I have moved offices down the street and extended my office network via > vlan (old office was using VLAN also). I have two networks one internal and > one wireless that I sell wireless service with. > > FusionPBX (In the cloud) -> Cable Modem -> PFSense -> VLAN -> Phone. > > my phone works fine for a few days on either network then calls stop > working, if I switch the phone VLAN to the other VLAN it starts working > again. I noticed this issue from my office that is connect of a Ubiquiti > wireless link and VLAN extended to my new office. > > What happens is the calls ring and I can pickup but no audio in either > direction. > > here is the pastebin of a call failing then I plugged the same phone into > the other vlan and reboot and made a successful call. > > http://pastebin.freeswitch.org/18675 > > Regards, > > 206.905.9520 > http://salmonbaytechnology.com > > > > ---------- Forwarded message ---------- > From: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Fri, 16 Mar 2012 13:39:55 -0700 > Subject: Re: [Freeswitch-users] Getting real-time amplitude of a channel > Post the bounty on the wiki and then shoot an email to the -dev list > letting them know there's a new bounty available... > > -MC > > On Fri, Mar 16, 2012 at 9:59 AM, Matt Stockton wrote: > >> Hi Michael, >> >> Thanks for the info!! Is there information I can find on how the bounties >> work? I'm interesting in getting this feature, but it would probably take >> me much longer to implement than someone who knows the code well. I see a >> bounty wiki entry...is that active? or is there a better place to post such >> things? >> >> Thanks, >> Matt >> >> >> On Wed, Mar 14, 2012 at 12:30 PM, Michael Collins wrote: >> >>> AFAIK there isn't anything pre-rolled that you can use. However, I know >>> that Tony's auto gain control (agc) logic checks the "volume" level coming >>> in on a channel. You might be able to look in mod_conference.c and see >>> where the agc calculates the incoming volume level and write an API that >>> displays that value. >>> >>> I'd start by looking at the function check_agc_levels() and for >>> references to member->agc_volume_in_level. Hopefully that will give you a >>> place to start. >>> >>> -MC >>> >>> On Wed, Mar 14, 2012 at 10:26 AM, Matt Stockton wrote: >>> >>>> Hi all, >>>> >>>> I am interested in providing a visualization of a channel in a >>>> conference. I am able to process the start talking and stop talking events >>>> from the conference, but I'm wondering, has anyone done anything that can >>>> get the real-time amplitude of a channel? I didn't see any built-in API >>>> commands to do this, but wondering if there's any modules or something else >>>> that I'm overlooking? Or maybe now is the time for me to dig into the API >>>> documentation? >>>> >>>> Just curious if anyone has done something like this before, and if so, >>>> if you could guide me in the right direction. >>>> >>>> Thanks! >>>> Matt >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati at gmail.com Skype id:- virbhati2 Hyderabad(India) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/2f2c0614/attachment-0001.html From bote_radio at botecomm.com Sat Mar 17 11:34:02 2012 From: bote_radio at botecomm.com (Bote Man) Date: Sat, 17 Mar 2012 04:34:02 -0400 Subject: [Freeswitch-users] endconf flag not working In-Reply-To: References: Message-ID: <00aa01cd0418$b60e9e70$222bdb50$@com> I have a Windows installation (yes, FML) that I can not change to linux, I'm stuck with it. It is running the .msi build from Feb. 29 that otherwise works great. This past weekend's build was declared unusable by Windows installer, go figure. I'm using the conference_auto_outcall feature to initiate an outbound group conference call to 4 endpoints. It works OK, except that because some of them are FXO gateways to analog lines they essentially ring until the initiator disconnects. Unfortunately, the endconf flag is not being honored by mod_conference so the initiator hangs up and the outbound calls keep on ringing because the number of participants is not less than 1. I would expect the endconf flag to drop the entire conference when the extension that created it released, as described by the wiki page. As I am pulling down sources I?m hoping somebody with experience in mod_conference can verify that the 'endconf' flag should or should not work? Have you played with this setting before and what were your results? I am brand new with FreeSWITCH and need all the help I can get. Thanks. Bote -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/78fc1dd5/attachment.html From b2m at a-cti.com Sat Mar 17 13:21:29 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Sat, 17 Mar 2012 15:51:29 +0530 Subject: [Freeswitch-users] Need Help to handle message Message-ID: where can I find more details about nua? tport_wakeup_pri(0xa7c160): events IN tport_recv_event(0xa7c160) tport_recv_iovec(0xa7c160) msg 0x7f18080037b0 from (udp/10.248.13.3:5060) has 487 bytes, veclen = 1 tport_deliver(0xa7c160): msg 0x7f18080037b0 (487 bytes) from udp/ 23.21.230.74:5060/sip next=(nil) nta: received MESSAGE sip:+15032721367 at 50.12.112.39 SIP/2.0 (CSeq 102) nta: Via check: received=23.21.230.74 nta: canonizing sip:+15032721367 at 50.12.112.39 with contact nta: MESSAGE (102) 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::0xa69c50, 0xa69490, 0x7f17f80f5180) called soa_set_params(static::0x7f17f80f52a0, ...) called nta_leg_tcreate(0x7f17f8088d30) tport_tsend(0xa7c160) tpn = UDP/23.211.230.74:5060 tport_resolve addrinfo = 23.211.230.74:5060 tport_by_addrinfo(0xa7c160): not found by name UDP/23.211.230.74:5060 tport_vsend(0xa7c160): 539 bytes of 539 to udp/23.211.230.74:5060 tport_vsend returned 539 nta: sent 200 OK for MESSAGE (102) nua(0x7f17f80f5180): event i_message 200 OK nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_respond: entering nua(0x7f17f80f5180): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x7f17f80f5180): sent signal r_destroy nua(0x7f17f80f5180): recv signal r_respond 202 Accepted nua(0x7f17f80f5180): event i_error 500 Responding to a Non-Existing Request nua(0x7f17f80f5180): recv signal r_destroy nta_leg_destroy(0x7f17f8088d30) soa_destroy(static::0x7f17f80f52a0) called nta: timer J fired, terminate 200 response incoming_reclaim_all((nil), (nil), 0x7f180752ec40) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 14002 ms tport_wakeup_pri(0xa7c160): events IN tport_recv_event(0xa7c160) tport_recv_iovec(0xa7c160) msg 0x7f1808147e10 from (udp/10.248.13.3:5060) has 4 bytes, veclen = 1 tport_deliver(0xa7c160): bad msg 0x7f1808147e10 (4 bytes) from udp/ 115.11.5.34:5060/sip next=(nil) nta: timer J fired, terminate 200 response incoming_reclaim_all((nil), (nil), 0x7f180752ec40) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set tport_wakeup_pri(0xa7c160): events IN tport_recv_event(0xa7c160) tport_recv_iovec(0xa7c160) msg 0x7f1808147e10 from (udp/10.248.13.3:5060) has 4 bytes, veclen = 1 tport_deliver(0xa7c160): bad msg 0x7f1808147e10 (4 bytes) from udp/ 115.11.5.34:5060/sip next=(nil) freeswitch at internal> Thanks, Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/00c9973d/attachment.html From trever.adams at gmail.com Sat Mar 17 17:08:45 2012 From: trever.adams at gmail.com (Trever L. Adams) Date: Sat, 17 Mar 2012 08:08:45 -0600 Subject: [Freeswitch-users] FreeSWITCH 1.2 is on the horizon. In-Reply-To: References: Message-ID: <4F649AED.4080802@gmail.com> 12 ????? 2012 ?. 23:19 ???????????? Ken Rice ???????: > > The FreeSWITCH Development Team is happy to announce that 1.2 is > officially on the horizon. > Starting Wed March 14th 2012, the Development branch of FreeSWITCH > will reach a Feature Freeze. > > What does this mean for you the user? It means we will have a > stable known feature set heading into the Release Candidate Cycle. > Only patches that Fix Bugs will be accepted. If you have a feature you > would like to see included, get us a Jira with a full patch set so we > can evaluate its inclusion with 1.2. > > What we need from the community: Testing, Testing, more Testing, > and Documentation Updates. > > The freeze will last 2 to 4 weeks as we spool up testing and > everything else we need to get the Release Candidates ready for prime > time. > > If you have outstanding bugs on Jira, please help us help you > during this time by making sure all information on them is up to date. > Grab the latest GIT Head and see if your bugs have been resolved and > someone forgot to close your Jira. > > If you want to help or need some help diagnosing and issue visit > us on IRC via irc.freenode.net/#freeswitch any time. > > The FreeSWITCH Dev Team > The following is a bug which is affecting several people. Several have just given up and gone FXS free. I do not have that option. I am willing to help out if I can. http://jira.freeswitch.org/browse/OPENZAP-173 Trever -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/52c4bb22/attachment.bin From freeswitch-list at puzzled.xs4all.nl Sat Mar 17 17:42:43 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Sat, 17 Mar 2012 15:42:43 +0100 Subject: [Freeswitch-users] Any help me to installed Luac compiler (was: FreeSWITCH-users Digest, Vol 69, Issue 127) In-Reply-To: References: Message-ID: <4F64A2E3.30007@puzzled.xs4all.nl> On 03/17/2012 08:26 AM, virendra bhati wrote: > Hi Patrick , > > I tryed a lot effort on google but no help i got and even i sent e-mail > to lua mail list as well. but no one replied on it. Please do not top-post and properly trim your replies (remove unrelated stuff). If you have the lua rpm installed on your CentOS system then luac is already available. Check with: $ rpm -q lua lua-5.1.4-4.1.el6.x86_64 <-- so it's installed If you don't have lua installed then install it as root with: # yum install lua luac is part of the lua package: $ rpm -ql lua | grep luac /usr/bin/luac /usr/share/doc/lua-5.1.4/luac.html /usr/share/man/man1/luac.1.gz Since you seem to be lacking basic Red Hat/CentOS knowledge and skills may I suggest this book: RHCSA/RHCE Red Hat Linux Certification Study Guide (Exams EX200 & EX300), 6th Edition written by Michael Jang http://www.amazon.co.uk/RHCSA-Linux-Certification-Study-Edition/dp/0071765654/ref=sr_1_1?s=books&ie=UTF8&qid=1331995110&sr=1-1 Regards, Patrick From freeswitch-list at puzzled.xs4all.nl Sat Mar 17 17:44:59 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Sat, 17 Mar 2012 15:44:59 +0100 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: References: <4F63B627.5070206@telefaks.de> Message-ID: <4F64A36B.2060801@puzzled.xs4all.nl> On 03/17/2012 03:30 AM, Brian Foster wrote: [snip] > > are there any dependencies for the opal and libpt versions? > > Those should be taken care of by the build script (if there are any) That's assuming that the build script does the proper thing. Last time I looked, it pulled svn trunk versions of ptlib and opal which are too new for mod_opal. Afaik it needs the Sirius or maybe the Luyten release. In the past I built mod_opal without a problem (not using that script) with the Sirius release of ptlib and opal installed. Regarrds, Patrick From david.villasmil.work at gmail.com Sat Mar 17 20:00:21 2012 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 17 Mar 2012 18:00:21 +0100 Subject: [Freeswitch-users] ODBC instalation Message-ID: Hello guys, I just installed odbc support on my debian and want to hace FS use it for lcr. I followed the steps on; http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core the installation is OK in terms of isql but i don't know whether FS is able to connect as I just can't see any reference to ODBC in the logs. Is there any way of testing FS's connectivity to MySQL-ODBC from the CLI? Thanks David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/10276702/attachment-0001.html From nasida at live.ru Sat Mar 17 20:32:52 2012 From: nasida at live.ru (Yuriy Nasida) Date: Sat, 17 Mar 2012 21:32:52 +0400 Subject: [Freeswitch-users] ODBC instalation In-Reply-To: References: Message-ID: Just check freeswitch database and look if exist some tables. In first start, FS has to create tables automatically. From: david.villasmil.work at gmail.com Date: Sat, 17 Mar 2012 18:00:21 +0100 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] ODBC instalation Hello guys, I just installed odbc support on my debian and want to hace FS use it for lcr. I followed the steps on; http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core the installation is OK in terms of isql but i don't know whether FS is able to connect as I just can't see any reference to ODBC in the logs. Is there any way of testing FS's connectivity to MySQL-ODBC from the CLI? Thanks David _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/0cb8bed7/attachment.html From david.villasmil.work at gmail.com Sat Mar 17 20:56:33 2012 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 17 Mar 2012 18:56:33 +0100 Subject: [Freeswitch-users] ODBC instalation In-Reply-To: References: Message-ID: Yeah, there's no table... :( the documentation is not very clear about this: onf/autoload_configs/switch.conf.xml (WHERE IN THE SWITCH.CONF?) Add or uncomment the following line in appropriate config file within <-- this is clear thanks David 2012/3/17 Yuriy Nasida > Just check freeswitch database and look if exist some tables. In first > start, FS has to create tables automatically. > > ------------------------------ > From: david.villasmil.work at gmail.com > Date: Sat, 17 Mar 2012 18:00:21 +0100 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] ODBC instalation > > > Hello guys, > > I just installed odbc support on my debian and want to hace FS use it for > lcr. > I followed the steps on; > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core > > the installation is OK in terms of isql but i don't know whether FS is > able to connect as I just can't see any reference to ODBC in the logs. > > Is there any way of testing FS's connectivity to MySQL-ODBC from the CLI? > > Thanks > > David > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/b2384cb9/attachment.html From avi at avimarcus.net Sat Mar 17 21:06:17 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 17 Mar 2012 20:06:17 +0200 Subject: [Freeswitch-users] ODBC instalation In-Reply-To: References: Message-ID: If it's not there, add it. Order doesn't matter. -Avi On Sat, Mar 17, 2012 at 7:56 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Yeah, there's no table... :( > > the documentation is not very clear about this: > > onf/autoload_configs/switch.conf.xml > > (WHERE IN THE SWITCH.CONF?) > > Add or uncomment the following line in appropriate config file within > <-- this is clear > > > > > > thanks > David > > > 2012/3/17 Yuriy Nasida > >> Just check freeswitch database and look if exist some tables. In first >> start, FS has to create tables automatically. >> >> ------------------------------ >> From: david.villasmil.work at gmail.com >> Date: Sat, 17 Mar 2012 18:00:21 +0100 >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] ODBC instalation >> >> >> Hello guys, >> >> I just installed odbc support on my debian and want to hace FS use it for >> lcr. >> I followed the steps on; >> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core >> >> the installation is OK in terms of isql but i don't know whether FS is >> able to connect as I just can't see any reference to ODBC in the logs. >> >> Is there any way of testing FS's connectivity to MySQL-ODBC from the CLI? >> >> Thanks >> >> David >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server Official FreeSWITCH >> Sites http://www.freeswitch.org http://wiki.freeswitch.org >> http://www.cluecon.com FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/4e137020/attachment.html From me at nevian.org Sat Mar 17 21:19:53 2012 From: me at nevian.org (Serge S. Yuriev) Date: Sat, 17 Mar 2012 22:19:53 +0400 Subject: [Freeswitch-users] Strange gw behavior Message-ID: <531341332008393@web134.yandex.ru> Hello, I have few GWs on my external profile and all work fine except one. When I register GW 'multifon' it breaks all outside calling - this GW is never used in DP as outbound BUT all calls trying to flow trough it with attributes from other GWs! Pls see http://pastebin.freeswitch.org/18691 Can be this my configuration fault or should I file it to JIRA? I'm on few days old GIT but this seen at least from middle of Nov'11 -- wbr, Serge From kjoseph.us at gmail.com Sat Mar 17 21:28:25 2012 From: kjoseph.us at gmail.com (Joseph Khoury) Date: Sat, 17 Mar 2012 11:28:25 -0700 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: Message-ID: Hi, I can not see the source code of the lua script, you have it in compiled form only! I thought this is an Open source. Muhammad any plan to make it open source? Other than that, good job :) Thanks Joseph On Fri, Mar 16, 2012 at 3:01 PM, Muhammad Naseer Bhatti wrote: > > Hello community, > I feel glad to announce the new release of *vBilling*, version 0.1.3. The > first open source billing platform for FreeSWITCH. This versions has some > major enhancements and supports many new features and bug fixes. Some of > the highlights are: > > - *Complete reseller support* > - Manage your resellers upto 3 levels. Once created by admin, a > reseller can add their sub resellers depending on their permissions > - *Mass updates of rates* > - Edit your rates in bulk using the mass rate update module > - *Built-in Help system* > - vBilling now comes with built-in help system. Every element on > each form has their own description and a tooltip. Just hover the mouse on > the desired element and you will get a tooltip explaining what it means. > - *Language localization* > - Translate vBilling in your own language. You don't have to edit > the bulky HTML/PHP templates. Just edit one language file and you have > vBilling talk to you in your own language :) > - *Customization of forms/output text* > - Looking to customize messages on each screen? This can be done > easily by editing one single file with your new messages. > - *Carrier Management* > - New and improved carrier management module > - *Call detail records* > - More enhanced CDR views > - *New Module for Billing and Invoicing* > - Enhanced module for billing and invoicing. > > But this is not all what we have. Take a look at our website, > http://www.vbilling.org/ to explore some more serious features. Upgrading > an existing install can never be so easier. Just browse to > http://vbilling.org/get-started/ and download the install file, run it > and relax. We will take care of the rest. If you have not installed yet, > please do visit our website to get started with vBilling. > > If you face any issues, or do you have any questions, feel free to visit > our forum @ http://forum.vbilling.org/ or send an email to > support at vbilling.org. Our dedicated staff works round the clock and will > try to help you as much as we can. > > Thanks and regards, > Muhammad Naseer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/3cdcad3f/attachment-0001.html From lazyvirus at gmx.com Sat Mar 17 21:37:16 2012 From: lazyvirus at gmx.com (Bzzz) Date: Sat, 17 Mar 2012 19:37:16 +0100 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: Message-ID: <20120317193716.14569724@anubis.defcon1> On Sat, 17 Mar 2012 11:28:25 -0700 Joseph Khoury wrote: > I can not see the source code of the lua script, you have it in compiled > form only! Take a look at: http://luadec51.luaforge.net/ -- Elliptical, n.: The feel of a kiss. From gabe at gundy.org Sat Mar 17 22:07:43 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 17 Mar 2012 13:07:43 -0600 Subject: [Freeswitch-users] Do trim, do not top post, please. Was: Re: Any help me to installed Luac compiler (was: FreeSWITCH-users Digest, Vol 69, Issue 127) Message-ID: On Sat, Mar 17, 2012 at 8:42 AM, Patrick Lists wrote: > Please do not top-post and properly trim your replies (remove unrelated > stuff). Everyone... more of this please! See how easy it was to follow the conversation here? Gabe From lazyvirus at gmx.com Sat Mar 17 22:23:01 2012 From: lazyvirus at gmx.com (Bzzz) Date: Sat, 17 Mar 2012 20:23:01 +0100 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: Message-ID: <20120317202301.1d80d7f1@anubis.defcon1> On Sat, 17 Mar 2012 01:01:21 +0300 Muhammad Naseer Bhatti wrote: > I feel glad to announce the new release of *vBilling*, version 0.1.3. The > first open source billing platform for FreeSWITCH. This versions has some > major enhancements and supports many new features and bug fixes. ...and will not fail until the first (*noticed*) glitch. At that time, the admin have chances to spend many hours/days to manually track all errors - as the database makes no use of any referential integrity... Jean-Yves -- From deryl at fs-specs.org Sat Mar 17 22:23:21 2012 From: deryl at fs-specs.org (Deryl R. Doucette) Date: Sat, 17 Mar 2012 15:23:21 -0400 Subject: [Freeswitch-users] Do trim, do not top post, please. Was: Re: Any help me to installed Luac compiler (was: FreeSWITCH-users Digest, Vol 69, Issue 127) In-Reply-To: References: Message-ID: The days of top vs. bottom posting are over. -- Deryl R. Doucette This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. On 2012-03-17, at 3:07 PM, Gabriel Gunderson wrote: > On Sat, Mar 17, 2012 at 8:42 AM, Patrick Lists > wrote: >> Please do not top-post and properly trim your replies (remove unrelated >> stuff). > > Everyone... more of this please! > > See how easy it was to follow the conversation here? > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/7e5d57ea/attachment.html From acrow at integrafin.co.uk Sat Mar 17 22:33:55 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Sat, 17 Mar 2012 19:33:55 +0000 Subject: [Freeswitch-users] Context problem with xml_curl directory Message-ID: <4F64E723.5090604@integrafin.co.uk> Hi, I've just tried to get mod_xml_curl working against a PHP script connecting to an OpenLDAP server. Things are working mostly OK, apart from the fact that calls from extensions defined in LDAP are always initiated in context "public", even though the XML returned at auth request includes the variable "user_context=default". I've traced in wireshark and the responses all seem correct. This gets caught by the static XML public dialplan, but I still feel the behaviour is not correct. I have attached the PHP and a tcpdump trace in case anyone can see something obviously incorrect. Many thanks Alex -------------- next part -------------- A non-text attachment was scrubbed... Name: foo.dmp Type: application/octet-stream Size: 11706 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/cb885b8e/attachment-0001.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: xmlcurl.php Type: application/x-php Size: 12596 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/cb885b8e/attachment-0001.bin From potxoka at gmail.com Sat Mar 17 23:13:52 2012 From: potxoka at gmail.com (Anto) Date: Sat, 17 Mar 2012 21:13:52 +0100 Subject: [Freeswitch-users] Codec negotiation with carriers In-Reply-To: References: Message-ID: Hello I do not want to bother them, nor give me the solution. I would like to learn how to configure it for myself (in fact one of the settings works), but I do not understand the functioning correctly, for so not to disturb in the future. I attached the various configurations and their sip traces, as well as the logs (debug): First scenario ( Call isn?t established ) http://pastebin.freeswitch.org/18685 Second scenario ( Call isn?t established ) http://pastebin.freeswitch.org/18686 Third scenario ( Call isn?t established ) http://pastebin.freeswitch.org/18687 Fourth scenario ( Call isn?t established ) http://pastebin.freeswitch.org/18688 Fifth scenario ( Call established ) http://pastebin.freeswitch.org/18689 Sixth scenario ( Fail transcoding ) http://pastebin.freeswitch.org/18690 General settings: vars.xml ----------------------------------------------------------------------------------------------------------------------------------------------- dialplan\outbound.xml Thanks ! Regards Anto 2012/3/16 Michael Collins : > Well, this is a little better, however you don't have proper freeswitch logs > on all these calls. For example, only the first call has freeswitch debug > output. The other calls have sip traces, but not the first call. One call > has what appears to be info-level output, but not debug-level output. > > I'd recommend that if you have this much information it might be good to put > each call example in its own pastebin. Also, be sure to give a detailed > description of what kind of call you are documenting. Some of your > traces/debugs have no information explaining what the call is doing. Whether > you are reporting a working or failed call, be sure to mention what kind of > call it is. In the case of a failed call, be sure to mention what it is you > are trying to do and what call result you expected to see. > > Thanks! > > -MC > > > On Thu, Mar 15, 2012 at 2:22 PM, Anto wrote: >> >> Hi >> >> If, upload a file to trace and explanation to this address >> http://pastebin.freeswitch.org/18599 >> >> I do not want disturb watching this ;-), I prefer to use a system to >> understand, this scenario and for future projects. >> >> With everything I've read do not really understand what the codecs :-S >> , but if I had been good to understand the rest of the operation of >> FreeSWITCH (or so I think). Thanks ! >> >> Regards >> Anto >> >> 2012/3/14 Michael Collins : >> > Did you get sip traces and logs of working vs. non-working calls and put >> > them on pastebin? Most likely there is an explanation but it will take >> > some >> > time and effort to figure it out. >> > >> > -MC >> > >> > >> > On Wed, Mar 14, 2012 at 2:18 PM, Anto wrote: >> >> >> >> Hello >> >> >> >> I have searched previous messages in the list, I consulted the book of >> >> FreeSWITCH (which I bought over a year), wiki and so on. I still do >> >> not understand how and why in some cases I work. Also I downloaded >> >> frontend to consult your code if there was something about this, but >> >> still the same. I have several weeks with this question and I can not >> >> find it. In the end I decided to spend the gateway to Asterisk, and >> >> you at least understand its operation. Thank you very much to all :-) >> >> >> >> Best regards >> >> Anto >> >> >> >> 2012/3/11 Anto : >> >> > Hi >> >> > >> >> > I still do not find the solution and not really understanding, >> >> > because >> >> > it works:-S >> >> > >> >> > regards >> >> > anto >> >> > >> >> > 2012/3/7 Anto : >> >> >> Hello >> >> >> >> >> >> Attached file, with the traces of the different tests (with >> >> >> different >> >> >> configurations). >> >> >> >> >> >> http://pastebin.freeswitch.org/18599 >> >> >> >> >> >> I have read the url that you mentioned, the initial guide >> >> >> FreeSWITCH, >> >> >> that of mod_sofia, applications, etc.. I believe that most of the >> >> >> wiki >> >> >> (maybe when do not give the solution, read as much documentation is >> >> >> worse idea :-S, lock me more). >> >> >> >> >> >> I made a configuration that works (I have not tested the audio), but >> >> >> earlier (before I started "touch" the profiles) if I could talk to a >> >> >> physical phone (several times). The problem is that I can not >> >> >> understand why it works and sometimes not, and I would like to learn >> >> >> :-). Not only do and forget, so I would like to learn and less >> >> >> disturbing to the mail list and (maybe in the future) to help other >> >> >> newbies like me :-). Thanks ! >> >> >> >> >> >> Best regards >> >> >> Anto >> >> >> >> >> >> 2012/3/7 Michael Collins : >> >> >>> You may want to read up on codec negotiation: >> >> >>> http://wiki.freeswitch.org/wiki/Codec_negotiation >> >> >>> >> >> >>> There are different ways to handle codecs depending on your needs. >> >> >>> I'd >> >> >>> read >> >> >>> that page first and then try out some of the suggestions. If you're >> >> >>> still >> >> >>> having trouble then I'd recommend getting SIP traces of the traffic >> >> >>> and >> >> >>> putting them on pastebin.freeswitch.org. The gang here is pretty >> >> >>> good >> >> >>> at >> >> >>> looking over logs and helping with diagnosing problems. :) >> >> >>> >> >> >>> -MC >> >> >>> >> >> >>> On Tue, Mar 6, 2012 at 2:30 PM, Anto wrote: >> >> >>>> >> >> >>>> Hi >> >> >>>> >> >> >>>> I am following the steps in this direction >> >> >>>> "http://wiki.freeswitch.org/wiki/SBC_Setup" and >> >> >>>> >> >> >>>> "http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice", >> >> >>>> I reread the whole entire wiki (or so I lack), but do not quite >> >> >>>> assimilate or finding the right formula to operate the bridge :-S. >> >> >>>> >> >> >>>> I captured traffic with ngrep, I enabled sip-trace, console >> >> >>>> logconsole >> >> >>>> 8, etc., but unless the transcoding error (only two of the >> >> >>>> hundreds >> >> >>>> of >> >> >>>> combinations of settings that I have), I have not seen anything >> >> >>>> "weird" :-S >> >> >>>> >> >> >>>> I have 3 suppliers, each with this codec: >> >> >>>> >> >> >>>> 1) ? ? ? ? ? 2) ? ? ? ? ? ? ?3) >> >> >>>> G729 ? ? ? ?G729 ? ? ? ?G729 >> >> >>>> G711u ? ? ?G711A ? ? ?G711A >> >> >>>> G711A ? ? G711u ? ? ? G711u >> >> >>>> ? ? ? ? ? ? ? ?G723 ? ? ? ? G723 >> >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?G722 >> >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?GSM >> >> >>>> >> >> >>>> I think I understand that when making an outside call, FreeSWITCH >> >> >>>> follow these steps: >> >> >>>> >> >> >>>> USER -> ( ? Dialplan -> profile (internal) -> bridge (external) -> >> >> >>>> profile (external) ? ) -> PROVIDER >> >> >>>> >> >> >>>> PROVIDER -> ( ? Dialplan -> profile (external) -> bridge >> >> >>>> (internal) >> >> >>>> -> >> >> >>>> profile (internal) ?) -> USER >> >> >>>> >> >> >>>> right? >> >> >>>> >> >> >>>> Internal and external I set as follows (and not many changes have >> >> >>>> done, and not remember it, because I've been testing days). If >> >> >>>> outbound (outbound-codec-prefs) all codecs specified system does >> >> >>>> not >> >> >>>> handle the call, I have to specify these by hand. If active >> >> >>>> inbound-proxy-media, not the caller. Some of the time I worked, >> >> >>>> but >> >> >>>> gave me an error that it can do transcoding G729 codec (I do >> >> >>>> passthrough), but the proxy does not work half. >> >> >>>> >> >> >>>> If the outbound property (outbound-codec-prefs) all codecs >> >> >>>> specified >> >> >>>> system does not handle the call, I have to specify these by hand. >> >> >>>> If >> >> >>>> active inbound-proxy-media, not the caller. Some of the time I >> >> >>>> worked, >> >> >>>> but gave me an error that it can do transcoding G729 codec (I want >> >> >>>> to >> >> >>>> make passthrough), but the "proxy media" does not work. >> >> >>>> >> >> >>>> Basically, what I do is that local users can use all the codecs >> >> >>>> allowed (iLBC, GSM, ...) and make an outside call, use the carrier >> >> >>>> that will indicate the priority but the free codec. >> >> >>>> >> >> >>>> With this configuration works for me, but I would like to >> >> >>>> understand >> >> >>>> why so if it works and otherwise no. Coming to understand how to >> >> >>>> configure properly and so as not to disturb the mail list ;-). >> >> >>>> Thanks >> >> >>>> ! >> >> >>>> >> >> >>>> Best regards >> >> >>>> Anto >> >> >>>> >> >> >>>> vars.xml >> >> >>>> >> >> >>>> > >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> data="global_codec_prefs=iLBC,G7221,speex,PCMU,PCMA,BV16,G726-32,GSM,G729,G723,AMR"/> >> >> >>>> > >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,iLBC,G7221,speex,BV16,G726-32,GSM"/> >> >> >>>> >> >> >>>> internal.xml >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> external.xml >> >> >>>> >> >> >>>> >> >> >>>> > >> >>>> value="$${carriers_codec_prefs}"/> >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> dialplan/outbound.xml >> >> >>>> >> >> >>>> >> >> >>>> ? ? ? ? >> >> >>>> ? ? ? ? ? ? ? ? >> >> >>>> ? ? ? ? ? ? ? ? ?> >> >>>> expression="^(\d+)$"> >> >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? >> >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? >> >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? >> >> >>>> ? ? ? ? ? ? ? ? ? ? ? ?> >> >>>> data="sofia/gateway/provider-2/$1"/> >> >> >>>> ? ? ? ? ? ? ? ? ? >> >> >>>> ? ? ? ? ? ? ? ? >> >> >>>> ? ? ? ? >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> _________________________________________________________________________ >> >> >>>> Professional FreeSWITCH Consulting Services: >> >> >>>> consulting at freeswitch.org >> >> >>>> http://www.freeswitchsolutions.com >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> >> >> >>>> Official FreeSWITCH Sites >> >> >>>> http://www.freeswitch.org >> >> >>>> http://wiki.freeswitch.org >> >> >>>> http://www.cluecon.com >> >> >>>> >> >> >>>> FreeSWITCH-users mailing list >> >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>>> >> >> >>>> >> >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>> http://www.freeswitch.org >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> _________________________________________________________________________ >> >> >>> Professional FreeSWITCH Consulting Services: >> >> >>> consulting at freeswitch.org >> >> >>> http://www.freeswitchsolutions.com >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> Official FreeSWITCH Sites >> >> >>> http://www.freeswitch.org >> >> >>> http://wiki.freeswitch.org >> >> >>> http://www.cluecon.com >> >> >>> >> >> >>> FreeSWITCH-users mailing list >> >> >>> FreeSWITCH-users at lists.freeswitch.org >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> >> >> >>> >> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> http://www.freeswitch.org >> >> >>> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From bdfoster at endigotech.com Sat Mar 17 23:57:16 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 17 Mar 2012 16:57:16 -0400 Subject: [Freeswitch-users] Do trim, do not top post, please. Was: Re: Any help me to installed Luac compiler (was: FreeSWITCH-users Digest, Vol 69, Issue 127) In-Reply-To: References: Message-ID: Huh? -BDF On Mar 17, 2012 3:24 PM, "Deryl R. Doucette" wrote: > > The days of top vs. bottom posting are over. > > -- > Deryl R. Doucette > > This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. > > On 2012-03-17, at 3:07 PM, Gabriel Gunderson wrote: > >> On Sat, Mar 17, 2012 at 8:42 AM, Patrick Lists >> wrote: >>> >>> Please do not top-post and properly trim your replies (remove unrelated >>> >>> stuff). >> >> >> Everyone... more of this please! >> >> See how easy it was to follow the conversation here? >> >> >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/bf89008e/attachment.html From brian at freeswitch.org Sun Mar 18 00:00:38 2012 From: brian at freeswitch.org (Brian West) Date: Sat, 17 Mar 2012 16:00:38 -0500 Subject: [Freeswitch-users] Do trim, do not top post, please. Was: Re: Any help me to installed Luac compiler (was: FreeSWITCH-users Digest, Vol 69, Issue 127) In-Reply-To: References: Message-ID: <-5889381104577349245@unknownmsgid> I always top post! /b Sent from my iPad On Mar 17, 2012, at 2:09 PM, Gabriel Gunderson wrote: > Everyone... more of this please! > > See how easy it was to follow the conversation here? > > > Gabe From brian at freeswitch.org Sun Mar 18 00:01:16 2012 From: brian at freeswitch.org (Brian West) Date: Sat, 17 Mar 2012 16:01:16 -0500 Subject: [Freeswitch-users] Do trim, do not top post, please. Was: Re: Any help me to installed Luac compiler (was: FreeSWITCH-users Digest, Vol 69, Issue 127) In-Reply-To: References: Message-ID: <6735331441911256577@unknownmsgid> Trim please before I get the lawn mower out! /b Sent from my iPad On Mar 17, 2012, at 3:58 PM, Brian Foster wrote: Huh? -BDF -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/a346ac1f/attachment.html From brian at freeswitch.org Sun Mar 18 00:02:37 2012 From: brian at freeswitch.org (Brian West) Date: Sat, 17 Mar 2012 16:02:37 -0500 Subject: [Freeswitch-users] Do trim, do not top post, please. Was: Re: Any help me to installed Luac compiler (was: FreeSWITCH-users Digest, Vol 69, Issue 127) In-Reply-To: References: Message-ID: <-9094136478972477221@unknownmsgid> So is the ability to enforce these stupid disclaimers! Please remove them when posting to a public list! /b. :) Sent from my iPad On Mar 17, 2012, at 2:24 PM, "Deryl R. Doucette" wrote: The days of top vs. bottom posting are over. -- Deryl R. Doucette This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/e3bb556d/attachment.html From andrew at cassidywebservices.co.uk Sun Mar 18 00:17:10 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Sat, 17 Mar 2012 21:17:10 +0000 Subject: [Freeswitch-users] Do trim, do not top post, please. Was: Re: Any help me to installed Luac compiler (was: FreeSWITCH-users Digest, Vol 69, Issue 127) In-Reply-To: <-9094136478972477221@unknownmsgid> References: <-9094136478972477221@unknownmsgid> Message-ID: Gmail top-posts by default, and trims your replies for me, so I can see the conversation just fine ;) On 17 March 2012 21:02, Brian West wrote: > So is the ability to enforce these stupid disclaimers! > > Please remove them when posting to a public list! > > /b. :) > > Sent from my iPad > > On Mar 17, 2012, at 2:24 PM, "Deryl R. Doucette" > wrote: > > The days of top vs. bottom posting are over. > > -- > Deryl R. Doucette > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/78b52e83/attachment.html From david.villasmil.work at gmail.com Sun Mar 18 00:17:15 2012 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 17 Mar 2012 22:17:15 +0100 Subject: [Freeswitch-users] ODBC instalation In-Reply-To: References: Message-ID: the core-db-dsn value is "dsn-in-odbc.ini:user:password" correct? On Sat, Mar 17, 2012 at 7:06 PM, Avi Marcus wrote: > If it's not there, add it. Order doesn't matter. > -Avi > > > On Sat, Mar 17, 2012 at 7:56 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Yeah, there's no table... :( >> >> the documentation is not very clear about this: >> >> onf/autoload_configs/switch.conf.xml >> >> (WHERE IN THE SWITCH.CONF?) >> >> Add or uncomment the following line in appropriate config file within >> <-- this is clear >> >> >> >> >> >> thanks >> David >> >> >> 2012/3/17 Yuriy Nasida >> >>> Just check freeswitch database and look if exist some tables. In first >>> start, FS has to create tables automatically. >>> >>> ------------------------------ >>> From: david.villasmil.work at gmail.com >>> Date: Sat, 17 Mar 2012 18:00:21 +0100 >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: [Freeswitch-users] ODBC instalation >>> >>> >>> Hello guys, >>> >>> I just installed odbc support on my debian and want to hace FS use it >>> for lcr. >>> I followed the steps on; >>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core >>> >>> the installation is OK in terms of isql but i don't know whether FS is >>> able to connect as I just can't see any reference to ODBC in the logs. >>> >>> Is there any way of testing FS's connectivity to MySQL-ODBC from the CLI? >>> >>> Thanks >>> >>> David >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >>> CudaTel Communication Server Official FreeSWITCH >>> Sites http://www.freeswitch.org http://wiki.freeswitch.org >>> http://www.cluecon.com FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/9930dcd9/attachment-0001.html From garbytrash at gmail.com Sun Mar 18 01:10:16 2012 From: garbytrash at gmail.com (Zenny) Date: Sat, 17 Mar 2012 22:10:16 +0000 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: <20120317202301.1d80d7f1@anubis.defcon1> References: <20120317202301.1d80d7f1@anubis.defcon1> Message-ID: Isn't this open source? But the very front page of the vbilling.org site categorically claims the following: "vBilling is the the first OPEN SOURCE billing platform for FreeSWITCH. With vBilling you can start your own calling card services, send SMS, conference call and voice mail and much more. You can even manage your existing wholesale or retail VoIP business. Because it?s all Open Source, vBilling will always be free and fast evolving. We have developed vBilling using simple architecture that is very easy to configure, so with just a few steps you can create and deploy a very strong billing platform utilizing the power of FreeSWITCH. You can also deploy wide range of telephony applications and services, including IP PBXs, VoIP gateways, call center, IVR systems and much much more." And the header "why OPEN SOURCE?" in the same website states: "We believe that OPEN SOURCE can simply create better and more powerful software. While everyone contributes, the software becomes more mature and stable. OPEN SOURCE software might be cheaper than alternatives, but it has many other business benefits, too. It is often said that open source software wins because it is cheaper. However, the bigger factor in the success of open source software in industry has been performance. Collaborative development projects have opened the door to much wider input than is the case in a closed development environment with subsequent improvements in price and performance. OPEN SOURCE reduces R&D costs, increases productivity, improves efficiency, facilitates interoperability and encourages innovation. This lead us for the open source model." In that case, I am also interested to review the code, any links or hints about the source? Thanks! Or On 3/17/12, Bzzz wrote: > On Sat, 17 Mar 2012 01:01:21 +0300 > Muhammad Naseer Bhatti wrote: > >> I feel glad to announce the new release of *vBilling*, version 0.1.3. The >> first open source billing platform for FreeSWITCH. This versions has some >> major enhancements and supports many new features and bug fixes. > > ...and will not fail until the first (*noticed*) glitch. At that > time, the admin have chances to spend many hours/days to > manually track all errors - as the database makes no use of any > referential integrity... > > Jean-Yves > -- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From freeswitch at earthspike.net Sun Mar 18 01:13:53 2012 From: freeswitch at earthspike.net (John) Date: Sat, 17 Mar 2012 22:13:53 +0000 Subject: [Freeswitch-users] Do trim, do not top post, please. Was: Re: Any help me to installed Luac compiler (was: FreeSWITCH-users Digest, Vol 69, Issue 127) In-Reply-To: <-5889381104577349245@unknownmsgid> References: <-5889381104577349245@unknownmsgid> Message-ID: <4F650CA1.1080604@earthspike.net> Top vs Bottom posting is a long-running debate with advocacy from both camps. I remember such debates when posting on Usenet groups 25 years ago and it would appear that they have yet to be resolved... There isn't much of a trimming debate; no-one wants to read past 12 copies of the FreeSWITCH mailing list footer. And there really is no debate around the ridiculous pseudo-legal disclaimers which have no place nor meaning on a mailing list (although I appreciate that for some the footer is imposed by some corporate mail server). Finally, see what I've done here? I'm going to annoy everyone... ;) John On 17/03/12 21:00, Brian West wrote: > I always top post! > > /b > > Sent from my iPad > > On Mar 17, 2012, at 2:09 PM, Gabriel Gunderson wrote: > >> Everyone... more of this please! >> >> See how easy it was to follow the conversation here? >> >> >> Gabe Top vs Bottom posting is a long-running debate with advocacy from both camps. I remember such debates when posting on Usenet groups 25 years ago and it would appear that they have yet to be resolved... There isn't much of a trimming debate; no-one wants to read past 12 copies of the FreeSWITCH mailing list footer. And there really is no debate around the ridiculous pseudo-legal disclaimers which have no place nor meaning on a mailing list (although I appreciate that for some the footer is imposed by some corporate mail server). Finally, see what I've done here? I'm going to annoy everyone... ;) John From lazyvirus at gmx.com Sun Mar 18 01:26:15 2012 From: lazyvirus at gmx.com (Bzzz) Date: Sat, 17 Mar 2012 23:26:15 +0100 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: <20120317202301.1d80d7f1@anubis.defcon1> Message-ID: <20120317232615.58cb3993@anubis.defcon1> On Sat, 17 Mar 2012 22:10:16 +0000 Zenny wrote: > Isn't this open source? But the very front page of the vbilling.org > site categorically claims the following: It is easy these days to be fooled by words, especially because corporations, as usual, have exploited the flaws into basic OSS definition to push the confusion. This lead to OS, FS, FOSS & FLOSS (2 or 3 more definitions and we'll reach ROTFLOL:) See: http://en.wikipedia.org/wiki/Free_and_open_source_software and stay aware. JY -- BOFH excuse #342: HTTPD Error 4004 : very old Intel cpu - insufficient processing power From bote_radio at botecomm.com Sun Mar 18 01:55:45 2012 From: bote_radio at botecomm.com (Bote Man) Date: Sat, 17 Mar 2012 18:55:45 -0400 Subject: [Freeswitch-users] Do trim, do not top post, please. Was: Re: Any help me to installed Luac compiler (was: FreeSWITCH-users Digest, Vol 69, Issue 127) In-Reply-To: <4F650CA1.1080604@earthspike.net> References: <-5889381104577349245@unknownmsgid> <4F650CA1.1080604@earthspike.net> Message-ID: <007501cd0491$1763fc10$462bf430$@com> I'm reading this on C-news via my uucp link to media cybernetics. Looks fine to me. My space bar is broken on this keyboard, so this will be the last message I'm able to read. Bote > -----Original Message----- > From: John > Sent: Saturday, 17 March, 2012 18:14 > > Top vs Bottom posting is a long-running debate with advocacy from both > camps. I remember such debates when posting on Usenet groups 25 years > ago and it would appear that they have yet to be resolved... > > > John > > > On 17/03/12 21:00, Brian West wrote: > > I always top post! > > > > /b > > > > > > On Mar 17, 2012, at 2:09 PM, Gabriel Gunderson wrote: > > > >> Everyone... more of this please! > >> > >> See how easy it was to follow the conversation here? > >> > >> > >> Gabe > Top vs Bottom posting is a long-running debate with advocacy from both > camps. I remember such debates when posting on Usenet groups 25 years > ago and it would appear that they have yet to be resolved... From freeswitch at zoho.com Sat Mar 17 16:40:17 2012 From: freeswitch at zoho.com (freeswitch) Date: Sat, 17 Mar 2012 06:40:17 -0700 Subject: [Freeswitch-users] FreeSwitch & ODBC errors Message-ID: <13620e31ef5.4902843172079284242.1157408376614637062@zoho.com> Hello all, can you please help me with my settings? trying to setup pg and FS via odbc,not sure what i missed, tried to follow this guide - http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#5.__Compile_Freeswitch_.26_Set_Modules_to_Use_ODBC and here's what i have http://pastebin.freeswitch.org/18684 isql test ------------------------------------------- [root at echo-nh-voip bin]# isql -v freeswitch superadmin s3cret0w +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> dingdong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/879f1e8c/attachment.html From rgarrett at garrettnet.net Sat Mar 17 19:37:58 2012 From: rgarrett at garrettnet.net (Robbie A. Garrett) Date: Sat, 17 Mar 2012 16:37:58 +0000 Subject: [Freeswitch-users] firewall ports Message-ID: <6981B8DC-0B89-4FA3-BF79-670B177857AD@garrettnet.net> hey all, I'm using free switch with google voice for testing. outbound works great but inbound does not appear too be hitting the free switch box. is there a list of ports I need too nat / open on the firewall ? Sent from a mobile device. Please excuse shorthand and spelling. From bdfoster at endigotech.com Sun Mar 18 02:11:04 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 17 Mar 2012 19:11:04 -0400 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: <20120317232615.58cb3993@anubis.defcon1> References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> Message-ID: Who said that com On Sat, Mar 17, 2012 at 6:26 PM, Bzzz wrote: > On Sat, 17 Mar 2012 22:10:16 +0000 > Zenny wrote: > > > Isn't this open source? But the very front page of the vbilling.org > > site categorically claims the following: > > It is easy these days to be fooled by words, especially because > corporations, as usual, have exploited the flaws into basic OSS > definition to push the confusion. This lead to OS, FS, FOSS & FLOSS > (2 or 3 more definitions and we'll reach ROTFLOL:) > See: http://en.wikipedia.org/wiki/Free_and_open_source_software > and stay aware. > > JY > -- > BOFH excuse #342: > HTTPD Error 4004 : very old Intel cpu - insufficient processing power > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Again, just because it's compiled doesn't mean that it's open source... You do realize most if not all of freeswitch is compiled, right? -BDF -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/fce8afda/attachment-0001.html From bdfoster at endigotech.com Sun Mar 18 02:18:10 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 17 Mar 2012 19:18:10 -0400 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> Message-ID: Whoops... Just because it is compiled doesn't have anything to do with it being open source or not. There are likely a few reasons why it's compiled (like being faster). You all do realize that most if not all of FreeSWITCH is compiled, right? -BDF On Mar 17, 2012 7:11 PM, "Brian Foster" wrote: > Who said that com > > On Sat, Mar 17, 2012 at 6:26 PM, Bzzz wrote: > >> On Sat, 17 Mar 2012 22:10:16 +0000 >> Zenny wrote: >> >> > Isn't this open source? But the very front page of the vbilling.org >> > site categorically claims the following: >> >> It is easy these days to be fooled by words, especially because >> corporations, as usual, have exploited the flaws into basic OSS >> definition to push the confusion. This lead to OS, FS, FOSS & FLOSS >> (2 or 3 more definitions and we'll reach ROTFLOL:) >> See: http://en.wikipedia.org/wiki/Free_and_open_source_software >> and stay aware. >> >> JY >> -- >> BOFH excuse #342: >> HTTPD Error 4004 : very old Intel cpu - insufficient processing power >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > Again, just because it's compiled doesn't mean that it's open source... > > You do realize most if not all of freeswitch is compiled, right? > > -BDF > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/2dc85a8f/attachment.html From bdfoster at endigotech.com Sun Mar 18 02:27:16 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 17 Mar 2012 19:27:16 -0400 Subject: [Freeswitch-users] Do trim, do not top post, please. Was: Re: Any help me to installed Luac compiler (was: FreeSWITCH-users Digest, Vol 69, Issue 127) In-Reply-To: <007501cd0491$1763fc10$462bf430$@com> References: <-5889381104577349245@unknownmsgid> <4F650CA1.1080604@earthspike.net> <007501cd0491$1763fc10$462bf430$@com> Message-ID: Hey, for what it's worth, there are many on here that have the disclaimer imposed by their admins (including me, but I'm the admin... soo...) -BDF P.S.: That's why I do the "-BDF" thing... well, one of the reasons ;-) On Sat, Mar 17, 2012 at 6:55 PM, Bote Man wrote: > I'm reading this on C-news via my uucp link to media cybernetics. Looks > fine to me. > > My space bar is broken on this keyboard, so this will be the last > message I'm able to read. > > Bote > > > > -----Original Message----- > > From: John > > Sent: Saturday, 17 March, 2012 18:14 > > > > Top vs Bottom posting is a long-running debate with advocacy from both > > camps. I remember such debates when posting on Usenet groups 25 years > > ago and it would appear that they have yet to be resolved... > > > > > > John > > > > > > On 17/03/12 21:00, Brian West wrote: > > > I always top post! > > > > > > /b > > > > > > > > > On Mar 17, 2012, at 2:09 PM, Gabriel Gunderson > wrote: > > > > > >> Everyone... more of this please! > > >> > > >> See how easy it was to follow the conversation here? > > >> > > >> > > >> Gabe > > Top vs Bottom posting is a long-running debate with advocacy from both > > camps. I remember such debates when posting on Usenet groups 25 years > > ago and it would appear that they have yet to be resolved... > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/d8c4e935/attachment.html From bdfoster at endigotech.com Sun Mar 18 02:28:54 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 17 Mar 2012 19:28:54 -0400 Subject: [Freeswitch-users] firewall ports In-Reply-To: <6981B8DC-0B89-4FA3-BF79-670B177857AD@garrettnet.net> References: <6981B8DC-0B89-4FA3-BF79-670B177857AD@garrettnet.net> Message-ID: I don't think so... I do know that google voice never worked well with asterisk, though. And that machine wasn't behind a NAT. On Sat, Mar 17, 2012 at 12:37 PM, Robbie A. Garrett wrote: > hey all, > > I'm using free switch with google voice for testing. outbound works great > but inbound does not appear too be hitting the free switch box. > > is there a list of ports I need too nat / open on the firewall ? > > Sent from a mobile device. > Please excuse shorthand and spelling. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/f5bc8af3/attachment.html From bdfoster at endigotech.com Sun Mar 18 02:32:31 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 17 Mar 2012 19:32:31 -0400 Subject: [Freeswitch-users] FreeSwitch & ODBC errors In-Reply-To: <13620e31ef5.4902843172079284242.1157408376614637062@zoho.com> References: <13620e31ef5.4902843172079284242.1157408376614637062@zoho.com> Message-ID: I don't actually see a problem here. -BDF On Sat, Mar 17, 2012 at 9:40 AM, freeswitch wrote: > ** > Hello all, > > can you please help me with my settings? trying to setup pg and FS via > odbc,not sure what i missed, > tried to follow this guide - > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#5.__Compile_Freeswitch_.26_Set_Modules_to_Use_ODBC > > > and here's what i have > > http://pastebin.freeswitch.org/18684 > > > isql test > ------------------------------------------- > [root at echo-nh-voip bin]# isql -v freeswitch superadmin s3cret0w > +---------------------------------------+ > | Connected! | > | | > | sql-statement | > | help [tablename] | > | quit | > | | > +---------------------------------------+ > SQL> > > > dingdong > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/b7122329/attachment-0001.html From bdfoster at endigotech.com Sun Mar 18 02:40:05 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 17 Mar 2012 19:40:05 -0400 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: <20120317202301.1d80d7f1@anubis.defcon1> Message-ID: The source code IS available. Please do not spread FUD. You should probably take a look at https://github.com/digitallinx/vBilling before posting anymore about this really great software that a developer has graciously set into the world, free of charge and open source. -BDF (And, please, if you are going to rant about someone's software, get your facts right first. Also, show some respect and don't do it on the developer's release announcement. It's just common courtesy.) On Sat, Mar 17, 2012 at 6:10 PM, Zenny wrote: > Isn't this open source? But the very front page of the vbilling.org > site categorically claims the following: > > "vBilling is the the first OPEN SOURCE billing platform for > FreeSWITCH. With vBilling you can start your own calling card > services, send SMS, conference call and voice mail and much more. You > can even manage your existing wholesale or retail VoIP business. > Because it?s all Open Source, vBilling will always be free and fast > evolving. We have developed vBilling using simple architecture that is > very easy to configure, so with just a few steps you can create and > deploy a very strong billing platform utilizing the power of > FreeSWITCH. You can also deploy wide range of telephony applications > and services, including IP PBXs, VoIP gateways, call center, IVR > systems and much much more." > > And the header "why OPEN SOURCE?" in the same website states: > > "We believe that OPEN SOURCE can simply create better and more > powerful software. While everyone contributes, the software becomes > more mature and stable. > OPEN SOURCE software might be cheaper than alternatives, but it has > many other business benefits, too. It is often said that open source > software wins because it is cheaper. However, the bigger factor in the > success of open source software in industry has been performance. > Collaborative development projects have opened the door to much wider > input than is the case in a closed development environment with > subsequent improvements in price and performance. OPEN SOURCE reduces > R&D costs, increases productivity, improves efficiency, facilitates > interoperability and encourages innovation. This lead us for the open > source model." > > In that case, I am also interested to review the code, any links or > hints about the source? Thanks! > > Or > > On 3/17/12, Bzzz wrote: > > On Sat, 17 Mar 2012 01:01:21 +0300 > > Muhammad Naseer Bhatti wrote: > > > >> I feel glad to announce the new release of *vBilling*, version 0.1.3. > The > >> first open source billing platform for FreeSWITCH. This versions has > some > >> major enhancements and supports many new features and bug fixes. > > > > ...and will not fail until the first (*noticed*) glitch. At that > > time, the admin have chances to spend many hours/days to > > manually track all errors - as the database makes no use of any > > referential integrity... > > > > Jean-Yves > > -- > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/1b348c42/attachment.html From bdfoster at endigotech.com Sun Mar 18 02:42:09 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 17 Mar 2012 19:42:09 -0400 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: <20120317232615.58cb3993@anubis.defcon1> References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> Message-ID: Same goes for you. Please read my last email. -BDF On Sat, Mar 17, 2012 at 6:26 PM, Bzzz wrote: > On Sat, 17 Mar 2012 22:10:16 +0000 > Zenny wrote: > > > Isn't this open source? But the very front page of the vbilling.org > > site categorically claims the following: > > It is easy these days to be fooled by words, especially because > corporations, as usual, have exploited the flaws into basic OSS > definition to push the confusion. This lead to OS, FS, FOSS & FLOSS > (2 or 3 more definitions and we'll reach ROTFLOL:) > See: http://en.wikipedia.org/wiki/Free_and_open_source_software > and stay aware. > > JY > -- > BOFH excuse #342: > HTTPD Error 4004 : very old Intel cpu - insufficient processing power > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/24a9b7a5/attachment.html From lazyvirus at gmx.com Sun Mar 18 03:00:00 2012 From: lazyvirus at gmx.com (Bzzz) Date: Sun, 18 Mar 2012 01:00:00 +0100 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> Message-ID: <20120318010000.67f91b94@anubis.defcon1> On Sat, 17 Mar 2012 19:18:10 -0400 Brian Foster wrote: > Just because it is compiled doesn't have anything to do with it being open > source or not. There are likely a few reasons why it's compiled (like being > faster). You all do realize that most if not all of FreeSWITCH is compiled, > right? But either docs & source are crystal clear: when you download the FS source, you got a... source; when you purchase a g729 license, doc warns you before it is a proprietary binary. So, everything's blunt from top to bottom without any "surprise". JY -- This document should be read only by those persons to whom it is addressed. If you have received this message it was obviously addressed to you and therefore you can read it, even it we didnt mean to send it to you. However, if the contents of this email make no sense whatsoever then you probably were not the intended recipient, or, you are a mindless cretin; either way, you should immediately delete yourself & destroy your computer! Once you have taken this action please contact us.. no you idiot, you cant use your computer, you just destroyed it, and by the way, you are also deleted, but we digress...... The Originator of this email is not liable for the transmission of the information contained in this communication, unless they are the originator in which case they probably are liable and rightly so considering the content of the aforementioned communication. In the event that the originator did not send this email to you, then please return it to us and attach a scanned-in picture of your mothers brothers wife wearing nothing but cami-knickers, and we will immediately refund you exactly half of what you paid for the can of Pal Meaty-Bites you bought when you went to Woolies yesterday. We take no responsibility for non-receipt of this email because we are running Windows NT & everyone knows how glitchy that can be. In the event that you do get this message then please note that we take no responsibility for that either. Nor will we accept any liability, tacit or implied, for any damage you may or may not incur as a result of receiving, or not, as the case may be, from time to time, notwithstanding all liabilities implied or otherwise, ummm, shit, where was I..umm, no matter what happens, IT's NOT, and NEVER WILL BE, OUR FAULT! From bdfoster at endigotech.com Sun Mar 18 03:03:47 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 17 Mar 2012 20:03:47 -0400 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: <20120318010000.67f91b94@anubis.defcon1> References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> Message-ID: Once again, just because it's compiled doesn't mean it is or it isn't open source. The source is on github, in uncompiled form. Please check one of my last emails. Also, that particular file is compiled LUA, which, believe it or not, isn't proprietary. -BDF On Sat, Mar 17, 2012 at 8:00 PM, Bzzz wrote: > On Sat, 17 Mar 2012 19:18:10 -0400 > Brian Foster wrote: > > > Just because it is compiled doesn't have anything to do with it being > open > > source or not. There are likely a few reasons why it's compiled (like > being > > faster). You all do realize that most if not all of FreeSWITCH is > compiled, > > right? > > But either docs & source are crystal clear: when you download the > FS source, you got a... source; when you purchase a g729 license, > doc warns you before it is a proprietary binary. > So, everything's blunt from top to bottom without any "surprise". > > JY > -- > This document should be read only by those persons to whom it is > addressed. If you have received this message it was obviously > addressed to you and therefore you can read it, even it we didnt > mean to send it to you. However, if the contents of this email > make no sense whatsoever then you probably were not the intended > recipient, or, you are a mindless cretin; either way, you should > immediately delete yourself & destroy your computer! Once you > have taken this action please contact us.. no you idiot, you cant > use your computer, you just destroyed it, and by the way, you are > also deleted, but we digress...... > The Originator of this email is not liable for the transmission > of the information contained in this communication, unless they > are the originator in which case they probably are liable and > rightly so considering the content of the aforementioned > communication. > In the event that the originator did not send this email to you, > then please return it to us and attach a scanned-in picture of > your mothers brothers wife wearing nothing but cami-knickers, and > we will immediately refund you exactly half of what you paid for > the can of Pal Meaty-Bites you bought when you went to Woolies > yesterday. > We take no responsibility for non-receipt of this email because > we are running Windows NT & everyone knows how glitchy that can be. > In the event that you do get this message then please note that > we take no responsibility for that either. Nor will we accept any > liability, tacit or implied, for any damage you may or may not > incur as a result of receiving, or not, as the case may be, from > time to time, notwithstanding all liabilities implied or otherwise, > ummm, shit, where was I..umm, no matter what happens, IT's NOT, > and NEVER WILL BE, OUR FAULT! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/b3ec6b36/attachment-0001.html From lazyvirus at gmx.com Sun Mar 18 03:19:50 2012 From: lazyvirus at gmx.com (Bzzz) Date: Sun, 18 Mar 2012 01:19:50 +0100 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> Message-ID: <20120318011950.70feee05@anubis.defcon1> On Sat, 17 Mar 2012 20:03:47 -0400 Brian Foster wrote: > Once again, just because it's compiled doesn't mean it is or it isn't open > source. The source is on github, in uncompiled form. Please check one of my > last emails. Also, that particular file is compiled LUA, which, believe it > or not, isn't proprietary. Hmm, I cloned the source, made a check, but LUA sources isn't in trunk - but as you say this is not a non-OS proof, may be a memory lapse. Anyway, open or closed it has the default of 99% of web applications: database consistence isn't enforced by referential integrities, which makes it production unusable. JY -- From virbhati at gmail.com Sun Mar 18 03:40:16 2012 From: virbhati at gmail.com (virendra bhati) Date: Sun, 18 Mar 2012 06:10:16 +0530 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 69, Issue 130 In-Reply-To: References: Message-ID: thanks Patrick, Lua was installed on server but luac command was not working. I re-installed lua-5.1.4-1.el6.x86_64.rpm and now everything working fine. thanks for help .... On Sat, Mar 17, 2012 at 10:31 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. endconf flag not working (Bote Man) > 2. Need Help to handle message (Balamurugan Mahendran) > 3. Re: FreeSWITCH 1.2 is on the horizon. (Trever L. Adams) > 4. Re: Any help me to installed Luac compiler (was: > FreeSWITCH-users Digest, Vol 69, Issue 127) (Patrick Lists) > 5. Re: Advice how to start with H.323 (Patrick Lists) > 6. ODBC instalation (David Villasmil) > > > ---------- Forwarded message ---------- > From: "Bote Man" > To: "'FreeSWITCH Users Help'" > Cc: > Date: Sat, 17 Mar 2012 04:34:02 -0400 > Subject: [Freeswitch-users] endconf flag not working > > I have a Windows installation (yes, FML) that I can not change to linux, > I'm stuck with it. It is running the .msi build from Feb. 29 that otherwise > works great. This past weekend's build was declared unusable by Windows > installer, go figure.**** > > ** ** > > I'm using the conference_auto_outcall feature to initiate an outbound > group conference call to 4 endpoints. It works OK, except that because some > of them are FXO gateways to analog lines they essentially ring until the > initiator disconnects. **** > > ** ** > > Unfortunately, the endconf flag is not being honored by mod_conference so > the initiator hangs up and the outbound calls keep on ringing because the > number of participants is not less than 1. I would expect the endconf flag > to drop the entire conference when the extension that created it released, > as described by the wiki page.**** > > ** ** > > As I am pulling down sources I?m hoping somebody with experience in > mod_conference can verify that the 'endconf' flag should or should not > work? Have you played with this setting before and what were your results? > **** > > ** ** > > I am brand new with FreeSWITCH and need all the help I can get.**** > > ** ** > > Thanks.**** > > ** ** > > ** ** > > Bote**** > > ** ** > > ** ** > > > ---------- Forwarded message ---------- > From: Balamurugan Mahendran > To: FreeSWITCH Users Help > Cc: > Date: Sat, 17 Mar 2012 15:51:29 +0530 > Subject: [Freeswitch-users] Need Help to handle message > where can I find more details about nua? > > > tport_wakeup_pri(0xa7c160): events IN > tport_recv_event(0xa7c160) > tport_recv_iovec(0xa7c160) msg 0x7f18080037b0 from (udp/10.248.13.3:5060) > has 487 bytes, veclen = 1 > tport_deliver(0xa7c160): msg 0x7f18080037b0 (487 bytes) from udp/ > 23.21.230.74:5060/sip next=(nil) > nta: received MESSAGE sip:+15032721367 at 50.12.112.39 SIP/2.0 (CSeq 102) > nta: Via check: received=23.21.230.74 > nta: canonizing sip:+15032721367 at 50.12.112.39 with contact > nta: MESSAGE (102) 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::0xa69c50, 0xa69490, 0x7f17f80f5180) called > soa_set_params(static::0x7f17f80f52a0, ...) called > nta_leg_tcreate(0x7f17f8088d30) > tport_tsend(0xa7c160) tpn = UDP/23.211.230.74:5060 > tport_resolve addrinfo = 23.211.230.74:5060 > tport_by_addrinfo(0xa7c160): not found by name UDP/23.211.230.74:5060 > tport_vsend(0xa7c160): 539 bytes of 539 to udp/23.211.230.74:5060 > tport_vsend returned 539 > nta: sent 200 OK for MESSAGE (102) > nua(0x7f17f80f5180): event i_message 200 OK > nua: nua_application_event: entering > nua: nua_handle_magic: entering > nua: nua_respond: entering > nua(0x7f17f80f5180): sent signal r_respond > nua: nua_handle_magic: entering > nua: nua_handle_destroy: entering > nua(0x7f17f80f5180): sent signal r_destroy > nua(0x7f17f80f5180): recv signal r_respond 202 Accepted > nua(0x7f17f80f5180): event i_error 500 Responding to a Non-Existing Request > nua(0x7f17f80f5180): recv signal r_destroy > nta_leg_destroy(0x7f17f8088d30) > soa_destroy(static::0x7f17f80f52a0) called > nta: timer J fired, terminate 200 response > incoming_reclaim_all((nil), (nil), 0x7f180752ec40) > nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free > nta: timer set next to 14002 ms > tport_wakeup_pri(0xa7c160): events IN > tport_recv_event(0xa7c160) > tport_recv_iovec(0xa7c160) msg 0x7f1808147e10 from (udp/10.248.13.3:5060) > has 4 bytes, veclen = 1 > tport_deliver(0xa7c160): bad msg 0x7f1808147e10 (4 bytes) from udp/ > 115.11.5.34:5060/sip next=(nil) > nta: timer J fired, terminate 200 response > incoming_reclaim_all((nil), (nil), 0x7f180752ec40) > nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free > nta: timer not set > tport_wakeup_pri(0xa7c160): events IN > tport_recv_event(0xa7c160) > tport_recv_iovec(0xa7c160) msg 0x7f1808147e10 from (udp/10.248.13.3:5060) > has 4 bytes, veclen = 1 > tport_deliver(0xa7c160): bad msg 0x7f1808147e10 (4 bytes) from udp/ > 115.11.5.34:5060/sip next=(nil) > freeswitch at internal> > > > Thanks, > Bala > > > ---------- Forwarded message ---------- > From: "Trever L. Adams" > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Sat, 17 Mar 2012 08:08:45 -0600 > Subject: Re: [Freeswitch-users] FreeSWITCH 1.2 is on the horizon. > 12 ????? 2012 ?. 23:19 ???????????? Ken Rice > ???????: > > > > The FreeSWITCH Development Team is happy to announce that 1.2 is > > officially on the horizon. > > Starting Wed March 14th 2012, the Development branch of FreeSWITCH > > will reach a Feature Freeze. > > > > What does this mean for you the user? It means we will have a > > stable known feature set heading into the Release Candidate Cycle. > > Only patches that Fix Bugs will be accepted. If you have a feature you > > would like to see included, get us a Jira with a full patch set so we > > can evaluate its inclusion with 1.2. > > > > What we need from the community: Testing, Testing, more Testing, > > and Documentation Updates. > > > > The freeze will last 2 to 4 weeks as we spool up testing and > > everything else we need to get the Release Candidates ready for prime > > time. > > > > If you have outstanding bugs on Jira, please help us help you > > during this time by making sure all information on them is up to date. > > Grab the latest GIT Head and see if your bugs have been resolved and > > someone forgot to close your Jira. > > > > If you want to help or need some help diagnosing and issue visit > > us on IRC via irc.freenode.net/#freeswitch any time. > > > > The FreeSWITCH Dev Team > > > The following is a bug which is affecting several people. Several have > just given up and gone FXS free. I do not have that option. I am willing > to help out if I can. > > http://jira.freeswitch.org/browse/OPENZAP-173 > > Trever > > > > > > ---------- Forwarded message ---------- > From: Patrick Lists > To: FreeSWITCH Users Help > Cc: > Date: Sat, 17 Mar 2012 15:42:43 +0100 > Subject: Re: [Freeswitch-users] Any help me to installed Luac compiler > (was: FreeSWITCH-users Digest, Vol 69, Issue 127) > On 03/17/2012 08:26 AM, virendra bhati wrote: > >> Hi Patrick , >> >> I tryed a lot effort on google but no help i got and even i sent e-mail >> to lua mail list as well. but no one replied on it. >> > > Please do not top-post and properly trim your replies (remove unrelated > stuff). > > If you have the lua rpm installed on your CentOS system then luac is > already available. Check with: > > $ rpm -q lua > > lua-5.1.4-4.1.el6.x86_64 <-- so it's installed > > If you don't have lua installed then install it as root with: > # yum install lua > > luac is part of the lua package: > > $ rpm -ql lua | grep luac > > /usr/bin/luac > /usr/share/doc/lua-5.1.4/luac.**html > /usr/share/man/man1/luac.1.gz > > Since you seem to be lacking basic Red Hat/CentOS knowledge and skills may > I suggest this book: > > RHCSA/RHCE Red Hat Linux Certification Study Guide (Exams EX200 & EX300), > 6th Edition written by Michael Jang > > http://www.amazon.co.uk/RHCSA-**Linux-Certification-Study-** > Edition/dp/0071765654/ref=sr_**1_1?s=books&ie=UTF8&qid=**1331995110&sr=1-1 > > Regards, > Patrick > > > > > ---------- Forwarded message ---------- > From: Patrick Lists > To: FreeSWITCH Users Help > Cc: > Date: Sat, 17 Mar 2012 15:44:59 +0100 > Subject: Re: [Freeswitch-users] Advice how to start with H.323 > On 03/17/2012 03:30 AM, Brian Foster wrote: > [snip] > >> > are there any dependencies for the opal and libpt versions? >> >> Those should be taken care of by the build script (if there are any) >> > > That's assuming that the build script does the proper thing. Last time I > looked, it pulled svn trunk versions of ptlib and opal which are too new > for mod_opal. Afaik it needs the Sirius or maybe the Luyten release. > In the past I built mod_opal without a problem (not using that script) > with the Sirius release of ptlib and opal installed. > > Regarrds, > Patrick > > > > > ---------- Forwarded message ---------- > From: David Villasmil > To: FreeSWITCH Users Help > Cc: > Date: Sat, 17 Mar 2012 18:00:21 +0100 > Subject: [Freeswitch-users] ODBC instalation > Hello guys, > > I just installed odbc support on my debian and want to hace FS use it for > lcr. > I followed the steps on; > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core > > the installation is OK in terms of isql but i don't know whether FS is > able to connect as I just can't see any reference to ODBC in the logs. > > Is there any way of testing FS's connectivity to MySQL-ODBC from the CLI? > > Thanks > > David > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati at gmail.com Skype id:- virbhati2 Hyderabad(India) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/53cf623c/attachment-0001.html From bdfoster at endigotech.com Sun Mar 18 04:18:33 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 17 Mar 2012 21:18:33 -0400 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: <20120318011950.70feee05@anubis.defcon1> References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> <20120318011950.70feee05@anubis.defcon1> Message-ID: You're more than welcome to offer your help to him, if you would like that particular feature. This isn't really the place to do it though. I believe he has a forum hosted somewhere, you might want to check there or send him an email if you feel that's appropriate. -BDF On Mar 17, 2012 8:21 PM, "Bzzz" wrote: > On Sat, 17 Mar 2012 20:03:47 -0400 > Brian Foster wrote: > > > Once again, just because it's compiled doesn't mean it is or it isn't > open > > source. The source is on github, in uncompiled form. Please check one of > my > > last emails. Also, that particular file is compiled LUA, which, believe > it > > or not, isn't proprietary. > > Hmm, I cloned the source, made a check, but LUA sources isn't in > trunk - but as you say this is not a non-OS proof, may be a memory > lapse. > > Anyway, open or closed it has the default of 99% of web > applications: database consistence isn't enforced by referential > integrities, which makes it production unusable. > > JY > -- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/2ff1906b/attachment.html From lazyvirus at gmx.com Sun Mar 18 04:28:23 2012 From: lazyvirus at gmx.com (Bzzz) Date: Sun, 18 Mar 2012 02:28:23 +0100 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> <20120318011950.70feee05@anubis.defcon1> Message-ID: <20120318022823.4766c4ec@anubis.defcon1> On Sat, 17 Mar 2012 21:18:33 -0400 Brian Foster wrote: > You're more than welcome to offer your help to him, if you would like that > particular feature. I surely would, project's very interesting and IMHO very important to FS future spreading, if only the OP used a real RDBMS. -- From kjoseph.us at gmail.com Sun Mar 18 05:11:43 2012 From: kjoseph.us at gmail.com (Joseph Khoury) Date: Sat, 17 Mar 2012 19:11:43 -0700 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: <20120318022823.4766c4ec@anubis.defcon1> References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> <20120318011950.70feee05@anubis.defcon1> <20120318022823.4766c4ec@anubis.defcon1> Message-ID: I like the project and I would like to contribute. But with a compiled code without the source( FreeSwitch comes compiled and you have the full source and can compile it yourself). Having said that Open starts with O , Compiled = Closed starts with C and ends with luaC :) On Sat, Mar 17, 2012 at 6:28 PM, Bzzz wrote: > On Sat, 17 Mar 2012 21:18:33 -0400 > Brian Foster wrote: > > > You're more than welcome to offer your help to him, if you would like > that > > particular feature. > > I surely would, project's very interesting and IMHO very important to > FS future spreading, if only the OP used a real RDBMS. > > -- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/3235c9b8/attachment.html From bdfoster at endigotech.com Sun Mar 18 05:32:39 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 17 Mar 2012 22:32:39 -0400 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> <20120318011950.70feee05@anubis.defcon1> <20120318022823.4766c4ec@anubis.defcon1> Message-ID: Ever heard of gcc? Probably should go over the source files of freeswitch, and compare it to what you see in /usr/local/freeswitch. On Mar 17, 2012 10:12 PM, "Joseph Khoury" wrote: > I like the project and I would like to contribute. But with a compiled > code without the source( FreeSwitch comes compiled and you have the full > source and can compile it yourself). > > Having said that Open starts with O , Compiled = Closed starts with C and > ends with luaC :) > > > > On Sat, Mar 17, 2012 at 6:28 PM, Bzzz wrote: > >> On Sat, 17 Mar 2012 21:18:33 -0400 >> Brian Foster wrote: >> >> > You're more than welcome to offer your help to him, if you would like >> that >> > particular feature. >> >> I surely would, project's very interesting and IMHO very important to >> FS future spreading, if only the OP used a real RDBMS. >> >> -- >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/b7ccc4a6/attachment.html From anton.jugatsu at gmail.com Sun Mar 18 07:46:06 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Sun, 18 Mar 2012 08:46:06 +0400 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: <4F63B627.5070206@telefaks.de> References: <4F63B627.5070206@telefaks.de> Message-ID: Try using Yate instead of FS for h323 gateway. 17.03.2012 1:54 ???????????? "Peter Steinbach" > ** > I want to create a H323 Gateway in order to receive calls from an external > H323 Server and transfer them to another Freeswitch server. > > I started on my Ubuntu 10.04 with mod_opal and spent some hours with > trying the following 2 ways: > > - Using buildopal.sh, I had to adapt the script slightly and had to do > some symlinks in order to progress. At the end I got stuck with a compile > error "class PSTUNClient' has no member named 'InvalidateCache'" during > compilation of opal manager > - I then installed the libpt and libopal packes from apt and started > to compile mod_opal. I then got stuck with a compile error "errror: no > matching function for call to > 'OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, void*&, > unsigned int&, OpalConnection::StringOptions*&)'" > > So this did not seem to be the right way as regards to future ability to > compile it with system updates. > > Therefore I would like to ask you for some advice: > > - what is the best way to go: mod_opal or mod_h323? I think the > feature set will be somehow the same for my purposes (I only need voice, > PCMA), so I am asking, which one is more stable and compiles better. > - what is best preferred platform (CentOS/Debian/??) for compiling > this? > - are there any dependencies for the opal and libpt versions? > - Who had recently made running this solution and can give me some > advice? > > -- > > With kind regards > Peter Steinbach > > Telefaks Services GmbHmailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/010d267e/attachment-0001.html From garbytrash at gmail.com Sun Mar 18 09:23:38 2012 From: garbytrash at gmail.com (Zenny) Date: Sun, 18 Mar 2012 06:23:38 +0000 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> <20120318011950.70feee05@anubis.defcon1> <20120318022823.4766c4ec@anubis.defcon1> Message-ID: Yep, I checked the "source" at https://github.com/digitallinx/vBilling and git cloned it to check the source. But I could not find the sources to the *.bin files in the "source". If I am wrong, please point to the source of the binaries. Like Brian stated, I also believe vBilling may help popularize FreeSWITCH only if it a truly open sourced without gimmicks. Thanks! On 3/18/12, Brian Foster wrote: > Ever heard of gcc? Probably should go over the source files of freeswitch, > and compare it to what you see in /usr/local/freeswitch. > On Mar 17, 2012 10:12 PM, "Joseph Khoury" wrote: > >> I like the project and I would like to contribute. But with a compiled >> code without the source( FreeSwitch comes compiled and you have the full >> source and can compile it yourself). >> >> Having said that Open starts with O , Compiled = Closed starts with C and >> ends with luaC :) >> >> >> >> On Sat, Mar 17, 2012 at 6:28 PM, Bzzz wrote: >> >>> On Sat, 17 Mar 2012 21:18:33 -0400 >>> Brian Foster wrote: >>> >>> > You're more than welcome to offer your help to him, if you would like >>> that >>> > particular feature. >>> >>> I surely would, project's very interesting and IMHO very important to >>> FS future spreading, if only the OP used a real RDBMS. >>> >>> -- >>> From gcd at i.ph Sun Mar 18 11:45:49 2012 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 18 Mar 2012 16:45:49 +0800 Subject: [Freeswitch-users] gsmopen cable, nokia pop port plug Message-ID: hello gang, i'd like make my own GSMopen audio-data cable but i can't find Nokia FBUS pop port plug from cellshops here in the Phils. any links where can i buy this stuff? can i order assembled cable, too? i appreciate for your help. tks, nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/9badf2a4/attachment.html From bote_radio at botecomm.com Sun Mar 18 14:14:22 2012 From: bote_radio at botecomm.com (Bote Man) Date: Sun, 18 Mar 2012 07:14:22 -0400 Subject: [Freeswitch-users] reverse time skew warning Message-ID: <00e201cd04f8$46a6a7e0$d3f3f7a0$@com> What kind of problems can arise from the FreeSWITCH error message "Reverse time skew!"? I notice in the system logs that at the time of the FreeSWITCH complaint, ntp adjusted the system time backwards from 1:00:13.982493600Z to 1:00:13.982000000Z which is less than half a millisecond. That sounds negligible to you and mean, but that is an eternity for an android. I just hope it doesn't cause a serious problem down the road like a segfault or something every time ntp adjusts the time. Any comments? Thanks. Bote -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/e1b7fb0f/attachment.html From admin at blindi.net Sun Mar 18 16:10:13 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Sun, 18 Mar 2012 14:10:13 +0100 (CET) Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> <20120318011950.70feee05@anubis.defcon1> <20120318022823.4766c4ec@anubis.defcon1> Message-ID: Hi Zenny The question: Is vbilling really opensource? Open source products can be installed on all operating systems. I found the *.bin files only for debian ubuntu and centos. For example: gentoo, slackware, bsds, and others are not supportet. So much for open source. :-) --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From nbhatti at gmail.com Sun Mar 18 16:45:09 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Sun, 18 Mar 2012 16:45:09 +0300 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> <20120318011950.70feee05@anubis.defcon1> <20120318022823.4766c4ec@anubis.defcon1> Message-ID: Folks, vBilling is a real open source application. The fact that lua source was not published yet because we are rapidly working on updates. We will push the lua source in a few days once the product becomes more stable. *.bin files were compiled to get better performance. It is compiled bytecode only. Currently we support CentOS and Debian based distros like Debian/Ubuntu. In future we are planning to add other distributions as well. Feel free to contribute if you would like to port vBilling to any other systems. It simply needs MySQL, a web server with php and FreeSWITCH to run. Open Source does not means the product is supported by all operating systems. It may or may not be supported by just one or many operating systems. Can you or would like to install Open Source "Linux" on a windows operating system? :-) I would appreciate if we can have constructive comments and community support to make this a success project. Thanks, Muhammad On Sun, Mar 18, 2012 at 4:10 PM, Thomas Hoellriegel wrote: > Hi Zenny > The question: > Is vbilling really opensource? > Open source products can be installed on all operating systems. > I found the *.bin files only for debian ubuntu and centos. > For example: gentoo, slackware, bsds, and others are not supportet. > So much for open source. > :-) > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/d53a2a65/attachment.html From bdfoster at endigotech.com Sun Mar 18 17:29:00 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 18 Mar 2012 10:29:00 -0400 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: <4F64A36B.2060801@puzzled.xs4all.nl> References: <4F63B627.5070206@telefaks.de> <4F64A36B.2060801@puzzled.xs4all.nl> Message-ID: That patch probably fixes that issue. If not, it's easily fixed. On Mar 17, 2012 10:45 AM, "Patrick Lists" wrote: > On 03/17/2012 03:30 AM, Brian Foster wrote: > [snip] > > > are there any dependencies for the opal and libpt versions? > > > > Those should be taken care of by the build script (if there are any) > > That's assuming that the build script does the proper thing. Last time I > looked, it pulled svn trunk versions of ptlib and opal which are too new > for mod_opal. Afaik it needs the Sirius or maybe the Luyten release. > In the past I built mod_opal without a problem (not using that script) > with the Sirius release of ptlib and opal installed. > > Regarrds, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/d0baa119/attachment-0001.html From peter.olsson at visionutveckling.se Sun Mar 18 17:43:05 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 18 Mar 2012 14:43:05 +0000 Subject: [Freeswitch-users] reverse time skew warning In-Reply-To: <00e201cd04f8$46a6a7e0$d3f3f7a0$@com> References: <00e201cd04f8$46a6a7e0$d3f3f7a0$@com> Message-ID: <1FFF97C269757C458224B7C895F35F1505ED2C@cantor.std.visionutv.se> It might cause bad audio during a very short time - however, as you say it's a very short time so it shouldn't cause any real problems. The timer loop will always detect this though, no matter how small the time change was. The best way is to use a monotonic clock. On Linux this should be default, on Windows it requires a specific startup param, on Mac OS X it's not available at this time. The monotonic clock doesn't change when time changes, so it's always a better choice. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Bote Man [bote_radio at botecomm.com] Skickat: den 18 mars 2012 12:14 Till: 'FreeSWITCH Users Help' ?mne: [Freeswitch-users] reverse time skew warning What kind of problems can arise from the FreeSWITCH error message "Reverse time skew!"? I notice in the system logs that at the time of the FreeSWITCH complaint, ntp adjusted the system time backwards from 1:00:13.982493600Z to 1:00:13.982000000Z which is less than half a millisecond. That sounds negligible to you and mean, but that is an eternity for an android. I just hope it doesn't cause a serious problem down the road like a segfault or something every time ntp adjusts the time. Any comments? Thanks. Bote !DSPAM:4f65c2f932761179111321! From peter.olsson at visionutveckling.se Sun Mar 18 17:45:32 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 18 Mar 2012 14:45:32 +0000 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: References: <4F63B627.5070206@telefaks.de> <4F64A36B.2060801@puzzled.xs4all.nl>, Message-ID: <1FFF97C269757C458224B7C895F35F15060D3D@cantor.std.visionutv.se> I'm using mod_h323. It works quite well - even though it's not 100% perfect, but the same goes for mod_opal... I've used them both, but I actually think mod_h323 is a better and more stable choice. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Brian Foster [bdfoster at endigotech.com] Skickat: den 18 mars 2012 15:29 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Advice how to start with H.323 That patch probably fixes that issue. If not, it's easily fixed. On Mar 17, 2012 10:45 AM, "Patrick Lists" > wrote: On 03/17/2012 03:30 AM, Brian Foster wrote: [snip] > > are there any dependencies for the opal and libpt versions? > > Those should be taken care of by the build script (if there are any) That's assuming that the build script does the proper thing. Last time I looked, it pulled svn trunk versions of ptlib and opal which are too new for mod_opal. Afaik it needs the Sirius or maybe the Luyten release. In the past I built mod_opal without a problem (not using that script) with the Sirius release of ptlib and opal installed. Regarrds, Patrick _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f65f20a32765192515336! From bdfoster at endigotech.com Sun Mar 18 17:55:51 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 18 Mar 2012 10:55:51 -0400 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> <20120318011950.70feee05@anubis.defcon1> <20120318022823.4766c4ec@anubis.defcon1> Message-ID: I think we need a conference call session on what open source is. It's actually pretty simple, though. The difference between compiled and open source? -They are two different things. Open source software runs on any operating system -Not true, as stated above. Believe it or not there's open source software that runs on Windows only. Or Free/NetBSD only. I could go on. It's not open source unless the developers have no way to make money from it. -Absolutely wrong. Many projects offer support services, training, and sometimes they have a community edition and a paid edition. If anyone has any questions concerning the differences between open and closed source let me/us know. But, please please please, do not discredit a developer and his project until you know the facts. Right now, this project is in it's early days, and needs everything to go just right to make this a successful project. -BDF On Mar 18, 2012 9:47 AM, "Muhammad Naseer Bhatti" wrote: > Folks, > vBilling is a real open source application. The fact that lua source was > not published yet because we are rapidly working on updates. We will push > the lua source in a few days once the product becomes more stable. *.bin > files were compiled to get better performance. It is compiled bytecode > only. > > Currently we support CentOS and Debian based distros like Debian/Ubuntu. > In future we are planning to add other distributions as well. Feel free to > contribute if you would like to port vBilling to any other systems. It > simply needs MySQL, a web server with php and FreeSWITCH to run. > > Open Source does not means the product is supported by all operating > systems. It may or may not be supported by just one or many operating > systems. Can you or would like to install Open Source "Linux" on a windows > operating system? :-) > > I would appreciate if we can have constructive comments and community > support to make this a success project. > > Thanks, > Muhammad > > On Sun, Mar 18, 2012 at 4:10 PM, Thomas Hoellriegel wrote: > >> Hi Zenny >> The question: >> Is vbilling really opensource? >> Open source products can be installed on all operating systems. >> I found the *.bin files only for debian ubuntu and centos. >> For example: gentoo, slackware, bsds, and others are not supportet. >> So much for open source. >> :-) >> >> --------------- >> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: >> http://www.blindi.net/callback >> homepage: http://www.blindi.net >> blinde-misc mailingliste f?r blinde. anmeldung unter: >> http://www.blindi.net/mailman/**listinfo/blinde-misc >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/efeda293/attachment.html From garbytrash at gmail.com Sun Mar 18 18:18:26 2012 From: garbytrash at gmail.com (Zenny) Date: Sun, 18 Mar 2012 15:18:26 +0000 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> <20120318011950.70feee05@anubis.defcon1> <20120318022823.4766c4ec@anubis.defcon1> Message-ID: Thanks Mr. Bhatti for your confirmation and commitment to OS. I look forward to the sources of the binaries from lua. Indeed, you are doing a great job. Keep it on! :-) PS: I am seeing a deviation of this thread. It is MPL licensed from what I read it here: https://github.com/digitallinx/vBilling/tree/master/htdocs. When the sources of the lua binaries be released, it satisfies the claim. ;-) On 3/18/12, Muhammad Naseer Bhatti wrote: > Folks, > vBilling is a real open source application. The fact that lua source was > not published yet because we are rapidly working on updates. We will push > the lua source in a few days once the product becomes more stable. *.bin > files were compiled to get better performance. It is compiled bytecode > only. > > Currently we support CentOS and Debian based distros like Debian/Ubuntu. In > future we are planning to add other distributions as well. Feel free to > contribute if you would like to port vBilling to any other systems. It > simply needs MySQL, a web server with php and FreeSWITCH to run. > > Open Source does not means the product is supported by all operating > systems. It may or may not be supported by just one or many operating > systems. Can you or would like to install Open Source "Linux" on a windows > operating system? :-) > > I would appreciate if we can have constructive comments and community > support to make this a success project. > > Thanks, > Muhammad > > On Sun, Mar 18, 2012 at 4:10 PM, Thomas Hoellriegel wrote: > >> Hi Zenny >> The question: >> Is vbilling really opensource? >> Open source products can be installed on all operating systems. >> I found the *.bin files only for debian ubuntu and centos. >> For example: gentoo, slackware, bsds, and others are not supportet. >> So much for open source. >> :-) >> >> --------------- >> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: >> http://www.blindi.net/callback >> homepage: http://www.blindi.net >> blinde-misc mailingliste f?r blinde. anmeldung unter: >> http://www.blindi.net/mailman/**listinfo/blinde-misc >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From admin at blindi.net Sun Mar 18 19:03:07 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Sun, 18 Mar 2012 17:03:07 +0100 (CET) Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> <20120318011950.70feee05@anubis.defcon1> <20120318022823.4766c4ec@anubis.defcon1> Message-ID: Am 18.03.12 um 16:45 schrieb Muhammad Naseer Bhatti: > Open Source does not means the product is supported by all operating > systems. It may or may not be supported by just one or many operating > systems. Can you or would like to install Open Source "Linux" on a windows > operating system? :-) I mean: vBilling is "opensource". for Example: i can install Fs, apache, mysql-server and php on windows systems. vBilling is a Web-application. You can.t execute linux-binary on Windowsplatforms. I find this very interesting product. Since I am a system administrator, I like to test this product under various conditions. I am also certified LPI. Since I am blind, I always put great value to Accessibility Statement. -------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From admin at blindi.net Sun Mar 18 19:17:06 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Sun, 18 Mar 2012 17:17:06 +0100 (CET) Subject: [Freeswitch-users] questions mod_opal Message-ID: Hi guys, mod_opal support iax2. I don.t find a setup for Iax2 server or client configuration. For example: i connect Fs to a iax2 asterisk trunk. or a asterisk iax2 client to a fs server. Can your help please? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From curriegrad2004 at gmail.com Sun Mar 18 19:29:18 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 18 Mar 2012 09:29:18 -0700 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: <1FFF97C269757C458224B7C895F35F15060D3D@cantor.std.visionutv.se> References: <4F63B627.5070206@telefaks.de> <4F64A36B.2060801@puzzled.xs4all.nl> <1FFF97C269757C458224B7C895F35F15060D3D@cantor.std.visionutv.se> Message-ID: Personally, a native H.323 solution for FreeSWITCH is better than having yate to act as a H323 and a SIP gateway. mod_opal has it's own problems for being finicky to build with specific combinations of OPAL and pt-lib. The same could be said for mod_H323 too. It's great to see people on the mailing list to ask questions about h323 because this module isn't in use by a lot of FreeSWITCHers out there. On Sun, Mar 18, 2012 at 7:45 AM, Peter Olsson wrote: > I'm using mod_h323. It works quite well - even though it's not 100% perfect, but the same goes for mod_opal... > > I've used them both, but I actually think mod_h323 is a better and more stable choice. > > /Peter > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Brian Foster [bdfoster at endigotech.com] > Skickat: den 18 mars 2012 15:29 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Advice how to start with H.323 > > > That patch probably fixes that issue. If not, it's easily fixed. > > On Mar 17, 2012 10:45 AM, "Patrick Lists" > wrote: > On 03/17/2012 03:30 AM, Brian Foster wrote: > [snip] >> ?> are there any dependencies for the opal and libpt versions? >> >> Those should be taken care of by the build script (if there are any) > > That's assuming that the build script does the proper thing. Last time I > looked, it pulled svn trunk versions of ptlib and opal which are too new > for mod_opal. Afaik it needs the Sirius or maybe the Luyten release. > In the past I built mod_opal without a problem (not using that script) > with the Sirius release of ptlib and opal installed. > > Regarrds, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > !DSPAM:4f65f20a32765192515336! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From curriegrad2004 at gmail.com Sun Mar 18 19:31:33 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 18 Mar 2012 09:31:33 -0700 Subject: [Freeswitch-users] questions mod_opal In-Reply-To: References: Message-ID: IAX is documented on the wiki. A hint here for your bridge syntax: opal/iax2:foo at bar.com On Sun, Mar 18, 2012 at 9:17 AM, Thomas Hoellriegel wrote: > Hi guys, mod_opal support iax2. > I don.t find a setup for Iax2 server or client configuration. > For example: i connect Fs to a iax2 asterisk trunk. > or a asterisk iax2 client ?to a fs server. > > Can your help please? > Thanks. > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From lists at telefaks.de Sun Mar 18 21:52:23 2012 From: lists at telefaks.de (Peter Steinbach) Date: Sun, 18 Mar 2012 19:52:23 +0100 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: <1FFF97C269757C458224B7C895F35F15060D3D@cantor.std.visionutv.se> References: <4F63B627.5070206@telefaks.de> <4F64A36B.2060801@puzzled.xs4all.nl>, <1FFF97C269757C458224B7C895F35F15060D3D@cantor.std.visionutv.se> Message-ID: <4F662EE7.6050607@telefaks.de> I tried to use mod_h323 now and followed the procedure on the wiki page. After fixing problems with - not finding /usr/bin/ptlib-config (copied from /usr/local/bin/, linking did not work) - not finding the h323 include files (linking from /usr/local/include) I now get the following problem: root at fs01:/usr/src/freeswitch# make mod_h323 making all mod_h323 Compiling /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp... quiet_libtool: compile: g++ -g -ITLIBDIR -I/usr/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exceptions -DP_64BIT -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp -fPIC -DPIC -o .libs/mod_h323.o In file included from /usr/include/openh323/h323.h:36, from /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:43, from /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:40: /usr/include/openh323/openh323buildopts.h:37:34: error: ptlib/../../revision.h: No such file or directory /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp: In member function ?virtual PBoolean FSH323_ExternalRTPChannel::Start()?: /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:2125: warning: format ?%lu? expects type ?long unsigned int?, but argument 8 has type ?unsigned int? /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:2126: warning: format ?%lu? expects type ?long unsigned int?, but argument 8 has type ?unsigned int? make[4]: *** [mod_h323.lo] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_h323-all] Error 1 make[1]: *** [mod_h323] Error 2 make: *** [mod_h323] Error 2 This is really tiring, as I run from one problem to the next one. And I am not sure whether this will lead me to a stable system at the end. So am I back to my one of my previous questions: Maybe I am on the wrong distribution? What do you recommend? Best regards Peter Am 18.03.2012 15:45, schrieb Peter Olsson: > I'm using mod_h323. It works quite well - even though it's not 100% perfect, but the same goes for mod_opal... > > I've used them both, but I actually think mod_h323 is a better and more stable choice. > > /Peter > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Brian Foster [bdfoster at endigotech.com] > Skickat: den 18 mars 2012 15:29 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Advice how to start with H.323 > > > That patch probably fixes that issue. If not, it's easily fixed. > > On Mar 17, 2012 10:45 AM, "Patrick Lists"> wrote: > On 03/17/2012 03:30 AM, Brian Foster wrote: > [snip] >> > are there any dependencies for the opal and libpt versions? >> >> Those should be taken care of by the build script (if there are any) > That's assuming that the build script does the proper thing. Last time I > looked, it pulled svn trunk versions of ptlib and opal which are too new > for mod_opal. Afaik it needs the Sirius or maybe the Luyten release. > In the past I built mod_opal without a problem (not using that script) > with the Sirius release of ptlib and opal installed. > > Regarrds, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > !DSPAM:4f65f20a32765192515336! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From kjoseph.us at gmail.com Sun Mar 18 21:57:37 2012 From: kjoseph.us at gmail.com (Joseph Khoury) Date: Sun, 18 Mar 2012 11:57:37 -0700 Subject: [Freeswitch-users] vBilling release announcement. Version 0.1.3 In-Reply-To: References: <20120317202301.1d80d7f1@anubis.defcon1> <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> <20120318011950.70feee05@anubis.defcon1> <20120318022823.4766c4ec@anubis.defcon1> Message-ID: Thanks Muhammad for the clarification. I'll be more than happy to help once the source code is published. Regards, Joseph On Sun, Mar 18, 2012 at 9:03 AM, Thomas Hoellriegel wrote: > Am 18.03.12 um 16:45 schrieb Muhammad Naseer Bhatti: > > > Open Source does not means the product is supported by all operating >> systems. It may or may not be supported by just one or many operating >> systems. Can you or would like to install Open Source "Linux" on a windows >> operating system? :-) >> > > I mean: vBilling is "opensource". for Example: > i can install Fs, apache, mysql-server and php on windows systems. > vBilling is a Web-application. You can.t execute linux-binary on > Windowsplatforms. > I find this very interesting product. > Since I am a system administrator, I like to test this product under > various conditions. I am also certified LPI. > Since I am blind, I always put great value to Accessibility Statement. > > > > -------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/d9b7ced9/attachment.html From peter.olsson at visionutveckling.se Sun Mar 18 22:33:40 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 18 Mar 2012 19:33:40 +0000 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: <4F662EE7.6050607@telefaks.de> References: <4F63B627.5070206@telefaks.de> <4F64A36B.2060801@puzzled.xs4all.nl>, <1FFF97C269757C458224B7C895F35F15060D3D@cantor.std.visionutv.se>, <4F662EE7.6050607@telefaks.de> Message-ID: <1FFF97C269757C458224B7C895F35F15060D86@cantor.std.visionutv.se> What Linux distribution are you using? It seems it can't find revision.h, I have no idea why though. Also, what versions of h323plus and ptlib did you use? I try to stay out of the trunk versions. I've built successfully on CentOS 5.7 and 6.2 (with some minor modifications to Makefile for mod_h323), and I've also built the same on Windows. /Peter ________________________________________ Fr?n: Peter Steinbach [lists at telefaks.de] Skickat: den 18 mars 2012 19:52 Till: FreeSWITCH Users Help Cc: Peter Olsson ?mne: Re: [Freeswitch-users] Advice how to start with H.323 I tried to use mod_h323 now and followed the procedure on the wiki page. After fixing problems with - not finding /usr/bin/ptlib-config (copied from /usr/local/bin/, linking did not work) - not finding the h323 include files (linking from /usr/local/include) I now get the following problem: root at fs01:/usr/src/freeswitch# make mod_h323 making all mod_h323 Compiling /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp... quiet_libtool: compile: g++ -g -ITLIBDIR -I/usr/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exceptions -DP_64BIT -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp -fPIC -DPIC -o .libs/mod_h323.o In file included from /usr/include/openh323/h323.h:36, from /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:43, from /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:40: /usr/include/openh323/openh323buildopts.h:37:34: error: ptlib/../../revision.h: No such file or directory /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp: In member function ?virtual PBoolean FSH323_ExternalRTPChannel::Start()?: /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:2125: warning: format ?%lu? expects type ?long unsigned int?, but argument 8 has type ?unsigned int? /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:2126: warning: format ?%lu? expects type ?long unsigned int?, but argument 8 has type ?unsigned int? make[4]: *** [mod_h323.lo] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_h323-all] Error 1 make[1]: *** [mod_h323] Error 2 make: *** [mod_h323] Error 2 This is really tiring, as I run from one problem to the next one. And I am not sure whether this will lead me to a stable system at the end. So am I back to my one of my previous questions: Maybe I am on the wrong distribution? What do you recommend? Best regards Peter Am 18.03.2012 15:45, schrieb Peter Olsson: > I'm using mod_h323. It works quite well - even though it's not 100% perfect, but the same goes for mod_opal... > > I've used them both, but I actually think mod_h323 is a better and more stable choice. > > /Peter > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Brian Foster [bdfoster at endigotech.com] > Skickat: den 18 mars 2012 15:29 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Advice how to start with H.323 > > > That patch probably fixes that issue. If not, it's easily fixed. > > On Mar 17, 2012 10:45 AM, "Patrick Lists"> wrote: > On 03/17/2012 03:30 AM, Brian Foster wrote: > [snip] >> > are there any dependencies for the opal and libpt versions? >> >> Those should be taken care of by the build script (if there are any) > That's assuming that the build script does the proper thing. Last time I > looked, it pulled svn trunk versions of ptlib and opal which are too new > for mod_opal. Afaik it needs the Sirius or maybe the Luyten release. > In the past I built mod_opal without a problem (not using that script) > with the Sirius release of ptlib and opal installed. > > Regarrds, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de !DSPAM:4f662c5632761804157004! From bote_radio at botecomm.com Sun Mar 18 23:47:26 2012 From: bote_radio at botecomm.com (Bote Man) Date: Sun, 18 Mar 2012 16:47:26 -0400 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: References: <4F63B627.5070206@telefaks.de> <4F64A36B.2060801@puzzled.xs4all.nl> <1FFF97C269757C458224B7C895F35F15060D3D@cantor.std.visionutv.se> Message-ID: I was mentioning H.323 to a vendor of hardware gateways last week and he replied, "Does anybody still use H.323???" like I was from Mars or something. With the increasing prevalence of SIP I'm betting that support for H.323 is not seen as a priority by most people, except those who use it of course. I agree that native support sounds preferable. FWIW, Bote On Sun, Mar 18, 2012 at 12:29 PM, curriegrad2004 wrote: > Personally, a native H.323 solution for FreeSWITCH is better than > having yate to act as a H323 and a SIP gateway. > > mod_opal has it's own problems for being finicky to build with > specific combinations of OPAL and pt-lib. The same could be said for > mod_H323 too. It's great to see people on the mailing list to ask > questions about h323 because this module isn't in use by a lot of > FreeSWITCHers out there. > > On Sun, Mar 18, 2012 at 7:45 AM, Peter Olsson > wrote: > > I'm using mod_h323. It works quite well - even though it's not 100% > perfect, but the same goes for mod_opal... > > > > I've used them both, but I actually think mod_h323 is a better and more > stable choice. > > > > /Peter > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/c3643015/attachment.html From bote_radio at botecomm.com Sun Mar 18 23:43:07 2012 From: bote_radio at botecomm.com (Bote Man) Date: Sun, 18 Mar 2012 16:43:07 -0400 Subject: [Freeswitch-users] endconf flag not working In-Reply-To: <00aa01cd0418$b60e9e70$222bdb50$@com> References: <00aa01cd0418$b60e9e70$222bdb50$@com> Message-ID: Apologies. This is not a problem after all, at least it's not a problem with FS. :- ) The font my browser or the FS wiki pages uses make parenthesis and braces look very similar. I was using the +flags() with parens instead of +flags{} with braces. That made all the difference in the world. Cancel red alert. I just wish mod_conference spit out a LOT more debug-level messages so that I could have figured it out two days ago. I'll see about that when I return. Thanks. Bote On Sat, Mar 17, 2012 at 4:34 AM, Bote Man wrote: > ... ** > > Unfortunately, the endconf flag is not being honored by mod_conference so > the initiator hangs up and the outbound calls keep on ringing because the > number of participants is not less than 1. I would expect the endconf flag > to drop the entire conference when the extension that created it released, > as described by the wiki page.**** > > ** ** > > As I am pulling down sources I?m hoping somebody with experience in > mod_conference can verify that the 'endconf' flag should or should not > work? Have you played with this setting before and what were your results? > **** > > ** ** > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/0d6287f0/attachment.html From moises.silva at gmail.com Mon Mar 19 02:00:05 2012 From: moises.silva at gmail.com (Moises Silva) Date: Sun, 18 Mar 2012 19:00:05 -0400 Subject: [Freeswitch-users] FreeSWITCH 1.2 is on the horizon. In-Reply-To: <4F649AED.4080802@gmail.com> References: <4F649AED.4080802@gmail.com> Message-ID: On Sat, Mar 17, 2012 at 10:08 AM, Trever L. Adams wrote: > > If you want to help or need some help diagnosing and issue visit > > us on IRC via irc.freenode.net/#freeswitch any time. > > > > The FreeSWITCH Dev Team > > > The following is a bug which is affecting several people. Several have > just given up and gone FXS free. I do not have that option. I am willing > to help out if I can. > > http://jira.freeswitch.org/browse/OPENZAP-173 > > I just updated that ticket requesting testing with real git HEAD (today's GIT HEAD). *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/a6f6d2b5/attachment.html From david.villasmil.work at gmail.com Mon Mar 19 02:13:10 2012 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 19 Mar 2012 00:13:10 +0100 Subject: [Freeswitch-users] ODBC instalation In-Reply-To: References: Message-ID: Hey, Thanks for the reply. That, and adding the xml in the actual sofia profile did the trick. It seems it is not used until you set it in the "settings" block. David On Sat, Mar 17, 2012 at 10:17 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > the core-db-dsn value is "dsn-in-odbc.ini:user:password" correct? > > > > On Sat, Mar 17, 2012 at 7:06 PM, Avi Marcus wrote: > >> If it's not there, add it. Order doesn't matter. >> -Avi >> >> >> On Sat, Mar 17, 2012 at 7:56 PM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Yeah, there's no table... :( >>> >>> the documentation is not very clear about this: >>> >>> onf/autoload_configs/switch.conf.xml >>> >>> (WHERE IN THE SWITCH.CONF?) >>> >>> Add or uncomment the following line in appropriate config file within >>> <-- this is clear >>> >>> >>> >>> >>> >>> thanks >>> David >>> >>> >>> 2012/3/17 Yuriy Nasida >>> >>>> Just check freeswitch database and look if exist some tables. In first >>>> start, FS has to create tables automatically. >>>> >>>> ------------------------------ >>>> From: david.villasmil.work at gmail.com >>>> Date: Sat, 17 Mar 2012 18:00:21 +0100 >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: [Freeswitch-users] ODBC instalation >>>> >>>> >>>> Hello guys, >>>> >>>> I just installed odbc support on my debian and want to hace FS use it >>>> for lcr. >>>> I followed the steps on; >>>> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core >>>> >>>> the installation is OK in terms of isql but i don't know whether FS is >>>> able to connect as I just can't see any reference to ODBC in the logs. >>>> >>>> Is there any way of testing FS's connectivity to MySQL-ODBC from the >>>> CLI? >>>> >>>> Thanks >>>> >>>> David >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >>>> CudaTel Communication Server Official >>>> FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org >>>> http://www.cluecon.com FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/022b2beb/attachment-0001.html From brian at freeswitch.org Mon Mar 19 02:26:28 2012 From: brian at freeswitch.org (Brian West) Date: Sun, 18 Mar 2012 18:26:28 -0500 Subject: [Freeswitch-users] reverse time skew warning In-Reply-To: <00e201cd04f8$46a6a7e0$d3f3f7a0$@com> References: <00e201cd04f8$46a6a7e0$d3f3f7a0$@com> Message-ID: <4801538409773451182@unknownmsgid> Broken flux capacitor? /b Sent from my iPad On Mar 18, 2012, at 6:16 AM, Bote Man wrote: I just hope it doesn't cause a serious problem down the road like a segfault or something every time ntp adjusts the time. Any comments? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/71c0d427/attachment.html From admin at blindi.net Mon Mar 19 04:11:11 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Mon, 19 Mar 2012 02:11:11 +0100 (CET) Subject: [Freeswitch-users] ptlib compile error In-Reply-To: References: <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> <20120318011950.70feee05@anubis.defcon1> <20120318022823.4766c4ec@anubis.defcon1> Message-ID: Hi Guys, i use these instructions from: http://wiki.freeswitch.org/wiki/Mod_opal My Os is debian squeeze. I install the needed packages and git The commands i type: git clone git://git.freeswitch.org/freeswitch.git And i run: cd /usr/src/freeswitch ./build/buildopal.sh The compilation fails. Here is the last output: checking ptlib version... 2.11.2 checking PTLIB has URL... yes checking PTLIB has STUN... no ERROR: compulsory feature from PTLib disabled. Make fails. Can Your help please? Thanks --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From curriegrad2004 at gmail.com Mon Mar 19 04:36:36 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 18 Mar 2012 18:36:36 -0700 Subject: [Freeswitch-users] ptlib compile error In-Reply-To: References: <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> <20120318011950.70feee05@anubis.defcon1> <20120318022823.4766c4ec@anubis.defcon1> Message-ID: Fix the script so STUN is enabled in ptlib. On Sun, Mar 18, 2012 at 6:11 PM, Thomas Hoellriegel wrote: > Hi Guys, i use these instructions from: > http://wiki.freeswitch.org/wiki/Mod_opal > My Os is debian squeeze. > I install the needed packages and git > The commands i type: > git clone git://git.freeswitch.org/freeswitch.git > And i run: > cd /usr/src/freeswitch > ./build/buildopal.sh > The compilation fails. > Here is the last output: > checking ptlib version... 2.11.2 > checking PTLIB has URL... yes > checking PTLIB has STUN... no > ?ERROR: compulsory feature from PTLib disabled. > Make fails. > Can Your help please? Thanks > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From bdfoster at endigotech.com Mon Mar 19 05:38:35 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 18 Mar 2012 22:38:35 -0400 Subject: [Freeswitch-users] reverse time skew warning In-Reply-To: <4801538409773451182@unknownmsgid> References: <00e201cd04f8$46a6a7e0$d3f3f7a0$@com> <4801538409773451182@unknownmsgid> Message-ID: Well it is Windows... On Mar 18, 2012 7:27 PM, "Brian West" wrote: > Broken flux capacitor? > > /b > > Sent from my iPad > > On Mar 18, 2012, at 6:16 AM, Bote Man wrote: > > I just hope it doesn't cause a serious problem down the road like a > segfault or something every time ntp adjusts the time. Any comments? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/1924ffae/attachment.html From bdfoster at endigotech.com Mon Mar 19 05:39:30 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 18 Mar 2012 22:39:30 -0400 Subject: [Freeswitch-users] ptlib compile error In-Reply-To: References: <20120317232615.58cb3993@anubis.defcon1> <20120318010000.67f91b94@anubis.defcon1> <20120318011950.70feee05@anubis.defcon1> <20120318022823.4766c4ec@anubis.defcon1> Message-ID: Jeff, Can we get this on the patch we've been working on? On Mar 18, 2012 9:38 PM, "curriegrad2004" wrote: > Fix the script so STUN is enabled in ptlib. > > On Sun, Mar 18, 2012 at 6:11 PM, Thomas Hoellriegel > wrote: > > Hi Guys, i use these instructions from: > > http://wiki.freeswitch.org/wiki/Mod_opal > > My Os is debian squeeze. > > I install the needed packages and git > > The commands i type: > > git clone git://git.freeswitch.org/freeswitch.git > > And i run: > > cd /usr/src/freeswitch > > ./build/buildopal.sh > > The compilation fails. > > Here is the last output: > > checking ptlib version... 2.11.2 > > checking PTLIB has URL... yes > > checking PTLIB has STUN... no > > ERROR: compulsory feature from PTLib disabled. > > Make fails. > > Can Your help please? Thanks > > > > --------------- > > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > > http://www.blindi.net/callback > > homepage: http://www.blindi.net > > blinde-misc mailingliste f?r blinde. anmeldung unter: > > http://www.blindi.net/mailman/listinfo/blinde-misc > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/f73a9d51/attachment.html From bob.mccarthy at experient.com Mon Mar 19 07:37:09 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Sun, 18 Mar 2012 22:37:09 -0600 Subject: [Freeswitch-users] Setting CallerID for inbound SLA Barge in Message-ID: <017c01cd0589$f3891c40$da9b54c0$@mccarthy@experient.com> I have read where you can set the outbound Callerid name and number for calling out of a conference, but how do you set the Callerid name and number for phones that barge in on a conference (SLA in particular) ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/1d1944ec/attachment-0001.html From singhai.piyush at gmail.com Mon Mar 19 08:39:29 2012 From: singhai.piyush at gmail.com (piyush singhai) Date: Mon, 19 Mar 2012 11:09:29 +0530 Subject: [Freeswitch-users] Understanding of freeswitch Events Message-ID: Hello All, I want to understand then Event Handling in freeswitch. 1. How events flow in freeswitch, 2. How fs core handle the events. 3. What mechanism used is it push architecture or pullfor events. 4. Priorities of events in FS. So please help me. --Piyush -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/6b0b6f54/attachment.html From valery.kalinin at gmail.com Mon Mar 19 09:00:00 2012 From: valery.kalinin at gmail.com (Valery Kalinin) Date: Mon, 19 Mar 2012 12:00:00 +0600 Subject: [Freeswitch-users] FreeSWITCH panel - web utility update Message-ID: Hi all! I update my "FreeSWITCH panel" online web utility. Screenshot & help: https://sites.google.com/site/freeswitched/home/downloads/fspanel_help.png This application allows you to view and control online in web browser: - SIP registered subscribers - Sofia status profile commands: - rescan - restart - flush registered endpoints - flush and reboot registered endpoints gateway commands: - kill - current channels/calls/detailed_calls - hangup call - conferences - lock/unlock - pin/no pin - dial - dtmf conference member: - defa/undeaf - mute/unmute - hup - kick - transfer to - volume in/out - energy level - FreeTDM channels - span start/stop - chan info It is also possible simple Freeswitch management: - sofia status - reloadxml - reloadacl - status - version You can select the display fields. All settings are stored in a cookies. Site here: https://sites.google.com/site/freeswitched/home git: https://github.com/Slonik/FreeSWITCH-panel Tests and features are welcome. Short install instructions: Upload files to your web-server with PHP support. Check modules.conf and event_socket.conf for enable module mod_event_socket Change variables if needed in fscontrol.php file: $FreeSWITCHserver = '127.0.0.1'; // event_socket.xml param: listen-ip $FreeSWITCHport = 8021; // event_socket.xml param: listen-port $FreeSWITCHpassword = 'xexexe'; // event_socket.xml param: password $SofiaProfiles = array('internal'); // write empty array if not used: array(); $FreeTDMspans = array('pri'); // write empty array if not used: array(); $disableXML = false; // You can disable retrieve XML data if error messages received Enter in browser http://you-server-name/fspanel.html Click on menu item "settings" for choice items. Thanks. From bote_radio at botecomm.com Mon Mar 19 10:38:51 2012 From: bote_radio at botecomm.com (Bote Man) Date: Mon, 19 Mar 2012 03:38:51 -0400 Subject: [Freeswitch-users] reverse time skew warning In-Reply-To: References: <00e201cd04f8$46a6a7e0$d3f3f7a0$@com> <4801538409773451182@unknownmsgid> Message-ID: <00b601cd05a3$55266480$ff732d80$@com> Man! That's heavy. Bote From: Brian Foster Sent: Sunday, 18 March, 2012 22:39 Well it is Windows... On Mar 18, 2012 7:27 PM, "Brian West" wrote: Broken flux capacitor? /b On Mar 18, 2012, at 6:16 AM, Bote Man wrote: I just hope it doesn't cause a serious problem down the road like a segfault or something every time ntp adjusts the time. Any comments? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/4f658b3a/attachment.html From bote_radio at botecomm.com Mon Mar 19 10:38:51 2012 From: bote_radio at botecomm.com (Bote Man) Date: Mon, 19 Mar 2012 03:38:51 -0400 Subject: [Freeswitch-users] Setting CallerID for inbound SLA Barge in In-Reply-To: <4f66b919.84d2e00a.7bc2.61f1SMTPIN_ADDED@mx.google.com> References: <4f66b919.84d2e00a.7bc2.61f1SMTPIN_ADDED@mx.google.com> Message-ID: <00bb01cd05a3$563ae1c0$02b0a540$@com> That originates in the device itself and is contained in the caller_id_name and caller_id_number variables. I just tried the effective_ series of variables, but those are for the outbound B-leg calls and are empty at the time the conference is entered. But if a phone is barging into a conference how does Caller*ID have any impact, other than to test for it in the dialplan? Do you intend to say: the number or name? Bote From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob McCarthy Sent: Monday, 19 March, 2012 00:37 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Setting CallerID for inbound SLA Barge in I have read where you can set the outbound Callerid name and number for calling out of a conference, but how do you set the Callerid name and number for phones that barge in on a conference (SLA in particular) ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/0026b57a/attachment.html From rgarrett at garrettnet.net Sun Mar 18 02:13:40 2012 From: rgarrett at garrettnet.net (Robbie A. Garrett) Date: Sat, 17 Mar 2012 23:13:40 +0000 Subject: [Freeswitch-users] Free switch + Google Voice + Cisco Call Manager Message-ID: <309A95007C8BF54293B52FF2068020BC808CEF@ex01.garrettnet.net> Hello Everyone, I have noticed a lack of documentation when it comes to cisco call manager and free switch integration. I would be willing to assist with documentation for the WiKI... I need to be point in the right direction though. I would like to create some documentation that would assist with the following 1) Google voice as the "carrier", Free switch as the "gateway", and call manager as the switch. 2) Call manager as the switch and free switch as the "gateway" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120317/0c84a4a3/attachment-0001.html From rgarrett at garrettnet.net Sun Mar 18 03:05:53 2012 From: rgarrett at garrettnet.net (Robbie A. Garrett) Date: Sun, 18 Mar 2012 00:05:53 +0000 Subject: [Freeswitch-users] firewall ports In-Reply-To: References: <6981B8DC-0B89-4FA3-BF79-670B177857AD@garrettnet.net> Message-ID: <309A95007C8BF54293B52FF2068020BC808DFA@ex01.garrettnet.net> Yes... GV is broke in asterisk. Free switch is about a 99% success call rate... Both answer and placing calls. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster Sent: Saturday, March 17, 2012 7:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] firewall ports I don't think so... I do know that google voice never worked well with asterisk, though. And that machine wasn't behind a NAT. On Sat, Mar 17, 2012 at 12:37 PM, Robbie A. Garrett > wrote: hey all, I'm using free switch with google voice for testing. outbound works great but inbound does not appear too be hitting the free switch box. is there a list of ports I need too nat / open on the firewall ? Sent from a mobile device. Please excuse shorthand and spelling. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/7c8584c5/attachment-0001.html From fuji246 at gmail.com Sun Mar 18 17:58:59 2012 From: fuji246 at gmail.com (Jeromy) Date: Sun, 18 Mar 2012 14:58:59 +0000 (UTC) Subject: [Freeswitch-users] TDM400P spans References: <5227389.xYDaNXzQut@axp> Message-ID: Jeremy Johnson writes: > > > I'm trying to convert from Asterisk to Freeswitch, and have a question about spans. > ? > I read the freetdm configuration example for TDM400 > and am wondering what exactly is a span. > ? > I have 2 TDM400P cards in my computer, > one card has 2 FXS modules for connecting analog phones. > The other card has 3 FXO modules for connecting to 3 PSTN trunk lines. > ? > # lsdahdi > ### Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" > 1 FXS FXOKS (In use) (EC: OSLEC - INACTIVE) > 2 FXS FXOKS (In use) (EC: OSLEC - INACTIVE) > 3 unknown Reserved > 4 unknown Reserved > ### Span 2: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER) > 5 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) > 6 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) > 7 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) > 8 unknown Reserved > ? > Do I configure freetdm as in the example for TDM400 > with separate spans for each module like the following? > or should I have just two spans as shown by lsdahsi? > Thanks. > ? Hi Jeremy Do you find the answer to this question? I'm also very confuse on the span and channel in both asterisk and freeswitch, and I've found little explaination on this. " A span is a logical unit that represents a group of channels. With digital telephony, a span usually represents a physical port on the card. If the system has only one such card with a single port, so it is referred to as span 1. " Could anyone help on this? Thanks in advance! From freeswitch at zoho.com Mon Mar 19 06:43:36 2012 From: freeswitch at zoho.com (freeswitch) Date: Sun, 18 Mar 2012 20:43:36 -0700 Subject: [Freeswitch-users] FreeSWITCH 1.2 is on the horizon. In-Reply-To: References: <4F649AED.4080802@gmail.com> Message-ID: <136290d902c.-7300161349838142370.-3404847615666721735@zoho.com> could this be what happened to our install? we have like 4 extensions, using 2 cisco spa405g and 2 yealinks, yealink works fine,until i use cisco to call the other yealinks. it just keeps on ringing even if fs_cli says says its hanged up.keeps on ringing ringing riing. i just had this installed like 2 or 3 days ago. not sure where to get those logs though. i dont really have access to the T1 lines and its now disconnected so options are limited.(and it is remotely located too) i have Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card with a PRI T1 installed on it. ---- On Sun, 18 Mar 2012 16:00:05 -0700 Moises Silva <moises.silva at gmail.com> wrote ---- On Sat, Mar 17, 2012 at 10:08 AM, Trever L. Adams <trever.adams at gmail.com> wrote: > If you want to help or need some help diagnosing and issue visit > us on IRC via irc.freenode.net/#freeswitch any time. > > The FreeSWITCH Dev Team > The following is a bug which is affecting several people. Several have just given up and gone FXS free. I do not have that option. I am willing to help out if I can. http://jira.freeswitch.org/browse/OPENZAP-173 I just updated that ticket requesting testing with real git HEAD (today's GIT HEAD). Moises Silva Manager, Software Engineering msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/dc35023b/attachment-0001.html From freeswitch at zoho.com Mon Mar 19 06:46:49 2012 From: freeswitch at zoho.com (freeswitch) Date: Sun, 18 Mar 2012 20:46:49 -0700 Subject: [Freeswitch-users] FreeSwitch & ODBC errors In-Reply-To: References: <13620e31ef5.4902843172079284242.1157408376614637062@zoho.com> Message-ID: <1362910809c.-5214290268468074844.3377733292056733808@zoho.com> yeah my bad. its the first time i set that up FS using ODBC. turned out that it was just doing its thing,tried to poke around pg and the db was actually populated. Freeswitch started without errors the next time around. thanks ---- On Sat, 17 Mar 2012 16:32:31 -0700 Brian Foster<bdfoster at endigotech.com> wrote ---- I don't actually see a problem here. -BDF On Sat, Mar 17, 2012 at 9:40 AM, freeswitch <freeswitch at zoho.com> wrote: Hello all, can you please help me with my settings? trying to setup pg and FS via odbc,not sure what i missed, tried to follow this guide - http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#5.__Compile_Freeswitch_.26_Set_Modules_to_Use_ODBC and here's what i have http://pastebin.freeswitch.org/18684 isql test ------------------------------------------- [root at echo-nh-voip bin]# isql -v freeswitch superadmin s3cret0w +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> dingdong _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120318/1878c15a/attachment-0001.html From freeswitch at zoho.com Mon Mar 19 10:23:26 2012 From: freeswitch at zoho.com (freeswitch) Date: Mon, 19 Mar 2012 00:23:26 -0700 Subject: [Freeswitch-users] FreeSWITCH panel - web utility update In-Reply-To: References: Message-ID: <13629d6d4f2.465495304447425147.7543463568927770653@zoho.com> nice start.very simple and easy to understand suggestions 1.] make a way to add an extension via web? 2.] make a way to have a control panel for mod_callcenter ?\ 3.]maybe an auto - dialer? - since you have that windows gadget that i assume can be turned to agent popup? 4.]extensive 'callcenter reporting' (who hanged up first, blah blah etc etc) 5.] unclutter? -control for admins / and stuff for non admin separated. lots of people are actually looking for that vicidial replacement out of the box. :) ---- On Sun, 18 Mar 2012 23:00:00 -0700 Valery Kalinin<valery.kalinin at gmail.com> wrote ---- Hi all! I update my "FreeSWITCH panel" online web utility. Screenshot & help: https://sites.google.com/site/freeswitched/home/downloads/fspanel_help.png This application allows you to view and control online in web browser: - SIP registered subscribers - Sofia status profile commands: - rescan - restart - flush registered endpoints - flush and reboot registered endpoints gateway commands: - kill - current channels/calls/detailed_calls - hangup call - conferences - lock/unlock - pin/no pin - dial - dtmf conference member: - defa/undeaf - mute/unmute - hup - kick - transfer to - volume in/out - energy level - FreeTDM channels - span start/stop - chan info It is also possible simple Freeswitch management: - sofia status - reloadxml - reloadacl - status - version You can select the display fields. All settings are stored in a cookies. Site here: https://sites.google.com/site/freeswitched/home git: https://github.com/Slonik/FreeSWITCH-panel Tests and features are welcome. Short install instructions: Upload files to your web-server with PHP support. Check modules.conf and event_socket.conf for enable module mod_event_socket Change variables if needed in fscontrol.php file: $FreeSWITCHserver = '127.0.0.1'; // event_socket.xml param: listen-ip $FreeSWITCHport = 8021; // event_socket.xml param: listen-port $FreeSWITCHpassword = 'xexexe'; // event_socket.xml param: password $SofiaProfiles = array('internal'); // write empty array if not used: array(); $FreeTDMspans = array('pri'); // write empty array if not used: array(); $disableXML = false; // You can disable retrieve XML data if error messages received Enter in browser http://you-server-name/fspanel.html Click on menu item "settings" for choice items. Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/2d4ee7d2/attachment.html From avi at avimarcus.net Mon Mar 19 10:48:52 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 19 Mar 2012 09:48:52 +0200 Subject: [Freeswitch-users] FreeSWITCH panel - web utility update In-Reply-To: <13629d6d4f2.465495304447425147.7543463568927770653@zoho.com> References: <13629d6d4f2.465495304447425147.7543463568927770653@zoho.com> Message-ID: This is actually an update. What you're asking for is a way to control everything.. this panel seems to be just real-time monitoring and basic FS commands. GUI for basic commands and views seems very nice. I cloned the GIT, set up the password.... but even though I can run status & version commands, I don't see any active registrations or calls/channels. What might be going on? Thanks for sharing your work! -Avi On Mon, Mar 19, 2012 at 9:23 AM, freeswitch wrote: > ** > nice start.very simple and easy to understand > > suggestions > > 1.] make a way to add an extension via web? > 2.] make a way to have a control panel for mod_callcenter ?\ > 3.]maybe an auto - dialer? - since you have that windows gadget that i > assume can be turned to agent popup? > 4.]extensive 'callcenter reporting' (who hanged up first, blah blah etc > etc) > 5.] unclutter? -control for admins / and stuff for non admin separated. > > > lots of people are actually looking for that vicidial replacement out of > the box. > > :) > ---- On Sun, 18 Mar 2012 23:00:00 -0700 *Valery Kalinin< > valery.kalinin at gmail.com>* wrote ---- > > Hi all! > > I update my "FreeSWITCH panel" online web utility. > > Screenshot & help: > https://sites.google.com/site/freeswitched/home/downloads/fspanel_help.png > > This application allows you to view and control online in web browser: > - SIP registered subscribers > - Sofia status > profile commands: > - rescan > - restart > - flush registered endpoints > - flush and reboot registered endpoints > gateway commands: > - kill > - current channels/calls/detailed_calls > - hangup call > - conferences > - lock/unlock > - pin/no pin > - dial > - dtmf > conference member: > - defa/undeaf > - mute/unmute > - hup > - kick > - transfer to > - volume in/out > - energy level > - FreeTDM channels > - span start/stop > - chan info > It is also possible simple Freeswitch management: > - sofia status > - reloadxml > - reloadacl > - status > - version > You can select the display fields. > All settings are stored in a cookies. > > Site here: https://sites.google.com/site/freeswitched/home > git: https://github.com/Slonik/FreeSWITCH-panel > > Tests and features are welcome. > > Short install instructions: > Upload files to your web-server with PHP support. > Check modules.conf and event_socket.conf for enable module > mod_event_socket > Change variables if needed in fscontrol.php file: > $FreeSWITCHserver = '127.0.0.1'; // event_socket.xml param: > listen-ip > $FreeSWITCHport = 8021; // event_socket.xml param: > listen-port > $FreeSWITCHpassword = 'xexexe'; // event_socket.xml param: > password > $SofiaProfiles = array('internal'); // write empty array if not > used: array(); > $FreeTDMspans = array('pri'); // write empty array if not > used: array(); > $disableXML = false; // You can disable retrieve XML > data if error > messages received > Enter in browser http://you-server-name/fspanel.html > Click on menu item "settings" for choice items. > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/19b1a34d/attachment.html From peter.olsson at visionutveckling.se Mon Mar 19 10:48:58 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 19 Mar 2012 07:48:58 +0000 Subject: [Freeswitch-users] reverse time skew warning Message-ID: <1FFF97C269757C458224B7C895F35F1506238D@cantor.std.visionutv.se> If you get that warning in Windows, just start FS with "-monotonic-clock", then it won't be dependant of system time (as done by default on Linux systems). /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian Foster Skickat: den 19 mars 2012 03:39 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] reverse time skew warning Well it is Windows... On Mar 18, 2012 7:27 PM, "Brian West" > wrote: Broken flux capacitor? /b Sent from my iPad On Mar 18, 2012, at 6:16 AM, Bote Man > wrote: I just hope it doesn't cause a serious problem down the road like a segfault or something every time ntp adjusts the time. Any comments? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f669b1732763860132602! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/753b783c/attachment-0001.html From avi at avimarcus.net Mon Mar 19 10:59:55 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 19 Mar 2012 09:59:55 +0200 Subject: [Freeswitch-users] FreeSWITCH panel - web utility update In-Reply-To: References: <13629d6d4f2.465495304447425147.7543463568927770653@zoho.com> Message-ID: Ah. Uhm. After a few minutes a dialogue with a bunch of json popped up and the profiles/gateways showed up. Registered users just showed up as "undefined" though. But looking at TOP I see php5-fpm is using nearly a whole core (88%), with no data showing. So something odd is going on. -Avi On Mon, Mar 19, 2012 at 9:48 AM, Avi Marcus wrote: > This is actually an update. What you're asking for is a way to control > everything.. this panel seems to be just real-time monitoring and basic FS > commands. > > GUI for basic commands and views seems very nice. > > I cloned the GIT, set up the password.... but even though I can run status > & version commands, I don't see any active registrations or calls/channels. > What might be going on? > > Thanks for sharing your work! > > -Avi > > > On Mon, Mar 19, 2012 at 9:23 AM, freeswitch wrote: > >> ** >> nice start.very simple and easy to understand >> >> suggestions >> >> 1.] make a way to add an extension via web? >> 2.] make a way to have a control panel for mod_callcenter ?\ >> 3.]maybe an auto - dialer? - since you have that windows gadget that i >> assume can be turned to agent popup? >> 4.]extensive 'callcenter reporting' (who hanged up first, blah blah etc >> etc) >> 5.] unclutter? -control for admins / and stuff for non admin separated. >> >> >> lots of people are actually looking for that vicidial replacement out of >> the box. >> >> :) >> ---- On Sun, 18 Mar 2012 23:00:00 -0700 *Valery Kalinin< >> valery.kalinin at gmail.com>* wrote ---- >> >> Hi all! >> >> I update my "FreeSWITCH panel" online web utility. >> >> Screenshot & help: >> https://sites.google.com/site/freeswitched/home/downloads/fspanel_help.png >> >> This application allows you to view and control online in web browser: >> - SIP registered subscribers >> - Sofia status >> profile commands: >> - rescan >> - restart >> - flush registered endpoints >> - flush and reboot registered endpoints >> gateway commands: >> - kill >> - current channels/calls/detailed_calls >> - hangup call >> - conferences >> - lock/unlock >> - pin/no pin >> - dial >> - dtmf >> conference member: >> - defa/undeaf >> - mute/unmute >> - hup >> - kick >> - transfer to >> - volume in/out >> - energy level >> - FreeTDM channels >> - span start/stop >> - chan info >> It is also possible simple Freeswitch management: >> - sofia status >> - reloadxml >> - reloadacl >> - status >> - version >> You can select the display fields. >> All settings are stored in a cookies. >> >> Site here: https://sites.google.com/site/freeswitched/home >> git: https://github.com/Slonik/FreeSWITCH-panel >> >> Tests and features are welcome. >> >> Short install instructions: >> Upload files to your web-server with PHP support. >> Check modules.conf and event_socket.conf for enable module >> mod_event_socket >> Change variables if needed in fscontrol.php file: >> $FreeSWITCHserver = '127.0.0.1'; // event_socket.xml param: >> listen-ip >> $FreeSWITCHport = 8021; // event_socket.xml param: >> listen-port >> $FreeSWITCHpassword = 'xexexe'; // event_socket.xml param: >> password >> $SofiaProfiles = array('internal'); // write empty array if >> not used: array(); >> $FreeTDMspans = array('pri'); // write empty array if not >> used: array(); >> $disableXML = false; // You can disable retrieve XML >> data if error >> messages received >> Enter in browser http://you-server-name/fspanel.html >> Click on menu item "settings" for choice items. >> >> Thanks. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/bb41256a/attachment.html From nbhatti at gmail.com Mon Mar 19 11:17:30 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Mon, 19 Mar 2012 11:17:30 +0300 Subject: [Freeswitch-users] FreeSWITCH panel - web utility update In-Reply-To: References: <13629d6d4f2.465495304447425147.7543463568927770653@zoho.com> Message-ID: Looks nice, but php is eating up all the CPU. Looks like it needs some debug and fine tuning. -B On Mon, Mar 19, 2012 at 10:59 AM, Avi Marcus wrote: > Ah. Uhm. After a few minutes a dialogue with a bunch of json popped up and > the profiles/gateways showed up. Registered users just showed up as > "undefined" though. > > But looking at TOP I see php5-fpm is using nearly a whole core (88%), with > no data showing. So something odd is going on. > > -Avi > > > On Mon, Mar 19, 2012 at 9:48 AM, Avi Marcus wrote: > >> This is actually an update. What you're asking for is a way to control >> everything.. this panel seems to be just real-time monitoring and basic FS >> commands. >> >> GUI for basic commands and views seems very nice. >> >> I cloned the GIT, set up the password.... but even though I can run >> status & version commands, I don't see any active registrations or >> calls/channels. >> What might be going on? >> >> Thanks for sharing your work! >> >> -Avi >> >> >> On Mon, Mar 19, 2012 at 9:23 AM, freeswitch wrote: >> >>> ** >>> nice start.very simple and easy to understand >>> >>> suggestions >>> >>> 1.] make a way to add an extension via web? >>> 2.] make a way to have a control panel for mod_callcenter ?\ >>> 3.]maybe an auto - dialer? - since you have that windows gadget that i >>> assume can be turned to agent popup? >>> 4.]extensive 'callcenter reporting' (who hanged up first, blah blah etc >>> etc) >>> 5.] unclutter? -control for admins / and stuff for non admin separated. >>> >>> >>> lots of people are actually looking for that vicidial replacement out of >>> the box. >>> >>> :) >>> ---- On Sun, 18 Mar 2012 23:00:00 -0700 *Valery Kalinin< >>> valery.kalinin at gmail.com>* wrote ---- >>> >>> Hi all! >>> >>> I update my "FreeSWITCH panel" online web utility. >>> >>> Screenshot & help: >>> >>> https://sites.google.com/site/freeswitched/home/downloads/fspanel_help.png >>> >>> This application allows you to view and control online in web browser: >>> - SIP registered subscribers >>> - Sofia status >>> profile commands: >>> - rescan >>> - restart >>> - flush registered endpoints >>> - flush and reboot registered endpoints >>> gateway commands: >>> - kill >>> - current channels/calls/detailed_calls >>> - hangup call >>> - conferences >>> - lock/unlock >>> - pin/no pin >>> - dial >>> - dtmf >>> conference member: >>> - defa/undeaf >>> - mute/unmute >>> - hup >>> - kick >>> - transfer to >>> - volume in/out >>> - energy level >>> - FreeTDM channels >>> - span start/stop >>> - chan info >>> It is also possible simple Freeswitch management: >>> - sofia status >>> - reloadxml >>> - reloadacl >>> - status >>> - version >>> You can select the display fields. >>> All settings are stored in a cookies. >>> >>> Site here: https://sites.google.com/site/freeswitched/home >>> git: https://github.com/Slonik/FreeSWITCH-panel >>> >>> Tests and features are welcome. >>> >>> Short install instructions: >>> Upload files to your web-server with PHP support. >>> Check modules.conf and event_socket.conf for enable module >>> mod_event_socket >>> Change variables if needed in fscontrol.php file: >>> $FreeSWITCHserver = '127.0.0.1'; // event_socket.xml param: >>> listen-ip >>> $FreeSWITCHport = 8021; // event_socket.xml param: >>> listen-port >>> $FreeSWITCHpassword = 'xexexe'; // event_socket.xml param: >>> password >>> $SofiaProfiles = array('internal'); // write empty array if >>> not used: array(); >>> $FreeTDMspans = array('pri'); // write empty array if not >>> used: array(); >>> $disableXML = false; // You can disable retrieve XML >>> data if error >>> messages received >>> Enter in browser http://you-server-name/fspanel.html >>> Click on menu item "settings" for choice items. >>> >>> Thanks. >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/5e5e2fcd/attachment-0001.html From valery.kalinin at gmail.com Mon Mar 19 11:38:15 2012 From: valery.kalinin at gmail.com (Valery Kalinin) Date: Mon, 19 Mar 2012 14:38:15 +0600 Subject: [Freeswitch-users] FreeSWITCH panel - web utility update Message-ID: > After a few minutes a dialogue with a bunch of json popped up and Please set variable in fscontrol.php $disableXML = true; > Registered users just showed up as "undefined" though. Click on header and select correct fields from list received from server. From bob.mccarthy at experient.com Mon Mar 19 11:43:21 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Mon, 19 Mar 2012 02:43:21 -0600 Subject: [Freeswitch-users] SLA unbuilding a conference Message-ID: <01d601cd05ac$58a8e570$09fab050$@mccarthy@experient.com> I have a customer who is using SLA who wishes to Blind Transfer a call after a conference has been built and has reduced to two parties. I have used uuid_bridge to "unconference". This action breaks any further SLA on the call, even though the lines are lit. Any other phone that tries to Barge in gets put into a single channel conference. The phone that had quit the conference fails on subsequent Barge attempts to the original call but can connect into a conference with the other phones if they have built the single channel. What I am asking is there a better way to accomplish this without breaking SLA ? I am basically trying to put the call back into the state it was in before the first SLA conference was built. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/bff794c5/attachment.html From avi at avimarcus.net Mon Mar 19 12:02:38 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 19 Mar 2012 11:02:38 +0200 Subject: [Freeswitch-users] FreeSWITCH panel - web utility update In-Reply-To: References: Message-ID: I set it to true but it's still eating major CPU (80%+). I didn't wait for anything to show up. -Avi On Mon, Mar 19, 2012 at 10:38 AM, Valery Kalinin wrote: > > After a few minutes a dialogue with a bunch of json popped up and > Please set variable in fscontrol.php > $disableXML = true; > > > Registered users just showed up as "undefined" though. > Click on header and select correct fields from list received from server. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/30616b5d/attachment.html From davidjbrazier at gmail.com Mon Mar 19 13:03:41 2012 From: davidjbrazier at gmail.com (David Brazier) Date: Mon, 19 Mar 2012 10:03:41 +0000 Subject: [Freeswitch-users] Early media buffered in bridge with ignore_early_media Message-ID: Hi All When we bridge one external call A to another B with ignore_early_media, then A hears nothing before B answers (as expected) but then hears ringing tones *after* B answers, and they can't otherwise hear each other until after that ringing has finished. The duration of this ringing appears to be related to the time B takes to answer, suggesting that what is happening is the early media from B is being buffered and not played back until after B answers. Easy to reproduce with: originate sofia/gateway/XXX/YYY &bridge({ignore_early_media=true}sofia/gateway/XXX/ZZZ) Doesn't happen with no ignore_early_media: originate sofia/gateway/XXX/YYY &bridge(sofia/gateway/XXX/ZZZ) Tested & problem is present on git-4276680 2012-03-14 10-08-41 -0500 but not an old build git-1086cba 2011-05-23 22-51-43 -0500 Have raised as http://jira.freeswitch.org/browse/FS-4009 but windering if anyone else has experienced this, or can reproduce it? Regards David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/999e29a3/attachment.html From lists at telefaks.de Mon Mar 19 13:53:31 2012 From: lists at telefaks.de (Peter Steinbach) Date: Mon, 19 Mar 2012 11:53:31 +0100 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: <1FFF97C269757C458224B7C895F35F15060D86@cantor.std.visionutv.se> References: <4F63B627.5070206@telefaks.de> <4F64A36B.2060801@puzzled.xs4all.nl>, <1FFF97C269757C458224B7C895F35F15060D3D@cantor.std.visionutv.se>, <4F662EE7.6050607@telefaks.de> <1FFF97C269757C458224B7C895F35F15060D86@cantor.std.visionutv.se> Message-ID: <4F67102B.1020305@telefaks.de> I am using Ubuntu 10.04. I followed the installation procedures on the wiki page and installed ptlib-2.8.2 + h323plus-trunk Best regards Peter Am 18.03.2012 20:33, schrieb Peter Olsson: > What Linux distribution are you using? > > It seems it can't find revision.h, I have no idea why though. Also, what versions of h323plus and ptlib did you use? I try to stay out of the trunk versions. > > I've built successfully on CentOS 5.7 and 6.2 (with some minor modifications to Makefile for mod_h323), and I've also built the same on Windows. > > /Peter > ________________________________________ > Fr?n: Peter Steinbach [lists at telefaks.de] > Skickat: den 18 mars 2012 19:52 > Till: FreeSWITCH Users Help > Cc: Peter Olsson > ?mne: Re: [Freeswitch-users] Advice how to start with H.323 > > I tried to use mod_h323 now and followed the procedure on the wiki page. > > After fixing problems with > - not finding /usr/bin/ptlib-config (copied from /usr/local/bin/, > linking did not work) > - not finding the h323 include files (linking from /usr/local/include) > I now get the following problem: > > root at fs01:/usr/src/freeswitch# make mod_h323 > making all mod_h323 > Compiling /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp... > quiet_libtool: compile: g++ -g -ITLIBDIR -I/usr/include/openh323 -I. > -DPTRACING=1 -D_REENTRANT -fno-exceptions -DP_64BIT > -I/usr/src/freeswitch/libs/curl/include > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE > -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp -fPIC -DPIC > -o .libs/mod_h323.o > In file included from /usr/include/openh323/h323.h:36, > from /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:43, > from /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:40: > /usr/include/openh323/openh323buildopts.h:37:34: error: > ptlib/../../revision.h: No such file or directory > /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp: In member > function ?virtual PBoolean FSH323_ExternalRTPChannel::Start()?: > /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:2125: > warning: format ?%lu? expects type ?long unsigned int?, but argument 8 > has type ?unsigned int? > /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:2126: > warning: format ?%lu? expects type ?long unsigned int?, but argument 8 > has type ?unsigned int? > make[4]: *** [mod_h323.lo] Error 1 > make[3]: *** [all] Error 1 > make[2]: *** [mod_h323-all] Error 1 > make[1]: *** [mod_h323] Error 2 > make: *** [mod_h323] Error 2 > > This is really tiring, as I run from one problem to the next one. And I > am not sure whether this will lead me to a stable system at the end. > > So am I back to my one of my previous questions: Maybe I am on the wrong > distribution? What do you recommend? > > Best regards > Peter > > > Am 18.03.2012 15:45, schrieb Peter Olsson: >> I'm using mod_h323. It works quite well - even though it's not 100% perfect, but the same goes for mod_opal... >> >> I've used them both, but I actually think mod_h323 is a better and more stable choice. >> >> /Peter >> >> ________________________________ >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Brian Foster [bdfoster at endigotech.com] >> Skickat: den 18 mars 2012 15:29 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] Advice how to start with H.323 >> >> >> That patch probably fixes that issue. If not, it's easily fixed. >> >> On Mar 17, 2012 10:45 AM, "Patrick Lists"> wrote: >> On 03/17/2012 03:30 AM, Brian Foster wrote: >> [snip] >>> > are there any dependencies for the opal and libpt versions? >>> >>> Those should be taken care of by the build script (if there are any) >> That's assuming that the build script does the proper thing. Last time I >> looked, it pulled svn trunk versions of ptlib and opal which are too new >> for mod_opal. Afaik it needs the Sirius or maybe the Luyten release. >> In the past I built mod_opal without a problem (not using that script) >> with the Sirius release of ptlib and opal installed. >> >> Regarrds, >> Patrick >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > > !DSPAM:4f662c5632761804157004! > > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From trever.adams at gmail.com Mon Mar 19 14:03:19 2012 From: trever.adams at gmail.com (Trever L. Adams) Date: Mon, 19 Mar 2012 05:03:19 -0600 Subject: [Freeswitch-users] FreeSWITCH 1.2 is on the horizon. In-Reply-To: <136290d902c.-7300161349838142370.-3404847615666721735@zoho.com> References: <4F649AED.4080802@gmail.com> <136290d902c.-7300161349838142370.-3404847615666721735@zoho.com> Message-ID: <4F671277.1090503@gmail.com> On 03/18/2012 09:43 PM, freeswitch wrote: > could this be what happened to our install? we have like 4 extensions, > using 2 cisco spa405g and 2 yealinks, yealink works fine,until i use > cisco to call the other yealinks. it just keeps on ringing even if > fs_cli says says its hanged up.keeps on ringing ringing riing. > > i just had this installed like 2 or 3 days ago. not sure where to get > those logs though. > i dont really have access to the T1 lines and its now disconnected so > options are limited.(and it is remotely located too) > > > > i have Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card > with a PRI T1 installed on it. > > Please look at the logs and descriptions at http://jira.freeswitch.org/browse/OPENZAP-173 It may very well be the same problem. If you have it up and running but ran freeswitch in the background, use fs_cli to reattach and then look at the logs on screen while seeing the problem. Trever -- "There are two ways to live your life. One is as though nothing is a miracle. The other is as though everything is a miracle." -- Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/c7f41e24/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/c7f41e24/attachment-0001.bin From valery.kalinin at gmail.com Mon Mar 19 14:24:09 2012 From: valery.kalinin at gmail.com (Valery Kalinin) Date: Mon, 19 Mar 2012 17:24:09 +0600 Subject: [Freeswitch-users] FreeSWITCH panel - web utility update Message-ID: > I set it to true but it's still eating major CPU (80%+). According to the profiler 98% of the CPU time spent on the php function "sleep". So really the processor does not overload, it just "hangs" on sleep. From freeswitch-list at puzzled.xs4all.nl Mon Mar 19 16:02:58 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Mon, 19 Mar 2012 14:02:58 +0100 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: <4F67102B.1020305@telefaks.de> References: <4F63B627.5070206@telefaks.de> <4F64A36B.2060801@puzzled.xs4all.nl>, <1FFF97C269757C458224B7C895F35F15060D3D@cantor.std.visionutv.se>, <4F662EE7.6050607@telefaks.de> <1FFF97C269757C458224B7C895F35F15060D86@cantor.std.visionutv.se> <4F67102B.1020305@telefaks.de> Message-ID: <4F672E82.3010603@puzzled.xs4all.nl> On 19-03-12 11:53, Peter Steinbach wrote: > I am using Ubuntu 10.04. > > I followed the installation procedures on the wiki page and installed > > ptlib-2.8.2 + h323plus-trunk Not sure if it's still valid but in the past the recommendation was to follow the versions listed at: http://www.gnugk.org/compiling-gnugk.html Which in this case means you should use H323Plus 1.22.0 and not trunk. Regards, Patrick From miha at softnet.si Mon Mar 19 16:33:29 2012 From: miha at softnet.si (Miha) Date: Mon, 19 Mar 2012 14:33:29 +0100 Subject: [Freeswitch-users] blocking destination number In-Reply-To: References: <4F5A1591.8090703@softnet.si> <4F6305C8.1070509@softnet.si> Message-ID: <4F6735A9.50303@softnet.si> On 3/16/2012 5:58 PM, Brian Foster wrote: > > The toll_allow variable.would be defined in the actual user xml file > located in /usr/local/freeswitch/conf/directory. Then you would > evaluate that variable in the dialplan that would end up making the > bridge to your gateway. > > -BDF > > On Mar 16, 2012 5:21 AM, "Miha" > wrote: > > Hi @Gabriel, > > I was looking on wiki how to define new xml or document where I will > have my restricted numbers. > Where are domestic,international,local defined? > > Thanks! > Miha > > On 3/10/2012 11:23 PM, Gabriel Gunderson wrote: > > On Fri, Mar 9, 2012 at 7:37 AM, Miha Zoubek > wrote: > >> what is the bast way to block destination number for certain user. > >> Is it possible to do it in user/dir? > > Ultimately, it would be a dialplan configuration, but you could set > > the variable that you match in the directory. See the example of > > 'toll_allow' in the default FreeSWITCH configuration... > > > > > > This is set on a per-user basis in the DIRECTORY: > > > > > > > > > > > > But it's considered while evaluating the DIALPLAN: > > > > > > > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > > > data="effective_caller_id_name=${outbound_caller_id_name}"/> > > > data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> > > > > > > > > > > > > You could easily do something similar by having a list of > numbers that > > they can't call listed in the directory (maybe something like > > 'restricted_numbers') and check to make sure the destination doesn't > > match it in the dialplan. > > > > Good luck! > > > > > > Gabe > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Thank you Brian! Regards, Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/bae5d5db/attachment.html From peter.olsson at visionutveckling.se Mon Mar 19 16:33:26 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 19 Mar 2012 13:33:26 +0000 Subject: [Freeswitch-users] Advice how to start with H.323 Message-ID: <1FFF97C269757C458224B7C895F35F150786C4@cantor.std.visionutv.se> Yes, I've used those recommendations - so 1.22.0 is probably a better choice. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Patrick Lists Skickat: den 19 mars 2012 14:03 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Advice how to start with H.323 On 19-03-12 11:53, Peter Steinbach wrote: > I am using Ubuntu 10.04. > > I followed the installation procedures on the wiki page and installed > > ptlib-2.8.2 + h323plus-trunk Not sure if it's still valid but in the past the recommendation was to follow the versions listed at: http://www.gnugk.org/compiling-gnugk.html Which in this case means you should use H323Plus 1.22.0 and not trunk. Regards, Patrick _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f672e0f32766345684320! From adam.kelloway at newpace.ca Mon Mar 19 17:02:58 2012 From: adam.kelloway at newpace.ca (Adam Kelloway) Date: Mon, 19 Mar 2012 11:02:58 -0300 Subject: [Freeswitch-users] is loopback really 'evil'? Message-ID: <4F673C92.6020303@newpace.ca> Hi there, Could someone please elaborate on this (from http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#loopback): "WARNING! Loopback is evil and should only be used as a last resort, when no other approach is possible." Need one only take the precautions noted in http://wiki.freeswitch.org/wiki/Loopback#Precautions in mind, or is there more to loopback than meets the eye? Thanks, Adam From andrew at cassidywebservices.co.uk Mon Mar 19 17:20:11 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 19 Mar 2012 14:20:11 +0000 Subject: [Freeswitch-users] is loopback really 'evil'? In-Reply-To: <4F673C92.6020303@newpace.ca> References: <4F673C92.6020303@newpace.ca> Message-ID: Good question, because the default voicemail setup uses loopback... On 19 March 2012 14:02, Adam Kelloway wrote: > Hi there, > > Could someone please elaborate on this (from > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#loopback): > > "WARNING! Loopback is evil and should only be used as a last resort, > when no other approach is possible." > > Need one only take the precautions noted in > http://wiki.freeswitch.org/wiki/Loopback#Precautions in mind, or is > there more to loopback than meets the eye? > > Thanks, > > Adam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/924dd71b/attachment-0001.html From admin at blindi.net Mon Mar 19 17:54:10 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Mon, 19 Mar 2012 15:54:10 +0100 (CET) Subject: [Freeswitch-users] FreeSWITCH panel - web utility update In-Reply-To: <13629d6d4f2.465495304447425147.7543463568927770653@zoho.com> References: <13629d6d4f2.465495304447425147.7543463568927770653@zoho.com> Message-ID: Your panel is very nice. My suggestions: 1. installation isstrutions for newbies. 2. Callforwarding. 3. Callback on busy. 4. withlist / Blacklist. 5. Wakeupcall to a extenion or phonenumber. 6. activate/ deactivate callscreening to a user. 7. callgroups. 8. conferencingfeatures: bridge a channel to a nother (privatecall). Move users to a conference. exprire a conference. Limit a conference to channels or phonenumbers. Mute / unmute user and conferences. Talkdetection. Play a sound or stop Playback record a conference. Say the usercount. User must record the name before entering the conference. Users can heare the name after enter the conference. Admin can heare a name, kick or transfer or mute these name. 9. Ivrs. 10. Gateways. 11. Timebasedrouting. 12. Follow me Functions (automatic location determination). 13. Identification numbers of oppressed 14. Connection of the incoming Saved call to other networks 15. Connection of the incoming call with third participants. 16. Faxservice. 17. E-mail for missed calls. 18. Call is forwarded to multiple recipients (ACD). 19. Callrecording. 20. Cdr. 21. Adressbook. 22.Display the caller name (automated number Phone book to look up) 23. Speeddial. 24. Dialprefix (Selection rules) 25. Online counter. 26. Least Cost Routing. 27. Push Service. 28. Conference invitation via e-mail. Thanks --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From tech at tech-invent.ru Mon Mar 19 11:42:54 2012 From: tech at tech-invent.ru (Dmitry Golubenko) Date: Mon, 19 Mar 2012 15:42:54 +0700 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: <4F662EE7.6050607@telefaks.de> References: <4F63B627.5070206@telefaks.de> <4F64A36B.2060801@puzzled.xs4all.nl>, <1FFF97C269757C458224B7C895F35F15060D3D@cantor.std.visionutv.se> <4F662EE7.6050607@telefaks.de> Message-ID: <4F66F18E.5020306@tech-invent.ru> 19.03.2012 01:52, Peter Steinbach ?????: > I tried to use mod_h323 now and followed the procedure on the wiki page. try to change mod_h323 Makefile to include correct path to your ptlib and h323plus folders, mine looks like this: BASE=../../../.. #export PTLIBDIR = $(shell /usr/local/bin/ptlib-config --ptlibdir) LOCAL_CFLAGS+=-g -I/usr/local/src/ptlib-v2_10_3/include -I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exc LOCAL_LDFLAGS= -L/usr/local/lib -lopenh323 -lpt -lrt ifeq ($(shell uname -m),x86_64) LOCAL_CFLAGS+=-DP_64BIT endif include $(BASE)/build/modmake.rules > > After fixing problems with > - not finding /usr/bin/ptlib-config (copied from /usr/local/bin/, > linking did not work) > - not finding the h323 include files (linking from /usr/local/include) > I now get the following problem: > > root at fs01:/usr/src/freeswitch# make mod_h323 > making all mod_h323 > Compiling /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp... > quiet_libtool: compile: g++ -g -ITLIBDIR -I/usr/include/openh323 -I. > -DPTRACING=1 -D_REENTRANT -fno-exceptions -DP_64BIT > -I/usr/src/freeswitch/libs/curl/include > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE > -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp -fPIC -DPIC > -o .libs/mod_h323.o > In file included from /usr/include/openh323/h323.h:36, > from /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:43, > from /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:40: > /usr/include/openh323/openh323buildopts.h:37:34: error: > ptlib/../../revision.h: No such file or directory > /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp: In member > function ?virtual PBoolean FSH323_ExternalRTPChannel::Start()?: > /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:2125: > warning: format ?%lu? expects type ?long unsigned int?, but argument 8 > has type ?unsigned int? > /usr/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:2126: > warning: format ?%lu? expects type ?long unsigned int?, but argument 8 > has type ?unsigned int? > make[4]: *** [mod_h323.lo] Error 1 > make[3]: *** [all] Error 1 > make[2]: *** [mod_h323-all] Error 1 > make[1]: *** [mod_h323] Error 2 > make: *** [mod_h323] Error 2 > > This is really tiring, as I run from one problem to the next one. And I > am not sure whether this will lead me to a stable system at the end. > > So am I back to my one of my previous questions: Maybe I am on the wrong > distribution? What do you recommend? > > Best regards > Peter > > > Am 18.03.2012 15:45, schrieb Peter Olsson: >> I'm using mod_h323. It works quite well - even though it's not 100% perfect, but the same goes for mod_opal... >> >> I've used them both, but I actually think mod_h323 is a better and more stable choice. >> >> /Peter >> >> ________________________________ >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Brian Foster [bdfoster at endigotech.com] >> Skickat: den 18 mars 2012 15:29 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] Advice how to start with H.323 >> >> >> That patch probably fixes that issue. If not, it's easily fixed. >> >> On Mar 17, 2012 10:45 AM, "Patrick Lists"> wrote: >> On 03/17/2012 03:30 AM, Brian Foster wrote: >> [snip] >>> > are there any dependencies for the opal and libpt versions? >>> >>> Those should be taken care of by the build script (if there are any) >> That's assuming that the build script does the proper thing. Last time I >> looked, it pulled svn trunk versions of ptlib and opal which are too new >> for mod_opal. Afaik it needs the Sirius or maybe the Luyten release. >> In the past I built mod_opal without a problem (not using that script) >> with the Sirius release of ptlib and opal installed. >> >> Regarrds, >> Patrick >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> !DSPAM:4f65f20a32765192515336! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > From bernard.david.murphy at gmail.com Mon Mar 19 12:30:13 2012 From: bernard.david.murphy at gmail.com (bernard.david.murphy at gmail.com) Date: Mon, 19 Mar 2012 09:30:13 +0000 Subject: [Freeswitch-users] SIP Notify for MWI on avaya 9630 Message-ID: <304277796-1332151706-cardhu_decombobulator_blackberry.rim.net-1910878399-@b13.c12.bise7.blackberry> I have an Avaya 9630 with SIP 2.6.6 firmware which will subscribe to freeswitch on ext login and then receives SIP notify messages when a voicemail is left. This works fine in freeswitch 1.0.6 but fails to light the light in any newer versions of freeswitch like 1.0 git head. In both cases the notify messages are definitely sent to the phone but anything other than 1.0.6 does not light the MWI light. Does anyone know of any reason. I will post the SIP messages shortly. Does anyone know how to control the format of these notify messages from within freeswitch, I.e. Is it possible in cml file to modify the content? Thanks Bernie Sent from my BlackBerry? wireless device From fredyg1965 at gmail.com Mon Mar 19 18:57:29 2012 From: fredyg1965 at gmail.com (Fredy Gonzales) Date: Mon, 19 Mar 2012 10:57:29 -0500 Subject: [Freeswitch-users] FreeSWITCH panel - web utility update References: Message-ID: Fabulous ... that good tool. FG ----- Original Message ----- From: "Valery Kalinin" To: Sent: Monday, March 19, 2012 1:00 AM Subject: [Freeswitch-users] FreeSWITCH panel - web utility update > Hi all! > > I update my "FreeSWITCH panel" online web utility. > > Screenshot & help: > https://sites.google.com/site/freeswitched/home/downloads/fspanel_help.png > > This application allows you to view and control online in web browser: > - SIP registered subscribers > - Sofia status > profile commands: > - rescan > - restart > - flush registered endpoints > - flush and reboot registered endpoints > gateway commands: > - kill > - current channels/calls/detailed_calls > - hangup call > - conferences > - lock/unlock > - pin/no pin > - dial > - dtmf > conference member: > - defa/undeaf > - mute/unmute > - hup > - kick > - transfer to > - volume in/out > - energy level > - FreeTDM channels > - span start/stop > - chan info > It is also possible simple Freeswitch management: > - sofia status > - reloadxml > - reloadacl > - status > - version > You can select the display fields. > All settings are stored in a cookies. > > Site here: https://sites.google.com/site/freeswitched/home > git: https://github.com/Slonik/FreeSWITCH-panel > > Tests and features are welcome. > > Short install instructions: > Upload files to your web-server with PHP support. > Check modules.conf and event_socket.conf for enable module > mod_event_socket > Change variables if needed in fscontrol.php file: > $FreeSWITCHserver = '127.0.0.1'; // event_socket.xml param: listen-ip > $FreeSWITCHport = 8021; // event_socket.xml param: listen-port > $FreeSWITCHpassword = 'xexexe'; // event_socket.xml param: password > $SofiaProfiles = array('internal'); // write empty array if not used: > array(); > $FreeTDMspans = array('pri'); // write empty array if not used: array(); > $disableXML = false; // You can disable retrieve XML data if error > messages received > Enter in browser http://you-server-name/fspanel.html > Click on menu item "settings" for choice items. > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From curriegrad2004 at gmail.com Mon Mar 19 19:17:17 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 19 Mar 2012 09:17:17 -0700 Subject: [Freeswitch-users] SIP Notify for MWI on avaya 9630 In-Reply-To: <304277796-1332151706-cardhu_decombobulator_blackberry.rim.net-1910878399-@b13.c12.bise7.blackberry> References: <304277796-1332151706-cardhu_decombobulator_blackberry.rim.net-1910878399-@b13.c12.bise7.blackberry> Message-ID: Have you tried checking out the older revisions of freeswitch git heads yet? iirc there were changes regarding to the BLF done a couple of weeks ago On Mon, Mar 19, 2012 at 2:30 AM, wrote: > > > I have an Avaya 9630 with SIP 2.6.6 firmware which will subscribe to freeswitch on ext login and then receives SIP notify messages when a voicemail is left. > > This works fine in freeswitch 1.0.6 but fails to light the light in any newer versions of freeswitch like 1.0 git head. > > In both cases the notify messages are definitely sent to the phone but anything other than 1.0.6 does not light the MWI light. > > Does anyone know of any reason. I will post the SIP messages shortly. > > Does anyone know how to control the format of these notify messages from within freeswitch, I.e. Is it possible in cml file to modify the content? > > Thanks > Bernie > > Sent from my BlackBerry? wireless device > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From avi at avimarcus.net Mon Mar 19 19:25:46 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 19 Mar 2012 18:25:46 +0200 Subject: [Freeswitch-users] FreeSWITCH panel - web utility update In-Reply-To: References: <13629d6d4f2.465495304447425147.7543463568927770653@zoho.com> Message-ID: Hi Thomas -- I just feel compelled to step in, in Valery's... defense? and say that he has released a free tool that *seems* to be an admin GUI. (e.g. it doesn't have any distinct user permissions, any persistent storage of it's own, etc.) Therefore most of the suggestions you are asking for would make it equivalent to fusionpbx, freepybx, blue.box (see Freeswitch-GUI) -- while it's always great to have options, these suggestions *seem* to be beyond the intent of this project. (Those projects already have many of what's in your list! Check them out!) If you want to utilize his framework to build on your requested features, he has open sourced it... -Avi Marcus On Mon, Mar 19, 2012 at 4:54 PM, Thomas Hoellriegel wrote: > Your panel is very nice. > My suggestions: > > 1. installation isstrutions for newbies. > 2. Callforwarding. > 3. Callback on busy. > 4. withlist / Blacklist. > 5. Wakeupcall to a extenion or phonenumber. > 6. activate/ deactivate callscreening to a user. > 7. callgroups. > 8. conferencingfeatures: > bridge a channel to a nother (privatecall). > Move users to a conference. > exprire a conference. > Limit a conference to channels or phonenumbers. > Mute / unmute user and conferences. > Talkdetection. > Play a sound or stop Playback > record a conference. > Say the usercount. > User must record the name before entering the conference. > Users can heare the name after enter the conference. > Admin can heare a name, kick or transfer or mute these name. > > > 9. Ivrs. > 10. Gateways. > 11. Timebasedrouting. > 12. Follow me Functions (automatic location determination). > 13. Identification numbers of oppressed > 14. Connection of the incoming Saved call to other networks > 15. Connection of the incoming call with third participants. > 16. Faxservice. > 17. E-mail for missed calls. > 18. Call is forwarded to multiple recipients (ACD). > 19. Callrecording. > 20. Cdr. > 21. Adressbook. > 22.Display the caller name (automated number Phone book to look up) > 23. Speeddial. > 24. Dialprefix (Selection rules) > 25. Online counter. > 26. Least Cost Routing. > 27. Push Service. > 28. Conference invitation via e-mail. > Thanks > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/14e8d548/attachment-0001.html From mitch.capper at gmail.com Mon Mar 19 19:46:51 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Mon, 19 Mar 2012 09:46:51 -0700 Subject: [Freeswitch-users] is loopback really 'evil'? In-Reply-To: References: <4F673C92.6020303@newpace.ca> Message-ID: I think its best to look at loopback as something of a hack, in most situations there is almost always another way to do what loopback does. As long as loopback behaves in your situation how it should you should be fine, but if when you go to use it you find oddities or issues generally taking loopback out of the equation will solve the problems. ~Mitch From mitch.capper at gmail.com Mon Mar 19 19:52:09 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Mon, 19 Mar 2012 09:52:09 -0700 Subject: [Freeswitch-users] FreeSWITCH panel - web utility update In-Reply-To: References: <13629d6d4f2.465495304447425147.7543463568927770653@zoho.com> Message-ID: I agree with Avi, In addition its important to note that this primarily a freeswitch status tool (showing whats going on) and it conducts itself over ESL. It definitely cannot modify dialplan configs, etc as its just over the event socket and not meant to be a config manager. It is certainly a nice HUD thanks for your contribution! ~Mitch From Kevin.Snow at ooma.com Mon Mar 19 21:00:01 2012 From: Kevin.Snow at ooma.com (Kevin Snow) Date: Mon, 19 Mar 2012 18:00:01 +0000 Subject: [Freeswitch-users] Using instant ringback. Message-ID: Hi, I'm experimenting with instant_ringback. It works perfectly but I'd like configure it to have a delay before starting. Making it not quite so instant. Many (maybe most) of our calls receive early media. Used in conjunction with instant_ringback it makes for a discontinuous ringback experience. First the user hears the instant_ringback, then it switches over to the early media ringback. Totally the right behavior but if I could configure instant ringback to start after X seconds I could avoid this for the user, except for cases where early media is slow to arrive. This would make a better customer experience. Is this possible? I looked into the teletone script syntax, hoping to insert a delay there but it didn't seem like I configure it with X seconds of silence then the ringback (and no more X seconds of silence). Thanks Kevin Snow -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/c5164f66/attachment.html From curriegrad2004 at gmail.com Mon Mar 19 21:20:25 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 19 Mar 2012 11:20:25 -0700 Subject: [Freeswitch-users] Using instant ringback. In-Reply-To: References: Message-ID: try ignoring early media and see what happens On Mon, Mar 19, 2012 at 11:00 AM, Kevin Snow wrote: > Hi, > > I'm experimenting with instant_ringback. It works perfectly but I'd like > configure it to have a delay before starting. Making it not quite so > instant. > > Many (maybe most) of our calls receive early media. Used in conjunction with > instant_ringback it makes for a discontinuous ringback experience. First the > user hears the instant_ringback, then it switches over to the early media > ringback. Totally the right behavior but if I could configure instant > ringback to start after X seconds I could avoid this for the user, except > for cases where early media is slow to arrive. This would make a better > customer experience. > > Is this possible? > > I looked into the teletone script syntax, hoping to insert a delay there but > it didn't seem like I configure it with X seconds of silence then the > ringback (and no more X seconds of silence). > > Thanks > > Kevin Snow > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From javieraristizabal at gmail.com Mon Mar 19 21:25:19 2012 From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=) Date: Mon, 19 Mar 2012 19:25:19 +0100 Subject: [Freeswitch-users] FreeSWITCH panel - web utility update In-Reply-To: References: <13629d6d4f2.465495304447425147.7543463568927770653@zoho.com> Message-ID: It looks nice :-) good job! On Mon, Mar 19, 2012 at 5:52 PM, Mitch Capper wrote: > I agree with Avi, > In addition its important to note that this primarily a freeswitch > status tool (showing whats going on) and it conducts itself over ESL. > It definitely cannot modify dialplan configs, etc as its just over the > event socket and not meant to be a config manager. It is certainly a > nice HUD thanks for your contribution! > > ~Mitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Javier Aristiz?bal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/4413a92f/attachment.html From Kevin.Snow at ooma.com Mon Mar 19 21:57:18 2012 From: Kevin.Snow at ooma.com (Kevin Snow) Date: Mon, 19 Mar 2012 18:57:18 +0000 Subject: [Freeswitch-users] Using instant ringback. In-Reply-To: Message-ID: That would do it, but it breaks some calls. There are some numbers that play messages during the early media phase, we need to play early media. Good idea though, thanks. Kevin On 3/19/12 11:20 AM, "curriegrad2004" wrote: >try ignoring early media and see what happens > >On Mon, Mar 19, 2012 at 11:00 AM, Kevin Snow wrote: >> Hi, >> >> I'm experimenting with instant_ringback. It works perfectly but I'd like >> configure it to have a delay before starting. Making it not quite so >> instant. >> >> Many (maybe most) of our calls receive early media. Used in conjunction >>with >> instant_ringback it makes for a discontinuous ringback experience. >>First the >> user hears the instant_ringback, then it switches over to the early >>media >> ringback. Totally the right behavior but if I could configure instant >> ringback to start after X seconds I could avoid this for the user, >>except >> for cases where early media is slow to arrive. This would make a better >> customer experience. >> >> Is this possible? >> >> I looked into the teletone script syntax, hoping to insert a delay >>there but >> it didn't seem like I configure it with X seconds of silence then the >> ringback (and no more X seconds of silence). >> >> Thanks >> >> Kevin Snow >> >> >> >>_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org From avi at avimarcus.net Mon Mar 19 22:10:56 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 19 Mar 2012 21:10:56 +0200 Subject: [Freeswitch-users] Using instant ringback. In-Reply-To: References: Message-ID: instant_ringback is a variable, not an application, so scheduling it seems to be out. What about Dialplan_Tools_ring_ready on the A-leg? You'd only want that to execute if there isn't already media though... and not sure what happens if the call is active, already... Maybe you can schedule it for 5 seconds after the bridge... http://wiki.freeswitch.org/wiki/Mod_commands#sched_api -Avi On Mon, Mar 19, 2012 at 8:57 PM, Kevin Snow wrote: > That would do it, but it breaks some calls. There are some numbers that > play messages during the early media phase, we need to play early media. > > Good idea though, thanks. > > Kevin > > > > > On 3/19/12 11:20 AM, "curriegrad2004" wrote: > > >try ignoring early media and see what happens > > > >On Mon, Mar 19, 2012 at 11:00 AM, Kevin Snow wrote: > >> Hi, > >> > >> I'm experimenting with instant_ringback. It works perfectly but I'd like > >> configure it to have a delay before starting. Making it not quite so > >> instant. > >> > >> Many (maybe most) of our calls receive early media. Used in conjunction > >>with > >> instant_ringback it makes for a discontinuous ringback experience. > >>First the > >> user hears the instant_ringback, then it switches over to the early > >>media > >> ringback. Totally the right behavior but if I could configure instant > >> ringback to start after X seconds I could avoid this for the user, > >>except > >> for cases where early media is slow to arrive. This would make a better > >> customer experience. > >> > >> Is this possible? > >> > >> I looked into the teletone script syntax, hoping to insert a delay > >>there but > >> it didn't seem like I configure it with X seconds of silence then the > >> ringback (and no more X seconds of silence). > >> > >> Thanks > >> > >> Kevin Snow > >> > >> > >> > >>_________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > >_________________________________________________________________________ > >Professional FreeSWITCH Consulting Services: > >consulting at freeswitch.org > >http://www.freeswitchsolutions.com > > > > > > > > > >Official FreeSWITCH Sites > >http://www.freeswitch.org > >http://wiki.freeswitch.org > >http://www.cluecon.com > > > >FreeSWITCH-users mailing list > >FreeSWITCH-users at lists.freeswitch.org > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/d0137b5b/attachment-0001.html From bdfoster at endigotech.com Mon Mar 19 22:12:41 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 19 Mar 2012 15:12:41 -0400 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds Message-ID: Alright, so I admit... I'm a little rusty when it comes to NAT, etc. I've only set up FS so far on machines with no NAT, so this is sort of a new experience for me. I have a FreeSWITCH server located on the same local network as all of my phones here at the house. When I try to make a call to Flowroute, after about 30 seconds the call drops. It also does the exact same thing when I call a buddy's server directly via SIP. Here's a siptrace of the call (I didn't think that the actual FS log would be much help): http://pastebin.freeswitch.org/18697 ...and here's a paste of 'sofia status': http://pastebin.freeswitch.org/18698 ...and just for good measure, here's a paste of vars.xml: http://pastebin.freeswitch.org/18699 -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/acac9f7d/attachment.html From bdfoster at endigotech.com Mon Mar 19 22:16:10 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 19 Mar 2012 15:16:10 -0400 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: Message-ID: Probably should include this... freeswitch at internal> version FreeSWITCH Version 1.0.head (git-fcab3de 2012-03-15 18-57-19 +0000) root at homeserver:~# ifconfig eth0 Link encap:Ethernet HWaddr 00:1a:a0:a4:c7:d8 inet addr:192.168.1.79 Bcast:192.168.1.255 Mask:255.255.255.0 inet6 addr: fe80::21a:a0ff:fea4:c7d8/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:1398614 errors:0 dropped:0 overruns:0 frame:0 TX packets:448607 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:743611560 (709.1 MiB) TX bytes:254563244 (242.7 MiB) Interrupt:16 lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:141954 errors:0 dropped:0 overruns:0 frame:0 TX packets:141954 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:25148012 (23.9 MiB) TX bytes:25148012 (23.9 MiB) On Mon, Mar 19, 2012 at 3:12 PM, Brian Foster wrote: > Alright, so I admit... I'm a little rusty when it comes to NAT, etc. I've > only set up FS so far on machines with no NAT, so this is sort of a new > experience for me. > > I have a FreeSWITCH server located on the same local network as all of my > phones here at the house. When I try to make a call to Flowroute, after > about 30 seconds the call drops. It also does the exact same thing when I > call a buddy's server directly via SIP. > > Here's a siptrace of the call (I didn't think that the actual FS log would > be much help): > http://pastebin.freeswitch.org/18697 > > ...and here's a paste of 'sofia status': > http://pastebin.freeswitch.org/18698 > > ...and just for good measure, here's a paste of vars.xml: > http://pastebin.freeswitch.org/18699 > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/bc0007d6/attachment.html From moises.silva at gmail.com Mon Mar 19 23:41:42 2012 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 19 Mar 2012 16:41:42 -0400 Subject: [Freeswitch-users] TDM400P spans In-Reply-To: References: <5227389.xYDaNXzQut@axp> Message-ID: On Sun, Mar 18, 2012 at 10:58 AM, Jeromy wrote: > > I'm trying to convert from Asterisk to Freeswitch, and have a question > about > spans. > > > > I read the freetdm configuration example for TDM400 > > and am wondering what exactly is a span. > > > > I have 2 TDM400P cards in my computer, > > one card has 2 FXS modules for connecting analog phones. > > The other card has 3 FXO modules for connecting to 3 PSTN trunk lines. > > > > # lsdahdi > > ### Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" > > 1 FXS FXOKS (In use) (EC: OSLEC - INACTIVE) > > 2 FXS FXOKS (In use) (EC: OSLEC - INACTIVE) > > 3 unknown Reserved > > 4 unknown Reserved > > ### Span 2: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER) > > 5 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) > > 6 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) > > 7 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) > > 8 unknown Reserved > > > > Do I configure freetdm as in the example for TDM400 > > with separate spans for each module like the following? > > or should I have just two spans as shown by lsdahsi? > > Thanks. > > > > Hi Jeremy > > Do you find the answer to this question? > > I'm also very confuse on the span and channel in both asterisk and > freeswitch, and I've found little explaination on this. > > " A span is a logical unit that represents a group of channels. > With digital telephony, a span usually represents a physical port > on the card. > If the system has only one such card with a single port, so it is > referred to > as span 1. " > > Could anyone help on this? > > At the very least you need 2 freetdm span configurations because you cannot mix FXS and FXO channels in the same FreeTDM logical span. http://wiki.freeswitch.org/wiki/FreeTDM#DAHDI_mode For the given hardware output, I'd simply create 2 spans, one for all the fxo channels and one for all the fxs channels. *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/a0df320d/attachment-0001.html From sdevoy at bizfocused.com Mon Mar 19 20:07:13 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Mon, 19 Mar 2012 13:07:13 -0400 Subject: [Freeswitch-users] Unable to TXFAX, but receiving is OK. Always result (49) Message-ID: <02a301cd05f2$bb00aaa0$3101ffe0$@bizfocused.com> Hi All, I can receive faxes and email them to their recipients now with rxfax!!!! I cannot send anything though. My test command is: (with number changed) /usr/local/freeswitch/bin/fs_cli -x "originate {ignore_early_media=true}sofia/gateway/voipinnovations/11234567890 &txfax(/usr/sean/output/txfax.tiff)" I have also tried it without the ignore_early_media. I do not have any V.38 enabled. When I did no incoming faxes would work. The receiving fax never sent tone, so sender never connected. Any help would be greatly appreciated. Thanks. Here is the FS log: freeswitch at internal> 2012-03-19 12:58:06.511937 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-19 12:58:06.511937 [DEBUG] switch_event.c:1521 Parsing variable [ignore_early_media]=[true] 2012-03-19 12:58:06.511937 [NOTICE] switch_channel.c:924 New Channel sofia/external_noauth/14108038876 [bc6cadc8-761b-42cb-8873-889320fc2658] 2012-03-19 12:58:06.511937 [DEBUG] mod_sofia.c:4674 (sofia/external_noauth/14108038876) State Change CS_NEW -> CS_INIT 2012-03-19 12:58:06.511937 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:06.511937 [DEBUG] switch_core_state_machine.c:362 (sofia/external_noauth/14108038876) Running State Change CS_INIT 2012-03-19 12:58:06.511937 [DEBUG] switch_core_state_machine.c:401 (sofia/external_noauth/14108038876) State INIT 2012-03-19 12:58:06.511937 [DEBUG] mod_sofia.c:85 sofia/external_noauth/14108038876 SOFIA INIT 2012-03-19 12:58:07.117896 [DEBUG] switch_nat.c:511 mapped public port 17322 protocol UDP to localport 17322 2012-03-19 12:58:07.602876 [DEBUG] switch_nat.c:511 mapped public port 17323 protocol UDP to localport 17323 2012-03-19 12:58:07.602876 [DEBUG] mod_sofia.c:125 (sofia/external_noauth/14108038876) State Change CS_INIT -> CS_ROUTING 2012-03-19 12:58:07.602876 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:07.602876 [DEBUG] switch_core_state_machine.c:401 (sofia/external_noauth/14108038876) State INIT going to sleep 2012-03-19 12:58:07.602876 [DEBUG] switch_core_state_machine.c:362 (sofia/external_noauth/14108038876) Running State Change CS_ROUTING 2012-03-19 12:58:07.602876 [DEBUG] switch_channel.c:1884 (sofia/external_noauth/14108038876) Callstate Change DOWN -> RINGING 2012-03-19 12:58:07.602876 [DEBUG] switch_core_state_machine.c:410 (sofia/external_noauth/14108038876) State ROUTING 2012-03-19 12:58:07.602876 [DEBUG] mod_sofia.c:148 sofia/external_noauth/14108038876 SOFIA ROUTING 2012-03-19 12:58:07.602876 [DEBUG] switch_ivr_originate.c:66 (sofia/external_noauth/14108038876) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-03-19 12:58:07.602876 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:07.602876 [DEBUG] switch_core_state_machine.c:410 (sofia/external_noauth/14108038876) State ROUTING going to sleep 2012-03-19 12:58:07.602876 [DEBUG] switch_core_state_machine.c:362 (sofia/external_noauth/14108038876) Running State Change CS_CONSUME_MEDIA 2012-03-19 12:58:07.602876 [DEBUG] switch_core_state_machine.c:429 (sofia/external_noauth/14108038876) State CONSUME_MEDIA 2012-03-19 12:58:07.602876 [DEBUG] switch_core_state_machine.c:429 (sofia/external_noauth/14108038876) State CONSUME_MEDIA going to sleep 2012-03-19 12:58:07.602876 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:07.602876 [DEBUG] sofia.c:5494 Channel sofia/external_noauth/14108038876 entering state [calling][0] 2012-03-19 12:58:10.613725 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:10.613725 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:10.623715 [DEBUG] sofia.c:5494 Channel sofia/external_noauth/14108038876 entering state [proceeding][183] 2012-03-19 12:58:10.623715 [DEBUG] sofia.c:5505 Remote SDP: v=0 o=Sansay-VSXi 188 1 IN IP4 64.136.174.30 s=Session Controller c=IN IP4 69.85.185.142 t=0 0 m=audio 26144 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:2919 Set Codec sofia/external_noauth/14108038876 PCMU/8000 20 ms 160 samples 64000 bits 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/external_noauth/14108038876] 10.10.40.185 port 17322 -> 69.85.185.142 port 26144 codec: 0 ms: 20 2012-03-19 12:58:10.623715 [DEBUG] switch_rtp.c:1659 Starting timer [soft] 160 bytes per 20ms 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send payload to 101 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive payload to 101 2012-03-19 12:58:10.623715 [NOTICE] sofia_glue.c:3945 Pre-Answer sofia/external_noauth/14108038876! 2012-03-19 12:58:10.623715 [DEBUG] switch_channel.c:2930 (sofia/external_noauth/14108038876) Callstate Change RINGING -> EARLY 2012-03-19 12:58:23.010042 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:23.010042 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:23.010042 [DEBUG] sofia.c:5494 Channel sofia/external_noauth/14108038876 entering state [completing][200] 2012-03-19 12:58:23.010042 [DEBUG] sofia.c:5502 Duplicate SDP v=0 o=Sansay-VSXi 188 1 IN IP4 64.136.174.30 s=Session Controller c=IN IP4 69.85.185.142 t=0 0 m=audio 26144 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-03-19 12:58:23.010042 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:23.010042 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:23.010042 [DEBUG] sofia.c:5494 Channel sofia/external_noauth/14108038876 entering state [ready][200] 2012-03-19 12:58:23.010042 [DEBUG] switch_channel.c:3188 (sofia/external_noauth/14108038876) Callstate Change EARLY -> ACTIVE 2012-03-19 12:58:23.010042 [NOTICE] sofia.c:6142 Channel [sofia/external_noauth/14108038876] has been answered 2012-03-19 12:58:23.010042 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [sofia/external_noauth/14108038876] 2012-03-19 12:58:23.010042 [INFO] switch_channel.c:2708 sofia/external_noauth/14108038876 Flipping CID from "" <0000000000> to "Outbound Call" <14108038876> 2012-03-19 12:58:23.010042 [DEBUG] mod_commands.c:3574 (sofia/external_noauth/14108038876) State Change CS_CONSUME_MEDIA -> CS_EXECUTE 2012-03-19 12:58:23.010042 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:23.010042 [DEBUG] switch_core_state_machine.c:362 (sofia/external_noauth/14108038876) Running State Change CS_EXECUTE 2012-03-19 12:58:23.010042 [DEBUG] switch_core_state_machine.c:417 (sofia/external_noauth/14108038876) State EXECUTE 2012-03-19 12:58:23.010042 [DEBUG] mod_sofia.c:241 sofia/external_noauth/14108038876 SOFIA EXECUTE 2012-03-19 12:58:23.010042 [DEBUG] switch_core_state_machine.c:192 sofia/external_noauth/14108038876 Standard EXECUTE EXECUTE sofia/external_noauth/14108038876 txfax(/usr/sean/output/txfax.tiff) 2012-03-19 12:58:23.010042 [DEBUG] mod_spandsp_fax.c:1355 Raw read codec activation Success L16 20000 2012-03-19 12:58:23.010042 [DEBUG] switch_core_codec.c:116 sofia/external_noauth/14108038876 Push codec L16:70 2012-03-19 12:58:23.010042 [DEBUG] mod_spandsp_fax.c:1371 Raw write codec activation Success L16 2012-03-19 12:58:23.070036 [DEBUG] switch_rtp.c:3204 Correct ip/port confirmed. 2012-03-19 12:58:25.539903 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:25.549903 [DEBUG] switch_channel.c:2846 (sofia/external_noauth/14108038876) Callstate Change ACTIVE -> HANGUP 2012-03-19 12:58:25.549903 [NOTICE] sofia.c:623 Hangup sofia/external_noauth/14108038876 [CS_EXECUTE] [NORMAL_CLEARING] 2012-03-19 12:58:25.549903 [DEBUG] switch_channel.c:2869 Send signal sofia/external_noauth/14108038876 [KILL] 2012-03-19 12:58:25.549903 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:489 ============================================================================ == 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:502 Fax processing not successful - result (49) The call dropped prematurely. 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:507 Remote station id: 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:508 Local station id: SpanDSP Fax Ident 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:509 Pages transferred: 0 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:511 Total fax pages: 0 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:512 Image resolution: 0x0 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:513 Transfer Rate: 14400 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:515 ECM status off 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:516 remote country: 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:517 remote vendor: 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:518 remote model: 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:520 ============================================================================ == 2012-03-19 12:58:25.549903 [DEBUG] switch_core_codec.c:141 sofia/external_noauth/14108038876 Restore previous codec PCMU:0. 2012-03-19 12:58:25.549903 [DEBUG] switch_core_session.c:2285 sofia/external_noauth/14108038876 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:417 (sofia/external_noauth/14108038876) State EXECUTE going to sleep 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:362 (sofia/external_noauth/14108038876) Running State Change CS_HANGUP 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:602 (sofia/external_noauth/14108038876) State HANGUP 2012-03-19 12:58:25.549903 [DEBUG] mod_sofia.c:463 sofia/external_noauth/14108038876 Overriding SIP cause 480 with 200 from the other leg 2012-03-19 12:58:25.549903 [DEBUG] mod_sofia.c:469 Channel sofia/external_noauth/14108038876 hanging up, cause: NORMAL_CLEARING 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:47 sofia/external_noauth/14108038876 Standard HANGUP, cause: NORMAL_CLEARING 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:602 (sofia/external_noauth/14108038876) State HANGUP going to sleep 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:393 (sofia/external_noauth/14108038876) State Change CS_HANGUP -> CS_REPORTING 2012-03-19 12:58:25.549903 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:362 (sofia/external_noauth/14108038876) Running State Change CS_REPORTING 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:662 (sofia/external_noauth/14108038876) State REPORTING 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:79 sofia/external_noauth/14108038876 Standard REPORTING, cause: NORMAL_CLEARING 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:662 (sofia/external_noauth/14108038876) State REPORTING going to sleep 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:387 (sofia/external_noauth/14108038876) State Change CS_REPORTING -> CS_DESTROY 2012-03-19 12:58:25.549903 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:25.549903 [DEBUG] switch_core_session.c:1380 Session 184 (sofia/external_noauth/14108038876) Locked, Waiting on external entities 2012-03-19 12:58:25.549903 [NOTICE] switch_core_session.c:1398 Session 184 (sofia/external_noauth/14108038876) Ended 2012-03-19 12:58:25.549903 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external_noauth/14108038876 [CS_DESTROY] 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:491 (sofia/external_noauth/14108038876) Callstate Change HANGUP -> DOWN 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:494 (sofia/external_noauth/14108038876) Running State Change CS_DESTROY 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:504 (sofia/external_noauth/14108038876) State DESTROY 2012-03-19 12:58:25.549903 [DEBUG] mod_sofia.c:374 sofia/external_noauth/14108038876 SOFIA DESTROY 2012-03-19 12:58:25.559907 [DEBUG] switch_nat.c:571 unmapped public port 17322 protocol UDP to localport 17322 2012-03-19 12:58:25.579904 [DEBUG] switch_nat.c:571 unmapped public port 17323 protocol UDP to localport 17323 2012-03-19 12:58:25.579904 [DEBUG] switch_core_state_machine.c:86 sofia/external_noauth/14108038876 Standard DESTROY 2012-03-19 12:58:25.579904 [DEBUG] switch_core_state_machine.c:504 (sofia/external_noauth/14108038876) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/4e9940ab/attachment-0001.html From robert.longfield at klinsight.com Mon Mar 19 21:13:31 2012 From: robert.longfield at klinsight.com (Robert Longfield) Date: Mon, 19 Mar 2012 14:13:31 -0400 Subject: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Message-ID: <0488BB5D11C94E8285DD956672994480@KITPC003> I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM. My dialplan looks like: What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn?t work either. The message I see when Freeswitch hangs up is: Channel sofia/internal/sip:1002 at 72.38.184.18:39042 hanging up, cause: USER_BUSY The full output from cli can be seen here: http://pastebin.freeswitch.org/18696 I would like to get the call to transfer properly. Thanks -Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/4168c5b4/attachment.html From andrew at cassidywebservices.co.uk Mon Mar 19 23:45:04 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 19 Mar 2012 20:45:04 +0000 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: Message-ID: Try either enabling upnp on your router or setting ext-sip-ip and ext-rtp-ip to stun:stun.freeswitch.org, and forward the relevant ports on your router. Those are the first things to check. On 19 March 2012 19:16, Brian Foster wrote: > Probably should include this... > > freeswitch at internal> version > FreeSWITCH Version 1.0.head (git-fcab3de 2012-03-15 18-57-19 +0000) > > root at homeserver:~# ifconfig > eth0 Link encap:Ethernet HWaddr 00:1a:a0:a4:c7:d8 > inet addr:192.168.1.79 Bcast:192.168.1.255 Mask:255.255.255.0 > inet6 addr: fe80::21a:a0ff:fea4:c7d8/64 Scope:Link > UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 > RX packets:1398614 errors:0 dropped:0 overruns:0 frame:0 > TX packets:448607 errors:0 dropped:0 overruns:0 carrier:0 > collisions:0 txqueuelen:1000 > RX bytes:743611560 (709.1 MiB) TX bytes:254563244 (242.7 MiB) > Interrupt:16 > > lo Link encap:Local Loopback > inet addr:127.0.0.1 Mask:255.0.0.0 > inet6 addr: ::1/128 Scope:Host > UP LOOPBACK RUNNING MTU:16436 Metric:1 > RX packets:141954 errors:0 dropped:0 overruns:0 frame:0 > TX packets:141954 errors:0 dropped:0 overruns:0 carrier:0 > collisions:0 txqueuelen:0 > RX bytes:25148012 (23.9 MiB) TX bytes:25148012 (23.9 MiB) > > > On Mon, Mar 19, 2012 at 3:12 PM, Brian Foster wrote: > >> Alright, so I admit... I'm a little rusty when it comes to NAT, etc. I've >> only set up FS so far on machines with no NAT, so this is sort of a new >> experience for me. >> >> I have a FreeSWITCH server located on the same local network as all of my >> phones here at the house. When I try to make a call to Flowroute, after >> about 30 seconds the call drops. It also does the exact same thing when I >> call a buddy's server directly via SIP. >> >> Here's a siptrace of the call (I didn't think that the actual FS log >> would be much help): >> http://pastebin.freeswitch.org/18697 >> >> ...and here's a paste of 'sofia status': >> http://pastebin.freeswitch.org/18698 >> >> ...and just for good measure, here's a paste of vars.xml: >> http://pastebin.freeswitch.org/18699 >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/2c50d0a3/attachment.html From gchen00 at insightbb.com Tue Mar 20 00:12:09 2012 From: gchen00 at insightbb.com (GCHEN00) Date: Mon, 19 Mar 2012 17:12:09 -0400 Subject: [Freeswitch-users] Question about mysql database engine type. Message-ID: When I use ODBC as core database, freeswitch auto create the database tables using engine=MyISAM. I have to change the table type to innodb for 'complete' table to avoid the Maximum index length ERROR. My question is should I change rest of the tables to innodb or just leave them as MyISAM. Which one is better suited for freeswitch? Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/60b045e8/attachment.html From avi at avimarcus.net Tue Mar 20 00:11:56 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 19 Mar 2012 23:11:56 +0200 Subject: [Freeswitch-users] Question about mysql database engine type. In-Reply-To: References: Message-ID: I didn't have that issue.. I actually changed the channels, calls, sip_subscriptions and anything else transitory to memory tables to they wouldn't write to disk. I kept regs as myisam so it would persist. -Avi On Mon, Mar 19, 2012 at 11:12 PM, GCHEN00 wrote: > ** > When I use ODBC as core database, freeswitch auto create the database > tables using engine=MyISAM. > I have to change the table type to innodb for 'complete' table to avoid > the Maximum index length ERROR. > My question is should I change rest of the tables to innodb or just leave > them as MyISAM. > > Which one is better suited for freeswitch? > > Gary > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/6cab5ea5/attachment-0001.html From lazyvirus at gmx.com Tue Mar 20 00:25:29 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 19 Mar 2012 22:25:29 +0100 Subject: [Freeswitch-users] [OT] sflphone and tls Message-ID: <20120319222529.0206f70c@anubis.defcon1> Hi list, Sorry for this off-topic, but I can't find any reliable doc on how to configure sflphone to work w/ tls and freeswitch. Does anybody has a mod'op for that? JY -- From curriegrad2004 at gmail.com Tue Mar 20 01:25:44 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 19 Mar 2012 15:25:44 -0700 Subject: [Freeswitch-users] Question about mysql database engine type. In-Reply-To: References: Message-ID: The wiki on FS does say how the driver should be configured iirc, but some of the heavy users here prefer postgresql for the high performance code, but at the end of the day all it boils down to is how you want to use FS as and what not. On Mon, Mar 19, 2012 at 2:11 PM, Avi Marcus wrote: > I didn't have that issue.. > I actually changed the channels, calls, sip_subscriptions and anything else > transitory to memory tables to they wouldn't write to disk. > I kept regs as myisam so it would persist. > > -Avi > > > On Mon, Mar 19, 2012 at 11:12 PM, GCHEN00 wrote: >> >> When I use ODBC as core database, freeswitch auto create the database >> tables using engine=MyISAM. >> I have to change the?table type?to innodb for 'complete' table to avoid >> the Maximum index length ERROR. >> My question is should I?change?rest of??the tables to innodb or just leave >> them as MyISAM. >> >> Which?one is better suited for freeswitch? >> >> Gary >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From bdfoster at endigotech.com Tue Mar 20 02:10:29 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 19 Mar 2012 19:10:29 -0400 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: Message-ID: UPnP isn't available on this network. Ports 16384-16406 are forwarded to the machine from the router. Port range in switch.conf.xml reflects that. Also 5080 is forwarded from the router to the server. Both of those variables (ext-sip-ip and ext-rtp-ip) are set to the stun server, see pastebin 18699 (third link in original email). -BDF On Mar 19, 2012 4:46 PM, "Andrew Cassidy" wrote: > Try either enabling upnp on your router or setting ext-sip-ip and > ext-rtp-ip to stun:stun.freeswitch.org, and forward the relevant ports on > your router. > > Those are the first things to check. > > On 19 March 2012 19:16, Brian Foster wrote: > >> Probably should include this... >> >> freeswitch at internal> version >> FreeSWITCH Version 1.0.head (git-fcab3de 2012-03-15 18-57-19 +0000) >> >> root at homeserver:~# ifconfig >> eth0 Link encap:Ethernet HWaddr 00:1a:a0:a4:c7:d8 >> inet addr:192.168.1.79 Bcast:192.168.1.255 Mask:255.255.255.0 >> inet6 addr: fe80::21a:a0ff:fea4:c7d8/64 Scope:Link >> UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 >> RX packets:1398614 errors:0 dropped:0 overruns:0 frame:0 >> TX packets:448607 errors:0 dropped:0 overruns:0 carrier:0 >> collisions:0 txqueuelen:1000 >> RX bytes:743611560 (709.1 MiB) TX bytes:254563244 (242.7 MiB) >> Interrupt:16 >> >> lo Link encap:Local Loopback >> inet addr:127.0.0.1 Mask:255.0.0.0 >> inet6 addr: ::1/128 Scope:Host >> UP LOOPBACK RUNNING MTU:16436 Metric:1 >> RX packets:141954 errors:0 dropped:0 overruns:0 frame:0 >> TX packets:141954 errors:0 dropped:0 overruns:0 carrier:0 >> collisions:0 txqueuelen:0 >> RX bytes:25148012 (23.9 MiB) TX bytes:25148012 (23.9 MiB) >> >> >> On Mon, Mar 19, 2012 at 3:12 PM, Brian Foster wrote: >> >>> Alright, so I admit... I'm a little rusty when it comes to NAT, etc. >>> I've only set up FS so far on machines with no NAT, so this is sort of a >>> new experience for me. >>> >>> I have a FreeSWITCH server located on the same local network as all of >>> my phones here at the house. When I try to make a call to Flowroute, after >>> about 30 seconds the call drops. It also does the exact same thing when I >>> call a buddy's server directly via SIP. >>> >>> Here's a siptrace of the call (I didn't think that the actual FS log >>> would be much help): >>> http://pastebin.freeswitch.org/18697 >>> >>> ...and here's a paste of 'sofia status': >>> http://pastebin.freeswitch.org/18698 >>> >>> ...and just for good measure, here's a paste of vars.xml: >>> http://pastebin.freeswitch.org/18699 >>> >>> >>> -- >>> Brian D. Foster >>> Endigo Computer LLC >>> Email: bdfoster at endigotech.com >>> Phone: 317-800-7876 >>> Indianapolis, Indiana, USA >>> >>> This message contains confidential information and is intended for those >>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >>> you are not the intended recipient you are notified that disclosing, >>> copying, distributing or taking any action in reliance on the contents of >>> this information is strictly prohibited. E-mail transmission cannot be >>> guaranteed to be secure or error-free as information could be intercepted, >>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >>> The sender therefore does not accept liability for any errors or omissions >>> in the contents of this message, which arise as a result of e-mail >>> transmission. If verification is required please request a hard-copy >>> version. >>> >>> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Andrew Cassidy BSc (Hons) MBCS > Managing Director; Cassidy Web Services Ltd > T: 03300 100 960 F: 03300 100 961 > E: andrew at cassidywebservices.co.uk > W: www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/997ed011/attachment.html From bdfoster at endigotech.com Tue Mar 20 02:12:17 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 19 Mar 2012 19:12:17 -0400 Subject: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring In-Reply-To: <0488BB5D11C94E8285DD956672994480@KITPC003> References: <0488BB5D11C94E8285DD956672994480@KITPC003> Message-ID: Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml On Mar 19, 2012 4:43 PM, "Robert Longfield" wrote: > I set up a group call for our support team in which all their phones > ring when someone needs to speak with them. If they are busy the call > should be transferred to a general extension which if not answered then > goes to that extensions VM. > > My dialplan looks like: > > > > > > > > > > > > > What is happening is a caller selects the support option from the IVR, > ever phone in the support group rings, which is what should happen. If no > one picks up the call Freeswitch hangs up instead of transferring the call > to extension 1000. You can see that I also tried to send the call directly > to voicemail but that didn?t work either. > > The message I see when Freeswitch hangs up is: > > Channel sofia/internal/sip:1002 at 72.38.184.18:39042 hanging up, cause: > USER_BUSY > > The full output from cli can be seen here: > http://pastebin.freeswitch.org/18696 > > I would like to get the call to transfer properly. > > Thanks > -Robert > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/d3ff0781/attachment-0001.html From admin at blindi.net Tue Mar 20 02:29:13 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Tue, 20 Mar 2012 00:29:13 +0100 (CET) Subject: [Freeswitch-users] FreeSWITCH panel - web utility update In-Reply-To: References: <13629d6d4f2.465495304447425147.7543463568927770653@zoho.com> Message-ID: Hi Mitch and Avi, > In addition its important to note that this primarily a freeswitch > status tool (showing whats going on) and it conducts itself over ESL. Sorry, I'm not a PHP programmer. I've assumed that you need sudo for the execution. I know it usual for PHP applications. In order to process the apache access to. I think that the suggestions of the conference features fit well. Thanks From jack at livecall.com Tue Mar 20 03:11:29 2012 From: jack at livecall.com (Jack) Date: Mon, 19 Mar 2012 17:11:29 -0700 Subject: [Freeswitch-users] FreeSWITCH panel - web utility update In-Reply-To: References: Message-ID: <4F67CB31.2010208@livecall.com> I have installed on windows ii6 server and when I launch the webpage I get an alert that says "Please debug or disable XML!" Any idea what I need? Thanks in advance.. Jack On 3/18/2012 11:00 PM, Valery Kalinin wrote: > Hi all! > > I update my "FreeSWITCH panel" online web utility. > > Screenshot& help: > https://sites.google.com/site/freeswitched/home/downloads/fspanel_help.png > > This application allows you to view and control online in web browser: > - SIP registered subscribers > - Sofia status > profile commands: > - rescan > - restart > - flush registered endpoints > - flush and reboot registered endpoints > gateway commands: > - kill > - current channels/calls/detailed_calls > - hangup call > - conferences > - lock/unlock > - pin/no pin > - dial > - dtmf > conference member: > - defa/undeaf > - mute/unmute > - hup > - kick > - transfer to > - volume in/out > - energy level > - FreeTDM channels > - span start/stop > - chan info > It is also possible simple Freeswitch management: > - sofia status > - reloadxml > - reloadacl > - status > - version > You can select the display fields. > All settings are stored in a cookies. > > Site here: https://sites.google.com/site/freeswitched/home > git: https://github.com/Slonik/FreeSWITCH-panel > > Tests and features are welcome. > > Short install instructions: > Upload files to your web-server with PHP support. > Check modules.conf and event_socket.conf for enable module mod_event_socket > Change variables if needed in fscontrol.php file: > $FreeSWITCHserver = '127.0.0.1'; // event_socket.xml param: listen-ip > $FreeSWITCHport = 8021; // event_socket.xml param: listen-port > $FreeSWITCHpassword = 'xexexe'; // event_socket.xml param: password > $SofiaProfiles = array('internal'); // write empty array if not used: array(); > $FreeTDMspans = array('pri'); // write empty array if not used: array(); > $disableXML = false; // You can disable retrieve XML data if error > messages received > Enter in browser http://you-server-name/fspanel.html > Click on menu item "settings" for choice items. > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From freeswitch-list at puzzled.xs4all.nl Tue Mar 20 04:20:17 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Tue, 20 Mar 2012 02:20:17 +0100 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: Message-ID: <4F67DB51.20900@puzzled.xs4all.nl> On 20-03-12 00:10, Brian Foster wrote: > UPnP isn't available on this network. Ports 16384-16406 are forwarded to > the machine from the router. Port range in switch.conf.xml reflects > that. Also 5080 is forwarded from the router to the server. > > Both of those variables (ext-sip-ip and ext-rtp-ip) are set to the stun > server, see pastebin 18699 (third link in original email). If the (ADSL) router has a (for lack of a better term cause I can't remember) "SIP helper" app, try disabling it. Similar to the Cisco "SMTP Fixup" functionality which does an excellent job at the opposite. Regards, Patrick From bdfoster at endigotech.com Tue Mar 20 05:14:35 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 19 Mar 2012 22:14:35 -0400 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: <4F67DB51.20900@puzzled.xs4all.nl> References: <4F67DB51.20900@puzzled.xs4all.nl> Message-ID: Yea I know what you're talking about. That has indeed been switched off on the RG. For reference purposes, the Residential Gateway is a 2wire AT&T-branded Uverse DSL modem/wireless router with a 4 port switch. I've also got two 8 port netgear prosafe switches. Phones and server are on the same subnet/netmask. Very simple network setup. No VLANS, etc. This was indeed working with Asterisk, so I'm almost positive it's a configuration issue with FreeSWITCH. If there's a need to look at more than the files/logs I've pb'd, please let be know! -BDF On Mar 19, 2012 9:20 PM, "Patrick Lists" wrote: > On 20-03-12 00:10, Brian Foster wrote: > > UPnP isn't available on this network. Ports 16384-16406 are forwarded to > > the machine from the router. Port range in switch.conf.xml reflects > > that. Also 5080 is forwarded from the router to the server. > > > > Both of those variables (ext-sip-ip and ext-rtp-ip) are set to the stun > > server, see pastebin 18699 (third link in original email). > > If the (ADSL) router has a (for lack of a better term cause I can't > remember) "SIP helper" app, try disabling it. Similar to the Cisco "SMTP > Fixup" functionality which does an excellent job at the opposite. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/2951230c/attachment.html From paul at iamfine.com Tue Mar 20 05:41:28 2012 From: paul at iamfine.com (Paul) Date: Mon, 19 Mar 2012 19:41:28 -0700 (PDT) Subject: [Freeswitch-users] Mod_flite wont load In-Reply-To: <1331919970183-7379705.post@n2.nabble.com> References: <1331842568647-7376926.post@n2.nabble.com> <1331919970183-7379705.post@n2.nabble.com> Message-ID: <1332211288645-7387869.post@n2.nabble.com> Jeff I did a Make current. Success. The changes you made in the makefile for flite fixed the problems on this OSX system as well as the production system on RedHat where I files a JIRA FS-4022 Many thanks -- CLOSED -- -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Mod-flite-wont-load-tp7376926p7387869.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mitch.capper at gmail.com Tue Mar 20 06:11:18 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Mon, 19 Mar 2012 20:11:18 -0700 Subject: [Freeswitch-users] Unable to TXFAX, but receiving is OK. Always result (49) In-Reply-To: <02a301cd05f2$bb00aaa0$3101ffe0$@bizfocused.com> References: <02a301cd05f2$bb00aaa0$3101ffe0$@bizfocused.com> Message-ID: Faxing is a B to try and get it to work reliably over VOIP. I have added a troubleshooting section at: http://wiki.freeswitch.com/wiki/Mod_spandsp#Troubleshooting with what I have found. It may help to track down the problem. ~mitch From anton.jugatsu at gmail.com Tue Mar 20 08:18:42 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Tue, 20 Mar 2012 09:18:42 +0400 Subject: [Freeswitch-users] Unable to TXFAX, but receiving is OK. Always result (49) In-Reply-To: References: <02a301cd05f2$bb00aaa0$3101ffe0$@bizfocused.com> Message-ID: Does your ITSP support T.38? Try to catch RE-INVITE ngrep -d eth0 -qt -W byline -i t38 20 ????? 2012 ?. 7:11 ???????????? Mitch Capper ???????: > Faxing is a B to try and get it to work reliably over VOIP. I have > added a troubleshooting section at: > http://wiki.freeswitch.com/wiki/Mod_spandsp#Troubleshooting with what > I have found. It may help to track down the problem. > > ~mitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/a2c93df6/attachment.html From anton.jugatsu at gmail.com Tue Mar 20 08:30:04 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Tue, 20 Mar 2012 09:30:04 +0400 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: <4F67DB51.20900@puzzled.xs4all.nl> Message-ID: Try to catch RTP streams tcpdump -i eth0 portrange 16384-16406 and see what happens. 20 ????? 2012 ?. 6:14 ???????????? Brian Foster ???????: > Yea I know what you're talking about. That has indeed been switched off on > the RG. For reference purposes, the Residential Gateway is a 2wire > AT&T-branded Uverse DSL modem/wireless router with a 4 port switch. I've > also got two 8 port netgear prosafe switches. Phones and server are on the > same subnet/netmask. Very simple network setup. No VLANS, etc. This was > indeed working with Asterisk, so I'm almost positive it's a configuration > issue with FreeSWITCH. > > If there's a need to look at more than the files/logs I've pb'd, please > let be know! > > -BDF > On Mar 19, 2012 9:20 PM, "Patrick Lists" < > freeswitch-list at puzzled.xs4all.nl> wrote: > >> On 20-03-12 00:10, Brian Foster wrote: >> > UPnP isn't available on this network. Ports 16384-16406 are forwarded to >> > the machine from the router. Port range in switch.conf.xml reflects >> > that. Also 5080 is forwarded from the router to the server. >> > >> > Both of those variables (ext-sip-ip and ext-rtp-ip) are set to the stun >> > server, see pastebin 18699 (third link in original email). >> >> If the (ADSL) router has a (for lack of a better term cause I can't >> remember) "SIP helper" app, try disabling it. Similar to the Cisco "SMTP >> Fixup" functionality which does an excellent job at the opposite. >> >> Regards, >> Patrick >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/a363c987/attachment-0001.html From bdfoster at endigotech.com Tue Mar 20 09:04:11 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 20 Mar 2012 02:04:11 -0400 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: <4F67DB51.20900@puzzled.xs4all.nl> Message-ID: Anton, I'm not really a pro at reading those, but I did post it here: http://da1.endigovoip.com/dump.txt ...I'm not sure if that will tell much. Where it says bdfoster-Precision-WorkStation-490, that is the FS server (192.168.1.79), the ATA is on 192.168.1.80 (Grandstream HT286) This problem has also occurred using a SPA921 and another HT286, all on the same LAN connected to the same server, etc. -BDF 2012/3/20 Anton Kvashenkin > Try to catch RTP streams > > tcpdump -i eth0 portrange 16384-16406 > > and see what happens. > > > 20 ????? 2012 ?. 6:14 ???????????? Brian Foster ???????: > > Yea I know what you're talking about. That has indeed been switched off on >> the RG. For reference purposes, the Residential Gateway is a 2wire >> AT&T-branded Uverse DSL modem/wireless router with a 4 port switch. I've >> also got two 8 port netgear prosafe switches. Phones and server are on the >> same subnet/netmask. Very simple network setup. No VLANS, etc. This was >> indeed working with Asterisk, so I'm almost positive it's a configuration >> issue with FreeSWITCH. >> >> If there's a need to look at more than the files/logs I've pb'd, please >> let be know! >> >> -BDF >> On Mar 19, 2012 9:20 PM, "Patrick Lists" < >> freeswitch-list at puzzled.xs4all.nl> wrote: >> >>> On 20-03-12 00:10, Brian Foster wrote: >>> > UPnP isn't available on this network. Ports 16384-16406 are forwarded >>> to >>> > the machine from the router. Port range in switch.conf.xml reflects >>> > that. Also 5080 is forwarded from the router to the server. >>> > >>> > Both of those variables (ext-sip-ip and ext-rtp-ip) are set to the stun >>> > server, see pastebin 18699 (third link in original email). >>> >>> If the (ADSL) router has a (for lack of a better term cause I can't >>> remember) "SIP helper" app, try disabling it. Similar to the Cisco "SMTP >>> Fixup" functionality which does an excellent job at the opposite. >>> >>> Regards, >>> Patrick >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/c44693cf/attachment.html From avi at avimarcus.net Tue Mar 20 09:41:28 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 20 Mar 2012 08:41:28 +0200 Subject: [Freeswitch-users] FreeSWITCH panel - web utility update In-Reply-To: <4F67CB31.2010208@livecall.com> References: <4F67CB31.2010208@livecall.com> Message-ID: Jack: try setting in the configuration file: $disableXML = true; -Avi On Tue, Mar 20, 2012 at 2:11 AM, Jack wrote: > I have installed on windows ii6 server and when I launch the webpage I > get an alert that says "Please debug or disable XML!" > Any idea what I need? > Thanks in advance.. > Jack -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/ee832a2b/attachment.html From singhai.piyush at gmail.com Tue Mar 20 11:16:29 2012 From: singhai.piyush at gmail.com (piyush singhai) Date: Tue, 20 Mar 2012 13:46:29 +0530 Subject: [Freeswitch-users] How can I change the moh_sound Message-ID: Hello, I want to change the moh sound at run time for conference. can i specify path on the basis of any ivr. --Piyush -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/e645a403/attachment.html From andrew at cassidywebservices.co.uk Tue Mar 20 12:53:35 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 20 Mar 2012 09:53:35 +0000 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: <4F67DB51.20900@puzzled.xs4all.nl> Message-ID: ---------- Forwarded message ---------- From: "Andrew Cassidy" Date: Mar 20, 2012 8:55 AM Subject: Re: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds To: "FreeSWITCH Users Help" When we get a solution (as I'm sure we will) Can we get a guide of things to try put up on the wiki? I've seen the same question asked at least twice already this week. I'll find the command line to do a more detailed packet dump that can be loaded into wireshark for further analysis. The main thing I'm wondering is if Nat is doing something funny with rewriting the rtp and rtcp port numbers. What type of NAT do you have? You can use WinStun or another basic stun client to find out. On Mar 20, 2012 6:07 AM, "Brian Foster" wrote: > Anton, > > I'm not really a pro at reading those, but I did post it here: > > http://da1.endigovoip.com/dump.txt > > ...I'm not sure if that will tell much. Where it says > bdfoster-Precision-WorkStation-490, that is the FS server (192.168.1.79), > the ATA is on 192.168.1.80 (Grandstream HT286) This problem has also > occurred using a SPA921 and another HT286, all on the same LAN connected to > the same server, etc. > > -BDF > > 2012/3/20 Anton Kvashenkin > >> Try to catch RTP streams >> >> tcpdump -i eth0 portrange 16384-16406 >> >> and see what happens. >> >> >> 20 ????? 2012 ?. 6:14 ???????????? Brian Foster ???????: >> >> Yea I know what you're talking about. That has indeed been switched off >>> on the RG. For reference purposes, the Residential Gateway is a 2wire >>> AT&T-branded Uverse DSL modem/wireless router with a 4 port switch. I've >>> also got two 8 port netgear prosafe switches. Phones and server are on the >>> same subnet/netmask. Very simple network setup. No VLANS, etc. This was >>> indeed working with Asterisk, so I'm almost positive it's a configuration >>> issue with FreeSWITCH. >>> >>> If there's a need to look at more than the files/logs I've pb'd, please >>> let be know! >>> >>> -BDF >>> On Mar 19, 2012 9:20 PM, "Patrick Lists" < >>> freeswitch-list at puzzled.xs4all.nl> wrote: >>> >>>> On 20-03-12 00:10, Brian Foster wrote: >>>> > UPnP isn't available on this network. Ports 16384-16406 are forwarded >>>> to >>>> > the machine from the router. Port range in switch.conf.xml reflects >>>> > that. Also 5080 is forwarded from the router to the server. >>>> > >>>> > Both of those variables (ext-sip-ip and ext-rtp-ip) are set to the >>>> stun >>>> > server, see pastebin 18699 (third link in original email). >>>> >>>> If the (ADSL) router has a (for lack of a better term cause I can't >>>> remember) "SIP helper" app, try disabling it. Similar to the Cisco "SMTP >>>> Fixup" functionality which does an excellent job at the opposite. >>>> >>>> Regards, >>>> Patrick >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/dee43074/attachment-0001.html From andrew at cassidywebservices.co.uk Tue Mar 20 13:32:00 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 20 Mar 2012 10:32:00 +0000 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: <4F67DB51.20900@puzzled.xs4all.nl> Message-ID: tcpdump -i -s 65535 -w tcpdump -s 65535 -w dump.pcap That should do the trick, but you can add port filters if you'd prefer. If you do, please include both sip and rtp traffic. The dump.pcap file will be compatible with wireshark to make analyzing it a little easier. 2012/3/20 Andrew Cassidy > ---------- Forwarded message ---------- > From: "Andrew Cassidy" > Date: Mar 20, 2012 8:55 AM > Subject: Re: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 > seconds > To: "FreeSWITCH Users Help" > > When we get a solution (as I'm sure we will) Can we get a guide of things > to try put up on the wiki? I've seen the same question asked at least twice > already this week. > > I'll find the command line to do a more detailed packet dump that can be > loaded into wireshark for further analysis. The main thing I'm wondering is > if Nat is doing something funny with rewriting the rtp and rtcp port > numbers. > > What type of NAT do you have? You can use WinStun or another basic stun > client to find out. > On Mar 20, 2012 6:07 AM, "Brian Foster" wrote: > >> Anton, >> >> I'm not really a pro at reading those, but I did post it here: >> >> http://da1.endigovoip.com/dump.txt >> >> ...I'm not sure if that will tell much. Where it says >> bdfoster-Precision-WorkStation-490, that is the FS server (192.168.1.79), >> the ATA is on 192.168.1.80 (Grandstream HT286) This problem has also >> occurred using a SPA921 and another HT286, all on the same LAN connected to >> the same server, etc. >> >> -BDF >> >> 2012/3/20 Anton Kvashenkin >> >>> Try to catch RTP streams >>> >>> tcpdump -i eth0 portrange 16384-16406 >>> >>> and see what happens. >>> >>> >>> 20 ????? 2012 ?. 6:14 ???????????? Brian Foster >> > ???????: >>> >>> Yea I know what you're talking about. That has indeed been switched off >>>> on the RG. For reference purposes, the Residential Gateway is a 2wire >>>> AT&T-branded Uverse DSL modem/wireless router with a 4 port switch. I've >>>> also got two 8 port netgear prosafe switches. Phones and server are on the >>>> same subnet/netmask. Very simple network setup. No VLANS, etc. This was >>>> indeed working with Asterisk, so I'm almost positive it's a configuration >>>> issue with FreeSWITCH. >>>> >>>> If there's a need to look at more than the files/logs I've pb'd, please >>>> let be know! >>>> >>>> -BDF >>>> On Mar 19, 2012 9:20 PM, "Patrick Lists" < >>>> freeswitch-list at puzzled.xs4all.nl> wrote: >>>> >>>>> On 20-03-12 00:10, Brian Foster wrote: >>>>> > UPnP isn't available on this network. Ports 16384-16406 are >>>>> forwarded to >>>>> > the machine from the router. Port range in switch.conf.xml reflects >>>>> > that. Also 5080 is forwarded from the router to the server. >>>>> > >>>>> > Both of those variables (ext-sip-ip and ext-rtp-ip) are set to the >>>>> stun >>>>> > server, see pastebin 18699 (third link in original email). >>>>> >>>>> If the (ADSL) router has a (for lack of a better term cause I can't >>>>> remember) "SIP helper" app, try disabling it. Similar to the Cisco >>>>> "SMTP >>>>> Fixup" functionality which does an excellent job at the opposite. >>>>> >>>>> Regards, >>>>> Patrick >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/18dad0dd/attachment.html From bernard.david.murphy at gmail.com Tue Mar 20 14:42:24 2012 From: bernard.david.murphy at gmail.com (bernard.david.murphy at gmail.com) Date: Tue, 20 Mar 2012 11:42:24 +0000 Subject: [Freeswitch-users] SIP Notify for MWI on avaya 9630 In-Reply-To: References: <304277796-1332151706-cardhu_decombobulator_blackberry.rim.net-1910878399-@b13.c12.bise7.blackberry> Message-ID: <1098514731-1332243747-cardhu_decombobulator_blackberry.rim.net-616519276-@b13.c12.bise7.blackberry> I've got various versions from git over the last 3 months and none of them send a sip NOTIFY that is compatible with the Avaya 9630. The only version that DOES work is a vanilla install of 1.0.6 Which part of the freeswitch config controls the format of that sip NOTIFY as a nay be able to figure out how to modify it so it works ? Thanks Sent from my BlackBerry? wireless device -----Original Message----- From: curriegrad2004 Sender: freeswitch-users-bounces at lists.freeswitch.org Date: Mon, 19 Mar 2012 09:17:17 To: FreeSWITCH Users Help Reply-To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP Notify for MWI on avaya 9630 Have you tried checking out the older revisions of freeswitch git heads yet? iirc there were changes regarding to the BLF done a couple of weeks ago On Mon, Mar 19, 2012 at 2:30 AM, wrote: > > > I have an Avaya 9630 with SIP 2.6.6 firmware which will subscribe to freeswitch on ext login and then receives SIP notify messages when a voicemail is left. > > This works fine in freeswitch 1.0.6 but fails to light the light in any newer versions of freeswitch like 1.0 git head. > > In both cases the notify messages are definitely sent to the phone but anything other than 1.0.6 does not light the MWI light. > > Does anyone know of any reason. I will post the SIP messages shortly. > > Does anyone know how to control the format of these notify messages from within freeswitch, I.e. Is it possible in cml file to modify the content? > > Thanks > Bernie > > Sent from my BlackBerry? wireless device > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From lists at telefaks.de Tue Mar 20 15:08:25 2012 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 20 Mar 2012 13:08:25 +0100 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: <1FFF97C269757C458224B7C895F35F150786C4@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F150786C4@cantor.std.visionutv.se> Message-ID: <4F687339.6040703@telefaks.de> Thanks to your help, I am now a step further with 1.22.0 and ptlib-2.8.2: mod_h323 has compiled successfully. After relinking libh323_linux_x86_64_.so.1.22.0 the mod_h323 module finally loaded. Thanks to all, who helped on this issue! Do you know of any gatekeeper avaliable for testing, where I could test connectivity and 2-way-audio? I tried with http://www.voxgratia.org/ but this gateway seems to be down. Best regards Peter Am 19.03.2012 14:33, schrieb Peter Olsson: > Yes, I've used those recommendations - so 1.22.0 is probably a better choice. > > /Peter > > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Patrick Lists > Skickat: den 19 mars 2012 14:03 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Advice how to start with H.323 > > On 19-03-12 11:53, Peter Steinbach wrote: >> I am using Ubuntu 10.04. >> >> I followed the installation procedures on the wiki page and installed >> >> ptlib-2.8.2 + h323plus-trunk > Not sure if it's still valid but in the past the recommendation was to follow the versions listed at: > > http://www.gnugk.org/compiling-gnugk.html > > Which in this case means you should use H323Plus 1.22.0 and not trunk. > > Regards, > Patrick > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f672e0f32766345684320! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From peter.olsson at visionutveckling.se Tue Mar 20 15:32:04 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 20 Mar 2012 12:32:04 +0000 Subject: [Freeswitch-users] Advice how to start with H.323 Message-ID: <1FFF97C269757C458224B7C895F35F15079D1B@cantor.std.visionutv.se> It would be very appreciated if you could take some time and update with some more information on the wiki - so others could benefit from this as well.. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Peter Steinbach Skickat: den 20 mars 2012 13:08 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Advice how to start with H.323 Thanks to your help, I am now a step further with 1.22.0 and ptlib-2.8.2: mod_h323 has compiled successfully. After relinking libh323_linux_x86_64_.so.1.22.0 the mod_h323 module finally loaded. Thanks to all, who helped on this issue! Do you know of any gatekeeper avaliable for testing, where I could test connectivity and 2-way-audio? I tried with http://www.voxgratia.org/ but this gateway seems to be down. Best regards Peter Am 19.03.2012 14:33, schrieb Peter Olsson: > Yes, I've used those recommendations - so 1.22.0 is probably a better choice. > > /Peter > > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Patrick > Lists > Skickat: den 19 mars 2012 14:03 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Advice how to start with H.323 > > On 19-03-12 11:53, Peter Steinbach wrote: >> I am using Ubuntu 10.04. >> >> I followed the installation procedures on the wiki page and installed >> >> ptlib-2.8.2 + h323plus-trunk > Not sure if it's still valid but in the past the recommendation was to follow the versions listed at: > > http://www.gnugk.org/compiling-gnugk.html > > Which in this case means you should use H323Plus 1.22.0 and not trunk. > > Regards, > Patrick > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f68720032761729395285! From tculjaga at gmail.com Tue Mar 20 15:33:52 2012 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 20 Mar 2012 13:33:52 +0100 Subject: [Freeswitch-users] TLS on FS Message-ID: hello, im new on TLS setup... and as usual, im having issues configuring a SIP client (Bria for windows) with FS. I guess i configured FS properly, but im not sure about certificates. FS conf: I created certificates using the commands on the wiki: ./gentls_cert setup -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org ./gentls_cert create_server -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org ./gentls_cert create_client -cn Client1 -out Client1 /usr/local/freeswitch/conf/ssl -rw-r----- 1 root root 3029 Mar 20 08:56 agent.pem drwxr-x--- 2 root root 4096 Mar 20 08:56 CA -rw-r----- 1 root root 1046 Mar 20 08:49 cafile.pem -rw-r----- 1 root root 3029 Mar 20 09:45 Client1 /usr/local/freeswitch/conf/ssl/CA -rw-r----- 1 root root 1046 Mar 20 08:49 cacert.pem -rw-r----- 1 root root 17 Mar 20 09:45 cacert.srl -rw-r----- 1 root root 1679 Mar 20 08:49 cakey.pem -rw-r----- 1 root root 579 Mar 20 08:49 config.tpl i deployed and installed cafile.pem on windows machine running Bria softphone. I did the same with Client1. Restarted but all im getting is this error in console: tport_wakeup_pri(0x85b9590): events IN tport_alloc_secondary(0x85b9590): new secondary tport 0x85ef610 tport_tls_accept(0x85ef610): new connection from tls/ 85.114.34.202:61030/sips tls_connect(0x85ef610): events NEGOTIATING tls_connect(0x85ef610): events NEGOTIATING tls_connect(0x85ef610): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport_close(0x85ef610): tls/85.114.34.202:61030/sips please, can anyone help ? this is a portion showing how FS loads mod_sofia: su_port_create(0x857c060): epoll_create() => 0: OK su_socket_port_init(0x857c060, 0xd9b400) called su_pthread_port_init(0x857c060, 0xd9b400) called su_port_create(0x85b1840): epoll_create() => 0: OK su_socket_port_init(0x85b1840, 0xd9b400) called su_pthread_port_init(0x85b1840, 0xd9b400) called nua: nua_create: entering su_port_create(0x85b3920): epoll_create() => 0: OK su_socket_port_init(0x85b3920, 0xd9b400) called su_pthread_port_init(0x85b3920, 0xd9b400) called nua: nua_stack_init: entering nua: nua_stack_set_params: entering soa_create("default", 0x85b0eb8, 0x85b0f70) called soa_set_params(static::0x85ae410, ...) called soa_set_params(static::0x85ae410, ...) called nta_agent_create: initialized hash tables nta_agent_create: initialized transports nua: nua_create: entering su_port_create(0x85b2368): epoll_create() => 0: OK su_socket_port_init(0x85b2368, 0xd9b400) called su_pthread_port_init(0x85b2368, 0xd9b400) called nua: nua_stack_init: entering nua: nua_stack_set_params: entering soa_create("default", 0x85b2198, 0x85b6440) called soa_set_params(static::0x85b6938, ...) called soa_set_params(static::0x85b6938, ...) called nta_agent_create: initialized hash tables nta_agent_create: initialized transports nta_agent_create: initialized random identifiers nta_agent_create: initialized timer nta_agent_create: initialized random identifiers nta_agent_create: initialized timer nta_agent_create: initialized resolver nta_agent_create: initialized resolver tport_create(): 0x85aefe8 tport_create(): 0xb780f688 nta: master transport created nta: master transport created tport_bind_server(0x85aefe8) to */85.114.35.241:5060/sip tport_bind_server(0xb780f688) to */85.114.35.241:5080/sip tport_bind_server(0xb780f688): calling tport_listen for udp tport_bind_server(0x85aefe8): calling tport_listen for udp tport_alloc_primary(0x85aefe8): new primary tport 0x85af2d0 tport_alloc_primary(0xb780f688): new primary tport 0xb780add0 tport_listen(0xb780add0): listening at udp/85.114.35.241:5080/sip tport_listen(0x85af2d0): listening at udp/85.114.35.241:5060/sip tport_bind_server(0xb780f688): calling tport_listen for tcp tport_bind_server(0x85aefe8): calling tport_listen for tcp tport_alloc_primary(0xb780f688): new primary tport 0xb7811af0 tport_alloc_primary(0x85aefe8): new primary tport 0x85af778 tport_listen(0x85af778): listening at tcp/85.114.35.241:5060/sip tport_listen(0xb7811af0): listening at tcp/85.114.35.241:5080/sip nta: bound to (85.114.35.241:5080;transport=*) nta: bound to (85.114.35.241:5060;transport=*) nta: agent_init_via: SIP/2.0/udp 85.114.35.241:5080 (sip) nta: agent_init_via: SIP/2.0/udp 85.114.35.241 (sip) nta: agent_init_via: SIP/2.0/tcp 85.114.35.241 (sip) nta: agent_init_via: SIP/2.0/tcp 85.114.35.241:5080 (sip) nta: Via fields initialized nta: Via fields initialized nta: Contact header created nta: Contact header created tport_bind_server(0x85aefe8) to tls/85.114.35.241:5061/sips tport_bind_server(0xb780f688) to tls/85.114.35.241:5081/sips tport_bind_server(0xb780f688): calling tport_listen for tls tport_bind_server(0x85aefe8): calling tport_listen for tls tport_alloc_primary(0xb780f688): new primary tport 0xb780e378 tport_alloc_primary(0x85aefe8): new primary tport 0x85b9590 tport_tls_init_master(0x85b9590): tls key = /usr/local/freeswitch/conf/ssl/agent.pem tport_tls_init_master(0xb780e378): tls key = /usr/local/freeswitch/conf/ssl/agent.pem tport_tls_init_master(0xb780e378): tls context initialized for [85.114.35.241]:5081 tport_tls_init_master(0x85b9590): tls context initialized for [85.114.35.241]:5061 tport_listen(0x85b9590): listening at tls/85.114.35.241:5061/sips tport_listen(0xb780e378): listening at tls/85.114.35.241:5081/sips nta: bound to (85.114.35.241:5061;transport=tls) nta: bound to (85.114.35.241:5081;transport=tls) nta: agent_init_via: SIP/2.0/udp 85.114.35.241 (sip) nta: agent_init_via: SIP/2.0/udp 85.114.35.241:5080 (sip) nta: agent_init_via: SIP/2.0/tcp 85.114.35.241 (sip) nta: agent_init_via: SIP/2.0/tcp 85.114.35.241:5080 (sip) nta: agent_init_via: SIP/2.0/tls 85.114.35.241 (sips) nta: agent_init_via: SIP/2.0/tls 85.114.35.241:5081 (sips) nta: Via fields initialized nta: Via fields initialized nta: Contact header created nta: Contact header created nua_register: Adding contact URL '85.114.35.241' to list. nua_register: Adding contact URL '85.114.35.241' to list. nua_register: Adding contact URL '85.114.35.241' to list. nua_register: Adding contact URL '85.114.35.241' to list. nua: nua_set_params: entering nua: nua_set_params: entering nua((nil)): sent signal r_set_params nua((nil)): sent signal r_set_params nua: nua_stack_set_params: entering soa_set_params(static::0x85b6938, ...) called 2012-03-20 10:57:48.859351 [NOTICE] sofia_reg.c:2969 Added gateway ' example.com' to profile 'external' nua: nua_stack_set_params: entering soa_set_params(static::0x85ae410, ...) called 2012-03-20 10:57:48.859745 [NOTICE] sofia.c:2710 Adding Alias [85.114.35.241] for profile [internal] tport_wakeup_pri(0x85b9590): events IN tport_alloc_secondary(0x85b9590): new secondary tport 0x85bae68 tport_tls_accept(0x85bae68): new connection from tls/ 85.114.34.202:60916/sips tls_connect(0x85bae68): events NEGOTIATING tls_connect(0x85bae68): events NEGOTIATING tls_connect(0x85bae68): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport_close(0x85bae68): tls/85.114.34.202:60916/sips nua: nua_application_event: entering nua: nua_application_event: entering 2012-03-20 10:57:49.959654 [CONSOLE] sofia.c:1214 MSG Thread Started nua: nua_handle_magic: entering nua: nua_handle_magic: entering 2012-03-20 10:57:50.360893 [CONSOLE] switch_loadable_module.c:1299 Successfully Loaded [mod_sofia] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/418605c7/attachment.html From garbytrash at gmail.com Tue Mar 20 15:35:50 2012 From: garbytrash at gmail.com (Zenny) Date: Tue, 20 Mar 2012 12:35:50 +0000 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv Message-ID: Hi: I tried to build FS from git and it halts with the following error. Could not figure out what could be wrong? Thanks in advance for any hints or imputs to overcome! [QUOTE] .... making all mod_cdr_pg_csv Compiling /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c... quiet_libtool: compile: gcc -I/usr/include/postgresql -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c -fPIC -DPIC -o .libs/mod_cdr_pg_csv.o /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:40:22: error: libpq-fe.h: No such file or directory /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:83: error: expected specifier-qualifier-list before ?PGconn? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function ?config_validate_spool_dir?: /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:110: error: ?struct ? has no member named ?spool_dir? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: At top level: /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:121: error: ?struct ? has no member named ?legs? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:122: error: ?struct ? has no member named ?spool_format? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:123: error: ?struct ? has no member named ?rotate? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:124: error: ?struct ? has no member named ?debug? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:127: error: ?struct ? has no member named ?spool_dir? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function ?do_rotate?: /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:167: error: ?struct ? has no member named ?rotate? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:187: error: ?struct ? has no member named ?rotate? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function ?insert_cdr?: /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: error: ?PGresult? undeclared (first use in this function) /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: error: (Each undeclared identifier is reported only once /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: error: for each function it appears in.) /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: error: ?res? undeclared (first use in this function) /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: error: invalid operands to binary * (have ?struct switch_xml_config_item_t *? and ?struct switch_xml_config_item_t *?) cc1: warnings being treated as errors /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: error: statement with no effect /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:255: error: ?struct ? has no member named ?debug? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:259: error: ?struct ? has no member named ?db_mutex? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:259: error: passing argument 1 of ?switch_mutex_lock? from incompatible pointer type /usr/src/freeswitch/src/include/switch_apr.h:400: note: expected ?struct switch_mutex_t *? but argument is of type ?struct switch_xml_config_item_t *? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: error: ?struct ? has no member named ?db_online? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: error: implicit declaration of function ?PQstatus? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: error: ?struct ? has no member named ?db_connection? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: error: ?CONNECTION_OK? undeclared (first use in this function) /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: error: comparison between pointer and integer /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:262: error: ?struct ? has no member named ?db_connection? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:262: error: implicit declaration of function ?PQconnectdb? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:262: error: statement with no effect /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:265: error: ?struct ? has no member named ?db_connection? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:265: error: comparison between pointer and integer /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:266: error: ?struct ? has no member named ?db_online? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:266: error: statement with no effect /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:268: error: implicit declaration of function ?PQerrorMessage? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:268: error: ?struct ? has no member named ?db_connection? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:268: error: format ?%s? expects type ?char *?, but argument 8 has type ?int? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:272: error: implicit declaration of function ?PQexec? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:272: error: ?struct ? has no member named ?db_connection? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:272: error: statement with no effect /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:273: error: implicit declaration of function ?PQresultStatus? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:273: error: ?PGRES_COMMAND_OK? undeclared (first use in this function) /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:273: error: comparison between pointer and integer /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:274: error: implicit declaration of function ?PQresultErrorMessage? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:274: error: format ?%s? expects type ?char *?, but argument 8 has type ?int? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:275: error: implicit declaration of function ?PQclear? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:282: error: ?struct ? has no member named ?db_mutex? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:282: error: passing argument 1 of ?switch_mutex_unlock? from incompatible pointer type /usr/src/freeswitch/src/include/switch_apr.h:406: note: expected ?struct switch_mutex_t *? but argument is of type ?struct switch_xml_config_item_t *? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:289: error: implicit declaration of function ?PQfinish? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:289: error: ?struct ? has no member named ?db_connection? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:290: error: ?struct ? has no member named ?db_online? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:290: error: statement with no effect /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:291: error: ?struct ? has no member named ?db_mutex? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:291: error: passing argument 1 of ?switch_mutex_unlock? from incompatible pointer type /usr/src/freeswitch/src/include/switch_apr.h:406: note: expected ?struct switch_mutex_t *? but argument is of type ?struct switch_xml_config_item_t *? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:294: error: ?struct ? has no member named ?spool_format? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:294: error: comparison between pointer and integer /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:295: error: ?struct ? has no member named ?spool_dir? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:299: error: ?struct ? has no member named ?spool_dir? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function ?my_on_reporting?: /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:323: error: ?struct ? has no member named ?legs? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:323: error: invalid operands to binary & (have ?struct switch_xml_config_item_t *? and ?int?) /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:323: error: ?struct ? has no member named ?legs? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:323: error: invalid operands to binary & (have ?struct switch_xml_config_item_t *? and ?int?) /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:324: error: ?struct ? has no member named ?legs? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:324: error: invalid operands to binary & (have ?struct switch_xml_config_item_t *? and ?int?) /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:335: error: ?struct ? has no member named ?spool_dir? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:335: error: passing argument 1 of ?switch_dir_make_recursive? from incompatible pointer type /usr/src/freeswitch/src/include/switch_apr.h:947: note: expected ?const char *? but argument is of type ?struct switch_xml_config_item_t *? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:336: error: ?struct ? has no member named ?spool_dir? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:336: error: format ?%s? expects type ?char *?, but argument 8 has type ?struct switch_xml_config_item_t *? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:340: error: ?struct ? has no member named ?debug? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:361: error: implicit declaration of function ?PQescapeString? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function ?event_handler?: /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:411: error: ?struct ? has no member named ?db_online? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:412: error: ?struct ? has no member named ?db_connection? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:413: error: ?struct ? has no member named ?db_online? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:413: error: statement with no effect /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function ?load_config?: /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:444: error: ?struct ? has no member named ?db_online? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:445: error: ?struct ? has no member named ?db_connection? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:446: error: ?struct ? has no member named ?db_mutex? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:446: error: passing argument 1 of ?switch_mutex_destroy? from incompatible pointer type /usr/src/freeswitch/src/include/switch_apr.h:393: note: expected ?struct switch_mutex_t *? but argument is of type ?struct switch_xml_config_item_t *? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:447: error: ?struct ? has no member named ?db_online? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:447: error: statement with no effect /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:452: error: ?struct ? has no member named ?db_mutex? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:452: error: passing argument 1 of ?switch_mutex_init? from incompatible pointer type /usr/src/freeswitch/src/include/switch_apr.h:386: note: expected ?struct switch_mutex_t **? but argument is of type ?struct switch_xml_config_item_t (*)[1]? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function ?mod_cdr_pg_csv_load?: /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:528: error: ?struct ? has no member named ?spool_dir? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:528: error: passing argument 1 of ?switch_dir_make_recursive? from incompatible pointer type /usr/src/freeswitch/src/include/switch_apr.h:947: note: expected ?const char *? but argument is of type ?struct switch_xml_config_item_t *? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:529: error: ?struct ? has no member named ?spool_dir? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:529: error: format ?%s? expects type ?char *?, but argument 8 has type ?struct switch_xml_config_item_t *? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function ?mod_cdr_pg_csv_shutdown?: /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:550: error: ?struct ? has no member named ?db_online? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:551: error: ?struct ? has no member named ?db_connection? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:552: error: ?struct ? has no member named ?db_online? /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:552: error: statement with no effect make[5]: *** [mod_cdr_pg_csv.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_cdr_pg_csv-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 [/QUOTE] zenny From peter.olsson at visionutveckling.se Tue Mar 20 15:46:34 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 20 Mar 2012 12:46:34 +0000 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv Message-ID: <1FFF97C269757C458224B7C895F35F15079D6F@cantor.std.visionutv.se> Seems like you're missing some PostgreSQL files; error: libpq-fe.h: No such file or directory /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Zenny Skickat: den 20 mars 2012 13:36 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv Hi: I tried to build FS from git and it halts with the following error. Could not figure out what could be wrong? Thanks in advance for any hints or imputs to overcome! [QUOTE] .... making all mod_cdr_pg_csv Compiling /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c... quiet_libtool: compile: gcc -I/usr/include/postgresql -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c -fPIC -DPIC -o .libs/mod_cdr_pg_csv.o /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:40:22: error: libpq-fe.h: No such file or directory /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:83: error: expected specifier-qualifier-list before 'PGconn' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function 'config_validate_spool_dir': /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:110: error: 'struct ' has no member named 'spool_dir' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: At top level: /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:121: error: 'struct ' has no member named 'legs' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:122: error: 'struct ' has no member named 'spool_format' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:123: error: 'struct ' has no member named 'rotate' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:124: error: 'struct ' has no member named 'debug' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:127: error: 'struct ' has no member named 'spool_dir' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function 'do_rotate': /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:167: error: 'struct ' has no member named 'rotate' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:187: error: 'struct ' has no member named 'rotate' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function 'insert_cdr': /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: error: 'PGresult' undeclared (first use in this function) /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: error: (Each undeclared identifier is reported only once /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: error: for each function it appears in.) /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: error: 'res' undeclared (first use in this function) /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: error: invalid operands to binary * (have 'struct switch_xml_config_item_t *' and 'struct switch_xml_config_item_t *') cc1: warnings being treated as errors /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: error: statement with no effect /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:255: error: 'struct ' has no member named 'debug' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:259: error: 'struct ' has no member named 'db_mutex' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:259: error: passing argument 1 of 'switch_mutex_lock' from incompatible pointer type /usr/src/freeswitch/src/include/switch_apr.h:400: note: expected 'struct switch_mutex_t *' but argument is of type 'struct switch_xml_config_item_t *' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: error: 'struct ' has no member named 'db_online' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: error: implicit declaration of function 'PQstatus' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: error: 'struct ' has no member named 'db_connection' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: error: 'CONNECTION_OK' undeclared (first use in this function) /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: error: comparison between pointer and integer /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:262: error: 'struct ' has no member named 'db_connection' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:262: error: implicit declaration of function 'PQconnectdb' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:262: error: statement with no effect /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:265: error: 'struct ' has no member named 'db_connection' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:265: error: comparison between pointer and integer /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:266: error: 'struct ' has no member named 'db_online' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:266: error: statement with no effect /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:268: error: implicit declaration of function 'PQerrorMessage' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:268: error: 'struct ' has no member named 'db_connection' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:268: error: format '%s' expects type 'char *', but argument 8 has type 'int' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:272: error: implicit declaration of function 'PQexec' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:272: error: 'struct ' has no member named 'db_connection' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:272: error: statement with no effect /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:273: error: implicit declaration of function 'PQresultStatus' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:273: error: 'PGRES_COMMAND_OK' undeclared (first use in this function) /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:273: error: comparison between pointer and integer /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:274: error: implicit declaration of function 'PQresultErrorMessage' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:274: error: format '%s' expects type 'char *', but argument 8 has type 'int' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:275: error: implicit declaration of function 'PQclear' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:282: error: 'struct ' has no member named 'db_mutex' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:282: error: passing argument 1 of 'switch_mutex_unlock' from incompatible pointer type /usr/src/freeswitch/src/include/switch_apr.h:406: note: expected 'struct switch_mutex_t *' but argument is of type 'struct switch_xml_config_item_t *' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:289: error: implicit declaration of function 'PQfinish' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:289: error: 'struct ' has no member named 'db_connection' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:290: error: 'struct ' has no member named 'db_online' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:290: error: statement with no effect /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:291: error: 'struct ' has no member named 'db_mutex' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:291: error: passing argument 1 of 'switch_mutex_unlock' from incompatible pointer type /usr/src/freeswitch/src/include/switch_apr.h:406: note: expected 'struct switch_mutex_t *' but argument is of type 'struct switch_xml_config_item_t *' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:294: error: 'struct ' has no member named 'spool_format' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:294: error: comparison between pointer and integer /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:295: error: 'struct ' has no member named 'spool_dir' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:299: error: 'struct ' has no member named 'spool_dir' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function 'my_on_reporting': /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:323: error: 'struct ' has no member named 'legs' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:323: error: invalid operands to binary & (have 'struct switch_xml_config_item_t *' and 'int') /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:323: error: 'struct ' has no member named 'legs' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:323: error: invalid operands to binary & (have 'struct switch_xml_config_item_t *' and 'int') /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:324: error: 'struct ' has no member named 'legs' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:324: error: invalid operands to binary & (have 'struct switch_xml_config_item_t *' and 'int') /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:335: error: 'struct ' has no member named 'spool_dir' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:335: error: passing argument 1 of 'switch_dir_make_recursive' from incompatible pointer type /usr/src/freeswitch/src/include/switch_apr.h:947: note: expected 'const char *' but argument is of type 'struct switch_xml_config_item_t *' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:336: error: 'struct ' has no member named 'spool_dir' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:336: error: format '%s' expects type 'char *', but argument 8 has type 'struct switch_xml_config_item_t *' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:340: error: 'struct ' has no member named 'debug' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:361: error: implicit declaration of function 'PQescapeString' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function 'event_handler': /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:411: error: 'struct ' has no member named 'db_online' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:412: error: 'struct ' has no member named 'db_connection' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:413: error: 'struct ' has no member named 'db_online' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:413: error: statement with no effect /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function 'load_config': /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:444: error: 'struct ' has no member named 'db_online' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:445: error: 'struct ' has no member named 'db_connection' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:446: error: 'struct ' has no member named 'db_mutex' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:446: error: passing argument 1 of 'switch_mutex_destroy' from incompatible pointer type /usr/src/freeswitch/src/include/switch_apr.h:393: note: expected 'struct switch_mutex_t *' but argument is of type 'struct switch_xml_config_item_t *' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:447: error: 'struct ' has no member named 'db_online' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:447: error: statement with no effect /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:452: error: 'struct ' has no member named 'db_mutex' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:452: error: passing argument 1 of 'switch_mutex_init' from incompatible pointer type /usr/src/freeswitch/src/include/switch_apr.h:386: note: expected 'struct switch_mutex_t **' but argument is of type 'struct switch_xml_config_item_t (*)[1]' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function 'mod_cdr_pg_csv_load': /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:528: error: 'struct ' has no member named 'spool_dir' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:528: error: passing argument 1 of 'switch_dir_make_recursive' from incompatible pointer type /usr/src/freeswitch/src/include/switch_apr.h:947: note: expected 'const char *' but argument is of type 'struct switch_xml_config_item_t *' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:529: error: 'struct ' has no member named 'spool_dir' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:529: error: format '%s' expects type 'char *', but argument 8 has type 'struct switch_xml_config_item_t *' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: In function 'mod_cdr_pg_csv_shutdown': /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:550: error: 'struct ' has no member named 'db_online' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:551: error: 'struct ' has no member named 'db_connection' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:552: error: 'struct ' has no member named 'db_online' /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:552: error: statement with no effect make[5]: *** [mod_cdr_pg_csv.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_cdr_pg_csv-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 [/QUOTE] zenny _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f68788832761165663031! From miha at softnet.si Tue Mar 20 15:51:39 2012 From: miha at softnet.si (Miha) Date: Tue, 20 Mar 2012 13:51:39 +0100 Subject: [Freeswitch-users] rad_auth faild to load modul Message-ID: <4F687D5B.7050200@softnet.si> Hi, I have installed Freeswitch (FreeSWITCH Version 1.0.head (git-9d3401e 2012-03-19 20-06-36 -0500)) on latest Centos (6.2). After I installed freeradius-client as is written on wiki, I tried to load it. I am getting this error: freeswitch at localhost.localdomain> load mod_rad_auth 2012-03-20 13:46:03.386561 [INFO] mod_enum.c:812 ENUM Reloaded 2012-03-20 13:46:03.386561 [INFO] switch_time.c:1128 Timezone reloaded 530 definitions 2012-03-20 13:46:03.386561 [CRIT] switch_loadable_module.c:1295 Error Loading module /usr/local/freeswitch/mod/mod_rad_auth.so **/usr/local/freeswitch/mod/mod_rad_auth.so: undefined symbol: rc_conf_str** I am having rad_auth installed on FS (FreeSWITCH Version 1.0.head (git-00de8e6 2011-11-01 17-27-13 -0600)), (Centos 5.6) and is working properly. Any suggestion how what could be causing problem that I can not load it on centos 6.2? Thank you! Regards, MIha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/63668561/attachment.html From garbytrash at gmail.com Tue Mar 20 16:12:20 2012 From: garbytrash at gmail.com (Zenny) Date: Tue, 20 Mar 2012 13:12:20 +0000 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: <1FFF97C269757C458224B7C895F35F15079D6F@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F15079D6F@cantor.std.visionutv.se> Message-ID: Thanks Peter. But when I checked I have: # find / -name libpq-fe.h /usr/pgsql-9.1/include/libpq-fe.h The file is there. It was compiled from the source in centos6. On 3/20/12, Peter Olsson wrote: > Seems like you're missing some PostgreSQL files; > > error: libpq-fe.h: No such file or directory > > /Peter > > > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Zenny > Skickat: den 20 mars 2012 13:36 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] FS compilation errored out at building > mod_cdr_pg_csv > > Hi: > > I tried to build FS from git and it halts with the following error. > Could not figure out what could be wrong? Thanks in advance for any hints or > imputs to overcome! > > [QUOTE] > .... > making all mod_cdr_pg_csv > Compiling > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c... > quiet_libtool: compile: gcc -I/usr/include/postgresql > -I/usr/src/freeswitch/libs/curl/include > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 > -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE > -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c > -fPIC -DPIC -o .libs/mod_cdr_pg_csv.o > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:40:22: > error: libpq-fe.h: No such file or directory > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:83: > error: expected specifier-qualifier-list before 'PGconn' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: > In function 'config_validate_spool_dir': > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:110: > error: 'struct ' has no member named 'spool_dir' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: > At top level: > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:121: > error: 'struct ' has no member named 'legs' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:122: > error: 'struct ' has no member named 'spool_format' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:123: > error: 'struct ' has no member named 'rotate' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:124: > error: 'struct ' has no member named 'debug' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:127: > error: 'struct ' has no member named 'spool_dir' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: > In function 'do_rotate': > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:167: > error: 'struct ' has no member named 'rotate' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:187: > error: 'struct ' has no member named 'rotate' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: > In function 'insert_cdr': > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: > error: 'PGresult' undeclared (first use in this function) > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: > error: (Each undeclared identifier is reported only once > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: > error: for each function it appears in.) > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: > error: 'res' undeclared (first use in this function) > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: > error: invalid operands to binary * (have 'struct switch_xml_config_item_t > *' and 'struct switch_xml_config_item_t *') > cc1: warnings being treated as errors > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:250: > error: statement with no effect > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:255: > error: 'struct ' has no member named 'debug' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:259: > error: 'struct ' has no member named 'db_mutex' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:259: > error: passing argument 1 of 'switch_mutex_lock' from incompatible pointer > type > /usr/src/freeswitch/src/include/switch_apr.h:400: note: expected 'struct > switch_mutex_t *' but argument is of type 'struct switch_xml_config_item_t > *' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: > error: 'struct ' has no member named 'db_online' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: > error: implicit declaration of function 'PQstatus' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: > error: 'struct ' has no member named 'db_connection' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: > error: 'CONNECTION_OK' undeclared (first use in this function) > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:261: > error: comparison between pointer and integer > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:262: > error: 'struct ' has no member named 'db_connection' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:262: > error: implicit declaration of function 'PQconnectdb' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:262: > error: statement with no effect > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:265: > error: 'struct ' has no member named 'db_connection' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:265: > error: comparison between pointer and integer > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:266: > error: 'struct ' has no member named 'db_online' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:266: > error: statement with no effect > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:268: > error: implicit declaration of function 'PQerrorMessage' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:268: > error: 'struct ' has no member named 'db_connection' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:268: > error: format '%s' expects type 'char *', but argument 8 has type 'int' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:272: > error: implicit declaration of function 'PQexec' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:272: > error: 'struct ' has no member named 'db_connection' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:272: > error: statement with no effect > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:273: > error: implicit declaration of function 'PQresultStatus' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:273: > error: 'PGRES_COMMAND_OK' undeclared (first use in this function) > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:273: > error: comparison between pointer and integer > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:274: > error: implicit declaration of function 'PQresultErrorMessage' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:274: > error: format '%s' expects type 'char *', but argument 8 has type 'int' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:275: > error: implicit declaration of function 'PQclear' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:282: > error: 'struct ' has no member named 'db_mutex' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:282: > error: passing argument 1 of 'switch_mutex_unlock' from incompatible pointer > type > /usr/src/freeswitch/src/include/switch_apr.h:406: note: expected 'struct > switch_mutex_t *' but argument is of type 'struct switch_xml_config_item_t > *' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:289: > error: implicit declaration of function 'PQfinish' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:289: > error: 'struct ' has no member named 'db_connection' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:290: > error: 'struct ' has no member named 'db_online' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:290: > error: statement with no effect > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:291: > error: 'struct ' has no member named 'db_mutex' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:291: > error: passing argument 1 of 'switch_mutex_unlock' from incompatible pointer > type > /usr/src/freeswitch/src/include/switch_apr.h:406: note: expected 'struct > switch_mutex_t *' but argument is of type 'struct switch_xml_config_item_t > *' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:294: > error: 'struct ' has no member named 'spool_format' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:294: > error: comparison between pointer and integer > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:295: > error: 'struct ' has no member named 'spool_dir' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:299: > error: 'struct ' has no member named 'spool_dir' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: > In function 'my_on_reporting': > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:323: > error: 'struct ' has no member named 'legs' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:323: > error: invalid operands to binary & (have 'struct switch_xml_config_item_t > *' and 'int') > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:323: > error: 'struct ' has no member named 'legs' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:323: > error: invalid operands to binary & (have 'struct switch_xml_config_item_t > *' and 'int') > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:324: > error: 'struct ' has no member named 'legs' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:324: > error: invalid operands to binary & (have 'struct switch_xml_config_item_t > *' and 'int') > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:335: > error: 'struct ' has no member named 'spool_dir' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:335: > error: passing argument 1 of 'switch_dir_make_recursive' from incompatible > pointer type > /usr/src/freeswitch/src/include/switch_apr.h:947: note: expected 'const char > *' but argument is of type 'struct switch_xml_config_item_t *' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:336: > error: 'struct ' has no member named 'spool_dir' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:336: > error: format '%s' expects type 'char *', but argument 8 has type 'struct > switch_xml_config_item_t *' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:340: > error: 'struct ' has no member named 'debug' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:361: > error: implicit declaration of function 'PQescapeString' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: > In function 'event_handler': > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:411: > error: 'struct ' has no member named 'db_online' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:412: > error: 'struct ' has no member named 'db_connection' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:413: > error: 'struct ' has no member named 'db_online' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:413: > error: statement with no effect > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: > In function 'load_config': > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:444: > error: 'struct ' has no member named 'db_online' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:445: > error: 'struct ' has no member named 'db_connection' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:446: > error: 'struct ' has no member named 'db_mutex' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:446: > error: passing argument 1 of 'switch_mutex_destroy' from incompatible > pointer type > /usr/src/freeswitch/src/include/switch_apr.h:393: note: expected 'struct > switch_mutex_t *' but argument is of type 'struct switch_xml_config_item_t > *' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:447: > error: 'struct ' has no member named 'db_online' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:447: > error: statement with no effect > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:452: > error: 'struct ' has no member named 'db_mutex' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:452: > error: passing argument 1 of 'switch_mutex_init' from incompatible pointer > type > /usr/src/freeswitch/src/include/switch_apr.h:386: note: expected 'struct > switch_mutex_t **' but argument is of type 'struct switch_xml_config_item_t > (*)[1]' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: > In function 'mod_cdr_pg_csv_load': > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:528: > error: 'struct ' has no member named 'spool_dir' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:528: > error: passing argument 1 of 'switch_dir_make_recursive' from incompatible > pointer type > /usr/src/freeswitch/src/include/switch_apr.h:947: note: expected 'const char > *' but argument is of type 'struct switch_xml_config_item_t *' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:529: > error: 'struct ' has no member named 'spool_dir' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:529: > error: format '%s' expects type 'char *', but argument 8 has type 'struct > switch_xml_config_item_t *' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c: > In function 'mod_cdr_pg_csv_shutdown': > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:550: > error: 'struct ' has no member named 'db_online' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:551: > error: 'struct ' has no member named 'db_connection' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:552: > error: 'struct ' has no member named 'db_online' > /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:552: > error: statement with no effect > make[5]: *** [mod_cdr_pg_csv.lo] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_cdr_pg_csv-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > [/QUOTE] > > zenny > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f68788832761165663031! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From lazyvirus at gmx.com Tue Mar 20 16:25:06 2012 From: lazyvirus at gmx.com (Bzzz) Date: Tue, 20 Mar 2012 14:25:06 +0100 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15079D6F@cantor.std.visionutv.se> Message-ID: <20120320142506.12c6eebe@anubis.defcon1> On Tue, 20 Mar 2012 13:12:20 +0000 Zenny wrote: > But when I checked I have: > > # find / -name libpq-fe.h > /usr/pgsql-9.1/include/libpq-fe.h > > The file is there. It was compiled from the source in centos6. As it was compiled, there's chances that this lib isn't accounted in the system; did you run ldconfig after installation? JY -- From garbytrash at gmail.com Tue Mar 20 16:28:39 2012 From: garbytrash at gmail.com (Zenny) Date: Tue, 20 Mar 2012 13:28:39 +0000 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: <20120320142506.12c6eebe@anubis.defcon1> References: <1FFF97C269757C458224B7C895F35F15079D6F@cantor.std.visionutv.se> <20120320142506.12c6eebe@anubis.defcon1> Message-ID: Yes, I did ldconfig after the installation. On 3/20/12, Bzzz wrote: > On Tue, 20 Mar 2012 13:12:20 +0000 > Zenny wrote: > >> But when I checked I have: >> >> # find / -name libpq-fe.h >> /usr/pgsql-9.1/include/libpq-fe.h >> >> The file is there. It was compiled from the source in centos6. > > As it was compiled, there's chances that this lib isn't accounted > in the system; did you run ldconfig after installation? > > JY > -- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From lazyvirus at gmx.com Tue Mar 20 16:42:09 2012 From: lazyvirus at gmx.com (Bzzz) Date: Tue, 20 Mar 2012 14:42:09 +0100 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15079D6F@cantor.std.visionutv.se> <20120320142506.12c6eebe@anubis.defcon1> Message-ID: <20120320144209.598d3940@anubis.defcon1> On Tue, 20 Mar 2012 13:28:39 +0000 Zenny wrote: > Yes, I did ldconfig after the installation. >> But when I checked I have: >> # find / -name libpq-fe.h >> /usr/pgsql-9.1/include/libpq-fe.h That don't look like a regular path (such as: /usr/include/.....) make a symlink: ln -s /usr/pgsql-9.1/ /usr/include/ re-ldconfig and retry. -- Love is the process of my leading you gently back to yourself. -- Saint Exupery From krice at freeswitch.org Tue Mar 20 16:57:13 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 20 Mar 2012 08:57:13 -0500 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: Message-ID: Check to see how those libs are getting specified in the Makefile for that module... You'll notice that the paths are hardcoded... This could really use some pg_config love K On 3/20/12 8:28 AM, "Zenny" wrote: > Yes, I did ldconfig after the installation. > > On 3/20/12, Bzzz wrote: >> On Tue, 20 Mar 2012 13:12:20 +0000 >> Zenny wrote: >> >>> But when I checked I have: >>> >>> # find / -name libpq-fe.h >>> /usr/pgsql-9.1/include/libpq-fe.h >>> >>> The file is there. It was compiled from the source in centos6. >> >> As it was compiled, there's chances that this lib isn't accounted >> in the system; did you run ldconfig after installation? >> >> JY >> -- >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at freeswitch.org Tue Mar 20 16:57:56 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 20 Mar 2012 08:57:56 -0500 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: <20120320144209.598d3940@anubis.defcon1> Message-ID: That's not a good way to solve the problem... The problem after a quick look is hardcoded paths for pgsql... On 3/20/12 8:42 AM, "Bzzz" wrote: > On Tue, 20 Mar 2012 13:28:39 +0000 > Zenny wrote: > >> Yes, I did ldconfig after the installation. > >>> But when I checked I have: >>> # find / -name libpq-fe.h >>> /usr/pgsql-9.1/include/libpq-fe.h > > That don't look like a regular path (such as: /usr/include/.....) > make a symlink: > ln -s /usr/pgsql-9.1/ /usr/include/ > re-ldconfig > and retry. From lazyvirus at gmx.com Tue Mar 20 17:13:04 2012 From: lazyvirus at gmx.com (Bzzz) Date: Tue, 20 Mar 2012 15:13:04 +0100 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: References: <20120320144209.598d3940@anubis.defcon1> Message-ID: <20120320151304.2e0c44b4@anubis.defcon1> On Tue, 20 Mar 2012 08:57:56 -0500 Ken Rice wrote: > That's not a good way to solve the problem... The problem after a quick look > is hardcoded paths for pgsql... Hmm, I didn't check but there's also usually a configure option to point to unusual paths (some' like: --with-thelibIwant=/....) -- From krice at freeswitch.org Tue Mar 20 17:36:59 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 20 Mar 2012 09:36:59 -0500 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: <20120320151304.2e0c44b4@anubis.defcon1> Message-ID: Actually there just needs to be a pg_config check in there... This will completely solve the problem in a cross platform sorta way Something like this in the Makefile LOCAL_CFLAGS=-I`pg_config --pkgincludedir` -I/usr/include ${EXTRA_FLAGS} LOCAL_LDFLAGS=`pg_config --libdir`/libpq.so Maybe a few extra things in there. But that will clean up that pathing issue and this is cross platform for properly installed postgresql K On 3/20/12 9:13 AM, "Bzzz" wrote: > On Tue, 20 Mar 2012 08:57:56 -0500 > Ken Rice wrote: > >> That's not a good way to solve the problem... The problem after a quick look >> is hardcoded paths for pgsql... > > Hmm, I didn't check but there's also usually a configure option to > point to unusual paths (some' like: --with-thelibIwant=/....) From lists at telefaks.de Tue Mar 20 17:39:13 2012 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 20 Mar 2012 15:39:13 +0100 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: <1FFF97C269757C458224B7C895F35F15079D1B@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F15079D1B@cantor.std.visionutv.se> Message-ID: <4F689691.4060602@telefaks.de> Wiki is updated accordingly. Best regards Peter Am 20.03.2012 13:32, schrieb Peter Olsson: > It would be very appreciated if you could take some time and update with some more information on the wiki - so others could benefit from this as well.. > > /Peter > > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Peter Steinbach > Skickat: den 20 mars 2012 13:08 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Advice how to start with H.323 > > Thanks to your help, I am now a step further with 1.22.0 and ptlib-2.8.2: > > mod_h323 has compiled successfully. > After relinking libh323_linux_x86_64_.so.1.22.0 the mod_h323 module finally loaded. > Thanks to all, who helped on this issue! > > Do you know of any gatekeeper avaliable for testing, where I could test connectivity and 2-way-audio? > I tried with http://www.voxgratia.org/ but this gateway seems to be down. > > Best regards > Peter > > > > > Am 19.03.2012 14:33, schrieb Peter Olsson: >> Yes, I've used those recommendations - so 1.22.0 is probably a better choice. >> >> /Peter >> >> >> -----Ursprungligt meddelande----- >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Patrick >> Lists >> Skickat: den 19 mars 2012 14:03 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] Advice how to start with H.323 >> >> On 19-03-12 11:53, Peter Steinbach wrote: >>> I am using Ubuntu 10.04. >>> >>> I followed the installation procedures on the wiki page and installed >>> >>> ptlib-2.8.2 + h323plus-trunk >> Not sure if it's still valid but in the past the recommendation was to follow the versions listed at: >> >> http://www.gnugk.org/compiling-gnugk.html >> >> Which in this case means you should use H323Plus 1.22.0 and not trunk. >> >> Regards, >> Patrick >> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org >> >> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org >> > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f68720032761729395285! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From kris at kriskinc.com Tue Mar 20 17:41:21 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 20 Mar 2012 10:41:21 -0400 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: <4F67DB51.20900@puzzled.xs4all.nl> Message-ID: Brian, Can you turn on siptrace (sofia global siptrace on) from fs_cli, replicate the issue, and upload the full fs_cli output (FreeSWITCH console and SIP trace) to pastebin.freeswitch.org? 2012/3/20 Brian Foster : > Anton, > > I'm not really a pro at reading those, but I did post it here: > > http://da1.endigovoip.com/dump.txt > > ...I'm not sure if that will tell much. Where it says > bdfoster-Precision-WorkStation-490, that is the FS server (192.168.1.79), > the ATA is on 192.168.1.80 (Grandstream HT286) This problem has also > occurred using a SPA921 and another HT286, all on the same LAN connected to > the same server, etc. > > -BDF > > > 2012/3/20 Anton Kvashenkin >> >> Try to catch RTP streams >> >> tcpdump -i eth0 portrange?16384-16406 >> >> and see what happens. >> >> >> 20 ????? 2012??. 6:14 ???????????? Brian Foster >> ???????: >> >>> Yea I know what you're talking about. That has indeed been switched off >>> on the RG. For reference purposes, the Residential Gateway is a 2wire >>> AT&T-branded Uverse DSL modem/wireless router with a 4 port switch. I've >>> also got two 8 port netgear prosafe switches. Phones and server are on the >>> same subnet/netmask. Very simple network setup. No VLANS, etc. This was >>> indeed working with Asterisk, so I'm almost positive it's a configuration >>> issue with FreeSWITCH. >>> >>> If there's a need to look at more than the files/logs I've pb'd, please >>> let be know! >>> >>> -BDF >>> >>> On Mar 19, 2012 9:20 PM, "Patrick Lists" >>> wrote: >>>> >>>> On 20-03-12 00:10, Brian Foster wrote: >>>> > UPnP isn't available on this network. Ports 16384-16406 are forwarded >>>> > to >>>> > the machine from the router. Port range in switch.conf.xml reflects >>>> > that. Also 5080 is forwarded from the router to the server. >>>> > >>>> > Both of those variables (ext-sip-ip and ext-rtp-ip) are set to the >>>> > stun >>>> > server, see pastebin 18699 (third link in original email). >>>> >>>> If the (ADSL) router has a (for lack of a better term cause I can't >>>> remember) "SIP helper" app, try disabling it. Similar to the Cisco "SMTP >>>> Fixup" functionality which does an excellent job at the opposite. >>>> >>>> Regards, >>>> Patrick >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From peter.olsson at visionutveckling.se Tue Mar 20 17:58:24 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 20 Mar 2012 14:58:24 +0000 Subject: [Freeswitch-users] Advice how to start with H.323 Message-ID: <1FFF97C269757C458224B7C895F35F1507AEE1@cantor.std.visionutv.se> Thanks! /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Peter Steinbach Skickat: den 20 mars 2012 15:39 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Advice how to start with H.323 Wiki is updated accordingly. Best regards Peter Am 20.03.2012 13:32, schrieb Peter Olsson: > It would be very appreciated if you could take some time and update with some more information on the wiki - so others could benefit from this as well.. > > /Peter > > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Peter > Steinbach > Skickat: den 20 mars 2012 13:08 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Advice how to start with H.323 > > Thanks to your help, I am now a step further with 1.22.0 and ptlib-2.8.2: > > mod_h323 has compiled successfully. > After relinking libh323_linux_x86_64_.so.1.22.0 the mod_h323 module finally loaded. > Thanks to all, who helped on this issue! > > Do you know of any gatekeeper avaliable for testing, where I could test connectivity and 2-way-audio? > I tried with http://www.voxgratia.org/ but this gateway seems to be down. > > Best regards > Peter > > > > > Am 19.03.2012 14:33, schrieb Peter Olsson: >> Yes, I've used those recommendations - so 1.22.0 is probably a better choice. >> >> /Peter >> >> >> -----Ursprungligt meddelande----- >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Patrick >> Lists >> Skickat: den 19 mars 2012 14:03 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] Advice how to start with H.323 >> >> On 19-03-12 11:53, Peter Steinbach wrote: >>> I am using Ubuntu 10.04. >>> >>> I followed the installation procedures on the wiki page and >>> installed >>> >>> ptlib-2.8.2 + h323plus-trunk >> Not sure if it's still valid but in the past the recommendation was to follow the versions listed at: >> >> http://www.gnugk.org/compiling-gnugk.html >> >> Which in this case means you should use H323Plus 1.22.0 and not trunk. >> >> Regards, >> Patrick >> >> >> >> _____________________________________________________________________ >> _ ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> e >> rs >> http://www.freeswitch.org >> >> >> >> >> _____________________________________________________________________ >> _ ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> e >> rs >> http://www.freeswitch.org >> > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f68950732764931015849! From sdevoy at bizfocused.com Tue Mar 20 00:49:56 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Mon, 19 Mar 2012 17:49:56 -0400 Subject: [Freeswitch-users] Unable to TXFAX, but receiving is OK. Always result (49) In-Reply-To: <02a301cd05f2$bb00aaa0$3101ffe0$@bizfocused.com> References: <02a301cd05f2$bb00aaa0$3101ffe0$@bizfocused.com> Message-ID: <03c801cd061a$395d8900$ac189b00$@bizfocused.com> DUH! The fax phone number is really critical whether you are dyslexic or not. From: Sean Devoy [mailto:sdevoy at bizfocused.com] Sent: Monday, March 19, 2012 1:07 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Unable to TXFAX, but receiving is OK. Always result (49) Hi All, I can receive faxes and email them to their recipients now with rxfax!!!! I cannot send anything though. My test command is: (with number changed) /usr/local/freeswitch/bin/fs_cli -x "originate {ignore_early_media=true}sofia/gateway/voipinnovations/11234567890 &txfax(/usr/sean/output/txfax.tiff)" I have also tried it without the ignore_early_media. I do not have any V.38 enabled. When I did no incoming faxes would work. The receiving fax never sent tone, so sender never connected. Any help would be greatly appreciated. Thanks. Here is the FS log: freeswitch at internal> 2012-03-19 12:58:06.511937 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-19 12:58:06.511937 [DEBUG] switch_event.c:1521 Parsing variable [ignore_early_media]=[true] 2012-03-19 12:58:06.511937 [NOTICE] switch_channel.c:924 New Channel sofia/external_noauth/14108038876 [bc6cadc8-761b-42cb-8873-889320fc2658] 2012-03-19 12:58:06.511937 [DEBUG] mod_sofia.c:4674 (sofia/external_noauth/14108038876) State Change CS_NEW -> CS_INIT 2012-03-19 12:58:06.511937 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:06.511937 [DEBUG] switch_core_state_machine.c:362 (sofia/external_noauth/14108038876) Running State Change CS_INIT 2012-03-19 12:58:06.511937 [DEBUG] switch_core_state_machine.c:401 (sofia/external_noauth/14108038876) State INIT 2012-03-19 12:58:06.511937 [DEBUG] mod_sofia.c:85 sofia/external_noauth/14108038876 SOFIA INIT 2012-03-19 12:58:07.117896 [DEBUG] switch_nat.c:511 mapped public port 17322 protocol UDP to localport 17322 2012-03-19 12:58:07.602876 [DEBUG] switch_nat.c:511 mapped public port 17323 protocol UDP to localport 17323 2012-03-19 12:58:07.602876 [DEBUG] mod_sofia.c:125 (sofia/external_noauth/14108038876) State Change CS_INIT -> CS_ROUTING 2012-03-19 12:58:07.602876 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:07.602876 [DEBUG] switch_core_state_machine.c:401 (sofia/external_noauth/14108038876) State INIT going to sleep 2012-03-19 12:58:07.602876 [DEBUG] switch_core_state_machine.c:362 (sofia/external_noauth/14108038876) Running State Change CS_ROUTING 2012-03-19 12:58:07.602876 [DEBUG] switch_channel.c:1884 (sofia/external_noauth/14108038876) Callstate Change DOWN -> RINGING 2012-03-19 12:58:07.602876 [DEBUG] switch_core_state_machine.c:410 (sofia/external_noauth/14108038876) State ROUTING 2012-03-19 12:58:07.602876 [DEBUG] mod_sofia.c:148 sofia/external_noauth/14108038876 SOFIA ROUTING 2012-03-19 12:58:07.602876 [DEBUG] switch_ivr_originate.c:66 (sofia/external_noauth/14108038876) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-03-19 12:58:07.602876 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:07.602876 [DEBUG] switch_core_state_machine.c:410 (sofia/external_noauth/14108038876) State ROUTING going to sleep 2012-03-19 12:58:07.602876 [DEBUG] switch_core_state_machine.c:362 (sofia/external_noauth/14108038876) Running State Change CS_CONSUME_MEDIA 2012-03-19 12:58:07.602876 [DEBUG] switch_core_state_machine.c:429 (sofia/external_noauth/14108038876) State CONSUME_MEDIA 2012-03-19 12:58:07.602876 [DEBUG] switch_core_state_machine.c:429 (sofia/external_noauth/14108038876) State CONSUME_MEDIA going to sleep 2012-03-19 12:58:07.602876 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:07.602876 [DEBUG] sofia.c:5494 Channel sofia/external_noauth/14108038876 entering state [calling][0] 2012-03-19 12:58:10.613725 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:10.613725 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:10.623715 [DEBUG] sofia.c:5494 Channel sofia/external_noauth/14108038876 entering state [proceeding][183] 2012-03-19 12:58:10.623715 [DEBUG] sofia.c:5505 Remote SDP: v=0 o=Sansay-VSXi 188 1 IN IP4 64.136.174.30 s=Session Controller c=IN IP4 69.85.185.142 t=0 0 m=audio 26144 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:2919 Set Codec sofia/external_noauth/14108038876 PCMU/8000 20 ms 160 samples 64000 bits 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/external_noauth/14108038876] 10.10.40.185 port 17322 -> 69.85.185.142 port 26144 codec: 0 ms: 20 2012-03-19 12:58:10.623715 [DEBUG] switch_rtp.c:1659 Starting timer [soft] 160 bytes per 20ms 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send payload to 101 2012-03-19 12:58:10.623715 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive payload to 101 2012-03-19 12:58:10.623715 [NOTICE] sofia_glue.c:3945 Pre-Answer sofia/external_noauth/14108038876! 2012-03-19 12:58:10.623715 [DEBUG] switch_channel.c:2930 (sofia/external_noauth/14108038876) Callstate Change RINGING -> EARLY 2012-03-19 12:58:23.010042 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:23.010042 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:23.010042 [DEBUG] sofia.c:5494 Channel sofia/external_noauth/14108038876 entering state [completing][200] 2012-03-19 12:58:23.010042 [DEBUG] sofia.c:5502 Duplicate SDP v=0 o=Sansay-VSXi 188 1 IN IP4 64.136.174.30 s=Session Controller c=IN IP4 69.85.185.142 t=0 0 m=audio 26144 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-03-19 12:58:23.010042 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:23.010042 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:23.010042 [DEBUG] sofia.c:5494 Channel sofia/external_noauth/14108038876 entering state [ready][200] 2012-03-19 12:58:23.010042 [DEBUG] switch_channel.c:3188 (sofia/external_noauth/14108038876) Callstate Change EARLY -> ACTIVE 2012-03-19 12:58:23.010042 [NOTICE] sofia.c:6142 Channel [sofia/external_noauth/14108038876] has been answered 2012-03-19 12:58:23.010042 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [sofia/external_noauth/14108038876] 2012-03-19 12:58:23.010042 [INFO] switch_channel.c:2708 sofia/external_noauth/14108038876 Flipping CID from "" <0000000000> to "Outbound Call" <14108038876> 2012-03-19 12:58:23.010042 [DEBUG] mod_commands.c:3574 (sofia/external_noauth/14108038876) State Change CS_CONSUME_MEDIA -> CS_EXECUTE 2012-03-19 12:58:23.010042 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:23.010042 [DEBUG] switch_core_state_machine.c:362 (sofia/external_noauth/14108038876) Running State Change CS_EXECUTE 2012-03-19 12:58:23.010042 [DEBUG] switch_core_state_machine.c:417 (sofia/external_noauth/14108038876) State EXECUTE 2012-03-19 12:58:23.010042 [DEBUG] mod_sofia.c:241 sofia/external_noauth/14108038876 SOFIA EXECUTE 2012-03-19 12:58:23.010042 [DEBUG] switch_core_state_machine.c:192 sofia/external_noauth/14108038876 Standard EXECUTE EXECUTE sofia/external_noauth/14108038876 txfax(/usr/sean/output/txfax.tiff) 2012-03-19 12:58:23.010042 [DEBUG] mod_spandsp_fax.c:1355 Raw read codec activation Success L16 20000 2012-03-19 12:58:23.010042 [DEBUG] switch_core_codec.c:116 sofia/external_noauth/14108038876 Push codec L16:70 2012-03-19 12:58:23.010042 [DEBUG] mod_spandsp_fax.c:1371 Raw write codec activation Success L16 2012-03-19 12:58:23.070036 [DEBUG] switch_rtp.c:3204 Correct ip/port confirmed. 2012-03-19 12:58:25.539903 [DEBUG] switch_core_session.c:875 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:25.549903 [DEBUG] switch_channel.c:2846 (sofia/external_noauth/14108038876) Callstate Change ACTIVE -> HANGUP 2012-03-19 12:58:25.549903 [NOTICE] sofia.c:623 Hangup sofia/external_noauth/14108038876 [CS_EXECUTE] [NORMAL_CLEARING] 2012-03-19 12:58:25.549903 [DEBUG] switch_channel.c:2869 Send signal sofia/external_noauth/14108038876 [KILL] 2012-03-19 12:58:25.549903 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:489 ============================================================================ == 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:502 Fax processing not successful - result (49) The call dropped prematurely. 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:507 Remote station id: 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:508 Local station id: SpanDSP Fax Ident 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:509 Pages transferred: 0 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:511 Total fax pages: 0 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:512 Image resolution: 0x0 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:513 Transfer Rate: 14400 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:515 ECM status off 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:516 remote country: 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:517 remote vendor: 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:518 remote model: 2012-03-19 12:58:25.549903 [DEBUG] mod_spandsp_fax.c:520 ============================================================================ == 2012-03-19 12:58:25.549903 [DEBUG] switch_core_codec.c:141 sofia/external_noauth/14108038876 Restore previous codec PCMU:0. 2012-03-19 12:58:25.549903 [DEBUG] switch_core_session.c:2285 sofia/external_noauth/14108038876 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:417 (sofia/external_noauth/14108038876) State EXECUTE going to sleep 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:362 (sofia/external_noauth/14108038876) Running State Change CS_HANGUP 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:602 (sofia/external_noauth/14108038876) State HANGUP 2012-03-19 12:58:25.549903 [DEBUG] mod_sofia.c:463 sofia/external_noauth/14108038876 Overriding SIP cause 480 with 200 from the other leg 2012-03-19 12:58:25.549903 [DEBUG] mod_sofia.c:469 Channel sofia/external_noauth/14108038876 hanging up, cause: NORMAL_CLEARING 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:47 sofia/external_noauth/14108038876 Standard HANGUP, cause: NORMAL_CLEARING 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:602 (sofia/external_noauth/14108038876) State HANGUP going to sleep 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:393 (sofia/external_noauth/14108038876) State Change CS_HANGUP -> CS_REPORTING 2012-03-19 12:58:25.549903 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:362 (sofia/external_noauth/14108038876) Running State Change CS_REPORTING 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:662 (sofia/external_noauth/14108038876) State REPORTING 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:79 sofia/external_noauth/14108038876 Standard REPORTING, cause: NORMAL_CLEARING 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:662 (sofia/external_noauth/14108038876) State REPORTING going to sleep 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:387 (sofia/external_noauth/14108038876) State Change CS_REPORTING -> CS_DESTROY 2012-03-19 12:58:25.549903 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14108038876 [BREAK] 2012-03-19 12:58:25.549903 [DEBUG] switch_core_session.c:1380 Session 184 (sofia/external_noauth/14108038876) Locked, Waiting on external entities 2012-03-19 12:58:25.549903 [NOTICE] switch_core_session.c:1398 Session 184 (sofia/external_noauth/14108038876) Ended 2012-03-19 12:58:25.549903 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external_noauth/14108038876 [CS_DESTROY] 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:491 (sofia/external_noauth/14108038876) Callstate Change HANGUP -> DOWN 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:494 (sofia/external_noauth/14108038876) Running State Change CS_DESTROY 2012-03-19 12:58:25.549903 [DEBUG] switch_core_state_machine.c:504 (sofia/external_noauth/14108038876) State DESTROY 2012-03-19 12:58:25.549903 [DEBUG] mod_sofia.c:374 sofia/external_noauth/14108038876 SOFIA DESTROY 2012-03-19 12:58:25.559907 [DEBUG] switch_nat.c:571 unmapped public port 17322 protocol UDP to localport 17322 2012-03-19 12:58:25.579904 [DEBUG] switch_nat.c:571 unmapped public port 17323 protocol UDP to localport 17323 2012-03-19 12:58:25.579904 [DEBUG] switch_core_state_machine.c:86 sofia/external_noauth/14108038876 Standard DESTROY 2012-03-19 12:58:25.579904 [DEBUG] switch_core_state_machine.c:504 (sofia/external_noauth/14108038876) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120319/6cbd0bb4/attachment-0001.html From zy at globnet.md Tue Mar 20 11:05:39 2012 From: zy at globnet.md (zy) Date: Tue, 20 Mar 2012 10:05:39 +0200 Subject: [Freeswitch-users] Freeswitch Digium TE220 and SS7 Message-ID: <1332230739.5556.2.camel@localhost.localdomain> Hi to every one! I have a card Digium TE220 and I want to ask if it is possible to configure it for working through SS7 on Freeswitch. I was searching on Internet on many sites and forums and couldn't find any thing, maybe someone has heard something or has a practice and can give me advise. Thanks for everyone. From fuji246 at gmail.com Tue Mar 20 12:20:13 2012 From: fuji246 at gmail.com (Fu Jiantao) Date: Tue, 20 Mar 2012 17:20:13 +0800 Subject: [Freeswitch-users] TDM400P spans In-Reply-To: References: <5227389.xYDaNXzQut@axp> Message-ID: Thanks Moises! So span is a logic concept, while channel is corresponding to the port(fxo or fxs) on the card, am i right? I'm new to the freeswitch and asterisk world, and found the most difficult part is relating to the PSTN, I'm not familiar with PSTN, and I've read asterisk book and freeswitch book, and learn the basic of PSTN, but that seems not enough, for example, I found I can't figure out how to use FreeTDM, little are relating to this topic, is there any good references on this topic? I'm trying to port the io module chan_fxin asterisk win32into freeTDM, but I found I need to know more about the above topic. 2012/3/20 Moises Silva > On Sun, Mar 18, 2012 at 10:58 AM, Jeromy wrote: > >> > I'm trying to convert from Asterisk to Freeswitch, and have a question >> about >> spans. >> > >> > I read the freetdm configuration example for TDM400 >> > and am wondering what exactly is a span. >> > >> > I have 2 TDM400P cards in my computer, >> > one card has 2 FXS modules for connecting analog phones. >> > The other card has 3 FXO modules for connecting to 3 PSTN trunk lines. >> > >> > # lsdahdi >> > ### Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" >> > 1 FXS FXOKS (In use) (EC: OSLEC - INACTIVE) >> > 2 FXS FXOKS (In use) (EC: OSLEC - INACTIVE) >> > 3 unknown Reserved >> > 4 unknown Reserved >> > ### Span 2: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER) >> > 5 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) >> > 6 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) >> > 7 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) >> > 8 unknown Reserved >> > >> > Do I configure freetdm as in the example for TDM400 >> > with separate spans for each module like the following? >> > or should I have just two spans as shown by lsdahsi? >> > Thanks. >> > >> >> Hi Jeremy >> >> Do you find the answer to this question? >> >> I'm also very confuse on the span and channel in both asterisk and >> freeswitch, and I've found little explaination on this. >> >> " A span is a logical unit that represents a group of channels. >> With digital telephony, a span usually represents a physical port >> on the card. >> If the system has only one such card with a single port, so it is >> referred to >> as span 1. " >> >> Could anyone help on this? >> >> > At the very least you need 2 freetdm span configurations because you > cannot mix FXS and FXO channels in the same FreeTDM logical span. > > http://wiki.freeswitch.org/wiki/FreeTDM#DAHDI_mode > > For the given hardware output, I'd simply create 2 spans, one for all the > fxo channels and one for all the fxs channels. > > *Moises Silva > **Manager, Software Engineering*** > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > ** > > Products > | Solutions > | Events > | Contact > | Wiki > | Facebook > | Twitter`| > | YouTube > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/2afd8c98/attachment-0001.html From apartment.help at me.com Tue Mar 20 13:57:14 2012 From: apartment.help at me.com (apartment.help at me.com) Date: Tue, 20 Mar 2012 10:57:14 +0000 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: <4F67DB51.20900@puzzled.xs4all.nl> Message-ID: <8C156396-CA07-4375-A37C-F6F7B3615247@me.com> I'm having the same issue. For info, I'm running static one-to-one NAT, have tried with and without STUN, and have checked the RTP ports etc. No vlans etc either. Worked fine with my other test system (3cx) so again think it must be a freeswitch config issue. - absolutely On 20 Mar 2012, at 10:32, Andrew Cassidy wrote: > tcpdump -i -s 65535 -w > > tcpdump -s 65535 -w dump.pcap > > That should do the trick, but you can add port filters if you'd prefer. If you do, please include both sip and rtp traffic. The dump.pcap file will be compatible with wireshark to make analyzing it a little easier. > > 2012/3/20 Andrew Cassidy > ---------- Forwarded message ---------- > From: "Andrew Cassidy" > Date: Mar 20, 2012 8:55 AM > Subject: Re: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds > To: "FreeSWITCH Users Help" > > When we get a solution (as I'm sure we will) Can we get a guide of things to try put up on the wiki? I've seen the same question asked at least twice already this week. > > I'll find the command line to do a more detailed packet dump that can be loaded into wireshark for further analysis. The main thing I'm wondering is if Nat is doing something funny with rewriting the rtp and rtcp port numbers. > > What type of NAT do you have? You can use WinStun or another basic stun client to find out. > > On Mar 20, 2012 6:07 AM, "Brian Foster" wrote: > Anton, > > I'm not really a pro at reading those, but I did post it here: > > http://da1.endigovoip.com/dump.txt > > ...I'm not sure if that will tell much. Where it says bdfoster-Precision-WorkStation-490, that is the FS server (192.168.1.79), the ATA is on 192.168.1.80 (Grandstream HT286) This problem has also occurred using a SPA921 and another HT286, all on the same LAN connected to the same server, etc. > > -BDF > > 2012/3/20 Anton Kvashenkin > Try to catch RTP streams > > tcpdump -i eth0 portrange 16384-16406 > > and see what happens. > > > 20 ????? 2012 ?. 6:14 ???????????? Brian Foster ???????: > > Yea I know what you're talking about. That has indeed been switched off on the RG. For reference purposes, the Residential Gateway is a 2wire AT&T-branded Uverse DSL modem/wireless router with a 4 port switch. I've also got two 8 port netgear prosafe switches. Phones and server are on the same subnet/netmask. Very simple network setup. No VLANS, etc. This was indeed working with Asterisk, so I'm almost positive it's a configuration issue with FreeSWITCH. > > If there's a need to look at more than the files/logs I've pb'd, please let be know! > > -BDF > > On Mar 19, 2012 9:20 PM, "Patrick Lists" wrote: > On 20-03-12 00:10, Brian Foster wrote: > > UPnP isn't available on this network. Ports 16384-16406 are forwarded to > > the machine from the router. Port range in switch.conf.xml reflects > > that. Also 5080 is forwarded from the router to the server. > > > > Both of those variables (ext-sip-ip and ext-rtp-ip) are set to the stun > > server, see pastebin 18699 (third link in original email). > > If the (ADSL) router has a (for lack of a better term cause I can't > remember) "SIP helper" app, try disabling it. Similar to the Cisco "SMTP > Fixup" functionality which does an excellent job at the opposite. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Andrew Cassidy BSc (Hons) MBCS > Managing Director; Cassidy Web Services Ltd > T: 03300 100 960 F: 03300 100 961 > E: andrew at cassidywebservices.co.uk > W: www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/24149b92/attachment-0001.html From sdevoy at bizfocused.com Tue Mar 20 18:08:10 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 20 Mar 2012 11:08:10 -0400 Subject: [Freeswitch-users] make current caused problems Message-ID: <065101cd06ab$43733390$ca599ab0$@bizfocused.com> Hi, I did a "make current" last night. Today none of my IVRs were working. .MP3 files were no longer recognized. I went back and did a "make mod_shout-install" and rectified the situation. This should be in the wiki. Is there something I can do to have all my user added modules recompiled and installed with a "make current". I am pretty sure I added one other module that is not the default, but I don't recall what it was. Thanks. Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/c8ac976a/attachment.html From krice at freeswitch.org Tue Mar 20 18:10:42 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 20 Mar 2012 10:10:42 -0500 Subject: [Freeswitch-users] Freeswitch Digium TE220 and SS7 In-Reply-To: <1332230739.5556.2.camel@localhost.localdomain> Message-ID: See Sangoma SS7 for SS7 with FreeSWITCH however I doubt it will work woth Digium Hardware.... There are no opensource SS7 stacks that are license compatible with FreeSWITCH at this time On 3/20/12 3:05 AM, "zy" wrote: > Hi to every one! I have a card Digium TE220 and I want to ask if it is > possible to configure it for working through SS7 on Freeswitch. I was > searching on Internet on many sites and forums and couldn't find any > thing, maybe someone has heard something or has a practice and can give > me advise. > Thanks for everyone. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From robert.longfield at klinsight.com Tue Mar 20 18:19:39 2012 From: robert.longfield at klinsight.com (Robert Longfield) Date: Tue, 20 Mar 2012 11:19:39 -0400 Subject: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring In-Reply-To: References: <0488BB5D11C94E8285DD956672994480@KITPC003> Message-ID: <80F4A0FF691F488F9BE7511BD0AA4BAC@KITPC003> Thanks for the tip Brian, I tried using a loopback using the example in /dialplan/default.xml and I am still experiencing the same problem. I?ve tried a loopback that looks like: Only instead of dropping the call it seems to sleep... From: Brian Foster Sent: Monday, March 19, 2012 7:12 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml On Mar 19, 2012 4:43 PM, "Robert Longfield" wrote: I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM. My dialplan looks like: What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn?t work either. The message I see when Freeswitch hangs up is: Channel sofia/internal/sip:1002 at 72.38.184.18:39042 hanging up, cause: USER_BUSY The full output from cli can be seen here: http://pastebin.freeswitch.org/18696 I would like to get the call to transfer properly. Thanks -Robert _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/d9f53fe0/attachment.html From msc at freeswitch.org Tue Mar 20 18:34:18 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 20 Mar 2012 08:34:18 -0700 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: Message-ID: We have scores of machines behind NAT talking to Flowroute with no problems, so there's got to be something potentially non-obvious but easy that needs to be set/unset. I noticed in the SIP trace that there are several calls. It's hard to know what's what. I think your best bet is a pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I also noticed that you redacted IP addrs - you won't be able to do this with a pcap. If security is an issue then I'd say get the pcap and let us know here on the list, then those who can have a look will email you privately and you can send the pcap file to them. -MC On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster wrote: > Alright, so I admit... I'm a little rusty when it comes to NAT, etc. I've > only set up FS so far on machines with no NAT, so this is sort of a new > experience for me. > > I have a FreeSWITCH server located on the same local network as all of my > phones here at the house. When I try to make a call to Flowroute, after > about 30 seconds the call drops. It also does the exact same thing when I > call a buddy's server directly via SIP. > > Here's a siptrace of the call (I didn't think that the actual FS log would > be much help): > http://pastebin.freeswitch.org/18697 > > ...and here's a paste of 'sofia status': > http://pastebin.freeswitch.org/18698 > > ...and just for good measure, here's a paste of vars.xml: > http://pastebin.freeswitch.org/18699 > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/fc9a9a41/attachment-0001.html From msc at freeswitch.org Tue Mar 20 18:44:13 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 20 Mar 2012 08:44:13 -0700 Subject: [Freeswitch-users] is loopback really 'evil'? In-Reply-To: References: <4F673C92.6020303@newpace.ca> Message-ID: Always ask this question: WHY do you need loopback? What does loopback give you that you can't get elsewhere? For example, in the default dialplan we use loopback for voicemail so that people can do an attended transfer to send a call into the target's voicemail. My recommendation is this: if you are tempted to use loopback but aren't sure if it's the only way to accomplish what you want, then ask here or on IRC. In many cases you can use loopback w/o any drama. However, some people abuse loopback by doing crazy things that Tony never dreamed of when he created mod_loopback. That's the primary reason that loopback is "evil." So... ask away! -MC On Mon, Mar 19, 2012 at 9:46 AM, Mitch Capper wrote: > I think its best to look at loopback as something of a hack, in most > situations there is almost always another way to do what loopback > does. As long as loopback behaves in your situation how it should > you should be fine, but if when you go to use it you find oddities or > issues generally taking loopback out of the equation will solve the > problems. > > ~Mitch > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/2567513e/attachment.html From mitch.capper at gmail.com Tue Mar 20 19:05:12 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Tue, 20 Mar 2012 09:05:12 -0700 Subject: [Freeswitch-users] TLS on FS In-Reply-To: References: Message-ID: Hi Tihomir, I added a section at http://wiki.freeswitch.com/wiki/SIP_TLS#Further_Debugging_Steps for help with sip debugging. if you can collect the logs mentioned there along with do a sofia status when sofia is running and paste that we can start there. You may also want to try one of the FS softphones to ensure you can get a client working first. ~Mitch From msc at freeswitch.org Tue Mar 20 19:19:46 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 20 Mar 2012 09:19:46 -0700 Subject: [Freeswitch-users] FreeSWITCH Community Conf Call Tomorrow Message-ID: Hey gang! Tomorrow's agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_21 We're going to be doing a community scrum. That means no specific topic, but instead we're going to put our heads together and work on ML questions and any other discussion items that come up. In preparation for tomorrow I'd like you all to review the questions that have been coming in to the mailing list. There are lots that haven't been touched. Pick one or two that you either know how to handle or would be able to research and then we'll tackle them tomorrow. Thanks guys! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/12f87723/attachment.html From msc at freeswitch.org Tue Mar 20 19:22:41 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 20 Mar 2012 09:22:41 -0700 Subject: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring In-Reply-To: <80F4A0FF691F488F9BE7511BD0AA4BAC@KITPC003> References: <0488BB5D11C94E8285DD956672994480@KITPC003> <80F4A0FF691F488F9BE7511BD0AA4BAC@KITPC003> Message-ID: Malfunction! Need input! Get debug log of call from start to finish and put on pastebin. -MC On Tue, Mar 20, 2012 at 8:19 AM, Robert Longfield < robert.longfield at klinsight.com> wrote: > Thanks for the tip Brian, > > I tried using a loopback using the example in /dialplan/default.xml and I > am still experiencing the same problem. > > I?ve tried a loopback that looks like: > > > > > > > Only instead of dropping the call it seems to sleep... > > > > *From:* Brian Foster > *Sent:* Monday, March 19, 2012 7:12 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Call sent to group call terminates when > group is busy instead of transferring > > > Try using loopback when you send the call to voicemail, also see the local > extensions dialplan located in conf/dialplan/default.xml > On Mar 19, 2012 4:43 PM, "Robert Longfield" < > robert.longfield at klinsight.com> wrote: > >> I set up a group call for our support team in which all their phones >> ring when someone needs to speak with them. If they are busy the call >> should be transferred to a general extension which if not answered then >> goes to that extensions VM. >> >> My dialplan looks like: >> >> >> >> >> >> >> >> >> >> >> >> >> What is happening is a caller selects the support option from the IVR, >> ever phone in the support group rings, which is what should happen. If no >> one picks up the call Freeswitch hangs up instead of transferring the call >> to extension 1000. You can see that I also tried to send the call directly >> to voicemail but that didn?t work either. >> >> The message I see when Freeswitch hangs up is: >> >> Channel sofia/internal/sip:1002 at 72.38.184.18:39042 hanging up, >> cause: USER_BUSY >> >> The full output from cli can be seen here: >> http://pastebin.freeswitch.org/18696 >> >> I would like to get the call to transfer properly. >> >> Thanks >> -Robert >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/b6028b9d/attachment.html From fdelawarde at wirelessmundi.com Tue Mar 20 19:37:03 2012 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 20 Mar 2012 17:37:03 +0100 Subject: [Freeswitch-users] is loopback really 'evil'? In-Reply-To: References: <4F673C92.6020303@newpace.ca> Message-ID: <1332261423.15458.1324.camel@luna.madrid.commsmundi.com> It is quite evil, but also a great tool in some cases. Use it only if you feel comfortable enough with FS to workaround strange issues. Fran?ois. On Tue, 2012-03-20 at 08:44 -0700, Michael Collins wrote: > Always ask this question: WHY do you need loopback? What does loopback > give you that you can't get elsewhere? > For example, in the default dialplan we use loopback for voicemail so > that people can do an attended transfer to send a call into the > target's voicemail. > > My recommendation is this: if you are tempted to use loopback but > aren't sure if it's the only way to accomplish what you want, then ask > here or on IRC. In many cases you can use loopback w/o any drama. > However, some people abuse loopback by doing crazy things that Tony > never dreamed of when he created mod_loopback. That's the primary > reason that loopback is "evil." > > So... ask away! > -MC > > On Mon, Mar 19, 2012 at 9:46 AM, Mitch Capper > wrote: > I think its best to look at loopback as something of a hack, > in most > situations there is almost always another way to do what > loopback > does. As long as loopback behaves in your situation how it > should > you should be fine, but if when you go to use it you find > oddities or > issues generally taking loopback out of the equation will > solve the > problems. > > ~Mitch > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From andrew at cassidywebservices.co.uk Tue Mar 20 19:37:03 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 20 Mar 2012 16:37:03 +0000 Subject: [Freeswitch-users] FreeSWITCH Community Conf Call Tomorrow In-Reply-To: References: Message-ID: I think I pointed these out before, but UK PSTN number: +44 3300 100 290 freenum.org ISN: 888*543 On 20 March 2012 16:19, Michael Collins wrote: > Hey gang! > > Tomorrow's agenda page is here: > > http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_21 > > We're going to be doing a community scrum. That means no specific topic, > but instead we're going to put our heads together and work on ML questions > and any other discussion items that come up. In preparation for tomorrow > I'd like you all to review the questions that have been coming in to the > mailing list. There are lots that haven't been touched. Pick one or two > that you either know how to handle or would be able to research and then > we'll tackle them tomorrow. > > Thanks guys! > > -Michael > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/05105d89/attachment-0001.html From msc at freeswitch.org Tue Mar 20 20:00:12 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 20 Mar 2012 10:00:12 -0700 Subject: [Freeswitch-users] How can I change the moh_sound In-Reply-To: References: Message-ID: MOH sound is a conference parameter specified in conference.conf.xml. I believe that the only way you can change it on the fly is to use xml_curl for your configs. -MC On Tue, Mar 20, 2012 at 1:16 AM, piyush singhai wrote: > Hello, > > I want to change the moh sound at run time for conference. can i specify > path on the basis of any ivr. > > --Piyush > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/fd6e6976/attachment.html From bdfoster at endigotech.com Tue Mar 20 20:00:13 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 20 Mar 2012 13:00:13 -0400 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: Message-ID: Andrew, root at homeserver:/usr/local/stund# ./client stunserver.org STUN client version 0.97 Primary: Independent Mapping, Independent Filter, preserves ports, will hairpin Return value is 0x000003 http://da1.endigovoip.com/dump.pcap Kristian, http://pastebin.freeswitch.org/18708 Michael, I did replace the IP's for security purposes, but now I've realized that it's needed and it's not really that big of a deal. I'll end up changing the Flowroute creds after this is fixed up. The prior siptrace is exactly one call (two legs). I don't think it's a carrier issue, as I've tried calling a buddy's server direct sip with the same issues. -BDF On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins wrote: > We have scores of machines behind NAT talking to Flowroute with no > problems, so there's got to be something potentially non-obvious but easy > that needs to be set/unset. I noticed in the SIP trace that there are > several calls. It's hard to know what's what. I think your best bet is a > pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I > also noticed that you redacted IP addrs - you won't be able to do this with > a pcap. If security is an issue then I'd say get the pcap and let us know > here on the list, then those who can have a look will email you privately > and you can send the pcap file to them. > > -MC > > > On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster wrote: > >> Alright, so I admit... I'm a little rusty when it comes to NAT, etc. I've >> only set up FS so far on machines with no NAT, so this is sort of a new >> experience for me. >> >> I have a FreeSWITCH server located on the same local network as all of my >> phones here at the house. When I try to make a call to Flowroute, after >> about 30 seconds the call drops. It also does the exact same thing when I >> call a buddy's server directly via SIP. >> >> Here's a siptrace of the call (I didn't think that the actual FS log >> would be much help): >> http://pastebin.freeswitch.org/18697 >> >> ...and here's a paste of 'sofia status': >> http://pastebin.freeswitch.org/18698 >> >> ...and just for good measure, here's a paste of vars.xml: >> http://pastebin.freeswitch.org/18699 >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/7dcb7df4/attachment.html From robert.longfield at klinsight.com Tue Mar 20 20:00:56 2012 From: robert.longfield at klinsight.com (Robert Longfield) Date: Tue, 20 Mar 2012 13:00:56 -0400 Subject: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring In-Reply-To: References: <0488BB5D11C94E8285DD956672994480@KITPC003><80F4A0FF691F488F9BE7511BD0AA4BAC@KITPC003> Message-ID: <9E00A259F85C4E2E9E4E0FF09A0CC03D@KITPC003> ugh, I can?t believe I forgot to include the pastebin http://pastebin.com/GYLvtDB3 The termination happens also when there is a single user in the call group. The call is transferred to the extension, that extension does not pickup and FS drops the call instead of the call going to VM. When you call any extension directly and the call is not answered you end up in that extensions VM like you should. -Rob From: Michael Collins Sent: Tuesday, March 20, 2012 12:22 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Malfunction! Need input! Get debug log of call from start to finish and put on pastebin. -MC On Tue, Mar 20, 2012 at 8:19 AM, Robert Longfield wrote: Thanks for the tip Brian, I tried using a loopback using the example in /dialplan/default.xml and I am still experiencing the same problem. I?ve tried a loopback that looks like: Only instead of dropping the call it seems to sleep... From: Brian Foster Sent: Monday, March 19, 2012 7:12 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml On Mar 19, 2012 4:43 PM, "Robert Longfield" wrote: I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM. My dialplan looks like: What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn?t work either. The message I see when Freeswitch hangs up is: Channel sofia/internal/sip:1002 at 72.38.184.18:39042 hanging up, cause: USER_BUSY The full output from cli can be seen here: http://pastebin.freeswitch.org/18696 I would like to get the call to transfer properly. Thanks -Robert -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/0b319454/attachment-0001.html From msc at freeswitch.org Tue Mar 20 20:15:20 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 20 Mar 2012 10:15:20 -0700 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: Message-ID: This is why I love Wireshark so much! Look at this purdee graph it makes: See all those 200 OK's that your FS is sending to the Grandstream? Guess what your GS is sending in response to those: NADA! If you look at the BYE that FS sends to the GS you'll even see the reason: SIP;cause=408;text=\"ACK Timeout\" FS never gets an ACK back from the GS. So the question is: why? I'm unfamiliar with the GS so I'll have to defer to those with more experience than I. However, I think you'll find that tcpdumps and analyzing w/ Wireshark is extremely helpful. (Open the pcap, click "Telephony > VoIP calls" and then a new dialog opens. In this case it shows two calls - meaning two call legs. Click "Select All" then click "Flow" and you'll get the cool graph. Click around and see what other stuff does. :) I'm thinking of doing a FreeSWITCH conference call presentation on the subject of collecting pcaps and doing Wireshark analysis. Let me know if you guys think that's a good presentation. -MC On Tue, Mar 20, 2012 at 10:00 AM, Brian Foster wrote: > Andrew, > > root at homeserver:/usr/local/stund# ./client stunserver.org > STUN client version 0.97 > Primary: Independent Mapping, Independent Filter, preserves ports, will > hairpin > Return value is 0x000003 > > http://da1.endigovoip.com/dump.pcap > > Kristian, > > http://pastebin.freeswitch.org/18708 > > Michael, > > I did replace the IP's for security purposes, but now I've realized that > it's needed and it's not really that big of a deal. I'll end up changing > the Flowroute creds after this is fixed up. The prior siptrace is exactly > one call (two legs). I don't think it's a carrier issue, as I've tried > calling a buddy's server direct sip with the same issues. > > -BDF > > On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins wrote: > >> We have scores of machines behind NAT talking to Flowroute with no >> problems, so there's got to be something potentially non-obvious but easy >> that needs to be set/unset. I noticed in the SIP trace that there are >> several calls. It's hard to know what's what. I think your best bet is a >> pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I >> also noticed that you redacted IP addrs - you won't be able to do this with >> a pcap. If security is an issue then I'd say get the pcap and let us know >> here on the list, then those who can have a look will email you privately >> and you can send the pcap file to them. >> >> -MC >> >> >> On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster wrote: >> >>> Alright, so I admit... I'm a little rusty when it comes to NAT, etc. >>> I've only set up FS so far on machines with no NAT, so this is sort of a >>> new experience for me. >>> >>> I have a FreeSWITCH server located on the same local network as all of >>> my phones here at the house. When I try to make a call to Flowroute, after >>> about 30 seconds the call drops. It also does the exact same thing when I >>> call a buddy's server directly via SIP. >>> >>> Here's a siptrace of the call (I didn't think that the actual FS log >>> would be much help): >>> http://pastebin.freeswitch.org/18697 >>> >>> ...and here's a paste of 'sofia status': >>> http://pastebin.freeswitch.org/18698 >>> >>> ...and just for good measure, here's a paste of vars.xml: >>> http://pastebin.freeswitch.org/18699 >>> >>> >>> -- >>> Brian D. Foster >>> Endigo Computer LLC >>> Email: bdfoster at endigotech.com >>> Phone: 317-800-7876 >>> Indianapolis, Indiana, USA >>> >>> This message contains confidential information and is intended for those >>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >>> you are not the intended recipient you are notified that disclosing, >>> copying, distributing or taking any action in reliance on the contents of >>> this information is strictly prohibited. E-mail transmission cannot be >>> guaranteed to be secure or error-free as information could be intercepted, >>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >>> The sender therefore does not accept liability for any errors or omissions >>> in the contents of this message, which arise as a result of e-mail >>> transmission. If verification is required please request a hard-copy >>> version. >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/69317b06/attachment-0001.html From andrew at cassidywebservices.co.uk Tue Mar 20 20:15:58 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 20 Mar 2012 17:15:58 +0000 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: Message-ID: Your flowroute credentials will be hashed in any case. Thanks for getting those details for us, I'll look into it as soon as I get a chance. On 20 March 2012 17:00, Brian Foster wrote: > Andrew, > > root at homeserver:/usr/local/stund# ./client stunserver.org > STUN client version 0.97 > Primary: Independent Mapping, Independent Filter, preserves ports, will > hairpin > Return value is 0x000003 > > http://da1.endigovoip.com/dump.pcap > > Kristian, > > http://pastebin.freeswitch.org/18708 > > Michael, > > I did replace the IP's for security purposes, but now I've realized that > it's needed and it's not really that big of a deal. I'll end up changing > the Flowroute creds after this is fixed up. The prior siptrace is exactly > one call (two legs). I don't think it's a carrier issue, as I've tried > calling a buddy's server direct sip with the same issues. > > -BDF > > On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins wrote: > >> We have scores of machines behind NAT talking to Flowroute with no >> problems, so there's got to be something potentially non-obvious but easy >> that needs to be set/unset. I noticed in the SIP trace that there are >> several calls. It's hard to know what's what. I think your best bet is a >> pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I >> also noticed that you redacted IP addrs - you won't be able to do this with >> a pcap. If security is an issue then I'd say get the pcap and let us know >> here on the list, then those who can have a look will email you privately >> and you can send the pcap file to them. >> >> -MC >> >> >> On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster wrote: >> >>> Alright, so I admit... I'm a little rusty when it comes to NAT, etc. >>> I've only set up FS so far on machines with no NAT, so this is sort of a >>> new experience for me. >>> >>> I have a FreeSWITCH server located on the same local network as all of >>> my phones here at the house. When I try to make a call to Flowroute, after >>> about 30 seconds the call drops. It also does the exact same thing when I >>> call a buddy's server directly via SIP. >>> >>> Here's a siptrace of the call (I didn't think that the actual FS log >>> would be much help): >>> http://pastebin.freeswitch.org/18697 >>> >>> ...and here's a paste of 'sofia status': >>> http://pastebin.freeswitch.org/18698 >>> >>> ...and just for good measure, here's a paste of vars.xml: >>> http://pastebin.freeswitch.org/18699 >>> >>> >>> -- >>> Brian D. Foster >>> Endigo Computer LLC >>> Email: bdfoster at endigotech.com >>> Phone: 317-800-7876 >>> Indianapolis, Indiana, USA >>> >>> This message contains confidential information and is intended for those >>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >>> you are not the intended recipient you are notified that disclosing, >>> copying, distributing or taking any action in reliance on the contents of >>> this information is strictly prohibited. E-mail transmission cannot be >>> guaranteed to be secure or error-free as information could be intercepted, >>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >>> The sender therefore does not accept liability for any errors or omissions >>> in the contents of this message, which arise as a result of e-mail >>> transmission. If verification is required please request a hard-copy >>> version. >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/6b4ec338/attachment.html From adam.kelloway at newpace.ca Tue Mar 20 20:16:41 2012 From: adam.kelloway at newpace.ca (Adam Kelloway) Date: Tue, 20 Mar 2012 14:16:41 -0300 Subject: [Freeswitch-users] is loopback really 'evil'? In-Reply-To: <1332261423.15458.1324.camel@luna.madrid.commsmundi.com> References: <4F673C92.6020303@newpace.ca> <1332261423.15458.1324.camel@luna.madrid.commsmundi.com> Message-ID: <4F68BB79.2040305@newpace.ca> Understood, thanks. I have two esl applications I connect calls between. In my testing with loopback, it gave me much better audio quality then if I bridged the call to another instance of freeswitch, as my freeswitch instances run on virtual machines (see my previous thread "Dialplan bridge results in choppy audio"). Of course, the difference is that I ran my two esl applications on the same freeswitch instance instead of different ones. It would not mind running them on the same one if I needed to, but I wanted to make sure that loopback was ok to use. Adam On 3:59 PM, Fran?ois Delawarde wrote: > It is quite evil, but also a great tool in some cases. Use it only if > you feel comfortable enough with FS to workaround strange issues. > > Fran?ois. > > > On Tue, 2012-03-20 at 08:44 -0700, Michael Collins wrote: >> Always ask this question: WHY do you need loopback? What does loopback >> give you that you can't get elsewhere? >> For example, in the default dialplan we use loopback for voicemail so >> that people can do an attended transfer to send a call into the >> target's voicemail. >> >> My recommendation is this: if you are tempted to use loopback but >> aren't sure if it's the only way to accomplish what you want, then ask >> here or on IRC. In many cases you can use loopback w/o any drama. >> However, some people abuse loopback by doing crazy things that Tony >> never dreamed of when he created mod_loopback. That's the primary >> reason that loopback is "evil." >> >> So... ask away! >> -MC >> >> On Mon, Mar 19, 2012 at 9:46 AM, Mitch Capper >> wrote: >> I think its best to look at loopback as something of a hack, >> in most >> situations there is almost always another way to do what >> loopback >> does. As long as loopback behaves in your situation how it >> should >> you should be fine, but if when you go to use it you find >> oddities or >> issues generally taking loopback out of the equation will >> solve the >> problems. >> >> ~Mitch >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/eadc3393/attachment.html From garbytrash at gmail.com Tue Mar 20 20:30:44 2012 From: garbytrash at gmail.com (Zenny) Date: Tue, 20 Mar 2012 17:30:44 +0000 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: References: <20120320151304.2e0c44b4@anubis.defcon1> Message-ID: Hi Ken: Thanks for your input, but even after inserting two lines you advised in Makefile pops out the same error. # grep -n libpq-fe.h make-error1.txt 4:/usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:40:22: error: libpq-fe.h: No such file or directory However, I did see the a problem with ./configure also. # grep -n libpq-fe.h configure-output1.txt 1440:checking libpq-fe.h usability... no 1441:checking libpq-fe.h presence... no 1442:checking for libpq-fe.h... no What I meant is if ./configure is reporting the unavailability of libpq-fe.h, is that possible by just making changes in the Makefile? I know I may sound pretty dumb! ;-) /z On 3/20/12, Ken Rice wrote: > Actually there just needs to be a pg_config check in there... This will > completely solve the problem in a cross platform sorta way > > Something like this in the Makefile > > LOCAL_CFLAGS=-I`pg_config --pkgincludedir` -I/usr/include ${EXTRA_FLAGS} > LOCAL_LDFLAGS=`pg_config --libdir`/libpq.so > > Maybe a few extra things in there. But that will clean up that pathing issue > and this is cross platform for properly installed postgresql > > K > > > On 3/20/12 9:13 AM, "Bzzz" wrote: > >> On Tue, 20 Mar 2012 08:57:56 -0500 >> Ken Rice wrote: >> >>> That's not a good way to solve the problem... The problem after a quick >>> look >>> is hardcoded paths for pgsql... >> >> Hmm, I didn't check but there's also usually a configure option to >> point to unusual paths (some' like: --with-thelibIwant=/....) > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From andrew at cassidywebservices.co.uk Tue Mar 20 20:35:26 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 20 Mar 2012 17:35:26 +0000 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: Message-ID: My thought at this stage is that the ATA is trying to send the ACK directly and not to FreeSWITCH using the Contact header. If I recall correctly, GS's can be configured with both a registrar and an outbound proxy. Try setting the outbound proxy to the FreeSWITCH box, see what happens. On 20 March 2012 17:15, Michael Collins wrote: > This is why I love Wireshark so much! Look at this purdee graph it makes: > > > > > See all those 200 OK's that your FS is sending to the Grandstream? Guess > what your GS is sending in response to those: NADA! If you look at the BYE > that FS sends to the GS you'll even see the reason: > > SIP;cause=408;text=\"ACK Timeout\" > > FS never gets an ACK back from the GS. So the question is: why? I'm > unfamiliar with the GS so I'll have to defer to those with more experience > than I. However, I think you'll find that tcpdumps and analyzing w/ > Wireshark is extremely helpful. (Open the pcap, click "Telephony > VoIP > calls" and then a new dialog opens. In this case it shows two calls - > meaning two call legs. Click "Select All" then click "Flow" and you'll get > the cool graph. Click around and see what other stuff does. :) > > I'm thinking of doing a FreeSWITCH conference call presentation on the > subject of collecting pcaps and doing Wireshark analysis. Let me know if > you guys think that's a good presentation. > > -MC > > > > On Tue, Mar 20, 2012 at 10:00 AM, Brian Foster wrote: > >> Andrew, >> >> root at homeserver:/usr/local/stund# ./client stunserver.org >> STUN client version 0.97 >> Primary: Independent Mapping, Independent Filter, preserves ports, will >> hairpin >> Return value is 0x000003 >> >> http://da1.endigovoip.com/dump.pcap >> >> Kristian, >> >> http://pastebin.freeswitch.org/18708 >> >> Michael, >> >> I did replace the IP's for security purposes, but now I've realized that >> it's needed and it's not really that big of a deal. I'll end up changing >> the Flowroute creds after this is fixed up. The prior siptrace is exactly >> one call (two legs). I don't think it's a carrier issue, as I've tried >> calling a buddy's server direct sip with the same issues. >> >> -BDF >> >> On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins wrote: >> >>> We have scores of machines behind NAT talking to Flowroute with no >>> problems, so there's got to be something potentially non-obvious but easy >>> that needs to be set/unset. I noticed in the SIP trace that there are >>> several calls. It's hard to know what's what. I think your best bet is a >>> pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I >>> also noticed that you redacted IP addrs - you won't be able to do this with >>> a pcap. If security is an issue then I'd say get the pcap and let us know >>> here on the list, then those who can have a look will email you privately >>> and you can send the pcap file to them. >>> >>> -MC >>> >>> >>> On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster wrote: >>> >>>> Alright, so I admit... I'm a little rusty when it comes to NAT, etc. >>>> I've only set up FS so far on machines with no NAT, so this is sort of a >>>> new experience for me. >>>> >>>> I have a FreeSWITCH server located on the same local network as all of >>>> my phones here at the house. When I try to make a call to Flowroute, after >>>> about 30 seconds the call drops. It also does the exact same thing when I >>>> call a buddy's server directly via SIP. >>>> >>>> Here's a siptrace of the call (I didn't think that the actual FS log >>>> would be much help): >>>> http://pastebin.freeswitch.org/18697 >>>> >>>> ...and here's a paste of 'sofia status': >>>> http://pastebin.freeswitch.org/18698 >>>> >>>> ...and just for good measure, here's a paste of vars.xml: >>>> http://pastebin.freeswitch.org/18699 >>>> >>>> >>>> -- >>>> Brian D. Foster >>>> Endigo Computer LLC >>>> Email: bdfoster at endigotech.com >>>> Phone: 317-800-7876 >>>> Indianapolis, Indiana, USA >>>> >>>> This message contains confidential information and is intended for >>>> those listed in the "To:", "CC:", and/or "BCC:" fields of the message >>>> header. If you are not the intended recipient you are notified that >>>> disclosing, copying, distributing or taking any action in reliance on the >>>> contents of this information is strictly prohibited. E-mail transmission >>>> cannot be guaranteed to be secure or error-free as information could be >>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or >>>> contain viruses. The sender therefore does not accept liability for any >>>> errors or omissions in the contents of this message, which arise as a >>>> result of e-mail transmission. If verification is required please request a >>>> hard-copy version. >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/e211e53f/attachment.html From bdfoster at endigotech.com Tue Mar 20 20:53:22 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 20 Mar 2012 13:53:22 -0400 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: Message-ID: Michael, I don't know if it's Grandstream specific. Here's a pcap using a SPA921 calling my buddy's server sip direct: http://da1.endigovoip.com/dump2.pcap (Warning: Listener discretion advised.) -BDF On Tue, Mar 20, 2012 at 1:15 PM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > Your flowroute credentials will be hashed in any case. Thanks for getting > those details for us, I'll look into it as soon as I get a chance. > > > On 20 March 2012 17:00, Brian Foster wrote: > >> Andrew, >> >> root at homeserver:/usr/local/stund# ./client stunserver.org >> STUN client version 0.97 >> Primary: Independent Mapping, Independent Filter, preserves ports, will >> hairpin >> Return value is 0x000003 >> >> http://da1.endigovoip.com/dump.pcap >> >> Kristian, >> >> http://pastebin.freeswitch.org/18708 >> >> Michael, >> >> I did replace the IP's for security purposes, but now I've realized that >> it's needed and it's not really that big of a deal. I'll end up changing >> the Flowroute creds after this is fixed up. The prior siptrace is exactly >> one call (two legs). I don't think it's a carrier issue, as I've tried >> calling a buddy's server direct sip with the same issues. >> >> -BDF >> >> On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins wrote: >> >>> We have scores of machines behind NAT talking to Flowroute with no >>> problems, so there's got to be something potentially non-obvious but easy >>> that needs to be set/unset. I noticed in the SIP trace that there are >>> several calls. It's hard to know what's what. I think your best bet is a >>> pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I >>> also noticed that you redacted IP addrs - you won't be able to do this with >>> a pcap. If security is an issue then I'd say get the pcap and let us know >>> here on the list, then those who can have a look will email you privately >>> and you can send the pcap file to them. >>> >>> -MC >>> >>> >>> On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster wrote: >>> >>>> Alright, so I admit... I'm a little rusty when it comes to NAT, etc. >>>> I've only set up FS so far on machines with no NAT, so this is sort of a >>>> new experience for me. >>>> >>>> I have a FreeSWITCH server located on the same local network as all of >>>> my phones here at the house. When I try to make a call to Flowroute, after >>>> about 30 seconds the call drops. It also does the exact same thing when I >>>> call a buddy's server directly via SIP. >>>> >>>> Here's a siptrace of the call (I didn't think that the actual FS log >>>> would be much help): >>>> http://pastebin.freeswitch.org/18697 >>>> >>>> ...and here's a paste of 'sofia status': >>>> http://pastebin.freeswitch.org/18698 >>>> >>>> ...and just for good measure, here's a paste of vars.xml: >>>> http://pastebin.freeswitch.org/18699 >>>> >>>> >>>> -- >>>> Brian D. Foster >>>> Endigo Computer LLC >>>> Email: bdfoster at endigotech.com >>>> Phone: 317-800-7876 >>>> Indianapolis, Indiana, USA >>>> >>>> This message contains confidential information and is intended for >>>> those listed in the "To:", "CC:", and/or "BCC:" fields of the message >>>> header. If you are not the intended recipient you are notified that >>>> disclosing, copying, distributing or taking any action in reliance on the >>>> contents of this information is strictly prohibited. E-mail transmission >>>> cannot be guaranteed to be secure or error-free as information could be >>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or >>>> contain viruses. The sender therefore does not accept liability for any >>>> errors or omissions in the contents of this message, which arise as a >>>> result of e-mail transmission. If verification is required please request a >>>> hard-copy version. >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Andrew Cassidy BSc (Hons) MBCS > Managing Director; Cassidy Web Services Ltd > T: 03300 100 960 F: 03300 100 961 > E: andrew at cassidywebservices.co.uk > W: www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/b94dacf7/attachment-0001.html From garbytrash at gmail.com Tue Mar 20 20:53:37 2012 From: garbytrash at gmail.com (Zenny) Date: Tue, 20 Mar 2012 17:53:37 +0000 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: References: <20120320151304.2e0c44b4@anubis.defcon1> Message-ID: Just to test, I directly specified /usr/pgsql-9.1/include/libpq-fe.h in /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c It seemed to have progressed a bit further, but spits out the error: Creating mod_cdr_pg_csv.la... /usr/bin/ld: cannot find -lpq collect2: ld returned 1 exit status cat: .libs/mod_cdr_pg_csv.log: No such file or directory make[5]: *** [mod_cdr_pg_csv.la] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_cdr_pg_csv-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 :-( On 3/20/12, Zenny wrote: > Hi Ken: > > Thanks for your input, but even after inserting two lines you advised > in Makefile pops out the same error. > > # grep -n libpq-fe.h make-error1.txt > 4:/usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:40:22: > error: libpq-fe.h: No such file or directory > > However, I did see the a problem with ./configure also. > # grep -n libpq-fe.h configure-output1.txt > 1440:checking libpq-fe.h usability... no > 1441:checking libpq-fe.h presence... no > 1442:checking for libpq-fe.h... no > > What I meant is if ./configure is reporting the unavailability of > libpq-fe.h, is that possible by just making changes in the Makefile? I > know I may sound pretty dumb! ;-) > > /z > > > > On 3/20/12, Ken Rice wrote: >> Actually there just needs to be a pg_config check in there... This will >> completely solve the problem in a cross platform sorta way >> >> Something like this in the Makefile >> >> LOCAL_CFLAGS=-I`pg_config --pkgincludedir` -I/usr/include ${EXTRA_FLAGS} >> LOCAL_LDFLAGS=`pg_config --libdir`/libpq.so >> >> Maybe a few extra things in there. But that will clean up that pathing >> issue >> and this is cross platform for properly installed postgresql >> >> K >> >> >> On 3/20/12 9:13 AM, "Bzzz" wrote: >> >>> On Tue, 20 Mar 2012 08:57:56 -0500 >>> Ken Rice wrote: >>> >>>> That's not a good way to solve the problem... The problem after a quick >>>> look >>>> is hardcoded paths for pgsql... >>> >>> Hmm, I didn't check but there's also usually a configure option to >>> point to unusual paths (some' like: --with-thelibIwant=/....) >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From buscom123+fs at gmail.com Tue Mar 20 21:00:53 2012 From: buscom123+fs at gmail.com (R H) Date: Tue, 20 Mar 2012 12:00:53 -0600 Subject: [Freeswitch-users] Mod_CallCenter and HA Failover Message-ID: Hey Guys, First I want to say that FreeSwitch has been an amazing platform to work on. Compared to asterisk I have been extremely impressed. In the callcenter.conf.xml I noticed a note that said " **" There is not much more elaboration on the topic anywhere to be found so I was wondering if there was ever a plan to support HA Failover in callcenter. I recently spent some time working on a successful HA failover setup in hopes to ensure uptime on our servers. Basic call flows work perfectly but CallCenter definitely shows negative results. Any feedback you can give me about plans for the module? Thanks! Ryan PS - If it's welcome, I'm planning on adding the notes of how I got HA Failover working with 2 servers to the FreeSwitch wiki. The documentation there is a bit incomplete. I'll try to do that as soon as I feel the notes are complete. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/fb2ebe28/attachment.html From lazyvirus at gmx.com Tue Mar 20 21:04:08 2012 From: lazyvirus at gmx.com (Bzzz) Date: Tue, 20 Mar 2012 19:04:08 +0100 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: References: <20120320151304.2e0c44b4@anubis.defcon1> Message-ID: <20120320190408.4ba4db24@anubis.defcon1> On Tue, 20 Mar 2012 17:53:37 +0000 Zenny wrote: > Just to test, I directly specified /usr/pgsql-9.1/include/libpq-fe.h > in /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c > > It seemed to have progressed a bit further, but spits out the error: > > Creating mod_cdr_pg_csv.la... > /usr/bin/ld: cannot find -lpq > collect2: ld returned 1 exit status > cat: .libs/mod_cdr_pg_csv.log: No such file or directory > make[5]: *** [mod_cdr_pg_csv.la] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_cdr_pg_csv-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 Seems normal: you give it the pq header, so it can compile, but it can't link against nothing. -- QOTD: All I want is more than my fair share. From garbytrash at gmail.com Tue Mar 20 21:09:49 2012 From: garbytrash at gmail.com (Zenny) Date: Tue, 20 Mar 2012 18:09:49 +0000 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: <20120320190408.4ba4db24@anubis.defcon1> References: <20120320151304.2e0c44b4@anubis.defcon1> <20120320190408.4ba4db24@anubis.defcon1> Message-ID: On 3/20/12, Bzzz wrote: > On Tue, 20 Mar 2012 17:53:37 +0000 > Zenny wrote: > >> Just to test, I directly specified /usr/pgsql-9.1/include/libpq-fe.h >> in >> /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c >> >> It seemed to have progressed a bit further, but spits out the error: >> >> Creating mod_cdr_pg_csv.la... >> /usr/bin/ld: cannot find -lpq >> collect2: ld returned 1 exit status >> cat: .libs/mod_cdr_pg_csv.log: No such file or directory >> make[5]: *** [mod_cdr_pg_csv.la] Error 1 >> make[4]: *** [all] Error 1 >> make[3]: *** [mod_cdr_pg_csv-all] Error 1 >> make[2]: *** [all-recursive] Error 1 >> make[1]: *** [all-recursive] Error 1 >> make: *** [all] Error 2 > > Seems normal: you give it the pq header, so it can compile, but it > can't link against nothing. Right. But how could that be solved? I am just struggling with this for a while. Linking directly the /usr/include with /user/pgsql-9.1/include? that may affect other applications I guess. > > -- > QOTD: > All I want is more than my fair share. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From bdfoster at endigotech.com Tue Mar 20 21:11:29 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 20 Mar 2012 14:11:29 -0400 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: Message-ID: Andrew, I did check this on the HT286, it was unset. I set it to the FS server IP, but I got the same results. -BDF On Tue, Mar 20, 2012 at 1:35 PM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > My thought at this stage is that the ATA is trying to send the ACK > directly and not to FreeSWITCH using the Contact header. If I recall > correctly, GS's can be configured with both a registrar and an outbound > proxy. Try setting the outbound proxy to the FreeSWITCH box, see what > happens. > > > On 20 March 2012 17:15, Michael Collins wrote: > >> This is why I love Wireshark so much! Look at this purdee graph it makes: >> >> >> >> >> See all those 200 OK's that your FS is sending to the Grandstream? Guess >> what your GS is sending in response to those: NADA! If you look at the BYE >> that FS sends to the GS you'll even see the reason: >> >> SIP;cause=408;text=\"ACK Timeout\" >> >> FS never gets an ACK back from the GS. So the question is: why? I'm >> unfamiliar with the GS so I'll have to defer to those with more experience >> than I. However, I think you'll find that tcpdumps and analyzing w/ >> Wireshark is extremely helpful. (Open the pcap, click "Telephony > VoIP >> calls" and then a new dialog opens. In this case it shows two calls - >> meaning two call legs. Click "Select All" then click "Flow" and you'll get >> the cool graph. Click around and see what other stuff does. :) >> >> I'm thinking of doing a FreeSWITCH conference call presentation on the >> subject of collecting pcaps and doing Wireshark analysis. Let me know if >> you guys think that's a good presentation. >> >> -MC >> >> >> >> On Tue, Mar 20, 2012 at 10:00 AM, Brian Foster wrote: >> >>> Andrew, >>> >>> root at homeserver:/usr/local/stund# ./client stunserver.org >>> STUN client version 0.97 >>> Primary: Independent Mapping, Independent Filter, preserves ports, will >>> hairpin >>> Return value is 0x000003 >>> >>> http://da1.endigovoip.com/dump.pcap >>> >>> Kristian, >>> >>> http://pastebin.freeswitch.org/18708 >>> >>> Michael, >>> >>> I did replace the IP's for security purposes, but now I've realized that >>> it's needed and it's not really that big of a deal. I'll end up changing >>> the Flowroute creds after this is fixed up. The prior siptrace is exactly >>> one call (two legs). I don't think it's a carrier issue, as I've tried >>> calling a buddy's server direct sip with the same issues. >>> >>> -BDF >>> >>> On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins wrote: >>> >>>> We have scores of machines behind NAT talking to Flowroute with no >>>> problems, so there's got to be something potentially non-obvious but easy >>>> that needs to be set/unset. I noticed in the SIP trace that there are >>>> several calls. It's hard to know what's what. I think your best bet is a >>>> pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I >>>> also noticed that you redacted IP addrs - you won't be able to do this with >>>> a pcap. If security is an issue then I'd say get the pcap and let us know >>>> here on the list, then those who can have a look will email you privately >>>> and you can send the pcap file to them. >>>> >>>> -MC >>>> >>>> >>>> On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster >>> > wrote: >>>> >>>>> Alright, so I admit... I'm a little rusty when it comes to NAT, etc. >>>>> I've only set up FS so far on machines with no NAT, so this is sort of a >>>>> new experience for me. >>>>> >>>>> I have a FreeSWITCH server located on the same local network as all of >>>>> my phones here at the house. When I try to make a call to Flowroute, after >>>>> about 30 seconds the call drops. It also does the exact same thing when I >>>>> call a buddy's server directly via SIP. >>>>> >>>>> Here's a siptrace of the call (I didn't think that the actual FS log >>>>> would be much help): >>>>> http://pastebin.freeswitch.org/18697 >>>>> >>>>> ...and here's a paste of 'sofia status': >>>>> http://pastebin.freeswitch.org/18698 >>>>> >>>>> ...and just for good measure, here's a paste of vars.xml: >>>>> http://pastebin.freeswitch.org/18699 >>>>> >>>>> >>>>> -- >>>>> Brian D. Foster >>>>> Endigo Computer LLC >>>>> Email: bdfoster at endigotech.com >>>>> Phone: 317-800-7876 >>>>> Indianapolis, Indiana, USA >>>>> >>>>> This message contains confidential information and is intended for >>>>> those listed in the "To:", "CC:", and/or "BCC:" fields of the message >>>>> header. If you are not the intended recipient you are notified that >>>>> disclosing, copying, distributing or taking any action in reliance on the >>>>> contents of this information is strictly prohibited. E-mail transmission >>>>> cannot be guaranteed to be secure or error-free as information could be >>>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or >>>>> contain viruses. The sender therefore does not accept liability for any >>>>> errors or omissions in the contents of this message, which arise as a >>>>> result of e-mail transmission. If verification is required please request a >>>>> hard-copy version. >>>>> >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Brian D. Foster >>> Endigo Computer LLC >>> Email: bdfoster at endigotech.com >>> Phone: 317-800-7876 >>> Indianapolis, Indiana, USA >>> >>> This message contains confidential information and is intended for those >>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >>> you are not the intended recipient you are notified that disclosing, >>> copying, distributing or taking any action in reliance on the contents of >>> this information is strictly prohibited. E-mail transmission cannot be >>> guaranteed to be secure or error-free as information could be intercepted, >>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >>> The sender therefore does not accept liability for any errors or omissions >>> in the contents of this message, which arise as a result of e-mail >>> transmission. If verification is required please request a hard-copy >>> version. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Andrew Cassidy BSc (Hons) MBCS > Managing Director; Cassidy Web Services Ltd > T: 03300 100 960 F: 03300 100 961 > E: andrew at cassidywebservices.co.uk > W: www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/a3085b03/attachment-0001.html From lazyvirus at gmx.com Tue Mar 20 21:18:57 2012 From: lazyvirus at gmx.com (Bzzz) Date: Tue, 20 Mar 2012 19:18:57 +0100 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: References: <20120320151304.2e0c44b4@anubis.defcon1> <20120320190408.4ba4db24@anubis.defcon1> Message-ID: <20120320191857.6abaac18@anubis.defcon1> On Tue, 20 Mar 2012 18:09:49 +0000 Zenny wrote: > > Right. But how could that be solved? I am just struggling with this for a while. > > Linking directly the /usr/include with /user/pgsql-9.1/include? that > may affect other applications I guess. Hmm, may be you should test the modifications Ken mentioned (into Makefile)? -- No matter how clever the hardware boys are, the software boys piss it away. From krice at freeswitch.org Tue Mar 20 21:27:38 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 20 Mar 2012 13:27:38 -0500 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: Message-ID: Those 2 lines were not guarenteed to fix the problem but an example of what the fix might look like... You'll need to adjust it for libpq-fe vs libpq And theres probably other flags that might need to be set as well... K On 3/20/12 12:30 PM, "Zenny" wrote: > Hi Ken: > > Thanks for your input, but even after inserting two lines you advised > in Makefile pops out the same error. > > # grep -n libpq-fe.h make-error1.txt > 4:/usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c:4 > 0:22: > error: libpq-fe.h: No such file or directory > > However, I did see the a problem with ./configure also. > # grep -n libpq-fe.h configure-output1.txt > 1440:checking libpq-fe.h usability... no > 1441:checking libpq-fe.h presence... no > 1442:checking for libpq-fe.h... no > > What I meant is if ./configure is reporting the unavailability of > libpq-fe.h, is that possible by just making changes in the Makefile? I > know I may sound pretty dumb! ;-) > > /z > > > > On 3/20/12, Ken Rice wrote: >> Actually there just needs to be a pg_config check in there... This will >> completely solve the problem in a cross platform sorta way >> >> Something like this in the Makefile >> >> LOCAL_CFLAGS=-I`pg_config --pkgincludedir` -I/usr/include ${EXTRA_FLAGS} >> LOCAL_LDFLAGS=`pg_config --libdir`/libpq.so >> >> Maybe a few extra things in there. But that will clean up that pathing issue >> and this is cross platform for properly installed postgresql >> >> K >> >> >> On 3/20/12 9:13 AM, "Bzzz" wrote: >> >>> On Tue, 20 Mar 2012 08:57:56 -0500 >>> Ken Rice wrote: >>> >>>> That's not a good way to solve the problem... The problem after a quick >>>> look >>>> is hardcoded paths for pgsql... >>> >>> Hmm, I didn't check but there's also usually a configure option to >>> point to unusual paths (some' like: --with-thelibIwant=/....) >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at freeswitch.org Tue Mar 20 21:29:43 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 20 Mar 2012 13:29:43 -0500 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: Message-ID: This error is common if ldd cant find libpq... Does it show up in " ldconfig -v |grep pq" if not you'll need to add the directory those libs live in to your ld.so.conf or ld.so.conf.d as is proper on your platform On 3/20/12 1:09 PM, "Zenny" wrote: > On 3/20/12, Bzzz wrote: >> On Tue, 20 Mar 2012 17:53:37 +0000 >> Zenny wrote: >> >>> Just to test, I directly specified /usr/pgsql-9.1/include/libpq-fe.h >>> in >>> /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c >>> >>> It seemed to have progressed a bit further, but spits out the error: >>> >>> Creating mod_cdr_pg_csv.la... >>> /usr/bin/ld: cannot find -lpq >>> collect2: ld returned 1 exit status >>> cat: .libs/mod_cdr_pg_csv.log: No such file or directory >>> make[5]: *** [mod_cdr_pg_csv.la] Error 1 >>> make[4]: *** [all] Error 1 >>> make[3]: *** [mod_cdr_pg_csv-all] Error 1 >>> make[2]: *** [all-recursive] Error 1 >>> make[1]: *** [all-recursive] Error 1 >>> make: *** [all] Error 2 >> >> Seems normal: you give it the pq header, so it can compile, but it >> can't link against nothing. > > Right. But how could that be solved? I am just struggling with this for a > while. > > Linking directly the /usr/include with /user/pgsql-9.1/include? that > may affect other applications I guess. > >> >> -- >> QOTD: >> All I want is more than my fair share. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From garbytrash at gmail.com Tue Mar 20 21:30:22 2012 From: garbytrash at gmail.com (Zenny) Date: Tue, 20 Mar 2012 18:30:22 +0000 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: <20120320191857.6abaac18@anubis.defcon1> References: <20120320151304.2e0c44b4@anubis.defcon1> <20120320190408.4ba4db24@anubis.defcon1> <20120320191857.6abaac18@anubis.defcon1> Message-ID: On 3/20/12, Bzzz wrote: > On Tue, 20 Mar 2012 18:09:49 +0000 > Zenny wrote: > >> >> Right. But how could that be solved? I am just struggling with this for a >> while. >> >> Linking directly the /usr/include with /user/pgsql-9.1/include? that >> may affect other applications I guess. > > Hmm, may be you should test the modifications Ken mentioned (into > Makefile)? Already did the changes as Ken suggested, but with the same error!!! :-( > > -- > No matter how clever the hardware boys are, the software boys piss > it away. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From garbytrash at gmail.com Tue Mar 20 21:51:47 2012 From: garbytrash at gmail.com (Zenny) Date: Tue, 20 Mar 2012 18:51:47 +0000 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: References: Message-ID: Thank Ken: I do see : # ldconfig -v | grep pq libpqwalreceiver.so -> libpqwalreceiver.so libpq.so.5 -> libpq.so.5.4 libipq.so.0 -> libipq.so.0.0.0 where do I need to add manually? /etc/ld.so.conf.d/? On 3/20/12, Ken Rice wrote: > This error is common if ldd cant find libpq... Does it show up in " ldconfig > -v |grep pq" if not you'll need to add the directory those libs live in to > your ld.so.conf or ld.so.conf.d as is proper on your platform > > > On 3/20/12 1:09 PM, "Zenny" wrote: > >> On 3/20/12, Bzzz wrote: >>> On Tue, 20 Mar 2012 17:53:37 +0000 >>> Zenny wrote: >>> >>>> Just to test, I directly specified /usr/pgsql-9.1/include/libpq-fe.h >>>> in >>>> /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c >>>> >>>> It seemed to have progressed a bit further, but spits out the error: >>>> >>>> Creating mod_cdr_pg_csv.la... >>>> /usr/bin/ld: cannot find -lpq >>>> collect2: ld returned 1 exit status >>>> cat: .libs/mod_cdr_pg_csv.log: No such file or directory >>>> make[5]: *** [mod_cdr_pg_csv.la] Error 1 >>>> make[4]: *** [all] Error 1 >>>> make[3]: *** [mod_cdr_pg_csv-all] Error 1 >>>> make[2]: *** [all-recursive] Error 1 >>>> make[1]: *** [all-recursive] Error 1 >>>> make: *** [all] Error 2 >>> >>> Seems normal: you give it the pq header, so it can compile, but it >>> can't link against nothing. >> >> Right. But how could that be solved? I am just struggling with this for a >> while. >> >> Linking directly the /usr/include with /user/pgsql-9.1/include? that >> may affect other applications I guess. >> >>> >>> -- >>> QOTD: >>> All I want is more than my fair share. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mitch.capper at gmail.com Tue Mar 20 22:31:57 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Tue, 20 Mar 2012 12:31:57 -0700 Subject: [Freeswitch-users] Mod_CallCenter and HA Failover In-Reply-To: References: Message-ID: Definitely write up the HA findings in the wiki when you can it certainly is always helpful and appreciated to get more documentation. ~Mitch From kris at kriskinc.com Tue Mar 20 23:52:34 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 20 Mar 2012 16:52:34 -0400 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: Message-ID: Expanding on this a bit: It appears as though the internal profile of FreeSWITCH is providing the external STUN discovered IP as its local IP in the Contact header in the 200 OK. ?The GS is more than likely sending the ACK to this address (which is correct). Try one (or both) of these things: 1) ?Check your localnet/localnet.auto ACL definition and make sure 192.168.1.0/24 (at least) is included. ?That should prevent FreeSWITCH from using ext-*-ip to those networks. ?Also check and make sure that local-network-acl is set to localnet or localnet.auto on your internal profile. 2) ?Disable external STUN address discovery completely on the internal profile. On Tue, Mar 20, 2012 at 1:15 PM, Michael Collins wrote: > > This is why I love Wireshark so much! Look at this purdee graph it makes: > > > > > See all those 200 OK's that your FS is sending to the Grandstream? Guess what your GS is sending in response to those: NADA! If you look at the BYE that FS sends to the GS you'll even see the reason: > > SIP;cause=408;text=\"ACK Timeout\" > > FS never gets an ACK back from the GS. So the question is: why? I'm unfamiliar with the GS so I'll have to defer to those with more experience than I. However, I think you'll find that tcpdumps and analyzing w/ Wireshark is extremely helpful. (Open the pcap, click "Telephony > VoIP calls" and then a new dialog opens. In this case it shows two calls - meaning two call legs. Click "Select All" then click "Flow" and you'll get the cool graph. Click around and see what other stuff does. :) > > I'm thinking of doing a FreeSWITCH conference call presentation on the subject of collecting pcaps and doing Wireshark analysis. Let me know if you guys think that's a good presentation. > > -MC > > > > On Tue, Mar 20, 2012 at 10:00 AM, Brian Foster wrote: >> >> Andrew, >> >> root at homeserver:/usr/local/stund# ./client stunserver.org >> STUN client version 0.97 >> Primary: Independent Mapping, Independent Filter, preserves ports, will hairpin >> Return value is 0x000003 >> >> http://da1.endigovoip.com/dump.pcap >> >> Kristian, >> >> http://pastebin.freeswitch.org/18708 >> >> Michael, >> >> I did replace the IP's for security purposes, but now I've realized that it's needed and it's not really that big of a deal. I'll end up changing the Flowroute creds after this is fixed up. The prior siptrace is exactly one call (two legs). I don't think it's a carrier issue, as I've tried calling a buddy's server direct sip with the same issues. >> >> -BDF >> >> On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins wrote: >>> >>> We have scores of machines behind NAT talking to Flowroute with no problems, so there's got to be something potentially non-obvious but easy that needs to be set/unset. I noticed in the SIP trace that there are several calls. It's hard to know what's what. I think your best bet is a pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I also noticed that you redacted IP addrs - you won't be able to do this with a pcap. If security is an issue then I'd say get the pcap and let us know here on the list, then those who can have a look will email you privately and you can send the pcap file to them. >>> >>> -MC >>> >>> >>> On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster wrote: >>>> >>>> Alright, so I admit... I'm a little rusty when it comes to NAT, etc. I've only set up FS so far on machines with no NAT, so this is sort of a new experience for me. >>>> >>>> I have a FreeSWITCH server located on the same local network as all of my phones here at the house. When I try to make a call to Flowroute, after about 30 seconds the call drops. It also does the exact same thing when I call a buddy's server directly via SIP. >>>> >>>> Here's a siptrace of the call (I didn't think that the actual FS log would be much help): >>>> http://pastebin.freeswitch.org/18697 >>>> >>>> ...and here's a paste of 'sofia status': >>>> http://pastebin.freeswitch.org/18698 >>>> >>>> ...and just for good measure, here's a paste of vars.xml: >>>> http://pastebin.freeswitch.org/18699 >>>> >>>> >>>> -- >>>> Brian D. Foster >>>> Endigo Computer LLC >>>> Email: bdfoster at endigotech.com >>>> Phone: 317-800-7876 >>>> Indianapolis, Indiana, USA >>>> >>>> This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From shane.harrison at paragon.co.nz Tue Mar 20 23:01:21 2012 From: shane.harrison at paragon.co.nz (Shane Harrison) Date: Wed, 21 Mar 2012 09:01:21 +1300 Subject: [Freeswitch-users] TDM400P spans In-Reply-To: References: <5227389.xYDaNXzQut@axp> Message-ID: I can't point you to any great source of documentation other than the FreeTDM Wiki pages which cover the configuration reasonably well as long as you understand the concepts of FXO and FXS. I too found it required a bit of studying. The key thing I think to remember is that FreeTDM justs talks to a hardware abstraction layer eg. dahdi. You need to ensure that this hardware abstraction layer is in place and working. In the case of dahdi I had to download and build the drivers. There are a set of tools to ensure that the hardware is seen correctly and is available via the dahdi interfaces. Once you have that working then it is just the FreeTDM config files (two I believe) that need to configured for the FreeTDM module to run correctly and make the analog "channels" available to your dialplans. The syntax for referencing the channels I had to guess at since I couldn't find any ultimate reference for that with FreeTDM examples. HTH Shane On Tue, Mar 20, 2012 at 10:20 PM, Fu Jiantao wrote: > Thanks Moises! > > So span is a logic concept, while channel is corresponding to the port(fxo > or fxs) on the card, am i right? > > I'm new to the freeswitch and asterisk world, and found the most difficult > part is relating to the PSTN, I'm not familiar with PSTN, and I've read > asterisk book and freeswitch book, and learn the basic of PSTN, but that > seems not enough, for example, I found I can't figure out how to use > FreeTDM, little are relating to this topic, is there any good references on > this topic? > > I'm trying to port the io module chan_fxin asterisk win32into freeTDM, but I found I need to know more about the above topic. > > > 2012/3/20 Moises Silva > >> On Sun, Mar 18, 2012 at 10:58 AM, Jeromy wrote: >> >>> > I'm trying to convert from Asterisk to Freeswitch, and have a question >>> about >>> spans. >>> > >>> > I read the freetdm configuration example for TDM400 >>> > and am wondering what exactly is a span. >>> > >>> > I have 2 TDM400P cards in my computer, >>> > one card has 2 FXS modules for connecting analog phones. >>> > The other card has 3 FXO modules for connecting to 3 PSTN trunk lines. >>> > >>> > # lsdahdi >>> > ### Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" >>> > 1 FXS FXOKS (In use) (EC: OSLEC - INACTIVE) >>> > 2 FXS FXOKS (In use) (EC: OSLEC - INACTIVE) >>> > 3 unknown Reserved >>> > 4 unknown Reserved >>> > ### Span 2: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER) >>> > 5 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) >>> > 6 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) >>> > 7 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) >>> > 8 unknown Reserved >>> > >>> > Do I configure freetdm as in the example for TDM400 >>> > with separate spans for each module like the following? >>> > or should I have just two spans as shown by lsdahsi? >>> > Thanks. >>> > >>> >>> Hi Jeremy >>> >>> Do you find the answer to this question? >>> >>> I'm also very confuse on the span and channel in both asterisk and >>> freeswitch, and I've found little explaination on this. >>> >>> " A span is a logical unit that represents a group of channels. >>> With digital telephony, a span usually represents a physical port >>> on the card. >>> If the system has only one such card with a single port, so it is >>> referred to >>> as span 1. " >>> >>> Could anyone help on this? >>> >>> >> At the very least you need 2 freetdm span configurations because you >> cannot mix FXS and FXO channels in the same FreeTDM logical span. >> >> http://wiki.freeswitch.org/wiki/FreeTDM#DAHDI_mode >> >> For the given hardware output, I'd simply create 2 spans, one for all the >> fxo channels and one for all the fxs channels. >> >> *Moises Silva >> **Manager, Software Engineering*** >> >> msilva at sangoma.com >> >> Sangoma Technologies >> >> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >> >> >> t. +1 800 388 2475 (N. America) >> >> t. +1 905 474 1990 x128 >> >> f. +1 905 474 9223 >> >> >> >> ** >> >> Products >> | Solutions >> | Events >> | Contact >> | Wiki >> | Facebook >> | Twitter`| >> | YouTube >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Paragon Electronic Design Ltd L6 Crest House 92 Queens Drive P0 Box 30449 Lower Hutt 5040 +64 4 5703870 Extn 875 +64 21 608919 (mobile) "Solving your problems with the right technology" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/2a15b1bb/attachment.html From tculjaga at gmail.com Wed Mar 21 00:51:33 2012 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 20 Mar 2012 22:51:33 +0100 Subject: [Freeswitch-users] TLS on FS In-Reply-To: References: Message-ID: hello Mitch, loglevel was already on 9... anyhow i made csipsimple on android to work. going to try with bria and FS softphones. i admit, im totally new to ssl/generating and deploying certs and its quite confusing to me.... anyhow once i figure out what i need to deploy on client and what on server, im going to update the wiki to help dummies as myself configure it. my problem with csipsimple was that i was keeping the usb cable plugged into the phone and by dong that android didn't have sdcard mounted!!! meaning csipsimple was unable to access cert files.... grrrr anyhow, going to give it a try tomorrow on a windows based client and hope to make it working. thanks, T. On Tue, Mar 20, 2012 at 5:05 PM, Mitch Capper wrote: > Hi Tihomir, > I added a section at > http://wiki.freeswitch.com/wiki/SIP_TLS#Further_Debugging_Steps for > help with sip debugging. if you can collect the logs mentioned there > along with do a sofia status when sofia is running and paste that we > can start there. You may also want to try one of the FS softphones to > ensure you can get a client working first. > > ~Mitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/4d775704/attachment-0001.html From tculjaga at gmail.com Wed Mar 21 00:55:36 2012 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 20 Mar 2012 22:55:36 +0100 Subject: [Freeswitch-users] rad_auth faild to load modul In-Reply-To: <4F687D5B.7050200@softnet.si> References: <4F687D5B.7050200@softnet.si> Message-ID: hi Miha, please always use freeradius-client-1.1.6 along with mod_rad_auth. looks like a different freeradius-client library version. On Tue, Mar 20, 2012 at 1:51 PM, Miha wrote: > Hi, > > I have installed Freeswitch (FreeSWITCH Version 1.0.head (git-9d3401e > 2012-03-19 20-06-36 -0500)) on latest Centos (6.2). > > After I installed freeradius-client as is written on wiki, I tried to load > it. I am getting this error: > > freeswitch at localhost.localdomain> load mod_rad_auth > 2012-03-20 13:46:03.386561 [INFO] mod_enum.c:812 ENUM Reloaded > 2012-03-20 13:46:03.386561 [INFO] switch_time.c:1128 Timezone reloaded 530 > definitions > 2012-03-20 13:46:03.386561 [CRIT] switch_loadable_module.c:1295 Error > Loading module /usr/local/freeswitch/mod/mod_rad_auth.so > **/usr/local/freeswitch/mod/mod_rad_auth.so: undefined symbol: > rc_conf_str** > > I am having rad_auth installed on FS (FreeSWITCH Version 1.0.head > (git-00de8e6 2011-11-01 17-27-13 -0600)), (Centos 5.6) and is working > properly. > > Any suggestion how what could be causing problem that I can not load it on > centos 6.2? > > Thank you! > > Regards, > MIha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/e63c4a06/attachment.html From freeswitch-list at puzzled.xs4all.nl Wed Mar 21 01:14:04 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Tue, 20 Mar 2012 23:14:04 +0100 Subject: [Freeswitch-users] rad_auth faild to load modul In-Reply-To: References: <4F687D5B.7050200@softnet.si> Message-ID: <4F69012C.5020403@puzzled.xs4all.nl> On 20-03-12 22:55, Tihomir Culjaga wrote: > hi Miha, > > please always use freeradius-client-1.1.6 along with mod_rad_auth. > looks like a different freeradius-client library version. RHEL 6.2 / CentOS 6.2 comes with freeradius 2.1.10 so that is going to be a problem. Even more when looking at the freeradius website where it clearly states: "Version 1.1 - No longer maintained! As of January 2008, the version 1.1.x releases are no longer actively maintained. Version 1.1.7 was the last release in that cycle. We recommend that everyone using Version 1.1.7 (or any earlier version) upgrade to the latest 2.x release as soon as possible. If there are any security isses found in Version 1.1.7, we will release a Version 1.1.8 to fix them. However, any new features or bug fixes will not be added to the 1.1.x releases." So the freeradius 1.1.6 version you mentioned is obsolete. Imho mod_rad_auth should be updated to work with freeradius 2.1.10. I don't use mod_rad_auth but it would make sense to me if someone filed a bug on Jira asap so maybe this could get fixed before FreeSWITCH 1.2 is released. Regards, Patrick From tculjaga at gmail.com Wed Mar 21 01:34:32 2012 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 20 Mar 2012 23:34:32 +0100 Subject: [Freeswitch-users] Advice how to start with H.323 In-Reply-To: <1FFF97C269757C458224B7C895F35F1507AEE1@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1507AEE1@cantor.std.visionutv.se> Message-ID: hello my test box is a small one but works :P [root at l01sipindirpp freeswitch.git]# cat /etc/issue CentOS release 5.5 (Final) Kernel \r on an \m [root at l01sipindirpp freeswitch.git]# uname -a Linux l01sipindirpp 2.6.18-194.32.1.el5 #1 SMP Wed Jan 5 17:53:09 EST 2011 i686 i686 i386 GNU/Linux [root at l01sipindirpp freeswitch.git]# please check pastebin for details.... http://pastebin.freeswitch.org/18709 On Tue, Mar 20, 2012 at 3:58 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Thanks! > > /Peter > > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] F?r Peter Steinbach > Skickat: den 20 mars 2012 15:39 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Advice how to start with H.323 > > Wiki is updated accordingly. > > Best regards > Peter > > Am 20.03.2012 13:32, schrieb Peter Olsson: > > It would be very appreciated if you could take some time and update with > some more information on the wiki - so others could benefit from this as > well.. > > > > /Peter > > > > > > -----Ursprungligt meddelande----- > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Peter > > Steinbach > > Skickat: den 20 mars 2012 13:08 > > Till: FreeSWITCH Users Help > > ?mne: Re: [Freeswitch-users] Advice how to start with H.323 > > > > Thanks to your help, I am now a step further with 1.22.0 and ptlib-2.8.2: > > > > mod_h323 has compiled successfully. > > After relinking libh323_linux_x86_64_.so.1.22.0 the mod_h323 module > finally loaded. > > Thanks to all, who helped on this issue! > > > > Do you know of any gatekeeper avaliable for testing, where I could test > connectivity and 2-way-audio? > > I tried with http://www.voxgratia.org/ but this gateway seems to be > down. > > > > Best regards > > Peter > > > > > > > > > > Am 19.03.2012 14:33, schrieb Peter Olsson: > >> Yes, I've used those recommendations - so 1.22.0 is probably a better > choice. > >> > >> /Peter > >> > >> > >> -----Ursprungligt meddelande----- > >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Patrick > >> Lists > >> Skickat: den 19 mars 2012 14:03 > >> Till: FreeSWITCH Users Help > >> ?mne: Re: [Freeswitch-users] Advice how to start with H.323 > >> > >> On 19-03-12 11:53, Peter Steinbach wrote: > >>> I am using Ubuntu 10.04. > >>> > >>> I followed the installation procedures on the wiki page and > >>> installed > >>> > >>> ptlib-2.8.2 + h323plus-trunk > >> Not sure if it's still valid but in the past the recommendation was to > follow the versions listed at: > >> > >> http://www.gnugk.org/compiling-gnugk.html > >> > >> Which in this case means you should use H323Plus 1.22.0 and not trunk. > >> > >> Regards, > >> Patrick > >> > >> > >> > >> _____________________________________________________________________ > >> _ ___ Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us > >> e > >> rs > >> http://www.freeswitch.org > >> > >> > >> > >> > >> _____________________________________________________________________ > >> _ ___ Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us > >> e > >> rs > >> http://www.freeswitch.org > >> > > > > -- > > With kind regards > > Peter Steinbach > > > > Telefaks Services GmbH > > mailto:lists (att) telefaks.de > > Internet: www.telefaks.de > > > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f68950732764931015849! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/381e26ef/attachment-0001.html From tculjaga at gmail.com Wed Mar 21 01:41:59 2012 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 20 Mar 2012 23:41:59 +0100 Subject: [Freeswitch-users] rad_auth faild to load modul In-Reply-To: <4F69012C.5020403@puzzled.xs4all.nl> References: <4F687D5B.7050200@softnet.si> <4F69012C.5020403@puzzled.xs4all.nl> Message-ID: yap, i totally agree on that... will move forward to update it! in the mean time you can use the old library... mod_rad_auth module uses an embedded config so you can build the library and copy it directly to fs/lib directory ... no need to be system available. T. On Tue, Mar 20, 2012 at 11:14 PM, Patrick Lists < freeswitch-list at puzzled.xs4all.nl> wrote: > On 20-03-12 22:55, Tihomir Culjaga wrote: > > hi Miha, > > > > please always use freeradius-client-1.1.6 along with mod_rad_auth. > > looks like a different freeradius-client library version. > > RHEL 6.2 / CentOS 6.2 comes with freeradius 2.1.10 so that is going to > be a problem. Even more when looking at the freeradius website where it > clearly states: > > "Version 1.1 - No longer maintained! > > As of January 2008, the version 1.1.x releases are no longer actively > maintained. Version 1.1.7 was the last release in that cycle. We > recommend that everyone using Version 1.1.7 (or any earlier version) > upgrade to the latest 2.x release as soon as possible. > > If there are any security isses found in Version 1.1.7, we will release > a Version 1.1.8 to fix them. However, any new features or bug fixes will > not be added to the 1.1.x releases." > > So the freeradius 1.1.6 version you mentioned is obsolete. Imho > mod_rad_auth should be updated to work with freeradius 2.1.10. > > I don't use mod_rad_auth but it would make sense to me if someone filed > a bug on Jira asap so maybe this could get fixed before FreeSWITCH 1.2 > is released. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/632db341/attachment.html From anita.hall at simmortel.com Wed Mar 21 02:21:17 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Wed, 21 Mar 2012 04:51:17 +0530 Subject: [Freeswitch-users] How can I change the moh_sound In-Reply-To: References: Message-ID: This question was answered sometimes back in detail by Tim St. Pierre To quote: Are you using xml_curl? If so, you can have mod_conference get it's configuration from your web server instead of the static XML file. In our setup here, every time a new conference is launched, mod_conference requests config, and this is returned by a PHP script. We look at what conference number was requested, and we generate a profile on-the-fly. We have a database that defines what music, rate, energy level, recording options, etc. on a per-conference basis. If you use the static files, the profiles you create only get parsed when the XML is refreshed, so you can't use variable expansion in there, other than pre-process variables that you might have put elsewhere in the XML (ie. vars.xml). -Tim regards, Anita On Tue, Mar 20, 2012 at 10:30 PM, Michael Collins wrote: > MOH sound is a conference parameter specified in conference.conf.xml. I > believe that the only way you can change it on the fly is to use xml_curl > for your configs. > > -MC > > On Tue, Mar 20, 2012 at 1:16 AM, piyush singhai wrote: > >> Hello, >> >> I want to change the moh sound at run time for conference. can i specify >> path on the basis of any ivr. >> >> --Piyush >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/f3f8ea81/attachment.html From mitch.capper at gmail.com Wed Mar 21 03:44:53 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Tue, 20 Mar 2012 17:44:53 -0700 Subject: [Freeswitch-users] TLS on FS In-Reply-To: References: Message-ID: If some clients work thats great, it most likely means you need to ensure the common name (cn) on the certificate is the same as the sip server you are having your client connect to. In addition ensure you have all the tls security settings (certificate validation etc) turned off/down at first on FS to allow bria to connect then you can start to dial them up to optimal. ~Mitch From anita.hall at simmortel.com Wed Mar 21 04:03:10 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Wed, 21 Mar 2012 06:33:10 +0530 Subject: [Freeswitch-users] good Fax T.38 Provider Message-ID: Hi I am looking for good T.38 Providers for both inbound and outbound Fax application. Any recommendation ? (I am using mod_spandsp) regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/26a9375b/attachment.html From ga at steadfasttelecom.com Wed Mar 21 04:05:49 2012 From: ga at steadfasttelecom.com (Gilad Abada) Date: Tue, 20 Mar 2012 21:05:49 -0400 Subject: [Freeswitch-users] good Fax T.38 Provider In-Reply-To: References: Message-ID: <-8474253985176721109@unknownmsgid> Flowroute is great with inbound and pretty good with outbound fax. Sent from my mobile device. On Mar 20, 2012, at 9:04 PM, Anita Hall wrote: > Hi > > I am looking for good T.38 Providers for both inbound and outbound Fax application. > > Any recommendation ? > > (I am using mod_spandsp) > > regards, > Anita > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From luis.daniel.lucio at gmail.com Wed Mar 21 05:06:05 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Tue, 20 Mar 2012 22:06:05 -0400 Subject: [Freeswitch-users] Cli command Message-ID: Just wondering where i can read about cli commands LD -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/9aeb426e/attachment.html From mitch.capper at gmail.com Wed Mar 21 05:06:34 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Tue, 20 Mar 2012 19:06:34 -0700 Subject: [Freeswitch-users] good Fax T.38 Provider In-Reply-To: <-8474253985176721109@unknownmsgid> References: <-8474253985176721109@unknownmsgid> Message-ID: Flowroute can get me to a good 95% or so following the wiki. Its possible to get better success rates which may just be using multiple providers. ~Mitch On Tue, Mar 20, 2012 at 6:05 PM, Gilad Abada wrote: > Flowroute is great with inbound and pretty good with outbound fax. > > Sent from my mobile device. > > On Mar 20, 2012, at 9:04 PM, Anita Hall wrote: > >> Hi >> >> I am looking for good T.38 Providers for both inbound and outbound Fax application. >> >> Any recommendation ? >> >> (I am using mod_spandsp) >> >> regards, >> Anita >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveu at coppice.org Wed Mar 21 05:20:21 2012 From: steveu at coppice.org (Steve Underwood) Date: Wed, 21 Mar 2012 10:20:21 +0800 Subject: [Freeswitch-users] good Fax T.38 Provider In-Reply-To: References: <-8474253985176721109@unknownmsgid> Message-ID: <4F693AE5.7080709@coppice.org> Hi Mitch, How you measure your 95% is critically important in assessing what to make of the number. When we quote success rates for iaxmodem or spandsp as 99.x%, that's the outcome of a lot of manual labour, going through failed calls to weed out the ones that had no potential to ever succeed. Steve On 03/21/2012 10:06 AM, Mitch Capper wrote: > Flowroute can get me to a good 95% or so following the wiki. Its > possible to get better success rates which may just be using multiple > providers. > > ~Mitch > > On Tue, Mar 20, 2012 at 6:05 PM, Gilad Abada wrote: >> Flowroute is great with inbound and pretty good with outbound fax. >> >> Sent from my mobile device. >> >> On Mar 20, 2012, at 9:04 PM, Anita Hall wrote: >> >>> Hi >>> >>> I am looking for good T.38 Providers for both inbound and outbound Fax application. >>> >>> Any recommendation ? >>> >>> (I am using mod_spandsp) >>> >>> regards, >>> Anita From singhai.piyush at gmail.com Wed Mar 21 05:51:59 2012 From: singhai.piyush at gmail.com (piyush singhai) Date: Wed, 21 Mar 2012 08:21:59 +0530 Subject: [Freeswitch-users] How can I change the moh_sound In-Reply-To: References: Message-ID: Thanks Anita, I will try xml_curl. On Wed, Mar 21, 2012 at 4:51 AM, Anita Hall wrote: > This question was answered sometimes back in detail by Tim St. Pierre > > To quote: > > Are you using xml_curl? > > If so, you can have mod_conference get it's configuration from your web > server instead of the static XML file. > > In our setup here, every time a new conference is launched, > mod_conference requests config, and this is returned by a PHP script. > We look at what conference number was requested, and we generate a > profile on-the-fly. We have a database that defines what music, rate, > energy level, recording options, etc. on a per-conference basis. > > If you use the static files, the profiles you create only get parsed > when the XML is refreshed, so you can't use variable expansion in there, > other than pre-process variables that you might have put elsewhere in > the XML (ie. vars.xml). > > -Tim > > > > regards, > Anita > > > > > On Tue, Mar 20, 2012 at 10:30 PM, Michael Collins wrote: > >> MOH sound is a conference parameter specified in conference.conf.xml. I >> believe that the only way you can change it on the fly is to use xml_curl >> for your configs. >> >> -MC >> >> On Tue, Mar 20, 2012 at 1:16 AM, piyush singhai > > wrote: >> >>> Hello, >>> >>> I want to change the moh sound at run time for conference. can i specify >>> path on the basis of any ivr. >>> >>> --Piyush >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/f90a3963/attachment.html From mitch.capper at gmail.com Wed Mar 21 07:44:21 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Tue, 20 Mar 2012 21:44:21 -0700 Subject: [Freeswitch-users] good Fax T.38 Provider In-Reply-To: <4F693AE5.7080709@coppice.org> References: <-8474253985176721109@unknownmsgid> <4F693AE5.7080709@coppice.org> Message-ID: I hear you, and I don't run a massive fax gateway or anything like that. Doing somewhere around just under dozen faxes a day or so to generally a different fax numbers each day so its mainly just seeing a variety of remote fax machines and input. Generally we try 3-4 times per fax, and fall back to a PSTN fax machine that always succeeds and just about on the first try every time. With FS/spandsp we get "Timed out waiting for the first message" or "Invalid response after sending a page" or "The call dropped prematurely" probably at least 15% of the time on the first try (t38 req on and ecm on). With successive tries though (t38 req off and ecm off) we generally get it to go through. We have also tried phaxio which I believe uses spandsp also. Frequently on numbers that we have an issue with they will have an issue with also, however after a few tries (sometimes upwards of 5 minutes later) they do seem to get them to go through. Part of the reason I think multiple carriers may just help the situation (as I am not sure what they change other than that to get it to eventually work). I have yet to find a number phaxio hasn't eventually succeed on however. Also obviously didn't mean to offend, the 95% was just to give people an idea that while it can be good it doesn't quite seem to be a replacement for a PSTN line yet. I love FS / spandsp and it really is excellent. ~Mitch From brian at freeswitch.org Wed Mar 21 07:48:03 2012 From: brian at freeswitch.org (Brian West) Date: Tue, 20 Mar 2012 23:48:03 -0500 Subject: [Freeswitch-users] good Fax T.38 Provider In-Reply-To: References: <-8474253985176721109@unknownmsgid> <4F693AE5.7080709@coppice.org> Message-ID: <8516601866571333876@unknownmsgid> Flowroute.com t38faxing.com Both work great with t.38 Sent from my iPad On Mar 20, 2012, at 11:46 PM, Mitch Capper wrote: > I hear you, and I don't run a massive fax gateway or anything like > that. Doing somewhere around just under dozen faxes a day or so to > generally a different fax numbers each day so its mainly just seeing a > variety of remote fax machines and input. Generally we try 3-4 times > per fax, and fall back to a PSTN fax machine that always succeeds and > just about on the first try every time. With FS/spandsp we get "Timed > out waiting for the first message" or "Invalid response after sending > a page" or "The call dropped prematurely" probably at least 15% of > the time on the first try (t38 req on and ecm on). With successive > tries though (t38 req off and ecm off) we generally get it to go > through. We have also tried phaxio which I believe uses spandsp > also. Frequently on numbers that we have an issue with they will have > an issue with also, however after a few tries (sometimes upwards of 5 > minutes later) they do seem to get them to go through. Part of the > reason I think multiple carriers may just help the situation (as I am > not sure what they change other than that to get it to eventually > work). I have yet to find a number phaxio hasn't eventually succeed > on however. > > Also obviously didn't mean to offend, the 95% was just to give people > an idea that while it can be good it doesn't quite seem to be a > replacement for a PSTN line yet. I love FS / spandsp and it really is > excellent. > > ~Mitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Wed Mar 21 08:51:24 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 20 Mar 2012 22:51:24 -0700 Subject: [Freeswitch-users] Cli command In-Reply-To: References: Message-ID: wiki.freeswitch.org left side under "navigation" click "API commands" -MC On Tue, Mar 20, 2012 at 7:06 PM, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > Just wondering where i can read about cli commands > > LD > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/b10844d3/attachment.html From freeswitch at zoho.com Wed Mar 21 06:18:01 2012 From: freeswitch at zoho.com (dingdong) Date: Tue, 20 Mar 2012 20:18:01 -0700 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after ~30 seconds In-Reply-To: References: Message-ID: <1363342dbcd.1079533506949741633.-8314378552953017400@zoho.com> id like to know more bout wireshark and sip debugging,please include this on the conf call topics ---- On Tue, 20 Mar 2012 10:15:20 -0700 Michael Collins <msc at freeswitch.org> wrote ---- This is why I love Wireshark so much! Look at this purdee graph it makes: See all those 200 OK's that your FS is sending to the Grandstream? Guess what your GS is sending in response to those: NADA! If you look at the BYE that FS sends to the GS you'll even see the reason: SIP;cause=408;text=\"ACK Timeout\" FS never gets an ACK back from the GS. So the question is: why? I'm unfamiliar with the GS so I'll have to defer to those with more experience than I. However, I think you'll find that tcpdumps and analyzing w/ Wireshark is extremely helpful. (Open the pcap, click "Telephony > VoIP calls" and then a new dialog opens. In this case it shows two calls - meaning two call legs. Click "Select All" then click "Flow" and you'll get the cool graph. Click around and see what other stuff does. :) I'm thinking of doing a FreeSWITCH conference call presentation on the subject of collecting pcaps and doing Wireshark analysis. Let me know if you guys think that's a good presentation. -MC On Tue, Mar 20, 2012 at 10:00 AM, Brian Foster <bdfoster at endigotech.com> wrote: Andrew, root at homeserver:/usr/local/stund# ./client stunserver.org STUN client version 0.97 Primary: Independent Mapping, Independent Filter, preserves ports, will hairpin Return value is 0x000003 http://da1.endigovoip.com/dump.pcap Kristian, http://pastebin.freeswitch.org/18708 Michael, I did replace the IP's for security purposes, but now I've realized that it's needed and it's not really that big of a deal. I'll end up changing the Flowroute creds after this is fixed up. The prior siptrace is exactly one call (two legs). I don't think it's a carrier issue, as I've tried calling a buddy's server direct sip with the same issues. -BDF On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins <msc at freeswitch.org> wrote: We have scores of machines behind NAT talking to Flowroute with no problems, so there's got to be something potentially non-obvious but easy that needs to be set/unset. I noticed in the SIP trace that there are several calls. It's hard to know what's what. I think your best bet is a pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I also noticed that you redacted IP addrs - you won't be able to do this with a pcap. If security is an issue then I'd say get the pcap and let us know here on the list, then those who can have a look will email you privately and you can send the pcap file to them. -MC On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster <bdfoster at endigotech.com> wrote: Alright, so I admit... I'm a little rusty when it comes to NAT, etc. I've only set up FS so far on machines with no NAT, so this is sort of a new experience for me. I have a FreeSWITCH server located on the same local network as all of my phones here at the house. When I try to make a call to Flowroute, after about 30 seconds the call drops. It also does the exact same thing when I call a buddy's server directly via SIP. Here's a siptrace of the call (I didn't think that the actual FS log would be much help): http://pastebin.freeswitch.org/18697 ...and here's a paste of 'sofia status': http://pastebin.freeswitch.org/18698 ...and just for good measure, here's a paste of vars.xml: http://pastebin.freeswitch.org/18699 -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120320/59095d0f/attachment-0001.html From fernandojdk at gmail.com Wed Mar 21 08:55:25 2012 From: fernandojdk at gmail.com (Fernando - NextBilling IP Solutions) Date: Wed, 21 Mar 2012 02:55:25 -0300 (Hora oficial do Brasil) Subject: [Freeswitch-users] Cli command References: Message-ID: <4F696D4D.000010.15156@FLIGHTPC> http://wiki.freeswitch.org/wiki/Mod_commands ^^ ? Atenciosamente, Importante: Esta mensagem, incluindo todo seu conte?do, cont?m informa??es confidenciais legalmente protegidas e destinadas a indiv?duo e prop?sito espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o mais breve poss?vel. As informa??es contidas nesta mensagem e em seu conte?do s?o de responsabilidade de seu autor, n?o representando necessariamente id?ias, opini?es, pensamentos ou qualquer forma de posicionamento por parte da NextBilling IP Solutions. P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." -------Original Message------- From: Michael Collins Date: 21/03/2012 02:53:02 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cli command wiki.freeswitch.org left side under "navigation" click "API commands" -MC On Tue, Mar 20, 2012 at 7:06 PM, Luis Daniel Lucio Quiroz wrote: Just wondering where i can read about cli commands LD _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/53963ccd/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 1596 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/53963ccd/attachment-0001.gif -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 20873 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/53963ccd/attachment-0001.jpe From fernandojdk at gmail.com Wed Mar 21 09:09:38 2012 From: fernandojdk at gmail.com (Fernando - NextBilling IP Solutions) Date: Wed, 21 Mar 2012 03:09:38 -0300 (Hora oficial do Brasil) Subject: [Freeswitch-users] NAT issues - Outbound Call drops after~30seconds References: <1363342dbcd.1079533506949741633.-8314378552953017400@zoho.com> Message-ID: <4F6970A2.000019.15156@FLIGHTPC> What you have in your external profile on variables ext-rtp-ip end ext-sip-ip? ? Best Regards, Fernando da Silva Santos NextBilling IP Solutions LTDA Phone: +55 21 2143-9000 MSN: fernandojdk at gmail.com www.nextbilling.com.br Rio de Janeiro, Brazil, BR Importante: Esta mensagem, incluindo todo seu conte?do, cont?m informa??es confidenciais legalmente protegidas e destinadas a indiv?duo e prop?sito espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o mais breve poss?vel. As informa??es contidas nesta mensagem e em seu conte?do s?o de responsabilidade de seu autor, n?o representando necessariamente id?ias, opini?es, pensamentos ou qualquer forma de posicionamento por parte da NextBilling IP Solutions. P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." -------Original Message------- From: dingdong Date: 21/03/2012 02:57:02 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] NAT issues - Outbound Call drops after~30seconds id like to know more bout wireshark and sip debugging,please include this on the conf call topics ---- On Tue, 20 Mar 2012 10:15:20 -0700 Michael Collins wrote ---- This is why I love Wireshark so much! Look at this purdee graph it makes: See all those 200 OK's that your FS is sending to the Grandstream? Guess what your GS is sending in response to those: NADA! If you look at the BYE that FS sends to the GS you'll even see the reason: SIP;cause=408;text=\"ACK Timeout\" FS never gets an ACK back from the GS. So the question is: why? I'm unfamiliar with the GS so I'll have to defer to those with more experience than I. However, I think you'll find that tcpdumps and analyzing w/ Wireshark is extremely helpful. (Open the pcap, click "Telephony > VoIP calls" and then a new dialog opens. In this case it shows two calls - meaning two call legs. Click "Select All" then click "Flow" and you'll get the cool graph. Click around and see what other stuff does. :) I'm thinking of doing a FreeSWITCH conference call presentation on the subject of collecting pcaps and doing Wireshark analysis. Let me know if you guys think that's a good presentation. -MC On Tue, Mar 20, 2012 at 10:00 AM, Brian Foster wrote: Andrew, root at homeserver:/usr/local/stund# ./client stunserver.org STUN client version 0.97 Primary: Independent Mapping, Independent Filter, preserves ports, will hairpin Return value is 0x000003 http://da1.endigovoip.com/dump.pcap Kristian, http://pastebin.freeswitch.org/18708 Michael, I did replace the IP's for security purposes, but now I've realized that it s needed and it's not really that big of a deal. I'll end up changing the Flowroute creds after this is fixed up. The prior siptrace is exactly one call (two legs). I don't think it's a carrier issue, as I've tried calling a buddy's server direct sip with the same issues. -BDF On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins wrote: We have scores of machines behind NAT talking to Flowroute with no problems, so there's got to be something potentially non-obvious but easy that needs to be set/unset. I noticed in the SIP trace that there are several calls. It s hard to know what's what. I think your best bet is a pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I also noticed that you redacted IP addrs - you won't be able to do this with a pcap. If security is an issue then I'd say get the pcap and let us know here on the list, then those who can have a look will email you privately and you can send the pcap file to them. -MC On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster wrote: Alright, so I admit... I'm a little rusty when it comes to NAT, etc. I've only set up FS so far on machines with no NAT, so this is sort of a new experience for me. I have a FreeSWITCH server located on the same local network as all of my phones here at the house. When I try to make a call to Flowroute, after about 30 seconds the call drops. It also does the exact same thing when I call a buddy's server directly via SIP. Here's a siptrace of the call (I didn't think that the actual FS log would be much help): http://pastebin.freeswitch.org/18697 ...and here's a paste of 'sofia status': http://pastebin.freeswitch.org/18698 ...and just for good measure, here's a paste of vars.xml: http://pastebin.freeswitch.org/18699 -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/fc8cd4e5/attachment.html From miha at softnet.si Wed Mar 21 10:10:37 2012 From: miha at softnet.si (Miha) Date: Wed, 21 Mar 2012 08:10:37 +0100 Subject: [Freeswitch-users] rad_auth faild to load modul In-Reply-To: <4F69012C.5020403@puzzled.xs4all.nl> References: <4F687D5B.7050200@softnet.si> <4F69012C.5020403@puzzled.xs4all.nl> Message-ID: <4F697EED.5030607@softnet.si> On 3/20/2012 11:14 PM, Patrick Lists wrote: > On 20-03-12 22:55, Tihomir Culjaga wrote: >> hi Miha, >> >> please always use freeradius-client-1.1.6 along with mod_rad_auth. >> looks like a different freeradius-client library version. > RHEL 6.2 / CentOS 6.2 comes with freeradius 2.1.10 so that is going to > be a problem. Even more when looking at the freeradius website where it > clearly states: > > "Version 1.1 - No longer maintained! > > As of January 2008, the version 1.1.x releases are no longer actively > maintained. Version 1.1.7 was the last release in that cycle. We > recommend that everyone using Version 1.1.7 (or any earlier version) > upgrade to the latest 2.x release as soon as possible. > > If there are any security isses found in Version 1.1.7, we will release > a Version 1.1.8 to fix them. However, any new features or bug fixes will > not be added to the 1.1.x releases." > > So the freeradius 1.1.6 version you mentioned is obsolete. Imho > mod_rad_auth should be updated to work with freeradius 2.1.10. > > I don't use mod_rad_auth but it would make sense to me if someone filed > a bug on Jira asap so maybe this could get fixed before FreeSWITCH 1.2 > is released. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Hi Patric, Freeradius is not causing the problem, the problem is freeradius-client as freeradius team is not longer developing it. I have download freeradius-client from wget ftp://ftp.freeradius.org/pub/freeradius/freeradius-client-1.1.6.tar.bz2 as is written on wiki. I will post this on jira. I have found one tip on google, that in make file I should change FREERADIUSLA=/usr/local/lib/libfreeradius-client.so to FREERADIUSLA=/usr/local/lib/libfreeradius-client.so. I did that and compile it again. Now it works. Do you maybe know Patric what this change? Regards, Miha From miha at softnet.si Wed Mar 21 10:11:40 2012 From: miha at softnet.si (Miha) Date: Wed, 21 Mar 2012 08:11:40 +0100 Subject: [Freeswitch-users] rad_auth faild to load modul In-Reply-To: References: <4F687D5B.7050200@softnet.si> Message-ID: <4F697F2C.7030807@softnet.si> Hi Thimor, I am also using this verison of radius-client. I have download version as is written on wiki. I will post this on Jira. Regards, Miha On 3/20/2012 10:55 PM, Tihomir Culjaga wrote: > hi Miha, > > please always use freeradius-client-1.1.6 along with mod_rad_auth. > looks like a different freeradius-client library version. > > > On Tue, Mar 20, 2012 at 1:51 PM, Miha > wrote: > > Hi, > > I have installed Freeswitch (FreeSWITCH Version 1.0.head > (git-9d3401e 2012-03-19 20-06-36 -0500)) on latest Centos (6.2). > > After I installed freeradius-client as is written on wiki, I tried > to load it. I am getting this error: > > freeswitch at localhost.localdomain > > load mod_rad_auth > 2012-03-20 13:46:03.386561 [INFO] mod_enum.c:812 ENUM Reloaded > 2012-03-20 13:46:03.386561 [INFO] switch_time.c:1128 Timezone > reloaded 530 definitions > 2012-03-20 13:46:03.386561 [CRIT] switch_loadable_module.c:1295 > Error Loading module /usr/local/freeswitch/mod/mod_rad_auth.so > **/usr/local/freeswitch/mod/mod_rad_auth.so: undefined symbol: > rc_conf_str** > > I am having rad_auth installed on FS (FreeSWITCH Version 1.0.head > (git-00de8e6 2011-11-01 17-27-13 -0600)), (Centos 5.6) and is > working properly. > > Any suggestion how what could be causing problem that I can not > load it on centos 6.2? > > Thank you! > > Regards, > MIha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/d080c890/attachment.html From oseslija at gmail.com Wed Mar 21 10:35:47 2012 From: oseslija at gmail.com (Ognjen Seslija) Date: Wed, 21 Mar 2012 08:35:47 +0100 Subject: [Freeswitch-users] rad_auth faild to load modul In-Reply-To: <4F69012C.5020403@puzzled.xs4all.nl> References: <4F687D5B.7050200@softnet.si> <4F69012C.5020403@puzzled.xs4all.nl> Message-ID: 1.1.6 is the last client version out there. 2.x are server versions. On Tue, Mar 20, 2012 at 11:14 PM, Patrick Lists < freeswitch-list at puzzled.xs4all.nl> wrote: > On 20-03-12 22:55, Tihomir Culjaga wrote: > > hi Miha, > > > > please always use freeradius-client-1.1.6 along with mod_rad_auth. > > looks like a different freeradius-client library version. > > RHEL 6.2 / CentOS 6.2 comes with freeradius 2.1.10 so that is going to > be a problem. Even more when looking at the freeradius website where it > clearly states: > > "Version 1.1 - No longer maintained! > > As of January 2008, the version 1.1.x releases are no longer actively > maintained. Version 1.1.7 was the last release in that cycle. We > recommend that everyone using Version 1.1.7 (or any earlier version) > upgrade to the latest 2.x release as soon as possible. > > If there are any security isses found in Version 1.1.7, we will release > a Version 1.1.8 to fix them. However, any new features or bug fixes will > not be added to the 1.1.x releases." > > So the freeradius 1.1.6 version you mentioned is obsolete. Imho > mod_rad_auth should be updated to work with freeradius 2.1.10. > > I don't use mod_rad_auth but it would make sense to me if someone filed > a bug on Jira asap so maybe this could get fixed before FreeSWITCH 1.2 > is released. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/dd463fba/attachment.html From miha at softnet.si Wed Mar 21 11:15:19 2012 From: miha at softnet.si (Miha) Date: Wed, 21 Mar 2012 09:15:19 +0100 Subject: [Freeswitch-users] rad_auth faild to load modul In-Reply-To: References: <4F687D5B.7050200@softnet.si> <4F69012C.5020403@puzzled.xs4all.nl> Message-ID: <4F698E17.7010805@softnet.si> Hi Tihormir, I have changed this and now works: FREERADIUSLA=/usr/local/lib/libfreeradius-client.so to FREERADIUSLA=/usr/local/lib/libfreeradius-client.la. Do you maybe know why (I found this on google). Regards, Miha On 3/20/2012 11:41 PM, Tihomir Culjaga wrote: > yap, i totally agree on that... will move forward to update it! > > in the mean time you can use the old library... mod_rad_auth module > uses an embedded config so you can build the library and copy it > directly to fs/lib directory ... no need to be system available. > > T. > > On Tue, Mar 20, 2012 at 11:14 PM, Patrick Lists > > wrote: > > On 20-03-12 22:55, Tihomir Culjaga wrote: > > hi Miha, > > > > please always use freeradius-client-1.1.6 along with mod_rad_auth. > > looks like a different freeradius-client library version. > > RHEL 6.2 / CentOS 6.2 comes with freeradius 2.1.10 so that is going to > be a problem. Even more when looking at the freeradius website > where it > clearly states: > > "Version 1.1 - No longer maintained! > > As of January 2008, the version 1.1.x releases are no longer actively > maintained. Version 1.1.7 was the last release in that cycle. We > recommend that everyone using Version 1.1.7 (or any earlier version) > upgrade to the latest 2.x release as soon as possible. > > If there are any security isses found in Version 1.1.7, we will > release > a Version 1.1.8 to fix them. However, any new features or bug > fixes will > not be added to the 1.1.x releases." > > So the freeradius 1.1.6 version you mentioned is obsolete. Imho > mod_rad_auth should be updated to work with freeradius 2.1.10. > > I don't use mod_rad_auth but it would make sense to me if someone > filed > a bug on Jira asap so maybe this could get fixed before FreeSWITCH 1.2 > is released. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/ffd4e149/attachment-0001.html From garbytrash at gmail.com Wed Mar 21 14:01:34 2012 From: garbytrash at gmail.com (Zenny) Date: Wed, 21 Mar 2012 11:01:34 +0000 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: References: Message-ID: I also tried to export with 'pg_config --includedir' instead of --pkgincludedir below before I executed ./configure: # export CFLAGS=-I'pg_config --includedir' [freeswitch]# ./configure checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for a thread-safe mkdir -p... /bin/mkdir -p checking for gawk... gawk checking whether make sets $(MAKE)... yes checking build system type... x86_64-unknown-linux-gnu checking host system type... x86_64-unknown-linux-gnu checking for gcc... gcc checking for C compiler default output file name... configure: error: in `/usr/src/freeswitch': configure: error: C compiler cannot create executables In this case, even ./configure failed. My include directories are: # pg_config | grep INCLUDE INCLUDEDIR = /usr/pgsql-9.1/include PKGINCLUDEDIR = /usr/pgsql-9.1/include INCLUDEDIR-SERVER = /usr/pgsql-9.1/include/server Where did I go wrong? On 3/20/12, Zenny wrote: > Thank Ken: > > I do see : > > # ldconfig -v | grep pq > libpqwalreceiver.so -> libpqwalreceiver.so > libpq.so.5 -> libpq.so.5.4 > libipq.so.0 -> libipq.so.0.0.0 > > where do I need to add manually? /etc/ld.so.conf.d/? > > On 3/20/12, Ken Rice wrote: >> This error is common if ldd cant find libpq... Does it show up in " >> ldconfig >> -v |grep pq" if not you'll need to add the directory those libs live in >> to >> your ld.so.conf or ld.so.conf.d as is proper on your platform >> >> >> On 3/20/12 1:09 PM, "Zenny" wrote: >> >>> On 3/20/12, Bzzz wrote: >>>> On Tue, 20 Mar 2012 17:53:37 +0000 >>>> Zenny wrote: >>>> >>>>> Just to test, I directly specified /usr/pgsql-9.1/include/libpq-fe.h >>>>> in >>>>> /usr/src/freeswitch/src/mod/event_handlers/mod_cdr_pg_csv/mod_cdr_pg_csv.c >>>>> >>>>> It seemed to have progressed a bit further, but spits out the error: >>>>> >>>>> Creating mod_cdr_pg_csv.la... >>>>> /usr/bin/ld: cannot find -lpq >>>>> collect2: ld returned 1 exit status >>>>> cat: .libs/mod_cdr_pg_csv.log: No such file or directory >>>>> make[5]: *** [mod_cdr_pg_csv.la] Error 1 >>>>> make[4]: *** [all] Error 1 >>>>> make[3]: *** [mod_cdr_pg_csv-all] Error 1 >>>>> make[2]: *** [all-recursive] Error 1 >>>>> make[1]: *** [all-recursive] Error 1 >>>>> make: *** [all] Error 2 >>>> >>>> Seems normal: you give it the pq header, so it can compile, but it >>>> can't link against nothing. >>> >>> Right. But how could that be solved? I am just struggling with this for >>> a >>> while. >>> >>> Linking directly the /usr/include with /user/pgsql-9.1/include? that >>> may affect other applications I guess. >>> >>>> >>>> -- >>>> QOTD: >>>> All I want is more than my fair share. >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From nasida at live.ru Wed Mar 21 14:49:28 2012 From: nasida at live.ru (Yuriy Nasida) Date: Wed, 21 Mar 2012 15:49:28 +0400 Subject: [Freeswitch-users] issue with freeswitch_licence_server Message-ID: Hello guys. Can anybody give me some advice with freeswitch_licence_server ? FS works as root. freeswitch_licence_server is started. But in fs_cli i see:freeswitch at internal> g729_infoCan't contact licence server. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/5f240f3c/attachment.html From nasida at live.ru Wed Mar 21 15:14:05 2012 From: nasida at live.ru (Yuriy Nasida) Date: Wed, 21 Mar 2012 16:14:05 +0400 Subject: [Freeswitch-users] issue with freeswitch_licence_server In-Reply-To: References: Message-ID: Solved :) From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Wed, 21 Mar 2012 15:49:28 +0400 Subject: [Freeswitch-users] issue with freeswitch_licence_server Hello guys. Can anybody give me some advice with freeswitch_licence_server ? FS works as root. freeswitch_licence_server is started. But in fs_cli i see:freeswitch at internal> g729_infoCan't contact licence server. Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/b74fe2fa/attachment.html From krice at freeswitch.org Wed Mar 21 15:17:25 2012 From: krice at freeswitch.org (Ken Rice) Date: Wed, 21 Mar 2012 07:17:25 -0500 Subject: [Freeswitch-users] issue with freeswitch_licence_server In-Reply-To: Message-ID: Solution for anyone else that might run across this? This seems to be a common occurance at times Thx K On 3/21/12 7:14 AM, "Yuriy Nasida" wrote: > Solved :) > > > > From: nasida at live.ru > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 21 Mar 2012 15:49:28 +0400 > Subject: [Freeswitch-users] issue with freeswitch_licence_server > > Hello guys. > > Can anybody give me some advice with freeswitch_licence_server ? > > FS works as root. freeswitch_licence_server is started. But in fs_cli i see: > freeswitch at internal> g729_info > Can't contact licence server. > > Thanks. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/92da72c8/attachment.html From freeswitch-list at puzzled.xs4all.nl Wed Mar 21 15:19:03 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 21 Mar 2012 13:19:03 +0100 Subject: [Freeswitch-users] rad_auth faild to load modul In-Reply-To: References: <4F687D5B.7050200@softnet.si> <4F69012C.5020403@puzzled.xs4all.nl> Message-ID: <4F69C737.4090708@puzzled.xs4all.nl> On 21-03-12 08:35, Ognjen Seslija wrote: > 1.1.6 is the last client version out there. 2.x are server versions. Yes you are right. I thought it was part of the main freeradius package on RHEL6/CentOS6 which it is not. Thanks for pointing that out. Regards, Patrick From freeswitch-list at puzzled.xs4all.nl Wed Mar 21 15:22:35 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 21 Mar 2012 13:22:35 +0100 Subject: [Freeswitch-users] rad_auth faild to load modul In-Reply-To: <4F697EED.5030607@softnet.si> References: <4F687D5B.7050200@softnet.si> <4F69012C.5020403@puzzled.xs4all.nl> <4F697EED.5030607@softnet.si> Message-ID: <4F69C80B.50606@puzzled.xs4all.nl> On 21-03-12 08:10, Miha wrote: > Hi Patric, > > Freeradius is not causing the problem, the problem is freeradius-client > as freeradius team is not longer developing it. I have download > freeradius-client from wget > ftp://ftp.freeradius.org/pub/freeradius/freeradius-client-1.1.6.tar.bz2 > as is written on wiki. > > I will post this on jira. I have found one tip on google, that in make > file I should change FREERADIUSLA=/usr/local/lib/libfreeradius-client.so > to FREERADIUSLA=/usr/local/lib/libfreeradius-client.so. I did that and > compile it again. Now it works. > > Do you maybe know Patric what this change? I guess you mean you changed it to FREERADIUSLA=/usr/local/lib/libfreeradius-client.la which makes more sense :) The FreeSWITCH build system is quite complex and I don't know very much about it. I guess you could keep using that old 1.1.6 version and make the change from .so to .la if that makes it work. Regards, Patrick From freeswitch-list at puzzled.xs4all.nl Wed Mar 21 15:30:29 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 21 Mar 2012 13:30:29 +0100 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: References: Message-ID: <4F69C9E5.5060309@puzzled.xs4all.nl> On 21-03-12 12:01, Zenny wrote: > I also tried to export with 'pg_config --includedir' instead of > --pkgincludedir below before I executed ./configure: > > # export CFLAGS=-I'pg_config --includedir' > [freeswitch]# ./configure > checking for a BSD-compatible install... /usr/bin/install -c > checking whether build environment is sane... yes > checking for a thread-safe mkdir -p... /bin/mkdir -p > checking for gawk... gawk > checking whether make sets $(MAKE)... yes > checking build system type... x86_64-unknown-linux-gnu > checking host system type... x86_64-unknown-linux-gnu > checking for gcc... gcc > checking for C compiler default output file name... > configure: error: in `/usr/src/freeswitch': > configure: error: C compiler cannot create executables > > In this case, even ./configure failed. My include directories are: > > # pg_config | grep INCLUDE > INCLUDEDIR = /usr/pgsql-9.1/include > PKGINCLUDEDIR = /usr/pgsql-9.1/include > INCLUDEDIR-SERVER = /usr/pgsql-9.1/include/server > > Where did I go wrong? I don't know which distro you use but that directory (/usr/pgsql-9.1) looks odd. If you use CentOS or RHEL why don't you just remove all stock postgresql RPMs from your box and install a decent PostgreSQL 9.1 RPM via http://yum.postgresql.org/ Regards, Patrick From markus.lindenberg at gmail.com Wed Mar 21 15:37:32 2012 From: markus.lindenberg at gmail.com (Markus Lindenberg) Date: Wed, 21 Mar 2012 13:37:32 +0100 Subject: [Freeswitch-users] Setting effective_caller_id_name before putting a user in a callcenter queue has no effect Message-ID: Hi, my dialplan goes like this: Setting/exporting effective_caller_id_name has no effect when the call is presented to a callback agent. the original callerid and calleridname will be used. for now i worked around the problem like this: while this works, it just doesn't look right to me and means i have to take additional actions if i want the dialplan to continue after bridging into the queue. To me the obvious solution would be to have mod_callcenter.c (line 2479 "INSERT INTO members...") check for origination/effective caller id variables and use them, if available. I'm already doing this kind of check over and over in some lua code i call before bridging (e.g. for sending xmpp notifications): cidnum = session:getVariable("effective_caller_id_number") if not cidnum then cidnum = session:getVariable ("origination_caller_id_number") end if not cidnum then cidnum = session:getVariable ("caller_id_number") end This doesn't feel entirely right either. Shouldn't there be a command that always gives me the computed callerid after possible manipulations? or should i generally do this stuff in the b leg, where the effective_caller_id_name is available as caller_id_name? Thanks, Markus From garbytrash at gmail.com Wed Mar 21 15:42:16 2012 From: garbytrash at gmail.com (Zenny) Date: Wed, 21 Mar 2012 12:42:16 +0000 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: <4F69C9E5.5060309@puzzled.xs4all.nl> References: <4F69C9E5.5060309@puzzled.xs4all.nl> Message-ID: On 3/21/12, Patrick Lists wrote: > On 21-03-12 12:01, Zenny wrote: >> I also tried to export with 'pg_config --includedir' instead of >> --pkgincludedir below before I executed ./configure: >> >> # export CFLAGS=-I'pg_config --includedir' >> [freeswitch]# ./configure >> checking for a BSD-compatible install... /usr/bin/install -c >> checking whether build environment is sane... yes >> checking for a thread-safe mkdir -p... /bin/mkdir -p >> checking for gawk... gawk >> checking whether make sets $(MAKE)... yes >> checking build system type... x86_64-unknown-linux-gnu >> checking host system type... x86_64-unknown-linux-gnu >> checking for gcc... gcc >> checking for C compiler default output file name... >> configure: error: in `/usr/src/freeswitch': >> configure: error: C compiler cannot create executables >> >> In this case, even ./configure failed. My include directories are: >> >> # pg_config | grep INCLUDE >> INCLUDEDIR = /usr/pgsql-9.1/include >> PKGINCLUDEDIR = /usr/pgsql-9.1/include >> INCLUDEDIR-SERVER = /usr/pgsql-9.1/include/server >> >> Where did I go wrong? > > I don't know which distro you use but that directory (/usr/pgsql-9.1) > looks odd. If you use CentOS or RHEL why don't you just remove all stock > postgresql RPMs from your box and install a decent PostgreSQL 9.1 RPM > via http://yum.postgresql.org/ I am using CentOS6. I installed the RPMs directly downloading the repos from: rpm -ivh http://yum.postgresql.org/9.1/redhat/rhel-6-x86:64/pgdg-centos91-9.1-4.noarch.rpm > > Regards, > Patrick > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From andrew at cassidywebservices.co.uk Wed Mar 21 15:42:30 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 21 Mar 2012 12:42:30 +0000 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after~30seconds In-Reply-To: <4F6970A2.000019.15156@FLIGHTPC> References: <1363342dbcd.1079533506949741633.-8314378552953017400@zoho.com> <4F6970A2.000019.15156@FLIGHTPC> Message-ID: I think we also need to check those variables on the internal profile. As said previously it looks like it's doing the tun on the internal profile which is why the ack is being send to the wrong place. On 21 March 2012 06:09, Fernando - NextBilling IP Solutions < fernandojdk at gmail.com> wrote: > What you have in your external profile on variables ext-rtp-ip end > ext-sip-ip? > ? > ** > > * > Best Regards, > Fernando da Silva Santos > NextBilling IP Solutions LTDA > Phone: +55 21 2143-9000 > MSN: fernandojdk at gmail.com > www.nextbilling.com.br > Rio de Janeiro, Brazil, BR > Importante: > Esta mensagem, incluindo todo seu conte?do, cont?m informa??es > confidenciais, legalmente protegidas e destinadas a indiv?duo e prop?sito > espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter > sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o > mais breve poss?vel. > As informa??es contidas nesta mensagem e em seu conte?do s?o de > responsabilidade de seu autor, n?o representando necessariamente id?ias, > opini?es, pensamentos ou qualquer forma de posicionamento por parte da > NextBilling IP Solutions. > P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." > * > *-------Original Message-------* > > *From:* dingdong > *Date:* 21/03/2012 02:57:02 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] NAT issues - Outbound Call drops > after~30seconds > > id like to know more bout wireshark and sip debugging,please include this > on the conf call topics > > ---- On Tue, 20 Mar 2012 10:15:20 -0700 *Michael Collins < > msc at freeswitch.org>* wrote ---- > > This is why I love Wireshark so much! Look at this purdee graph it makes: > > > > > See all those 200 OK's that your FS is sending to the Grandstream? Guess > what your GS is sending in response to those: NADA! If you look at the BYE > that FS sends to the GS you'll even see the reason: > > SIP;cause=408;text=\"ACK Timeout\" > > FS never gets an ACK back from the GS. So the question is: why? I'm > unfamiliar with the GS so I'll have to defer to those with more experience > than I. However, I think you'll find that tcpdumps and analyzing w/ > Wireshark is extremely helpful. (Open the pcap, click "Telephony > VoIP > calls" and then a new dialog opens. In this case it shows two calls - > meaning two call legs. Click "Select All" then click "Flow" and you'll get > the cool graph. Click around and see what other stuff does. :) > > I'm thinking of doing a FreeSWITCH conference call presentation on the > subject of collecting pcaps and doing Wireshark analysis. Let me know if > you guys think that's a good presentation. > > -MC > > > On Tue, Mar 20, 2012 at 10:00 AM, Brian Foster wrote: > Andrew, > > root at homeserver:/usr/local/stund# ./client stunserver.org > STUN client version 0.97 > Primary: Independent Mapping, Independent Filter, preserves ports, will > hairpin > Return value is 0x000003 > > http://da1.endigovoip.com/dump.pcap > > Kristian, > > http://pastebin.freeswitch.org/18708 > > Michael, > > I did replace the IP's for security purposes, but now I've realized that > it's needed and it's not really that big of a deal. I'll end up changing > the Flowroute creds after this is fixed up. The prior siptrace is exactly > one call (two legs). I don't think it's a carrier issue, as I've tried > calling a buddy's server direct sip with the same issues. > > -BDF > > On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins wrote: > We have scores of machines behind NAT talking to Flowroute with no > problems, so there's got to be something potentially non-obvious but easy > that needs to be set/unset. I noticed in the SIP trace that there are > several calls. It's hard to know what's what. I think your best bet is a > pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I > also noticed that you redacted IP addrs - you won't be able to do this with > a pcap. If security is an issue then I'd say get the pcap and let us know > here on the list, then those who can have a look will email you privately > and you can send the pcap file to them. > > -MC > > > On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster wrote: > Alright, so I admit... I'm a little rusty when it comes to NAT, etc. I've > only set up FS so far on machines with no NAT, so this is sort of a new > experience for me. > > I have a FreeSWITCH server located on the same local network as all of my > phones here at the house. When I try to make a call to Flowroute, after > about 30 seconds the call drops. It also does the exact same thing when I > call a buddy's server directly via SIP. > > Here's a siptrace of the call (I didn't think that the actual FS log would > be much help): > http://pastebin.freeswitch.org/18697 > > ...and here's a paste of 'sofia status': > http://pastebin.freeswitch.org/18698 > > ...and just for good measure, here's a paste of vars.xml: > http://pastebin.freeswitch.org/18699 > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/dca23589/attachment-0001.html From markus.lindenberg at gmail.com Wed Mar 21 15:43:03 2012 From: markus.lindenberg at gmail.com (Markus Lindenberg) Date: Wed, 21 Mar 2012 13:43:03 +0100 Subject: [Freeswitch-users] Setting effective_caller_id_name before putting a user in a callcenter queue has no effect In-Reply-To: References: Message-ID: I'v just noticed that this exact problem was mentioned a few days ago. Sorry for starting another thread. Still, this doesn't look right to me either. I assumed caller_id_name is considered read only? What's best practice here? On Fri, Mar 9, 2012 at 14:58, Mike wrote: > In case it's of use to anyone else: > > Thanks to the guys on the IRC - the solution to this is to set the > caller_id_name variable in the dialplan before calling the callcenter > application thuswise: > > > > > This works a treat. > > The other methods we tried - using effective_caller_id_name or > orignination_caller_id_name with either export, set or bridge_export > didn't work. > > Mike On Wed, Mar 21, 2012 at 13:37, Markus Lindenberg wrote: > Hi, > > my dialplan goes like this: > > > > > Setting/exporting effective_caller_id_name has no effect when the call > is presented to a callback agent. the original callerid and > calleridname will be used. > > for now i worked around the problem like this: > > > ? > ? ? data="effective_caller_id_name=HOTLINE ${caller_id_name}"/> > ? ? > ? > > > > ? > ? ? > ? > > > while this works, it just doesn't look right to me and means i have to > take additional actions if i want the dialplan to continue after > bridging into the queue. > > To me the obvious solution would be to have mod_callcenter.c (line > 2479 "INSERT INTO members...") check for origination/effective caller > id variables and use them, if available. > > I'm already doing this kind of check over and over in some lua code i > call before bridging (e.g. for sending xmpp notifications): > > ? ? ? ?cidnum = session:getVariable("effective_caller_id_number") > ? ? ? ?if not cidnum then > ? ? ? ? ? ? ? ?cidnum = session:getVariable ("origination_caller_id_number") > ? ? ? ?end > ? ? ? ?if not cidnum then > ? ? ? ? ? ? ? ?cidnum = session:getVariable ("caller_id_number") > ? ? ? ?end > > This doesn't feel entirely right either. Shouldn't there be a command > that always gives me the computed callerid after possible > manipulations? or should i generally do this stuff in the b leg, where > the effective_caller_id_name is available as caller_id_name? > > Thanks, > > Markus From freeswitch-list at puzzled.xs4all.nl Wed Mar 21 16:16:43 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 21 Mar 2012 14:16:43 +0100 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: References: <4F69C9E5.5060309@puzzled.xs4all.nl> Message-ID: <4F69D4BB.7090300@puzzled.xs4all.nl> On 21-03-12 13:42, Zenny wrote: >> I don't know which distro you use but that directory (/usr/pgsql-9.1) >> looks odd. If you use CentOS or RHEL why don't you just remove all stock >> postgresql RPMs from your box and install a decent PostgreSQL 9.1 RPM >> via http://yum.postgresql.org/ > > I am using CentOS6. I installed the RPMs directly downloading the repos from: > rpm -ivh http://yum.postgresql.org/9.1/redhat/rhel-6-x86:64/pgdg-centos91-9.1-4.noarch.rpm I just checked the postgresql-9.1 server RPM and you are right. Wow... The PostgreSQL packager could read up on the RPM Packaging Guide Lines and the FHS. I guess they packaged it so that it does not interfere with the RHEL/CentOS provided PostgreSQL RPMs but putting everything in a non-standard location clearly causes challenges as you are experiencing right now. I'm afraid I can't be of any help. Regards, Patrick From brett at launch3.net Wed Mar 21 16:28:06 2012 From: brett at launch3.net (Brett Wilson) Date: Wed, 21 Mar 2012 09:28:06 -0400 Subject: [Freeswitch-users] BLF/Presence Message-ID: Hey guys, Im looking to get BLF working on my GXP2100's. I have not been able to find much info on how to configure it. I have set the phones keys to BLF, so they constantly light up green. But they never change. I have those keys configured for the other extensions, ie. 101, 102, 103 etc. Can someone steer me in the right direction? Thanks. Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: Brett.Wilson at launch3.net AOL IM: Brett.Wilson at launch3.net www.Launch3.net www.Launch3telecom.com *************************** Description: Description: Blogger-logo Description: Description: FaceBook-Logo Description: Description: Twitter-Logo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/b00fc705/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 1815 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/b00fc705/attachment.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 1680 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/b00fc705/attachment-0001.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 1715 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/b00fc705/attachment-0002.jpe From gsamat at gmail.com Wed Mar 21 10:08:06 2012 From: gsamat at gmail.com (Samat Galimov) Date: Wed, 21 Mar 2012 11:08:06 +0400 Subject: [Freeswitch-users] Pulse dialing in freeswitch. Message-ID: Hello! Is it possible to make outgoing calls to PSTN via pulse dial? My PSTN provider uses old hardware, so they doesnt support tone dial, and I have some small SIP gateways (Linksys SPA3102) which dont support pulse dial. Can I generate tones on the freeswitch side? Thank you. From mytemike72 at gmail.com Wed Mar 21 16:48:49 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Wed, 21 Mar 2012 14:48:49 +0100 Subject: [Freeswitch-users] Originate using inbound socket connection Message-ID: Hi Guys, I am working on a .net ESL host (inbound mode) and connect to the ESL using ESLConnection. Now I am trying to originate a call using the ESLConnection.Api() function. But it does't work as expected. It always returns: === Content-Type: api/response Content-Length: 125 Content-Length: 125 -USAGE: |&() [] [] [] [] [] === But I am pretty sure my string is right: {origination_caller_id_number=31341.....,origination_caller_id_name=31341......}sofia/external/31634258... at xxx.xxx.xxx.xxx I also tried with double {{ and }} but same result. Even if I take out the whole {} string and just use "sofia/external/etc.." I get the same message back. My code: ESLconnection eslDial = new ESLconnection("x.x.x.x", "8021", "x"); if (eslDial.Connected() == ESL_SUCCESS) { // Create a uuid used to identify the b-leg. string legb_uuid = eslDial.Api("create_uuid", "").GetBody(); string cDialString = "{{origination_uuid=" + legb_uuid + ",origination_caller_id_number=" + thisAni + ",origination_caller_id_name=" + thisAniName + "}}sofia/external/" + thisDestination; // Send the command var eslEvent = eslDial.Api("originate", cDialString); // Write the result to the console Console.WriteLine(eslEvent.Serialize(string.Empty)); return true; } Thanks for your help!, Regards, Michael Lutz From curriegrad2004 at gmail.com Wed Mar 21 17:10:58 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 21 Mar 2012 07:10:58 -0700 Subject: [Freeswitch-users] Pulse dialing in freeswitch. In-Reply-To: References: Message-ID: To be quite honest, I'm quite surprised that your POTS provider doesn't support DTMF dialing. Unfortunately due to the way how pulse dialing works, I highly doubt FreeSWITCH has the capability of generating pulse dial tones. There are 56K modems out there that does pulse dialing, so you could probably get FS to interface with that using a shell script or whatnot and have the SPA3102 to go off hook as soon as the digits are dialed. That's an idea that comes to mind when speaking of getting pulse dialing to work. On Wed, Mar 21, 2012 at 12:08 AM, Samat Galimov wrote: > Hello! > ?Is it possible to make outgoing calls to PSTN via pulse dial? My PSTN > provider uses old hardware, so they doesnt support tone dial, and I > have some small SIP gateways (Linksys SPA3102) which dont support > pulse dial. > ?Can I generate tones on the freeswitch side? > > ?Thank you. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at freeswitch.org Wed Mar 21 17:14:03 2012 From: krice at freeswitch.org (Ken Rice) Date: Wed, 21 Mar 2012 09:14:03 -0500 Subject: [Freeswitch-users] Pulse dialing in freeswitch. In-Reply-To: Message-ID: The problem is pulse dialing is switch hook... I don't think freetdm even supports that On 3/21/12 9:10 AM, "curriegrad2004" wrote: > To be quite honest, I'm quite surprised that your POTS provider doesn't > support DTMF dialing. Unfortunately due to the way how pulse dialing works, I > highly doubt FreeSWITCH has the capability of generating pulse dial > tones. There are 56K modems out there that does pulse dialing, so you > could probably get FS to interface with that using a shell script or > whatnot and have the SPA3102 to go off hook as soon as the digits are > dialed. That's an idea that comes to mind when speaking of getting > pulse dialing to work. On Wed, Mar 21, 2012 at 12:08 AM, Samat Galimov > wrote: > Hello! > ?Is it possible to make outgoing calls to > PSTN via pulse dial? My PSTN > provider uses old hardware, so they doesnt > support tone dial, and I > have some small SIP gateways (Linksys SPA3102) > which dont support > pulse dial. > ?Can I generate tones on the freeswitch > side? > > ?Thank you. > > > _________________________________________________________________________> > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel > Communication Server > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > http://wiki.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org ___________________________________________________ > ______________________ Professional FreeSWITCH Consulting > Services: consulting at freeswitch.org http://www.freeswitchsolutions.com FreeSW > ITCH-powered IP PBX: The CudaTel Communication > Server Official FreeSWITCH > Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon. > com FreeSWITCH-users mailing > list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org From peter.olsson at visionutveckling.se Wed Mar 21 17:27:10 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 21 Mar 2012 14:27:10 +0000 Subject: [Freeswitch-users] Originate using inbound socket connection Message-ID: <1FFF97C269757C458224B7C895F35F1507B744@cantor.std.visionutv.se> You need to put the other end of the call somewhere The correct string is (example) "originate sofia/gateway/test/1002 &park()" This will call 1002 and then park the call. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz Skickat: den 21 mars 2012 14:49 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Originate using inbound socket connection Hi Guys, I am working on a .net ESL host (inbound mode) and connect to the ESL using ESLConnection. Now I am trying to originate a call using the ESLConnection.Api() function. But it does't work as expected. It always returns: === Content-Type: api/response Content-Length: 125 Content-Length: 125 -USAGE: |&() [] [] [] [] [] === But I am pretty sure my string is right: {origination_caller_id_number=31341.....,origination_caller_id_name=31341......}sofia/external/31634258... at xxx.xxx.xxx.xxx I also tried with double {{ and }} but same result. Even if I take out the whole {} string and just use "sofia/external/etc.." I get the same message back. My code: ESLconnection eslDial = new ESLconnection("x.x.x.x", "8021", "x"); if (eslDial.Connected() == ESL_SUCCESS) { // Create a uuid used to identify the b-leg. string legb_uuid = eslDial.Api("create_uuid", "").GetBody(); string cDialString = "{{origination_uuid=" + legb_uuid + ",origination_caller_id_number=" + thisAni + ",origination_caller_id_name=" + thisAniName + "}}sofia/external/" + thisDestination; // Send the command var eslEvent = eslDial.Api("originate", cDialString); // Write the result to the console Console.WriteLine(eslEvent.Serialize(string.Empty)); return true; } Thanks for your help!, Regards, Michael Lutz _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f69dabc32766734014540! From freeswitch at earthspike.net Wed Mar 21 17:58:25 2012 From: freeswitch at earthspike.net (John) Date: Wed, 21 Mar 2012 14:58:25 +0000 Subject: [Freeswitch-users] FreeSWITCH Community Conf Call Tomorrow In-Reply-To: References: Message-ID: <4F69EC91.5080108@earthspike.net> On 20/03/2012 16:19, Michael Collins wrote: > Hey gang! > > Tomorrow's agenda page is here: > > http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_21 Can you confirm the timings? UTC doesn't do DST but I think EST and PST do... Thanks, John From steveu at coppice.org Wed Mar 21 18:08:08 2012 From: steveu at coppice.org (Steve Underwood) Date: Wed, 21 Mar 2012 23:08:08 +0800 Subject: [Freeswitch-users] Pulse dialing in freeswitch. In-Reply-To: References: Message-ID: <4F69EED8.3030800@coppice.org> On 03/21/2012 03:08 PM, Samat Galimov wrote: > Hello! > Is it possible to make outgoing calls to PSTN via pulse dial? My PSTN > provider uses old hardware, so they doesnt support tone dial, and I > have some small SIP gateways (Linksys SPA3102) which dont support > pulse dial. > Can I generate tones on the freeswitch side? > > Thank you. The original zaptel supported both incoming and outgoing pulse dialing, and supported them both for analogue loop and T1 RBS circuits. I don't know if the current dahdi and wanpipe drivers still support it. Steve From cmrienzo at gmail.com Wed Mar 21 18:23:15 2012 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Wed, 21 Mar 2012 11:23:15 -0400 Subject: [Freeswitch-users] FreeSWITCH Community Conf Call Tomorrow In-Reply-To: <4F69EC91.5080108@earthspike.net> References: <4F69EC91.5080108@earthspike.net> Message-ID: It will be 1700 UTC this week. I updated the wiki. On Wed, Mar 21, 2012 at 10:58 AM, John wrote: > On 20/03/2012 16:19, Michael Collins wrote: > > Hey gang! > > > > Tomorrow's agenda page is here: > > > > http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_21 > Can you confirm the timings? UTC doesn't do DST but I think EST and PST > do... > > Thanks, > > John > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/65c99d38/attachment.html From andrew at cassidywebservices.co.uk Wed Mar 21 18:23:54 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 21 Mar 2012 15:23:54 +0000 Subject: [Freeswitch-users] FreeSWITCH Community Conf Call Tomorrow In-Reply-To: <4F69EC91.5080108@earthspike.net> References: <4F69EC91.5080108@earthspike.net> Message-ID: UTC does not do DST, EST and PST are in DST iirc, and BST does not start til this weekend. On 21 March 2012 14:58, John wrote: > On 20/03/2012 16:19, Michael Collins wrote: > > Hey gang! > > > > Tomorrow's agenda page is here: > > > > http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_21 > Can you confirm the timings? UTC doesn't do DST but I think EST and PST > do... > > Thanks, > > John > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/3959e7c2/attachment.html From lloyd.aloysius at gmail.com Wed Mar 21 18:54:09 2012 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Wed, 21 Mar 2012 11:54:09 -0400 Subject: [Freeswitch-users] Direct Call Pickup - Issue In-Reply-To: References: <040950C4-60EC-47AD-9230-D1EF1F4A759B@freeswitch.org> Message-ID: Brian, CallPIckup problem start happening again. I use the latest git 1. Extension 201 Ringing 2. Extension 202 dial **201 and pickup the call 3. Extension 203 Dial **201 4. Extension 203 steel the call which is connected to 202. I am using Default Dial plan application Thank you Lloyd * * On Mon, Feb 13, 2012 at 12:21 PM, Lloyd Aloysius wrote: > Thanks Brian. New version update fix the problem > > Lloyd > * > * > > > > On Fri, Feb 10, 2012 at 2:11 PM, Brian West wrote: > >> Please use the latest git head. >> >> /b >> >> On Feb 10, 2012, at 12:46 PM, Lloyd Aloysius wrote: >> >> FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) >> >> >> -- >> Brian West >> FreeSWITCH Solutions, LLC >> Phone: +1 (918) 420-9266 >> Fax: +1 (918) 420-9267 >> brian at freeswitch.org >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/7bbe7969/attachment-0001.html From msc at freeswitch.org Wed Mar 21 19:18:31 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Mar 2012 09:18:31 -0700 Subject: [Freeswitch-users] FreeSWITCH Community Conf Call Tomorrow In-Reply-To: <4F69EC91.5080108@earthspike.net> References: <4F69EC91.5080108@earthspike.net> Message-ID: 1pm EDT, 10am PDT. I think it's 1700 UTC during DST but if someone with more TZ savvy than I could chime in I would appreciate it. Thanks! -MC On Wed, Mar 21, 2012 at 7:58 AM, John wrote: > On 20/03/2012 16:19, Michael Collins wrote: > > Hey gang! > > > > Tomorrow's agenda page is here: > > > > http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_21 > Can you confirm the timings? UTC doesn't do DST but I think EST and PST > do... > > Thanks, > > John > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/d3827fb7/attachment.html From vetali100 at gmail.com Wed Mar 21 19:21:03 2012 From: vetali100 at gmail.com (Vitalie Colosov) Date: Wed, 21 Mar 2012 09:21:03 -0700 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: <4F69D4BB.7090300@puzzled.xs4all.nl> References: <4F69C9E5.5060309@puzzled.xs4all.nl> <4F69D4BB.7090300@puzzled.xs4all.nl> Message-ID: According to the FS wiki, on CentOS 6 you need to run ./configure --without-libcurl --without-pgsql. See if that helps. Release(es) 6 and Later ... Also, when using release 6 and later, make sure to configure with "./configure --without-libcurl --without-pgsql", this is to make sure that FreeSWITCH uses it's own curl library, instead of the system provided, and that it doesn't try to use the system provided postgresql libs. If using the system provided versions linking errors will occur. Hopefully this will be auto detected in a near future. Related Jira issues are: FS-3384, FS-3630 and FS-3393. 2012/3/21 Patrick Lists > On 21-03-12 13:42, Zenny wrote: > >> I don't know which distro you use but that directory (/usr/pgsql-9.1) > >> looks odd. If you use CentOS or RHEL why don't you just remove all stock > >> postgresql RPMs from your box and install a decent PostgreSQL 9.1 RPM > >> via http://yum.postgresql.org/ > > > > I am using CentOS6. I installed the RPMs directly downloading the repos > from: > > rpm -ivh > http://yum.postgresql.org/9.1/redhat/rhel-6-x86:64/pgdg-centos91-9.1-4.noarch.rpm > > I just checked the postgresql-9.1 server RPM and you are right. Wow... > The PostgreSQL packager could read up on the RPM Packaging Guide Lines > and the FHS. I guess they packaged it so that it does not interfere with > the RHEL/CentOS provided PostgreSQL RPMs but putting everything in a > non-standard location clearly causes challenges as you are experiencing > right now. I'm afraid I can't be of any help. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/1c09ffd8/attachment.html From avi at avimarcus.net Wed Mar 21 19:31:05 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 21 Mar 2012 18:31:05 +0200 Subject: [Freeswitch-users] FreeSWITCH Community Conf Call Tomorrow In-Reply-To: References: <4F69EC91.5080108@earthspike.net> Message-ID: This seems right: http://www.timeanddate.com/worldclock/fixedtime.html?msg=FreeSWITCH+Conference&iso=20120321T13&p1=179 (I let them figure out TZ stuff.) -Avi On Wed, Mar 21, 2012 at 6:18 PM, Michael Collins wrote: > 1pm EDT, 10am PDT. I think it's 1700 UTC during DST but if someone with > more TZ savvy than I could chime in I would appreciate it. > > Thanks! > -MC > > > On Wed, Mar 21, 2012 at 7:58 AM, John wrote: > >> On 20/03/2012 16:19, Michael Collins wrote: >> > Hey gang! >> > >> > Tomorrow's agenda page is here: >> > >> > http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_21 >> Can you confirm the timings? UTC doesn't do DST but I think EST and PST >> do... >> >> Thanks, >> >> John >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/34083e68/attachment.html From msc at freeswitch.org Wed Mar 21 19:33:26 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Mar 2012 09:33:26 -0700 Subject: [Freeswitch-users] make current caused problems In-Reply-To: <065101cd06ab$43733390$ca599ab0$@bizfocused.com> References: <065101cd06ab$43733390$ca599ab0$@bizfocused.com> Message-ID: Confirm that you have mod_shout and any other non-default modules enabled in modules.conf in your FreeSWITCH source directory. Remember that a make current will do a make clean. After you enable the items in modules.conf you may want to run another make current and capture the output for later analysis - there may be something unusual happening but you won't necessarily see it with all the thousands of lines of output from the build process. -MC On Tue, Mar 20, 2012 at 8:08 AM, Sean Devoy wrote: > Hi,**** > > ** ** > > I did a ?make current? last night. Today none of my IVRs were working. > .MP3 files were no longer recognized.**** > > ** ** > > I went back and did a ?make mod_shout-install? and rectified the > situation. This should be in the wiki.**** > > ** ** > > Is there something I can do to have all my user added modules recompiled > and installed with a ?make current?. I am pretty sure I added one other > module that is not the default, but I don?t recall what it was.**** > > ** ** > > Thanks.**** > > Sean**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/69b415c5/attachment-0001.html From basit.engg at gmail.com Wed Mar 21 19:35:23 2012 From: basit.engg at gmail.com (Abdul Basit) Date: Wed, 21 Mar 2012 21:35:23 +0500 Subject: [Freeswitch-users] FreeSWITCH Community Conf Call Tomorrow In-Reply-To: References: <4F69EC91.5080108@earthspike.net> Message-ID: +1 -- regards, abdul basit On Wed, Mar 21, 2012 at 9:31 PM, Avi Marcus wrote: > This seems right: > > > http://www.timeanddate.com/worldclock/fixedtime.html?msg=FreeSWITCH+Conference&iso=20120321T13&p1=179 > > (I let them figure out TZ stuff.) > -Avi > > > > On Wed, Mar 21, 2012 at 6:18 PM, Michael Collins wrote: > >> 1pm EDT, 10am PDT. I think it's 1700 UTC during DST but if someone with >> more TZ savvy than I could chime in I would appreciate it. >> >> Thanks! >> -MC >> >> >> On Wed, Mar 21, 2012 at 7:58 AM, John wrote: >> >>> On 20/03/2012 16:19, Michael Collins wrote: >>> > Hey gang! >>> > >>> > Tomorrow's agenda page is here: >>> > >>> > http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_21 >>> Can you confirm the timings? UTC doesn't do DST but I think EST and PST >>> do... >>> >>> Thanks, >>> >>> John >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/1c74734f/attachment.html From garbytrash at gmail.com Wed Mar 21 19:48:43 2012 From: garbytrash at gmail.com (Zenny) Date: Wed, 21 Mar 2012 16:48:43 +0000 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: References: <4F69C9E5.5060309@puzzled.xs4all.nl> <4F69D4BB.7090300@puzzled.xs4all.nl> Message-ID: On 3/21/12, Vitalie Colosov wrote: > According to the FS wiki, on CentOS 6 you need to run ./configure > --without-libcurl --without-pgsql. See if that helps. Tried also with above, but ends up in the same error saying libpq-fe.h not found! Hmmmmm! > > Release(es) 6 and Later > > ... > > Also, when using release 6 and later, make sure to configure with > "./configure --without-libcurl --without-pgsql", this is to make sure that > FreeSWITCH uses it's own curl library, instead of the system provided, and > that it doesn't try to use the system provided postgresql libs. If using > the system provided versions linking errors will occur. Hopefully this will > be auto detected in a near future. Related Jira issues are: FS-3384, > FS-3630 and FS-3393. > > > > > > > > > 2012/3/21 Patrick Lists > >> On 21-03-12 13:42, Zenny wrote: >> >> I don't know which distro you use but that directory (/usr/pgsql-9.1) >> >> looks odd. If you use CentOS or RHEL why don't you just remove all >> >> stock >> >> postgresql RPMs from your box and install a decent PostgreSQL 9.1 RPM >> >> via http://yum.postgresql.org/ >> > >> > I am using CentOS6. I installed the RPMs directly downloading the repos >> from: >> > rpm -ivh >> http://yum.postgresql.org/9.1/redhat/rhel-6-x86:64/pgdg-centos91-9.1-4.noarch.rpm >> >> I just checked the postgresql-9.1 server RPM and you are right. Wow... >> The PostgreSQL packager could read up on the RPM Packaging Guide Lines >> and the FHS. I guess they packaged it so that it does not interfere with >> the RHEL/CentOS provided PostgreSQL RPMs but putting everything in a >> non-standard location clearly causes challenges as you are experiencing >> right now. I'm afraid I can't be of any help. >> >> Regards, >> Patrick >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From lazyvirus at gmx.com Wed Mar 21 19:56:04 2012 From: lazyvirus at gmx.com (Bzzz) Date: Wed, 21 Mar 2012 17:56:04 +0100 Subject: [Freeswitch-users] FS compilation errored out at building mod_cdr_pg_csv In-Reply-To: References: <4F69C9E5.5060309@puzzled.xs4all.nl> <4F69D4BB.7090300@puzzled.xs4all.nl> Message-ID: <20120321175604.30f13b53@anubis.defcon1> On Wed, 21 Mar 2012 16:48:43 +0000 Zenny wrote: > > According to the FS wiki, on CentOS 6 you need to run ./configure > > --without-libcurl --without-pgsql. See if that helps. > > Tried also with above, but ends up in the same error saying libpq-fe.h > not found! > > Hmmmmm! So, use Debian! Ok->[] -- From msc at freeswitch.org Wed Mar 21 20:00:26 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Mar 2012 10:00:26 -0700 Subject: [Freeswitch-users] issue with freeswitch_licence_server In-Reply-To: References: Message-ID: The license server *MUST* be started as root. It will update its user id for the process after it starts. -MC On Wed, Mar 21, 2012 at 5:17 AM, Ken Rice wrote: > Solution for anyone else that might run across this? This seems to be a > common occurance at times > > Thx > K > > > > On 3/21/12 7:14 AM, "Yuriy Nasida" wrote: > > Solved :) > > > ------------------------------ > From: nasida at live.ru > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 21 Mar 2012 15:49:28 +0400 > Subject: [Freeswitch-users] issue with freeswitch_licence_server > > Hello guys. > > Can anybody give me some advice with freeswitch_licence_server ? > > FS works as root. freeswitch_licence_server is started. But in fs_cli i > see: > freeswitch at internal> g729_info > Can't contact licence server. > > Thanks. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/8412ec51/attachment-0001.html From msc at freeswitch.org Wed Mar 21 20:03:54 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Mar 2012 10:03:54 -0700 Subject: [Freeswitch-users] Understanding of freeswitch Events In-Reply-To: References: Message-ID: That's a tall order. The first thing you need to do is read chapter 9 of the first FreeSWITCH book. Then you probably will need to look at A LOT of source code. You probably will need to ask specific questions in the #freeswitch-dev channel on IRC or on the freeswitch-dev mailing list. These topics are way beyond the basic user list. -MC On Sun, Mar 18, 2012 at 10:39 PM, piyush singhai wrote: > Hello All, > > I want to understand then Event Handling in freeswitch. > > 1. How events flow in freeswitch, > 2. How fs core handle the events. > 3. What mechanism used is it push architecture or pullfor events. > 4. Priorities of events in FS. > > So please help me. > > > --Piyush > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/d8bdd347/attachment.html From msc at freeswitch.org Wed Mar 21 21:15:30 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Mar 2012 11:15:30 -0700 Subject: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring In-Reply-To: <9E00A259F85C4E2E9E4E0FF09A0CC03D@KITPC003> References: <0488BB5D11C94E8285DD956672994480@KITPC003> <80F4A0FF691F488F9BE7511BD0AA4BAC@KITPC003> <9E00A259F85C4E2E9E4E0FF09A0CC03D@KITPC003> Message-ID: Rob, Check out this pastebin: http://pastebin.freeswitch.org/18712 Look at line #424. There's no more dialplan actions parsed after the bridge. Look in the extension named 'group_dial_support' and see what's in there. Confirm that you actually have something in there after the bridge. -MC On Tue, Mar 20, 2012 at 10:00 AM, Robert Longfield < robert.longfield at klinsight.com> wrote: > ugh, I can?t believe I forgot to include the pastebin > > http://pastebin.com/GYLvtDB3 > > The termination happens also when there is a single user in the call > group. The call is transferred to the extension, that extension does not > pickup and FS drops the call instead of the call going to VM. When you call > any extension directly and the call is not answered you end up in that > extensions VM like you should. > > -Rob > > *From:* Michael Collins > *Sent:* Tuesday, March 20, 2012 12:22 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Call sent to group call terminates when > group is busy instead of transferring > > Malfunction! Need input! > Get debug log of call from start to finish and put on pastebin. > -MC > > On Tue, Mar 20, 2012 at 8:19 AM, Robert Longfield < > robert.longfield at klinsight.com> wrote: > >> Thanks for the tip Brian, >> >> I tried using a loopback using the example in /dialplan/default.xml and I >> am still experiencing the same problem. >> >> I?ve tried a loopback that looks like: >> >> >> >> >> >> >> Only instead of dropping the call it seems to sleep... >> >> >> >> *From:* Brian Foster >> *Sent:* Monday, March 19, 2012 7:12 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates >> when group is busy instead of transferring >> >> >> Try using loopback when you send the call to voicemail, also see the >> local extensions dialplan located in conf/dialplan/default.xml >> On Mar 19, 2012 4:43 PM, "Robert Longfield" < >> robert.longfield at klinsight.com> wrote: >> >>> I set up a group call for our support team in which all their phones >>> ring when someone needs to speak with them. If they are busy the call >>> should be transferred to a general extension which if not answered then >>> goes to that extensions VM. >>> >>> My dialplan looks like: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> What is happening is a caller selects the support option from the IVR, >>> ever phone in the support group rings, which is what should happen. If no >>> one picks up the call Freeswitch hangs up instead of transferring the call >>> to extension 1000. You can see that I also tried to send the call directly >>> to voicemail but that didn?t work either. >>> >>> The message I see when Freeswitch hangs up is: >>> >>> Channel sofia/internal/sip:1002 at 72.38.184.18:39042 hanging up, >>> cause: USER_BUSY >>> >>> The full output from cli can be seen here: >>> http://pastebin.freeswitch.org/18696 >>> >>> I would like to get the call to transfer properly. >>> >>> Thanks >>> -Robert >>> >>> ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/2e6882b0/attachment.html From msc at freeswitch.org Wed Mar 21 21:16:36 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Mar 2012 11:16:36 -0700 Subject: [Freeswitch-users] Direct Call Pickup - Issue In-Reply-To: References: <040950C4-60EC-47AD-9230-D1EF1F4A759B@freeswitch.org> Message-ID: Could you please open a new Jira on this? I know it's not a new issue but Jira will help us keep track of it. Thanks, MC On Wed, Mar 21, 2012 at 8:54 AM, Lloyd Aloysius wrote: > Brian, > > CallPIckup problem start happening again. I use the latest git > > 1. Extension 201 Ringing > > 2. Extension 202 dial **201 and pickup the call > > 3. Extension 203 Dial **201 > > 4. Extension 203 steel the call which is connected to 202. > > > I am using Default Dial plan application > > > > > > data="${hash(select/${domain_name}-last_dial_ext/$1)}"/> > > > > > > > Thank you > Lloyd > * > * > > > > On Mon, Feb 13, 2012 at 12:21 PM, Lloyd Aloysius > wrote: > >> Thanks Brian. New version update fix the problem >> >> Lloyd >> * >> * >> >> >> >> On Fri, Feb 10, 2012 at 2:11 PM, Brian West wrote: >> >>> Please use the latest git head. >>> >>> /b >>> >>> On Feb 10, 2012, at 12:46 PM, Lloyd Aloysius wrote: >>> >>> FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) >>> >>> >>> -- >>> Brian West >>> FreeSWITCH Solutions, LLC >>> Phone: +1 (918) 420-9266 >>> Fax: +1 (918) 420-9267 >>> brian at freeswitch.org >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/c03cd2b6/attachment-0001.html From robert.longfield at klinsight.com Wed Mar 21 21:33:35 2012 From: robert.longfield at klinsight.com (Robert Longfield) Date: Wed, 21 Mar 2012 14:33:35 -0400 Subject: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring In-Reply-To: References: <0488BB5D11C94E8285DD956672994480@KITPC003><80F4A0FF691F488F9BE7511BD0AA4BAC@KITPC003><9E00A259F85C4E2E9E4E0FF09A0CC03D@KITPC003> Message-ID: <5B977376AA2547E6935E7E2792702134@KITPC003> I think there was some confusion in the conf call and apparently all I was producing was static. When a caller is passed to extension 2001 which is our support group the extensions in the group all ring (extensions 1002 and 1003). At that point if the call is not answered the call is dropped or more recently it seems that FS is going to sleep and not dropping the call but giving only static. Here is a paste using the right one :) http://pastebin.freeswitch.org/18714 Thanks, -Rob From: Michael Collins Sent: Wednesday, March 21, 2012 2:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Rob, Check out this pastebin: http://pastebin.freeswitch.org/18712 Look at line #424. There's no more dialplan actions parsed after the bridge. Look in the extension named 'group_dial_support' and see what's in there. Confirm that you actually have something in there after the bridge. -MC On Tue, Mar 20, 2012 at 10:00 AM, Robert Longfield wrote: ugh, I can?t believe I forgot to include the pastebin http://pastebin.com/GYLvtDB3 The termination happens also when there is a single user in the call group. The call is transferred to the extension, that extension does not pickup and FS drops the call instead of the call going to VM. When you call any extension directly and the call is not answered you end up in that extensions VM like you should. -Rob From: Michael Collins Sent: Tuesday, March 20, 2012 12:22 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Malfunction! Need input! Get debug log of call from start to finish and put on pastebin. -MC On Tue, Mar 20, 2012 at 8:19 AM, Robert Longfield wrote: Thanks for the tip Brian, I tried using a loopback using the example in /dialplan/default.xml and I am still experiencing the same problem. I?ve tried a loopback that looks like: Only instead of dropping the call it seems to sleep... From: Brian Foster Sent: Monday, March 19, 2012 7:12 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml On Mar 19, 2012 4:43 PM, "Robert Longfield" wrote: I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM. My dialplan looks like: What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn?t work either. The message I see when Freeswitch hangs up is: Channel sofia/internal/sip:1002 at 72.38.184.18:39042 hanging up, cause: USER_BUSY The full output from cli can be seen here: http://pastebin.freeswitch.org/18696 I would like to get the call to transfer properly. Thanks -Robert ------------------------------------------------------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/abaecadd/attachment.html From vishal.kakkar at gmail.com Wed Mar 21 21:44:19 2012 From: vishal.kakkar at gmail.com (Vishal Kakkar) Date: Thu, 22 Mar 2012 00:14:19 +0530 Subject: [Freeswitch-users] Mod_CDR and FIFO Message-ID: Hi all, I am using fifo to forward an incoming call to another endpoint.. and using xml_cdr for CDR tracking. CDR works good when i have *param name="log-b-leg" value="false". *Only one leg is reported.* *It creates log for *CallerID->FSNumber *and correct duration.* * But the moment i change* param name="log-b-leg" value="true". It screws up both the CDRs. *In B-Leg CDR both the calling and called number are same. i.e. *FWDToNumber to **FWDToNumber .(Why same)* IN A-Leg duration is smaller than the B-leg so it seems as if both legs got switched.*It is being generated from CallerID ->**FSNumber. (Only duraton issue)* * Following is my config- * {fifo_member_wait=nowait,ignore_early_media=true,hangup_after_bridge=true,origination_caller_id_number= *FSNumber*}freetdm/wp2/a/*FWDToNumber* * *I thought it should be widely used scenario. Please help if i am doing anything wrong. Thanks, -Manav.. * * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/f6f03277/attachment.html From gchen00 at insightbb.com Wed Mar 21 22:05:52 2012 From: gchen00 at insightbb.com (GCHEN00) Date: Wed, 21 Mar 2012 15:05:52 -0400 Subject: [Freeswitch-users] Using mysql cluster as core db Message-ID: mysql cluster 7.2.4 and newest FS I am trying to use ODBC to connect mysql cluster database. When FS start, it has ERRORs to create the tables channel and interfaces, So I tried to manually create these two tables with schema from non-cluster mysql database. With engine=NDBCLUSTER for the tables, I got this error: "The maximum row size for the used table type, not counting BLOBs, is 14000. You have to change some columns to TEXT or BLOBs" I can use engine=InnoDB but not NDBCLUSTER. So I changed fields 'presence_id' and 'presence_data' in channel table from size 4096 to 512 and field 'description' in interfaces table from 4096 to 1024. And it worked. I am wordering will these table changes cause any problem down the road. Any other suggestion to solve this problem? Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/679f293e/attachment-0001.html From bdfoster at endigotech.com Wed Mar 21 23:37:53 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 21 Mar 2012 16:37:53 -0400 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after~30seconds In-Reply-To: References: <1363342dbcd.1079533506949741633.-8314378552953017400@zoho.com> <4F6970A2.000019.15156@FLIGHTPC> Message-ID: I'm going to go ahead and mark this issue as solved. The solution was to do a make current. There was a bug in FreeSWITCH that was fixed before Anthony went on vacation. If anyone was on conference or if your name is Brian West, if there was a specific revision that this was fixed on please chime in for anyone else that may have this issue. Fwiw I actually went ahead and started with a clean install (due to deleting the source folder and also not knowing what all I messed with to try and solve this issue). After I did the install, both the external and internal profile were getting the local ipv4 address. To fix this, I went to conf/sip_profiles/external.xml and changed the ext-sip-ip and ext-rtp-ip from $${local_ip_v4} to stun:stun.freeswitch.org. After restarting the profile, the external profile was getting my external IP and my internal profile was getting the local IP. Thanks to everyone on the conference and on the thread for helping out. Brian, for repayment, I'll get you that pic you've always wanted lol. -BDF On Mar 21, 2012 8:43 AM, "Andrew Cassidy" wrote: > I think we also need to check those variables on the internal profile. As > said previously it looks like it's doing the tun on the internal profile > which is why the ack is being send to the wrong place. > > On 21 March 2012 06:09, Fernando - NextBilling IP Solutions < > fernandojdk at gmail.com> wrote: > >> What you have in your external profile on variables ext-rtp-ip end >> ext-sip-ip? >> ? >> ** >> >> * >> Best Regards, >> Fernando da Silva Santos >> NextBilling IP Solutions LTDA >> Phone: +55 21 2143-9000 >> MSN: fernandojdk at gmail.com >> www.nextbilling.com.br >> Rio de Janeiro, Brazil, BR >> Importante: >> Esta mensagem, incluindo todo seu conte?do, cont?m informa??es >> confidenciais, legalmente protegidas e destinadas a indiv?duo e prop?sito >> espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter >> sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o >> mais breve poss?vel. >> As informa??es contidas nesta mensagem e em seu conte?do s?o de >> responsabilidade de seu autor, n?o representando necessariamente id?ias, >> opini?es, pensamentos ou qualquer forma de posicionamento por parte da >> NextBilling IP Solutions. >> P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." >> * >> *-------Original Message-------* >> >> *From:* dingdong >> *Date:* 21/03/2012 02:57:02 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] NAT issues - Outbound Call drops >> after~30seconds >> >> id like to know more bout wireshark and sip debugging,please include this >> on the conf call topics >> >> ---- On Tue, 20 Mar 2012 10:15:20 -0700 *Michael Collins < >> msc at freeswitch.org>* wrote ---- >> >> This is why I love Wireshark so much! Look at this purdee graph it makes: >> >> >> >> >> See all those 200 OK's that your FS is sending to the Grandstream? Guess >> what your GS is sending in response to those: NADA! If you look at the BYE >> that FS sends to the GS you'll even see the reason: >> >> SIP;cause=408;text=\"ACK Timeout\" >> >> FS never gets an ACK back from the GS. So the question is: why? I'm >> unfamiliar with the GS so I'll have to defer to those with more experience >> than I. However, I think you'll find that tcpdumps and analyzing w/ >> Wireshark is extremely helpful. (Open the pcap, click "Telephony > VoIP >> calls" and then a new dialog opens. In this case it shows two calls - >> meaning two call legs. Click "Select All" then click "Flow" and you'll get >> the cool graph. Click around and see what other stuff does. :) >> >> I'm thinking of doing a FreeSWITCH conference call presentation on the >> subject of collecting pcaps and doing Wireshark analysis. Let me know if >> you guys think that's a good presentation. >> >> -MC >> >> >> On Tue, Mar 20, 2012 at 10:00 AM, Brian Foster wrote: >> Andrew, >> >> root at homeserver:/usr/local/stund# ./client stunserver.org >> STUN client version 0.97 >> Primary: Independent Mapping, Independent Filter, preserves ports, will >> hairpin >> Return value is 0x000003 >> >> http://da1.endigovoip.com/dump.pcap >> >> Kristian, >> >> http://pastebin.freeswitch.org/18708 >> >> Michael, >> >> I did replace the IP's for security purposes, but now I've realized that >> it's needed and it's not really that big of a deal. I'll end up changing >> the Flowroute creds after this is fixed up. The prior siptrace is exactly >> one call (two legs). I don't think it's a carrier issue, as I've tried >> calling a buddy's server direct sip with the same issues. >> >> -BDF >> >> On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins wrote: >> We have scores of machines behind NAT talking to Flowroute with no >> problems, so there's got to be something potentially non-obvious but easy >> that needs to be set/unset. I noticed in the SIP trace that there are >> several calls. It's hard to know what's what. I think your best bet is a >> pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I >> also noticed that you redacted IP addrs - you won't be able to do this with >> a pcap. If security is an issue then I'd say get the pcap and let us know >> here on the list, then those who can have a look will email you privately >> and you can send the pcap file to them. >> >> -MC >> >> >> On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster wrote: >> Alright, so I admit... I'm a little rusty when it comes to NAT, etc. I've >> only set up FS so far on machines with no NAT, so this is sort of a new >> experience for me. >> >> I have a FreeSWITCH server located on the same local network as all of my >> phones here at the house. When I try to make a call to Flowroute, after >> about 30 seconds the call drops. It also does the exact same thing when I >> call a buddy's server directly via SIP. >> >> Here's a siptrace of the call (I didn't think that the actual FS log >> would be much help): >> http://pastebin.freeswitch.org/18697 >> >> ...and here's a paste of 'sofia status': >> http://pastebin.freeswitch.org/18698 >> >> ...and just for good measure, here's a paste of vars.xml: >> http://pastebin.freeswitch.org/18699 >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Andrew Cassidy BSc (Hons) MBCS > Managing Director; Cassidy Web Services Ltd > T: 03300 100 960 F: 03300 100 961 > E: andrew at cassidywebservices.co.uk > W: www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/f3352918/attachment-0001.html From bdfoster at endigotech.com Wed Mar 21 23:44:05 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 21 Mar 2012 16:44:05 -0400 Subject: [Freeswitch-users] make current caused problems In-Reply-To: References: <065101cd06ab$43733390$ca599ab0$@bizfocused.com> Message-ID: Going off of what MSC said, more than likely you installed the specific modules just by doing a make install on the specific module. If it was done that way, those changes aren't made in the modules.conf file. Usually when I want to install a module, I just uncomment it in modules.conf, and go ahead and do the make current. I know that some aren't really able to do that, but I just like staying in sync with git head as much as possible. -BDF On Mar 21, 2012 12:34 PM, "Michael Collins" wrote: > Confirm that you have mod_shout and any other non-default modules enabled > in modules.conf in your FreeSWITCH source directory. Remember that a make > current will do a make clean. After you enable the items in modules.conf > you may want to run another make current and capture the output for later > analysis - there may be something unusual happening but you won't > necessarily see it with all the thousands of lines of output from the build > process. > > -MC > > On Tue, Mar 20, 2012 at 8:08 AM, Sean Devoy wrote: > >> Hi,**** >> >> ** ** >> >> I did a ?make current? last night. Today none of my IVRs were working. >> .MP3 files were no longer recognized.**** >> >> ** ** >> >> I went back and did a ?make mod_shout-install? and rectified the >> situation. This should be in the wiki.**** >> >> ** ** >> >> Is there something I can do to have all my user added modules recompiled >> and installed with a ?make current?. I am pretty sure I added one other >> module that is not the default, but I don?t recall what it was.**** >> >> ** ** >> >> Thanks.**** >> >> Sean**** >> >> ** ** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/b2be0a30/attachment.html From tculjaga at gmail.com Thu Mar 22 00:00:49 2012 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 21 Mar 2012 22:00:49 +0100 Subject: [Freeswitch-users] rad_auth faild to load modul In-Reply-To: <4F69C80B.50606@puzzled.xs4all.nl> References: <4F687D5B.7050200@softnet.si> <4F69012C.5020403@puzzled.xs4all.nl> <4F697EED.5030607@softnet.si> <4F69C80B.50606@puzzled.xs4all.nl> Message-ID: and i was wondering where the heck did u find 2.x version of freeradius-client :P so mistery solved... On Wed, Mar 21, 2012 at 1:22 PM, Patrick Lists < freeswitch-list at puzzled.xs4all.nl> wrote: > On 21-03-12 08:10, Miha wrote: > > Hi Patric, > > > > Freeradius is not causing the problem, the problem is freeradius-client > > as freeradius team is not longer developing it. I have download > > freeradius-client from wget > > ftp://ftp.freeradius.org/pub/freeradius/freeradius-client-1.1.6.tar.bz2 > > as is written on wiki. > > > > I will post this on jira. I have found one tip on google, that in make > > file I should change FREERADIUSLA=/usr/local/lib/libfreeradius-client.so > > to FREERADIUSLA=/usr/local/lib/libfreeradius-client.so. I did that and > > compile it again. Now it works. > > > > Do you maybe know Patric what this change? > > I guess you mean you changed it to > FREERADIUSLA=/usr/local/lib/libfreeradius-client.la which makes more > sense :) The FreeSWITCH build system is quite complex and I don't know > very much about it. I guess you could keep using that old 1.1.6 version > and make the change from .so to .la if that makes it work. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/59ae21c3/attachment.html From mytemike72 at gmail.com Thu Mar 22 00:19:37 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Wed, 21 Mar 2012 22:19:37 +0100 Subject: [Freeswitch-users] Originate using inbound socket connection In-Reply-To: <1FFF97C269757C458224B7C895F35F1507B744@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1507B744@cantor.std.visionutv.se> Message-ID: Thanks, that worked. Now my next problem occurs... ;-) I am trying to bridge these calls to another allready established session in the switch (in a Lua script). I can reach all data of the newly created session using my created uuid which I specified in the originate. However, the uuid_bridge does not seem to work. (it says +OK and the the uuid of the bleg) but everything stays silent and no message is being wrtitten to console. When I check console, I do notice that the new session seems to use a different uuid than I specified, (which seems strange as i can access al sesssion data uusing uuid_ commands...) string cDialString = "{origination_uuid=" + legb_uuid + ",origination_caller_id_number=" + thisAni + ",origination_caller_id_name=" + thisAniName + "}sofia/external/" + thisDestination + " &park()"; var eslEvent = eslDial.Api("originate", cDialString); dispo = eslDial.Api("uuid_getvar", legb_uuid + " endpoint_disposition").GetBody(); while (eslDial.Api("uuid_exists", thisFrom).GetBody() == "true" && eslDial.Api("uuid_exists", legb_uuid).GetBody() == "true" && dispo != "ANSWER") { .. blabla } when I am connected I just do a "uuid_bridge legb_uuid" I get a response: +OK My console says: New Channel sofia/external/316....... at xxx.xxx.xxx.xxx[7cb13e5d-1dfe-483e-a8da-ec3a592700c7] and is a completely different uuid than my legb uuid i specified and use to get to the data...... Any ideas? Thanks, Mike 2012/3/21 Peter Olsson : > You need to put the other end of the call somewhere > > The correct string is (example) "originate sofia/gateway/test/1002 &park()" > > This will call 1002 and then park the call. > > /Peter > > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz > Skickat: den 21 mars 2012 14:49 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] Originate using inbound socket connection > > Hi Guys, > > I am working on a .net ?ESL host (inbound mode) and connect to the ESL using ESLConnection. > Now I am trying to originate a call using the ESLConnection.Api() function. But it does't work as expected. > > It always returns: > === > Content-Type: api/response > Content-Length: 125 > Content-Length: 125 > > -USAGE: |&() > [] [] [] [] [] === > > But I am pretty sure my string is right: > > {origination_caller_id_number=31341.....,origination_caller_id_name=31341......}sofia/external/31634258... at xxx.xxx.xxx.xxx > > I also tried with double {{ and }} but same result. Even if I take out the whole {} string and just use "sofia/external/etc.." I get the same message back. > > My code: > > ESLconnection eslDial = new ESLconnection("x.x.x.x", "8021", "x"); if (eslDial.Connected() == ESL_SUCCESS) { > ? // Create a uuid used to identify the b-leg. > ? string legb_uuid = eslDial.Api("create_uuid", "").GetBody(); > ? string cDialString = "{{origination_uuid=" + legb_uuid + ",origination_caller_id_number=" + thisAni + ",origination_caller_id_name=" + thisAniName + "}}sofia/external/" + thisDestination; > > ? // Send the command > ? var eslEvent = eslDial.Api("originate", cDialString); > > ? // Write the result to the console > ? Console.WriteLine(eslEvent.Serialize(string.Empty)); > ? return true; > } > > Thanks for your help!, > > Regards, > Michael Lutz > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f69dabc32766734014540! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Hector.Geraldino at ipsoft.com Thu Mar 22 00:34:32 2012 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Wed, 21 Mar 2012 17:34:32 -0400 Subject: [Freeswitch-users] Originate using inbound socket connection In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1507B744@cantor.std.visionutv.se> Message-ID: <6A6B4C284AD15042B429EB9D904544AD022D77CF5C@NY1-EXMB-01.ip-soft.net> What happens if you try to do the same operation from the fs_cli ? uuid_bridge legA_uuid legB_uuid Works on the console and not when you call it from your app? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Lutz Sent: Wednesday, March 21, 2012 5:20 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Originate using inbound socket connection Thanks, that worked. Now my next problem occurs... ;-) I am trying to bridge these calls to another allready established session in the switch (in a Lua script). I can reach all data of the newly created session using my created uuid which I specified in the originate. However, the uuid_bridge does not seem to work. (it says +OK and the the uuid of the bleg) but everything stays silent and no message is being wrtitten to console. When I check console, I do notice that the new session seems to use a different uuid than I specified, (which seems strange as i can access al sesssion data uusing uuid_ commands...) string cDialString = "{origination_uuid=" + legb_uuid + ",origination_caller_id_number=" + thisAni + ",origination_caller_id_name=" + thisAniName + "}sofia/external/" + thisDestination + " &park()"; var eslEvent = eslDial.Api("originate", cDialString); dispo = eslDial.Api("uuid_getvar", legb_uuid + " endpoint_disposition").GetBody(); while (eslDial.Api("uuid_exists", thisFrom).GetBody() == "true" && eslDial.Api("uuid_exists", legb_uuid).GetBody() == "true" && dispo != "ANSWER") { .. blabla } when I am connected I just do a "uuid_bridge legb_uuid" I get a response: +OK My console says: New Channel sofia/external/316....... at xxx.xxx.xxx.xxx[7cb13e5d-1dfe-483e-a8da-ec3a592700c7] and is a completely different uuid than my legb uuid i specified and use to get to the data...... Any ideas? Thanks, Mike 2012/3/21 Peter Olsson : > You need to put the other end of the call somewhere > > The correct string is (example) "originate sofia/gateway/test/1002 &park()" > > This will call 1002 and then park the call. > > /Peter > > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz > Skickat: den 21 mars 2012 14:49 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] Originate using inbound socket connection > > Hi Guys, > > I am working on a .net ?ESL host (inbound mode) and connect to the ESL using ESLConnection. > Now I am trying to originate a call using the ESLConnection.Api() function. But it does't work as expected. > > It always returns: > === > Content-Type: api/response > Content-Length: 125 > Content-Length: 125 > > -USAGE: |&() > [] [] [] [] [] === > > But I am pretty sure my string is right: > > {origination_caller_id_number=31341.....,origination_caller_id_name=31341......}sofia/external/31634258... at xxx.xxx.xxx.xxx > > I also tried with double {{ and }} but same result. Even if I take out the whole {} string and just use "sofia/external/etc.." I get the same message back. > > My code: > > ESLconnection eslDial = new ESLconnection("x.x.x.x", "8021", "x"); if (eslDial.Connected() == ESL_SUCCESS) { > ? // Create a uuid used to identify the b-leg. > ? string legb_uuid = eslDial.Api("create_uuid", "").GetBody(); > ? string cDialString = "{{origination_uuid=" + legb_uuid + ",origination_caller_id_number=" + thisAni + ",origination_caller_id_name=" + thisAniName + "}}sofia/external/" + thisDestination; > > ? // Send the command > ? var eslEvent = eslDial.Api("originate", cDialString); > > ? // Write the result to the console > ? Console.WriteLine(eslEvent.Serialize(string.Empty)); > ? return true; > } > > Thanks for your help!, > > Regards, > Michael Lutz > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f69dabc32766734014540! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From t.mahe at b-and-c.net Wed Mar 21 23:39:33 2012 From: t.mahe at b-and-c.net (=?ISO-8859-1?Q?Tristan_Mah=E9?=) Date: Wed, 21 Mar 2012 21:39:33 +0100 Subject: [Freeswitch-users] Using mysql cluster as core db In-Reply-To: References: Message-ID: <4F6A3C85.6060700@b-and-c.net> Hi Gary, You can always hack and recompile your mysql cluster src with larger row support ( beware it's not an easy hack ). I don't know the presence stuff in FS, but if it's storing xml msgs received from UA's, you may store truncated stuff using your modified scheme and this will break functionnality, same for the 'description' field, it's been too long I didn't take a look at FS src to answer precisely... Regards, Gled. Le 21/03/2012 20:05, GCHEN00 a ?crit : > mysql cluster 7.2.4 > and newest FS > I am trying to use ODBC to connect mysql cluster database. > When FS start, it has ERRORs to create the tables channel and > interfaces, So I tried to manually create these two tables with schema > from non-cluster mysql database. > With engine=NDBCLUSTER for the tables, I got this error: "The maximum > row size for the used table type, not counting BLOBs, is 14000. You > have to change some columns to TEXT or BLOBs" > I can use engine=InnoDB but not NDBCLUSTER. > So I changed fields 'presence_id' and 'presence_data' in channel table > from size 4096 to 512 and field 'description' in interfaces table from > 4096 to 1024. And it worked. > I am wordering will these table changes cause any problem down the road. > Any other suggestion to solve this problem? > Gary > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Tristan Mah? B&C 08.25.59.50.59 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/18a80bb1/attachment.html From mytemike72 at gmail.com Thu Mar 22 01:05:21 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Wed, 21 Mar 2012 23:05:21 +0100 Subject: [Freeswitch-users] Originate using inbound socket connection In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD022D77CF5C@NY1-EXMB-01.ip-soft.net> References: <1FFF97C269757C458224B7C895F35F1507B744@cantor.std.visionutv.se> <6A6B4C284AD15042B429EB9D904544AD022D77CF5C@NY1-EXMB-01.ip-soft.net> Message-ID: Nope, it's the same, it only says OK, but nothing else... 2012/3/21 Hector Geraldino : > What happens if you try to do the same operation from the fs_cli ? > > uuid_bridge legA_uuid legB_uuid > > Works on the console and not when you call it from your app? > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Lutz > Sent: Wednesday, March 21, 2012 5:20 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Originate using inbound socket connection > > Thanks, that worked. > > Now my next problem occurs... ;-) > > I am trying to bridge these calls to another allready established > session in the switch (in a Lua script). > > I can reach all data of the newly created session using my created > uuid which I specified in the originate. > > However, the uuid_bridge does not seem to work. (it says +OK and the > the uuid of the bleg) but everything stays silent and no message is > being wrtitten to console. > > When I check console, I do notice that the new session seems to use a > different uuid than I specified, (which seems strange as i can access > al sesssion data uusing uuid_ commands...) > > string cDialString = "{origination_uuid=" + legb_uuid + > ",origination_caller_id_number=" + thisAni + > ",origination_caller_id_name=" + thisAniName + "}sofia/external/" + > thisDestination + " &park()"; > > var eslEvent = eslDial.Api("originate", cDialString); > dispo = eslDial.Api("uuid_getvar", legb_uuid + " > endpoint_disposition").GetBody(); > while (eslDial.Api("uuid_exists", thisFrom).GetBody() == "true" && > eslDial.Api("uuid_exists", legb_uuid).GetBody() == "true" && dispo != > "ANSWER") > { > ? .. > ? blabla > } > > when I am connected I just do a "uuid_bridge legb_uuid" > > I get a response: > +OK > > My console says: > New Channel sofia/external/316....... at xxx.xxx.xxx.xxx[7cb13e5d-1dfe-483e-a8da-ec3a592700c7] > > and is a completely different uuid than my legb uuid i specified and > use to get to the data...... > > Any ideas? > > Thanks, > Mike > > 2012/3/21 Peter Olsson : >> You need to put the other end of the call somewhere >> >> The correct string is (example) "originate sofia/gateway/test/1002 &park()" >> >> This will call 1002 and then park the call. >> >> /Peter >> >> >> -----Ursprungligt meddelande----- >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz >> Skickat: den 21 mars 2012 14:49 >> Till: FreeSWITCH Users Help >> ?mne: [Freeswitch-users] Originate using inbound socket connection >> >> Hi Guys, >> >> I am working on a .net ?ESL host (inbound mode) and connect to the ESL using ESLConnection. >> Now I am trying to originate a call using the ESLConnection.Api() function. But it does't work as expected. >> >> It always returns: >> === >> Content-Type: api/response >> Content-Length: 125 >> Content-Length: 125 >> >> -USAGE: |&() >> [] [] [] [] [] === >> >> But I am pretty sure my string is right: >> >> {origination_caller_id_number=31341.....,origination_caller_id_name=31341......}sofia/external/31634258... at xxx.xxx.xxx.xxx >> >> I also tried with double {{ and }} but same result. Even if I take out the whole {} string and just use "sofia/external/etc.." I get the same message back. >> >> My code: >> >> ESLconnection eslDial = new ESLconnection("x.x.x.x", "8021", "x"); if (eslDial.Connected() == ESL_SUCCESS) { >> ? // Create a uuid used to identify the b-leg. >> ? string legb_uuid = eslDial.Api("create_uuid", "").GetBody(); >> ? string cDialString = "{{origination_uuid=" + legb_uuid + ",origination_caller_id_number=" + thisAni + ",origination_caller_id_name=" + thisAniName + "}}sofia/external/" + thisDestination; >> >> ? // Send the command >> ? var eslEvent = eslDial.Api("originate", cDialString); >> >> ? // Write the result to the console >> ? Console.WriteLine(eslEvent.Serialize(string.Empty)); >> ? return true; >> } >> >> Thanks for your help!, >> >> Regards, >> Michael Lutz >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> !DSPAM:4f69dabc32766734014540! >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ash at archerdrive.com Thu Mar 22 01:14:32 2012 From: ash at archerdrive.com (Ash) Date: Thu, 22 Mar 2012 09:14:32 +1100 Subject: [Freeswitch-users] Using mysql cluster as core db In-Reply-To: References: Message-ID: Hi Gary, I used to use mysql cluster on my tables for the freeswitch database. When I set it up I started freeswitch which created the tables as MyISAM, then I run an ALTER TABLE X type=ndbcluster on each table this worked for me. I thought you could tune the NDB config file to allow more elements. Did you try raising your Data Size and Index Sizes? There is a nab_size.pl script which can help you work things out I believe. I actually ended up moving away from NDB for my FS database as I was suffering strange lock issues and lock errors, NDB seems very limited in the locking mechanism it does and in some case I found the sip_registrations table was actually locked which caused an outage for my clients. I moved to a Innodb Active/Active replication scenario, in doing so I found that the locking issues went away and in fact the SQL queries got faster thanks to indexing. With the InnoDB setup you can still achieve high availability by using Active/Active replication and a virtual address that is monitored using heartbeat. Entirely up to you but after the experience I had I would strongly advise against NDB Cluster for your FS Core, initially it was great but as my volume increased I started seeing ODBC lock timeout messages from Freeswitch, as soon as I change the tables to Innodb the lock error messages disappeared. Cheers, Ash. On 22/03/2012, at 6:05 AM, GCHEN00 wrote: > mysql cluster 7.2.4 > and newest FS > > I am trying to use ODBC to connect mysql cluster database. > When FS start, it has ERRORs to create the tables channel and interfaces, So I tried to manually create these two tables with schema from non-cluster mysql database. > With engine=NDBCLUSTER for the tables, I got this error: "The maximum row size for the used table type, not counting BLOBs, is 14000. You have to change some columns to TEXT or BLOBs" > > I can use engine=InnoDB but not NDBCLUSTER. > > So I changed fields 'presence_id' and 'presence_data' in channel table from size 4096 to 512 and field 'description' in interfaces table from 4096 to 1024. And it worked. > > I am wordering will these table changes cause any problem down the road. > > Any other suggestion to solve this problem? > > Gary > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/727fa2a9/attachment.html From msc at freeswitch.org Thu Mar 22 03:11:21 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Mar 2012 17:11:21 -0700 Subject: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring In-Reply-To: <5B977376AA2547E6935E7E2792702134@KITPC003> References: <0488BB5D11C94E8285DD956672994480@KITPC003> <80F4A0FF691F488F9BE7511BD0AA4BAC@KITPC003> <9E00A259F85C4E2E9E4E0FF09A0CC03D@KITPC003> <5B977376AA2547E6935E7E2792702134@KITPC003> Message-ID: Okay, I have tried *really* hard and I cannot reproduce these symptoms. I've used your exact dialplan and it worked perfectly for me. I tried both group calling methods. (See default dialplan x2000 vs. x2001) I honestly think that this is a dialplan parsing issue. I suspect that you may have a rogue copy of the "group_dial_support" extension somewhere in your dialplan. The first thing I would do in your position is backup my configs, then clear out the /usr/local/freeswitch/conf directory tree. Run 'make samples' to get a fresh set of configs, then test with the stock x2000 and x2001 entries. FWIW, I've PB'd the changes I made to 2000 and 2001 and included a debug log of it working. See PB 18716, especially lines 163-165. Hope this helps... -MC On Wed, Mar 21, 2012 at 11:33 AM, Robert Longfield < robert.longfield at klinsight.com> wrote: > I think there was some confusion in the conf call and apparently all I > was producing was static. > > When a caller is passed to extension 2001 which is our support group the > extensions in the group all ring (extensions 1002 and 1003). > At that point if the call is not answered the call is dropped or more > recently it seems that FS is going to sleep and not dropping the call but > giving only static. > > Here is a paste using the right one :) > > http://pastebin.freeswitch.org/18714 > > Thanks, > -Rob > > *From:* Michael Collins > *Sent:* Wednesday, March 21, 2012 2:15 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Call sent to group call terminates when > group is busy instead of transferring > > Rob, > > Check out this pastebin: http://pastebin.freeswitch.org/18712 > > Look at line #424. There's no more dialplan actions parsed after the > bridge. Look in the extension named 'group_dial_support' and see what's in > there. Confirm that you actually have something in there after the bridge. > > -MC > > On Tue, Mar 20, 2012 at 10:00 AM, Robert Longfield < > robert.longfield at klinsight.com> wrote: > >> ugh, I can?t believe I forgot to include the pastebin >> >> http://pastebin.com/GYLvtDB3 >> >> The termination happens also when there is a single user in the call >> group. The call is transferred to the extension, that extension does not >> pickup and FS drops the call instead of the call going to VM. When you call >> any extension directly and the call is not answered you end up in that >> extensions VM like you should. >> >> -Rob >> >> *From:* Michael Collins >> *Sent:* Tuesday, March 20, 2012 12:22 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates >> when group is busy instead of transferring >> >> Malfunction! Need input! >> Get debug log of call from start to finish and put on pastebin. >> -MC >> >> On Tue, Mar 20, 2012 at 8:19 AM, Robert Longfield < >> robert.longfield at klinsight.com> wrote: >> >>> Thanks for the tip Brian, >>> >>> I tried using a loopback using the example in /dialplan/default.xml and >>> I am still experiencing the same problem. >>> >>> I?ve tried a loopback that looks like: >>> >>> >>> >>> >>> >>> >>> Only instead of dropping the call it seems to sleep... >>> >>> >>> >>> *From:* Brian Foster >>> *Sent:* Monday, March 19, 2012 7:12 PM >>> *To:* FreeSWITCH Users Help >>> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates >>> when group is busy instead of transferring >>> >>> >>> Try using loopback when you send the call to voicemail, also see the >>> local extensions dialplan located in conf/dialplan/default.xml >>> On Mar 19, 2012 4:43 PM, "Robert Longfield" < >>> robert.longfield at klinsight.com> wrote: >>> >>>> I set up a group call for our support team in which all their phones >>>> ring when someone needs to speak with them. If they are busy the call >>>> should be transferred to a general extension which if not answered then >>>> goes to that extensions VM. >>>> >>>> My dialplan looks like: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> What is happening is a caller selects the support option from the IVR, >>>> ever phone in the support group rings, which is what should happen. If no >>>> one picks up the call Freeswitch hangs up instead of transferring the call >>>> to extension 1000. You can see that I also tried to send the call directly >>>> to voicemail but that didn?t work either. >>>> >>>> The message I see when Freeswitch hangs up is: >>>> >>>> Channel sofia/internal/sip:1002 at 72.38.184.18:39042 hanging up, >>>> cause: USER_BUSY >>>> >>>> The full output from cli can be seen here: >>>> http://pastebin.freeswitch.org/18696 >>>> >>>> I would like to get the call to transfer properly. >>>> >>>> Thanks >>>> -Robert >>>> >>>> ------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/654a5bb0/attachment-0001.html From me at nybras.com Thu Mar 22 04:00:59 2012 From: me at nybras.com (Anderson Arboleya) Date: Wed, 21 Mar 2012 22:00:59 -0300 Subject: [Freeswitch-users] Gateway Configuration (asterisk example provided) Message-ID: Hello, I'm trying to connect freeswitch to a provider so I can call real numbers and reach my freeswitch server, but I'm kinda stuck in this process. My provider sent me this Asterisk's configs: ======================================================= # sip.conf: register => USERNAME:PASSWD at SERVER_IP [directcall12] context=default fromuser= USERNAME username= USERNAME secret= PASSWD type=friend host=SERVER_IP canreinvite=no nat=no dtmfmode=rfc2833 insecure=very disallow=all allow=g729 # extensions.conf: exten => 3131,1,Noop(ORIGEM - ${CALLERID(num)}) exten => 3131,n,Dial(${TRUNKPBX}/${RAMAL}) exten => 3131,n,Hangup() ** change these settings according you internal extensions configs ======================================================= Well, I tried adapting it to FreeSwitch like this (according some examples I found across the web): ======================================================= # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml # /usr/local/freeswitch/conf/dialplan/public/testing.xml ======================================================= In the end all I got is this log: 2012-03-17 01:41:23.565471 [NOTICE] sofia_reg.c:415 Registering testing 2012-03-17 01:41:55.565460 [ERR] sofia_reg.c:1962 testing Registration Failed with status Request Timeout [408]. All I want is to connect with my provider and route the calls made to the real number my provider gave me to my Dialplan, so my lua script will be called. Btw, I dont actually bought a "dedicated number" but an extension. I mean, I will call a number and an IVR-menu will ask me for an extension, then I type the extension I bought (3131) and their system will send the call for me. Is this clear? Sorry, I'm an experienced programmer but I'm just starting with all this telephony thing and I don't know what else I need to do from this point on, I have searched a lot and tried a lot of different approaches, but none succeed. Could anybody help me with this, please? Thanks in advance. Anderson Arboleya From fernandojdk at gmail.com Thu Mar 22 05:37:10 2012 From: fernandojdk at gmail.com (Fernando - NextBilling IP Solutions) Date: Wed, 21 Mar 2012 23:37:10 -0300 (Hora oficial do Brasil) Subject: [Freeswitch-users] Gateway Configuration (asterisk example provided) References: Message-ID: <4F6A9056.000003.01040@FLIGHTPC> Please read: http://wiki.freeswitch.org/wiki/Clarification:gateways You should "tell" freeswitch to register to your gateway: # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml Look the param register: Best Regards, Fernando da Silva Santos NextBilling IP Solutions LTDA Phone: +55 21 2143-9000 MSN: fernandojdk at gmail.com www.nextbilling.com.br Rio de Janeiro, Brazil, BR Importante: Esta mensagem, incluindo todo seu conte?do, cont?m informa??es confidenciais legalmente protegidas e destinadas a indiv?duo e prop?sito espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o mais breve poss?vel. As informa??es contidas nesta mensagem e em seu conte?do s?o de responsabilidade de seu autor, n?o representando necessariamente id?ias, opini?es, pensamentos ou qualquer forma de posicionamento por parte da NextBilling IP Solutions. P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." -------Original Message------- ? From: Anderson Arboleya Date: 21/03/2012 23:28:45 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Gateway Configuration (asterisk example provided) Hello, I'm trying to connect freeswitch to a provider so I can call real numbers and reach my freeswitch server, but I'm kinda stuck in this process. My provider sent me this Asterisk's configs: ======================================================= # sip.conf: register => USERNAME:PASSWD at SERVER_IP [directcall12] context=default fromuser= USERNAME username= USERNAME secret= PASSWD type=friend host=SERVER_IP canreinvite=no nat=no dtmfmode=rfc2833 insecure=very disallow=all allow=g729 # extensions.conf: exten => 3131,1,Noop(ORIGEM - ${CALLERID(num)}) exten => 3131,n,Dial(${TRUNKPBX}/${RAMAL}) exten => 3131,n,Hangup() ** change these settings according you internal extensions configs ======================================================= Well, I tried adapting it to FreeSwitch like this (according some examples I found across the web): ======================================================= # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml # /usr/local/freeswitch/conf/dialplan/public/testing.xml ======================================================= In the end all I got is this log: 2012-03-17 01:41:23.565471 [NOTICE] sofia_reg.c:415 Registering testing 2012-03-17 01:41:55.565460 [ERR] sofia_reg.c:1962 testing Registration Failed with status Request Timeout [408]. All I want is to connect with my provider and route the calls made to the real number my provider gave me to my Dialplan, so my lua script will be called. Btw, I dont actually bought a "dedicated number" but an extension. I mean, I will call a number and an IVR-menu will ask me for an extension, then I type the extension I bought (3131) and their system will send the call for me. Is this clear? Sorry, I'm an experienced programmer but I'm just starting with all this telephony thing and I don't know what else I need to do from this point on, I have searched a lot and tried a lot of different approaches, but none succeed. Could anybody help me with this, please? Thanks in advance. Anderson Arboleya _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/959f2179/attachment.html From bdfoster at endigotech.com Thu Mar 22 05:38:30 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 21 Mar 2012 22:38:30 -0400 Subject: [Freeswitch-users] Gateway Configuration (asterisk example provided) In-Reply-To: References: Message-ID: On Mar 21, 2012 10:27 PM, "Anderson Arboleya" wrote: > Hello, > > I'm trying to connect freeswitch to a provider so I can call real > numbers and reach my freeswitch server, but I'm kinda stuck in this > process. > > My provider sent me this Asterisk's configs: > > ======================================================= > # sip.conf: > register => USERNAME:PASSWD at SERVER_IP > > [directcall12] > context=default > fromuser= USERNAME > username= USERNAME > secret= PASSWD > type=friend > host=SERVER_IP > canreinvite=no > nat=no > dtmfmode=rfc2833 > insecure=very > disallow=all > allow=g729 > > # extensions.conf: > exten => 3131,1,Noop(ORIGEM - ${CALLERID(num)}) > exten => 3131,n,Dial(${TRUNKPBX}/${RAMAL}) > exten => 3131,n,Hangup() > > ** change these settings according you internal extensions configs > ======================================================= > > > Well, I tried adapting it to FreeSwitch like this (according some > examples I found across the web): > > ======================================================= > # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml > > > > > > > > > # /usr/local/freeswitch/conf/dialplan/public/testing.xml > > > > > > > > ======================================================= > > In the end all I got is this log: > > 2012-03-17 01:41:23.565471 [NOTICE] sofia_reg.c:415 Registering testing > 2012-03-17 01:41:55.565460 [ERR] sofia_reg.c:1962 testing Registration > Failed with status Request Timeout [408]. > > All I want is to connect with my provider and route the calls made to > the real number my provider gave me to my Dialplan, so my lua script > will be called. > > Btw, I dont actually bought a "dedicated number" but an extension. I > mean, I will call a number and an IVR-menu will ask me for an > extension, then I type the extension I bought (3131) and their system > will send the call for me. Is this clear? > > Sorry, I'm an experienced programmer but I'm just starting with all > this telephony thing and I don't know what else I need to do from this > point on, I have searched a lot and tried a lot of different > approaches, but none succeed. > > Could anybody help me with this, please? > > Thanks in advance. > > > Anderson Arboleya > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/4b102cf9/attachment-0001.html From bdfoster at endigotech.com Thu Mar 22 05:41:19 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 21 Mar 2012 22:41:19 -0400 Subject: [Freeswitch-users] Gateway Configuration (asterisk example provided) In-Reply-To: References: Message-ID: Whoops... By default it will try and register to the gateway name unless stated otherwise. So as your dialplan sits, its registering to "testing" literally. Should be "example.com" or the domain you're registering to. -BDF On Mar 21, 2012 10:27 PM, "Anderson Arboleya" wrote: > Hello, > > I'm trying to connect freeswitch to a provider so I can call real > numbers and reach my freeswitch server, but I'm kinda stuck in this > process. > > My provider sent me this Asterisk's configs: > > ======================================================= > # sip.conf: > register => USERNAME:PASSWD at SERVER_IP > > [directcall12] > context=default > fromuser= USERNAME > username= USERNAME > secret= PASSWD > type=friend > host=SERVER_IP > canreinvite=no > nat=no > dtmfmode=rfc2833 > insecure=very > disallow=all > allow=g729 > > # extensions.conf: > exten => 3131,1,Noop(ORIGEM - ${CALLERID(num)}) > exten => 3131,n,Dial(${TRUNKPBX}/${RAMAL}) > exten => 3131,n,Hangup() > > ** change these settings according you internal extensions configs > ======================================================= > > > Well, I tried adapting it to FreeSwitch like this (according some > examples I found across the web): > > ======================================================= > # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml > > > > > > > > > # /usr/local/freeswitch/conf/dialplan/public/testing.xml > > > > > > > > ======================================================= > > In the end all I got is this log: > > 2012-03-17 01:41:23.565471 [NOTICE] sofia_reg.c:415 Registering testing > 2012-03-17 01:41:55.565460 [ERR] sofia_reg.c:1962 testing Registration > Failed with status Request Timeout [408]. > > All I want is to connect with my provider and route the calls made to > the real number my provider gave me to my Dialplan, so my lua script > will be called. > > Btw, I dont actually bought a "dedicated number" but an extension. I > mean, I will call a number and an IVR-menu will ask me for an > extension, then I type the extension I bought (3131) and their system > will send the call for me. Is this clear? > > Sorry, I'm an experienced programmer but I'm just starting with all > this telephony thing and I don't know what else I need to do from this > point on, I have searched a lot and tried a lot of different > approaches, but none succeed. > > Could anybody help me with this, please? > > Thanks in advance. > > > Anderson Arboleya > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120321/1f01cc15/attachment.html From gregor at infomedia.si Thu Mar 22 11:02:52 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 22 Mar 2012 09:02:52 +0100 Subject: [Freeswitch-users] Myevents Message-ID: One explanation... If I use in outbound connection event plain , I get only subscribed events. But if I also use myevents command I get all events. So, does myevents command override event plain command? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/c34d464c/attachment.html From andrew at cassidywebservices.co.uk Thu Mar 22 12:41:08 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 22 Mar 2012 09:41:08 +0000 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after~30seconds In-Reply-To: References: <1363342dbcd.1079533506949741633.-8314378552953017400@zoho.com> <4F6970A2.000019.15156@FLIGHTPC> Message-ID: Glad you got your problem sorted in the end Brian. Do I also remember seeing in this thread someone else with a similar issue? On 21 March 2012 20:37, Brian Foster wrote: > I'm going to go ahead and mark this issue as solved. The solution was to > do a make current. There was a bug in FreeSWITCH that was fixed before > Anthony went on vacation. > > If anyone was on conference or if your name is Brian West, if there was a > specific revision that this was fixed on please chime in for anyone else > that may have this issue. > > Fwiw I actually went ahead and started with a clean install (due to > deleting the source folder and also not knowing what all I messed with to > try and solve this issue). After I did the install, both the external and > internal profile were getting the local ipv4 address. To fix this, I went > to conf/sip_profiles/external.xml and changed the ext-sip-ip and ext-rtp-ip > from $${local_ip_v4} to stun:stun.freeswitch.org. After restarting the > profile, the external profile was getting my external IP and my internal > profile was getting the local IP. > > Thanks to everyone on the conference and on the thread for helping out. > Brian, for repayment, I'll get you that pic you've always wanted lol. > > -BDF > On Mar 21, 2012 8:43 AM, "Andrew Cassidy" > wrote: > >> I think we also need to check those variables on the internal profile. As >> said previously it looks like it's doing the tun on the internal profile >> which is why the ack is being send to the wrong place. >> >> On 21 March 2012 06:09, Fernando - NextBilling IP Solutions < >> fernandojdk at gmail.com> wrote: >> >>> What you have in your external profile on variables ext-rtp-ip end >>> ext-sip-ip? >>> ? >>> ** >>> >>> * >>> Best Regards, >>> Fernando da Silva Santos >>> NextBilling IP Solutions LTDA >>> Phone: +55 21 2143-9000 >>> MSN: fernandojdk at gmail.com >>> www.nextbilling.com.br >>> Rio de Janeiro, Brazil, BR >>> Importante: >>> Esta mensagem, incluindo todo seu conte?do, cont?m informa??es >>> confidenciais, legalmente protegidas e destinadas a indiv?duo e prop?sito >>> espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter >>> sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o >>> mais breve poss?vel. >>> As informa??es contidas nesta mensagem e em seu conte?do s?o de >>> responsabilidade de seu autor, n?o representando necessariamente id?ias, >>> opini?es, pensamentos ou qualquer forma de posicionamento por parte da >>> NextBilling IP Solutions. >>> P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." >>> * >>> *-------Original Message-------* >>> >>> *From:* dingdong >>> *Date:* 21/03/2012 02:57:02 >>> *To:* FreeSWITCH Users Help >>> *Subject:* Re: [Freeswitch-users] NAT issues - Outbound Call drops >>> after~30seconds >>> >>> id like to know more bout wireshark and sip debugging,please include >>> this on the conf call topics >>> >>> ---- On Tue, 20 Mar 2012 10:15:20 -0700 *Michael Collins < >>> msc at freeswitch.org>* wrote ---- >>> >>> This is why I love Wireshark so much! Look at this purdee graph it makes: >>> >>> >>> >>> >>> See all those 200 OK's that your FS is sending to the Grandstream? Guess >>> what your GS is sending in response to those: NADA! If you look at the BYE >>> that FS sends to the GS you'll even see the reason: >>> >>> SIP;cause=408;text=\"ACK Timeout\" >>> >>> FS never gets an ACK back from the GS. So the question is: why? I'm >>> unfamiliar with the GS so I'll have to defer to those with more experience >>> than I. However, I think you'll find that tcpdumps and analyzing w/ >>> Wireshark is extremely helpful. (Open the pcap, click "Telephony > VoIP >>> calls" and then a new dialog opens. In this case it shows two calls - >>> meaning two call legs. Click "Select All" then click "Flow" and you'll get >>> the cool graph. Click around and see what other stuff does. :) >>> >>> I'm thinking of doing a FreeSWITCH conference call presentation on the >>> subject of collecting pcaps and doing Wireshark analysis. Let me know if >>> you guys think that's a good presentation. >>> >>> -MC >>> >>> >>> On Tue, Mar 20, 2012 at 10:00 AM, Brian Foster wrote: >>> Andrew, >>> >>> root at homeserver:/usr/local/stund# ./client stunserver.org >>> STUN client version 0.97 >>> Primary: Independent Mapping, Independent Filter, preserves ports, will >>> hairpin >>> Return value is 0x000003 >>> >>> http://da1.endigovoip.com/dump.pcap >>> >>> Kristian, >>> >>> http://pastebin.freeswitch.org/18708 >>> >>> Michael, >>> >>> I did replace the IP's for security purposes, but now I've realized that >>> it's needed and it's not really that big of a deal. I'll end up changing >>> the Flowroute creds after this is fixed up. The prior siptrace is exactly >>> one call (two legs). I don't think it's a carrier issue, as I've tried >>> calling a buddy's server direct sip with the same issues. >>> >>> -BDF >>> >>> On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins wrote: >>> We have scores of machines behind NAT talking to Flowroute with no >>> problems, so there's got to be something potentially non-obvious but easy >>> that needs to be set/unset. I noticed in the SIP trace that there are >>> several calls. It's hard to know what's what. I think your best bet is a >>> pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I >>> also noticed that you redacted IP addrs - you won't be able to do this with >>> a pcap. If security is an issue then I'd say get the pcap and let us know >>> here on the list, then those who can have a look will email you privately >>> and you can send the pcap file to them. >>> >>> -MC >>> >>> >>> On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster wrote: >>> Alright, so I admit... I'm a little rusty when it comes to NAT, etc. >>> I've only set up FS so far on machines with no NAT, so this is sort of a >>> new experience for me. >>> >>> I have a FreeSWITCH server located on the same local network as all of >>> my phones here at the house. When I try to make a call to Flowroute, after >>> about 30 seconds the call drops. It also does the exact same thing when I >>> call a buddy's server directly via SIP. >>> >>> Here's a siptrace of the call (I didn't think that the actual FS log >>> would be much help): >>> http://pastebin.freeswitch.org/18697 >>> >>> ...and here's a paste of 'sofia status': >>> http://pastebin.freeswitch.org/18698 >>> >>> ...and just for good measure, here's a paste of vars.xml: >>> http://pastebin.freeswitch.org/18699 >>> >>> >>> -- >>> Brian D. Foster >>> Endigo Computer LLC >>> Email: bdfoster at endigotech.com >>> Phone: 317-800-7876 >>> Indianapolis, Indiana, USA >>> >>> This message contains confidential information and is intended for those >>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >>> you are not the intended recipient you are notified that disclosing, >>> copying, distributing or taking any action in reliance on the contents of >>> this information is strictly prohibited. E-mail transmission cannot be >>> guaranteed to be secure or error-free as information could be intercepted, >>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >>> The sender therefore does not accept liability for any errors or omissions >>> in the contents of this message, which arise as a result of e-mail >>> transmission. If verification is required please request a hard-copy >>> version. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> -- >>> Brian D. Foster >>> Endigo Computer LLC >>> Email: bdfoster at endigotech.com >>> Phone: 317-800-7876 >>> Indianapolis, Indiana, USA >>> >>> This message contains confidential information and is intended for those >>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >>> you are not the intended recipient you are notified that disclosing, >>> copying, distributing or taking any action in reliance on the contents of >>> this information is strictly prohibited. E-mail transmission cannot be >>> guaranteed to be secure or error-free as information could be intercepted, >>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >>> The sender therefore does not accept liability for any errors or omissions >>> in the contents of this message, which arise as a result of e-mail >>> transmission. If verification is required please request a hard-copy >>> version. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Andrew Cassidy BSc (Hons) MBCS >> Managing Director; Cassidy Web Services Ltd >> T: 03300 100 960 F: 03300 100 961 >> E: andrew at cassidywebservices.co.uk >> W: www.cassidywebservices.co.uk >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/23b1617b/attachment-0001.html From me at nybras.com Thu Mar 22 13:39:21 2012 From: me at nybras.com (Anderson Arboleya) Date: Thu, 22 Mar 2012 07:39:21 -0300 Subject: [Freeswitch-users] Gateway Configuration (asterisk example provided) In-Reply-To: References: Message-ID: Hi Fernando, I added the register param too. Brian, I've change the name to the SERVER_IP my provider sent me. That's my configs now: ======================================================= # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml # /usr/local/freeswitch/conf/dialplan/public/testing.xml ======================================================= And now I got this error: ======================================================= 2012-03-22 10:28:00.125504 [NOTICE] sofia_reg.c:415 Registering 189.84.133.130 2012-03-22 10:28:32.125503 [ERR] sofia_reg.c:1962 189.84.133.130 Registration Failed with status Request Timeout [408]. failure #1 ======================================================= When I run a "sofia status" and the fs_cli I got: external::SERVER_IP gateway sip:USERNAME at SERVER_IP FAIL_WAIT Any other suggestions? Tks Anderson Arboleya On Wed, Mar 21, 2012 at 11:41 PM, Brian Foster wrote: > Whoops... > > By default it will try and register to the gateway name unless stated > otherwise. So as your dialplan sits, its registering to "testing" literally. > Should be "example.com" or the domain you're registering to. > > -BDF > > On Mar 21, 2012 10:27 PM, "Anderson Arboleya" wrote: >> >> Hello, >> >> I'm trying to connect freeswitch to a provider so I can call real >> numbers and reach my freeswitch server, but I'm kinda stuck in this >> process. >> >> My provider sent me this Asterisk's configs: >> >> ======================================================= >> # sip.conf: >> register ? => ? USERNAME:PASSWD at SERVER_IP >> >> [directcall12] >> context=default >> fromuser= USERNAME >> username= USERNAME >> secret= PASSWD >> type=friend >> host=SERVER_IP >> canreinvite=no >> nat=no >> dtmfmode=rfc2833 >> insecure=very >> disallow=all >> allow=g729 >> >> # extensions.conf: >> exten => 3131,1,Noop(ORIGEM - ${CALLERID(num)}) >> exten => 3131,n,Dial(${TRUNKPBX}/${RAMAL}) >> exten => 3131,n,Hangup() >> >> ** change these settings according you internal extensions configs >> ======================================================= >> >> >> Well, I tried adapting it to FreeSwitch like this (according some >> examples I found across the web): >> >> ======================================================= >> # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml >> >> ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? >> >> >> # /usr/local/freeswitch/conf/dialplan/public/testing.xml >> >> ? ? >> ? ? ? >> ? ? ? ? >> ? ? ? >> ? ? >> >> ======================================================= >> >> In the end all I got is this log: >> >> 2012-03-17 01:41:23.565471 [NOTICE] sofia_reg.c:415 Registering testing >> 2012-03-17 01:41:55.565460 [ERR] sofia_reg.c:1962 testing Registration >> Failed with status Request Timeout [408]. >> >> All I want is to connect with my provider and route the calls made to >> the real number my provider gave me to my Dialplan, so my lua script >> will be called. >> >> Btw, I dont actually bought a "dedicated number" but an extension. I >> mean, I will call a number and an IVR-menu will ask me for an >> extension, then I type the extension I bought (3131) and their system >> will send the call for me. Is this clear? >> >> Sorry, I'm an experienced programmer but I'm just starting with all >> this telephony thing and I don't know what else I need to do from this >> point on, I have searched a lot and tried a lot of different >> approaches, but none succeed. >> >> Could anybody help me with this, please? >> >> Thanks in advance. >> >> >> Anderson Arboleya From fernandojdk at gmail.com Thu Mar 22 13:45:40 2012 From: fernandojdk at gmail.com (Fernando - NextBilling IP Solutions) Date: Thu, 22 Mar 2012 07:45:40 -0300 (Hora oficial do Brasil) Subject: [Freeswitch-users] Gateway Configuration (asterisk exampleprovided) References: Message-ID: <4F6B02D4.000005.01040@FLIGHTPC> Its Seem that FS cannot connect to your gateway. Try the command "sofia profile external siptrace on" and see the response from your gateway. Post here the siptrace, maybe i can help. Best regards, Importante: Esta mensagem, incluindo todo seu conte?do, cont?m informa??es confidenciais legalmente protegidas e destinadas a indiv?duo e prop?sito espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o mais breve poss?vel. As informa??es contidas nesta mensagem e em seu conte?do s?o de responsabilidade de seu autor, n?o representando necessariamente id?ias, opini?es, pensamentos ou qualquer forma de posicionamento por parte da NextBilling IP Solutions. P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." -------Original Message------- From: Anderson Arboleya Date: 22/03/2012 07:40:53 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Gateway Configuration (asterisk exampleprovided) Hi Fernando, I added the register param too. Brian, I've change the name to the SERVER_IP my provider sent me. That's my configs now: ======================================================= # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml # /usr/local/freeswitch/conf/dialplan/public/testing.xml ======================================================= And now I got this error: ======================================================= 2012-03-22 10:28:00.125504 [NOTICE] sofia_reg.c:415 Registering 189.84.133 130 2012-03-22 10:28:32.125503 [ERR] sofia_reg.c:1962 189.84.133.130 Registration Failed with status Request Timeout [408]. failure #1 ======================================================= When I run a "sofia status" and the fs_cli I got: external::SERVER_IP gateway sip:USERNAME at SERVER_IP FAIL_WAIT Any other suggestions? Tks Anderson Arboleya On Wed, Mar 21, 2012 at 11:41 PM, Brian Foster wrote: > Whoops... > > By default it will try and register to the gateway name unless stated > otherwise. So as your dialplan sits, its registering to "testing" literally. > Should be "example.com" or the domain you're registering to. > > -BDF > > On Mar 21, 2012 10:27 PM, "Anderson Arboleya" wrote: >> >> Hello, >> >> I'm trying to connect freeswitch to a provider so I can call real >> numbers and reach my freeswitch server, but I'm kinda stuck in this >> process. >> >> My provider sent me this Asterisk's configs: >> >> ======================================================= >> # sip.conf: >> register => USERNAME:PASSWD at SERVER_IP >> >> [directcall12] >> context=default >> fromuser= USERNAME >> username= USERNAME >> secret= PASSWD >> type=friend >> host=SERVER_IP >> canreinvite=no >> nat=no >> dtmfmode=rfc2833 >> insecure=very >> disallow=all >> allow=g729 >> >> # extensions.conf: >> exten => 3131,1,Noop(ORIGEM - ${CALLERID(num)}) >> exten => 3131,n,Dial(${TRUNKPBX}/${RAMAL}) >> exten => 3131,n,Hangup() >> >> ** change these settings according you internal extensions configs >> ======================================================= >> >> >> Well, I tried adapting it to FreeSwitch like this (according some >> examples I found across the web): >> >> ======================================================= >> # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml >> >> >> >> >> >> >> >> >> # /usr/local/freeswitch/conf/dialplan/public/testing.xml >> >> >> >> >> >> >> >> ======================================================= >> >> In the end all I got is this log: >> >> 2012-03-17 01:41:23.565471 [NOTICE] sofia_reg.c:415 Registering testing >> 2012-03-17 01:41:55.565460 [ERR] sofia_reg.c:1962 testing Registration >> Failed with status Request Timeout [408]. >> >> All I want is to connect with my provider and route the calls made to >> the real number my provider gave me to my Dialplan, so my lua script >> will be called. >> >> Btw, I dont actually bought a "dedicated number" but an extension. I >> mean, I will call a number and an IVR-menu will ask me for an >> extension, then I type the extension I bought (3131) and their system >> will send the call for me. Is this clear? >> >> Sorry, I'm an experienced programmer but I'm just starting with all >> this telephony thing and I don't know what else I need to do from this >> point on, I have searched a lot and tried a lot of different >> approaches, but none succeed. >> >> Could anybody help me with this, please? >> >> Thanks in advance. >> >> >> Anderson Arboleya _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/b15f039f/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 20873 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/b15f039f/attachment-0001.jpe From me at nybras.com Thu Mar 22 13:55:43 2012 From: me at nybras.com (Anderson Arboleya) Date: Thu, 22 Mar 2012 07:55:43 -0300 Subject: [Freeswitch-users] Gateway Configuration (asterisk exampleprovided) In-Reply-To: <4F6B02D4.000005.01040@FLIGHTPC> References: <4F6B02D4.000005.01040@FLIGHTPC> Message-ID: Fernando, here's the result: send 662 bytes to udp/[189.84.133.130]:5060 at 10:49:40.725611: ? ?------------------------------------------------------------------------ ? ?REGISTER sip:PROVIDER_IP;transport=udp SIP/2.0 ? ?Via: SIP/2.0/UDP SERVER_IP:5080;rport;branch=z9hG4bKFHjr44Ug1NZ7e ? ?Max-Forwards: 70 ? ?From: ;tag=FN5Fjc2t9Be1K ? ?To: ? ?Call-ID: ae19b5d8-740c-11e1-a151-17725ca17356 ? ?CSeq: 25870616 REGISTER ? ?Contact: ? ?Expires: 3600 ? ?User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-914f6cb 2012-03-16 12-33-42 -0500 ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY ? ?Supported: timer, precondition, path, replaces ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ I've replaced the IPs according this aliases: PROVIDER_IP -> My Provider IP SERVER_IP -> My server IP (in which fs is running) Thank you very much for your help. Anderson Arboleya On Thu, Mar 22, 2012 at 7:45 AM, Fernando - NextBilling IP Solutions wrote: > > Its Seem that FS cannot connect to your gateway. > > Try the command "sofia profile ?external siptrace on" and see?the response from your?gateway. > > Post here the siptrace, maybe?i can help. > > Best regards, > Importante: > Esta mensagem, incluindo todo seu conte?do, cont?m informa??es confidenciais, legalmente protegidas e destinadas a indiv?duo e prop?sito espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o mais breve poss?vel. > As informa??es contidas nesta mensagem e em seu conte?do s?o de responsabilidade de seu autor, n?o representando necessariamente id?ias, opini?es, pensamentos ou qualquer forma de posicionamento por parte da NextBilling IP Solutions. > P?"Antes de imprimir pense em seu compromisso com o Meio Ambiente." > -------Original Message------- > > From: Anderson Arboleya > Date: 22/03/2012 07:40:53 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Gateway Configuration (asterisk exampleprovided) > > Hi Fernando, I added the register param too. > > Brian, I've change the name to the SERVER_IP my provider sent me. > > That's my configs now: > > ======================================================= > # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml > > ?????? > ?????????????? > ?????????????? > ?????????????? > ?????????????? > ?????? > > > # /usr/local/freeswitch/conf/dialplan/public/testing.xml > > ?? > ???? > ?????? > ???? > ?? > > ======================================================= > > And now I got this error: > > ======================================================= > 2012-03-22 10:28:00.125504 [NOTICE] sofia_reg.c:415 Registering 189.84.133.130 > 2012-03-22 10:28:32.125503 [ERR] sofia_reg.c:1962 189.84.133.130 > Registration Failed with status Request Timeout [408]. failure #1 > ======================================================= > > When I run a "sofia status" and the fs_cli I got: > > external::SERVER_IP?? gateway?? sip:USERNAME at SERVER_IP?? FAIL_WAIT > > Any other suggestions? > > Tks > > Anderson Arboleya > > > On Wed, Mar 21, 2012 at 11:41 PM, Brian Foster wrote: > > Whoops... > > > > By default it will try and register to the gateway name unless stated > > otherwise. So as your dialplan sits, its registering to "testing" literally. > > Should be "example.com" or the domain you're registering to. > > > > -BDF > > > > On Mar 21, 2012 10:27 PM, "Anderson Arboleya" wrote: > >> > >> Hello, > >> > >> I'm trying to connect freeswitch to a provider so I can call real > >> numbers and reach my freeswitch server, but I'm kinda stuck in this > >> process. > >> > >> My provider sent me this Asterisk's configs: > >> > >> ======================================================= > >> # sip.conf: > >> register ? => ? USERNAME:PASSWD at SERVER_IP > >> > >> [directcall12] > >> context=default > >> fromuser= USERNAME > >> username= USERNAME > >> secret= PASSWD > >> type=friend > >> host=SERVER_IP > >> canreinvite=no > >> nat=no > >> dtmfmode=rfc2833 > >> insecure=very > >> disallow=all > >> allow=g729 > >> > >> # extensions.conf: > >> exten => 3131,1,Noop(ORIGEM - ${CALLERID(num)}) > >> exten => 3131,n,Dial(${TRUNKPBX}/${RAMAL}) > >> exten => 3131,n,Hangup() > >> > >> ** change these settings according you internal extensions configs > >> ======================================================= > >> > >> > >> Well, I tried adapting it to FreeSwitch like this (according some > >> examples I found across the web): > >> > >> ======================================================= > >> # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml > >> > >> ? ? ? ? > >> ? ? ? ? ? ? ? ? > >> ? ? ? ? ? ? ? ? > >> ? ? ? ? ? ? ? ? > >> ? ? ? ? > >> > >> > >> # /usr/local/freeswitch/conf/dialplan/public/testing.xml > >> > >> ? ? > >> ? ? ? > >> ? ? ? ? > >> ? ? ? > >> ? ? > >> > >> ======================================================= > >> > >> In the end all I got is this log: > >> > >> 2012-03-17 01:41:23.565471 [NOTICE] sofia_reg.c:415 Registering testing > >> 2012-03-17 01:41:55.565460 [ERR] sofia_reg.c:1962 testing Registration > >> Failed with status Request Timeout [408]. > >> > >> All I want is to connect with my provider and route the calls made to > >> the real number my provider gave me to my Dialplan, so my lua script > >> will be called. > >> > >> Btw, I dont actually bought a "dedicated number" but an extension. I > >> mean, I will call a number and an IVR-menu will ask me for an > >> extension, then I type the extension I bought (3131) and their system > >> will send the call for me. Is this clear? > >> > >> Sorry, I'm an experienced programmer but I'm just starting with all > >> this telephony thing and I don't know what else I need to do from this > >> point on, I have searched a lot and tried a lot of different > >> approaches, but none succeed. > >> > >> Could anybody help me with this, please? > >> > >> Thanks in advance. > >> > >> > >> Anderson Arboleya From fernandojdk at gmail.com Thu Mar 22 14:04:18 2012 From: fernandojdk at gmail.com (Fernando - NextBilling IP Solutions) Date: Thu, 22 Mar 2012 08:04:18 -0300 (Hora oficial do Brasil) Subject: [Freeswitch-users] Gateway Configuration (asteriskexampleprovided) References: <4F6B02D4.000005.01040@FLIGHTPC> Message-ID: <4F6B0732.000008.01040@FLIGHTPC> This header has sent to your gateway by FS, where is the response? If has no response, maybe FS cannot reach your Gateway. ?Best Regards, Importante: Esta mensagem, incluindo todo seu conte?do, cont?m informa??es confidenciais legalmente protegidas e destinadas a indiv?duo e prop?sito espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o mais breve poss?vel. As informa??es contidas nesta mensagem e em seu conte?do s?o de responsabilidade de seu autor, n?o representando necessariamente id?ias, opini?es, pensamentos ou qualquer forma de posicionamento por parte da NextBilling IP Solutions. P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." -------Original Message------- From: Anderson Arboleya Date: 22/03/2012 07:57:15 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Gateway Configuration (asteriskexampleprovided) Fernando, here's the result: send 662 bytes to udp/[189.84.133.130]:5060 at 10:49:40.725611: ------------------------------------------------------------------------ REGISTER sip:PROVIDER_IP;transport=udp SIP/2.0 Via: SIP/2.0/UDP SERVER_IP:5080;rport;branch=z9hG4bKFHjr44Ug1NZ7e Max-Forwards: 70 From: ;tag=FN5Fjc2t9Be1K To: Call-ID: ae19b5d8-740c-11e1-a151-17725ca17356 CSeq: 25870616 REGISTER Contact: Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-914f6cb 2012-03-16 12-33-42 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ I've replaced the IPs according this aliases: PROVIDER_IP -> My Provider IP SERVER_IP -> My server IP (in which fs is running) Thank you very much for your help. Anderson Arboleya On Thu, Mar 22, 2012 at 7:45 AM, Fernando - NextBilling IP Solutions wrote: > > Its Seem that FS cannot connect to your gateway. > > Try the command "sofia profile external siptrace on" and see the response from your gateway. > > Post here the siptrace, maybe i can help. > > Best regards, > Importante: > Esta mensagem, incluindo todo seu conte?do, cont?m informa??es confidenciais, legalmente protegidas e destinadas a indiv?duo e prop?sito espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o mais breve poss?vel. > As informa??es contidas nesta mensagem e em seu conte?do s?o de responsabilidade de seu autor, n?o representando necessariamente id?ias, opini?es, pensamentos ou qualquer forma de posicionamento por parte da NextBilling IP Solutions. > P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." > -------Original Message------- > > From: Anderson Arboleya > Date: 22/03/2012 07:40:53 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Gateway Configuration (asterisk exampleprovided) > > Hi Fernando, I added the register param too. > > Brian, I've change the name to the SERVER_IP my provider sent me. > > That's my configs now: > > ======================================================= > # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml > > > > > > > > > > # /usr/local/freeswitch/conf/dialplan/public/testing.xml > > > > > > > > ======================================================= > > And now I got this error: > > ======================================================= > 2012-03-22 10:28:00.125504 [NOTICE] sofia_reg.c:415 Registering 189.84.133 130 > 2012-03-22 10:28:32.125503 [ERR] sofia_reg.c:1962 189.84.133.130 > Registration Failed with status Request Timeout [408]. failure #1 > ======================================================= > > When I run a "sofia status" and the fs_cli I got: > > external::SERVER_IP gateway sip:USERNAME at SERVER_IP FAIL_WAIT > > Any other suggestions? > > Tks > > Anderson Arboleya > > > On Wed, Mar 21, 2012 at 11:41 PM, Brian Foster wrote: > > Whoops... > > > > By default it will try and register to the gateway name unless stated > > otherwise. So as your dialplan sits, its registering to "testing" literally. > > Should be "example.com" or the domain you're registering to. > > > > -BDF > > > > On Mar 21, 2012 10:27 PM, "Anderson Arboleya" wrote: > >> > >> Hello, > >> > >> I'm trying to connect freeswitch to a provider so I can call real > >> numbers and reach my freeswitch server, but I'm kinda stuck in this > >> process. > >> > >> My provider sent me this Asterisk's configs: > >> > >> ======================================================= > >> # sip.conf: > >> register => USERNAME:PASSWD at SERVER_IP > >> > >> [directcall12] > >> context=default > >> fromuser= USERNAME > >> username= USERNAME > >> secret= PASSWD > >> type=friend > >> host=SERVER_IP > >> canreinvite=no > >> nat=no > >> dtmfmode=rfc2833 > >> insecure=very > >> disallow=all > >> allow=g729 > >> > >> # extensions.conf: > >> exten => 3131,1,Noop(ORIGEM - ${CALLERID(num)}) > >> exten => 3131,n,Dial(${TRUNKPBX}/${RAMAL}) > >> exten => 3131,n,Hangup() > >> > >> ** change these settings according you internal extensions configs > >> ======================================================= > >> > >> > >> Well, I tried adapting it to FreeSwitch like this (according some > >> examples I found across the web): > >> > >> ======================================================= > >> # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml > >> > >> > >> > >> > >> > >> > >> > >> > >> # /usr/local/freeswitch/conf/dialplan/public/testing.xml > >> > >> > >> > >> > >> > >> > >> > >> ======================================================= > >> > >> In the end all I got is this log: > >> > >> 2012-03-17 01:41:23.565471 [NOTICE] sofia_reg.c:415 Registering testing > >> 2012-03-17 01:41:55.565460 [ERR] sofia_reg.c:1962 testing Registration > >> Failed with status Request Timeout [408]. > >> > >> All I want is to connect with my provider and route the calls made to > >> the real number my provider gave me to my Dialplan, so my lua script > >> will be called. > >> > >> Btw, I dont actually bought a "dedicated number" but an extension. I > >> mean, I will call a number and an IVR-menu will ask me for an > >> extension, then I type the extension I bought (3131) and their system > >> will send the call for me. Is this clear? > >> > >> Sorry, I'm an experienced programmer but I'm just starting with all > >> this telephony thing and I don't know what else I need to do from this > >> point on, I have searched a lot and tried a lot of different > >> approaches, but none succeed. > >> > >> Could anybody help me with this, please? > >> > >> Thanks in advance. > >> > >> > >> Anderson Arboleya _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/jpeg Size: 20873 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/018fde20/attachment-0001.jpe From bwibowo at gmail.com Thu Mar 22 15:43:37 2012 From: bwibowo at gmail.com (budi wibowo) Date: Thu, 22 Mar 2012 19:43:37 +0700 Subject: [Freeswitch-users] mod h323 error Message-ID: hi i just tried to build mod h323 with ptlib 2.8.2 and h323plus-trunk, FreeSWITCH Version 1.0.head (git-9d3401e 2012-03-19 20-06-36 -0500) when i make mod_323 i got this error usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: error: 'OPAL_G7231_5k3' was not declared in this scope /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h: At global scope: /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: error: 'OPAL_G7231_5k3' has not been declared /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: error: expected ',' or '...' before string constant /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:624: error: expected constructor, destructor, or type conversion before 'class' make[4]: *** [mod_h323.lo] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_h323-all] Error 1 make[1]: *** [mod_h323] Error 2 make: *** [mod_h323] Error 2 detail error log http://pastebin.freeswitch.org/18717 help to solve this issue is welcome and appreciated regards budi wibowo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/4a2d6bcc/attachment.html From gchen00 at insightbb.com Thu Mar 22 16:06:19 2012 From: gchen00 at insightbb.com (GCHEN00) Date: Thu, 22 Mar 2012 09:06:19 -0400 Subject: [Freeswitch-users] Using mysql cluster as core db Message-ID: <785C4C8DF57F483EBBA8CAADFEC4A9BF@lightyeaaeqfyt> Thanks for the info. I am also having a lot of problems when trying to convert to mysql cluster. I may switch back to regular mysql with InnoDB. Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/c9bd5a13/attachment.html From dgarcia at anew.com.ve Thu Mar 22 16:12:19 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Thu, 22 Mar 2012 08:42:19 -0430 Subject: [Freeswitch-users] Gateway Configuration (asteriskexampleprovided) In-Reply-To: <4F6B0732.000008.01040@FLIGHTPC> References: <4F6B02D4.000005.01040@FLIGHTPC> <4F6B0732.000008.01040@FLIGHTPC> Message-ID: <4F6B2533.7030002@anew.com.ve> Hi, Have you tried the basic first? 1. Do a ping from your FS to your provider IP to check conectivity 2. Do a telnet: telne PROVIDER_IP PORT and see if it open conection 3. Have you a firewall installed? 4. Have you FS behind a router doing NAT or it has network interface conected directly to internet? On 3/22/2012 6:34 AM, Fernando - NextBilling IP Solutions wrote: > This header has sent to your gateway by FS, where is the response? > If has no response, maybe FS cannot reach your Gateway. > ?Best Regards, > ** > *Importante:* > Esta mensagem, incluindo todo seu conte?do, cont?m informa??es > confidenciais, legalmente protegidas e destinadas a indiv?duo e > prop?sito espec?ficos. Caso a tenha recebido por engano, lembramos do > car?ter sigiloso e solicitamos a gentileza de desconsider?-la e > comunicar-nos o mais breve poss?vel. > As informa??es contidas nesta mensagem e em seu conte?do s?o de > responsabilidade de seu autor, n?o representando necessariamente > id?ias, opini?es, pensamentos ou qualquer forma de posicionamento por > parte da NextBilling IP Solutions. > P* "Antes de imprimir pense em seu compromisso com o Meio Ambiente."* > /-------Original Message-------/ > /*From:*/ Anderson Arboleya > /*Date:*/ 22/03/2012 07:57:15 > /*To:*/ FreeSWITCH Users Help > > /*Subject:*/ Re: [Freeswitch-users] Gateway Configuration > (asteriskexampleprovided) > Fernando, here's the result: > send 662 bytes to udp/[189.84.133.130]:5060 at 10:49:40.725611: > > ------------------------------------------------------------------------ > REGISTER sip:PROVIDER_IP;transport=udp SIP/2.0 > Via: SIP/2.0/UDP SERVER_IP:5080;rport;branch=z9hG4bKFHjr44Ug1NZ7e > Max-Forwards: 70 > From: ;tag=FN5Fjc2t9Be1K > To: > Call-ID: ae19b5d8-740c-11e1-a151-17725ca17356 > CSeq: 25870616 REGISTER > Contact: > > > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-914f6cb 2012-03-16 > 12-33-42 -0500 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > I've replaced the IPs according this aliases: > PROVIDER_IP -> My Provider IP > SERVER_IP -> My server IP (in which fs is running) > Thank you very much for your help. > Anderson Arboleya > On Thu, Mar 22, 2012 at 7:45 AM, Fernando - NextBilling IP Solutions > > wrote: > > > > Its Seem that FS cannot connect to your gateway. > > > > Try the command "sofia profile external siptrace on" and see the > response from your gateway. > > > > Post here the siptrace, maybe i can help. > > > > Best regards, > > Importante: > > Esta mensagem, incluindo todo seu conte?do, cont?m informa??es > confidenciais, legalmente protegidas e destinadas a indiv?duo e > prop?sito espec?ficos. Caso a tenha recebido por engano, lembramos do > car?ter sigiloso e solicitamos a gentileza de desconsider?-la e > comunicar-nos o mais breve poss?vel. > > As informa??es contidas nesta mensagem e em seu conte?do s?o de > responsabilidade de seu autor, n?o representando necessariamente > id?ias, opini?es, pensamentos ou qualquer forma de posicionamento por > parte da NextBilling IP Solutions. > > P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." > > -------Original Message------- > > > > From: Anderson Arboleya > > Date: 22/03/2012 07:40:53 > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Gateway Configuration (asterisk > exampleprovided) > > > > Hi Fernando, I added the register param too. > > > > Brian, I've change the name to the SERVER_IP my provider sent me. > > > > That's my configs now: > > > > ======================================================= > > # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml > > > > > > > > > > > > > > > > > > > > # /usr/local/freeswitch/conf/dialplan/public/testing.xml > > > > > > > > > > > > > > > > ======================================================= > > > > And now I got this error: > > > > ======================================================= > > 2012-03-22 10:28:00.125504 [NOTICE] sofia_reg.c:415 Registering > 189.84.133.130 > > 2012-03-22 10:28:32.125503 [ERR] sofia_reg.c:1962 189.84.133.130 > > Registration Failed with status Request Timeout [408]. failure #1 > > ======================================================= > > > > When I run a "sofia status" and the fs_cli I got: > > > > external::SERVER_IP gateway sip:USERNAME at SERVER_IP FAIL_WAIT > > > > Any other suggestions? > > > > Tks > > > > Anderson Arboleya > > > > > > On Wed, Mar 21, 2012 at 11:41 PM, Brian Foster > > wrote: > > > Whoops... > > > > > > By default it will try and register to the gateway name unless stated > > > otherwise. So as your dialplan sits, its registering to "testing" > literally. > > > Should be "example.com" or the domain you're registering to. > > > > > > -BDF > > > > > > On Mar 21, 2012 10:27 PM, "Anderson Arboleya" > wrote: > > >> > > >> Hello, > > >> > > >> I'm trying to connect freeswitch to a provider so I can call real > > >> numbers and reach my freeswitch server, but I'm kinda stuck in this > > >> process. > > >> > > >> My provider sent me this Asterisk's configs: > > >> > > >> ======================================================= > > >> # sip.conf: > > >> register => USERNAME:PASSWD at SERVER_IP > > >> > > >> [directcall12] > > >> context=default > > >> fromuser= USERNAME > > >> username= USERNAME > > >> secret= PASSWD > > >> type=friend > > >> host=SERVER_IP > > >> canreinvite=no > > >> nat=no > > >> dtmfmode=rfc2833 > > >> insecure=very > > >> disallow=all > > >> allow=g729 > > >> > > >> # extensions.conf: > > >> exten => 3131,1,Noop(ORIGEM - ${CALLERID(num)}) > > >> exten => 3131,n,Dial(${TRUNKPBX}/${RAMAL}) > > >> exten => 3131,n,Hangup() > > >> > > >> ** change these settings according you internal extensions configs > > >> ======================================================= > > >> > > >> > > >> Well, I tried adapting it to FreeSwitch like this (according some > > >> examples I found across the web): > > >> > > >> ======================================================= > > >> # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> # /usr/local/freeswitch/conf/dialplan/public/testing.xml > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> ======================================================= > > >> > > >> In the end all I got is this log: > > >> > > >> 2012-03-17 01:41:23.565471 [NOTICE] sofia_reg.c:415 Registering > testing > > >> 2012-03-17 01:41:55.565460 [ERR] sofia_reg.c:1962 testing > Registration > > >> Failed with status Request Timeout [408]. > > >> > > >> All I want is to connect with my provider and route the calls made to > > >> the real number my provider gave me to my Dialplan, so my lua script > > >> will be called. > > >> > > >> Btw, I dont actually bought a "dedicated number" but an extension. I > > >> mean, I will call a number and an IVR-menu will ask me for an > > >> extension, then I type the extension I bought (3131) and their system > > >> will send the call for me. Is this clear? > > >> > > >> Sorry, I'm an experienced programmer but I'm just starting with all > > >> this telephony thing and I don't know what else I need to do from > this > > >> point on, I have searched a lot and tried a lot of different > > >> approaches, but none succeed. > > >> > > >> Could anybody help me with this, please? > > >> > > >> Thanks in advance. > > >> > > >> > > >> Anderson Arboleya > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1913 / Virus Database: 2114/4886 - Release Date: 03/22/12 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/5b5f42e2/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 20873 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/5b5f42e2/attachment-0001.jpe From bwibowo at gmail.com Thu Mar 22 16:14:08 2012 From: bwibowo at gmail.com (budi wibowo) Date: Thu, 22 Mar 2012 20:14:08 +0700 Subject: [Freeswitch-users] mod h323 error In-Reply-To: References: Message-ID: hi thx fixed with this tutorial try to change mod_h323 Makefile to include correct path to your ptlib and h323plus folders, mine looks like this: BASE=../../../.. #export PTLIBDIR = $(shell /usr/local/bin/ptlib-config --ptlibdir) LOCAL_CFLAGS+=-g -I/usr/local/src/ptlib-v2_10_3/include -I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exc LOCAL_LDFLAGS= -L/usr/local/lib -lopenh323 -lpt -lrt ifeq ($(shell uname -m),x86_64) LOCAL_CFLAGS+=-DP_64BIT endif include $(BASE)/build/modmake.rules again thx a lot On Thu, Mar 22, 2012 at 7:43 PM, budi wibowo wrote: > hi > i just tried to build mod h323 with ptlib 2.8.2 and h323plus-trunk, FreeSWITCH > Version 1.0.head (git-9d3401e 2012-03-19 20-06-36 -0500) > when i make mod_323 i got this error > usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: error: > 'OPAL_G7231_5k3' was not declared in this scope > /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h: At global > scope: > /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: > error: 'OPAL_G7231_5k3' has not been declared > /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: > error: expected ',' or '...' before string constant > /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:624: > error: expected constructor, destructor, or type conversion before 'class' > make[4]: *** [mod_h323.lo] Error 1 > make[3]: *** [all] Error 1 > make[2]: *** [mod_h323-all] Error 1 > make[1]: *** [mod_h323] Error 2 > make: *** [mod_h323] Error 2 > > detail error log http://pastebin.freeswitch.org/18717 > > help to solve this issue is welcome and appreciated > > > regards > > budi wibowo > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/a4cf0bf0/attachment.html From peter.olsson at visionutveckling.se Thu Mar 22 16:15:41 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 22 Mar 2012 13:15:41 +0000 Subject: [Freeswitch-users] mod h323 error Message-ID: <1FFF97C269757C458224B7C895F35F1507C35F@cantor.std.visionutv.se> It seems you might need to modify the Makefile, because of the errors reported. Please read through this information first: http://wiki.freeswitch.org/wiki/Mod_h323 I personally recommend not to use h323plus-trunk. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r budi wibowo Skickat: den 22 mars 2012 13:44 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] mod h323 error hi i just tried to build mod h323 with ptlib 2.8.2 and h323plus-trunk, FreeSWITCH Version 1.0.head (git-9d3401e 2012-03-19 20-06-36 -0500) when i make mod_323 i got this error usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: error: 'OPAL_G7231_5k3' was not declared in this scope /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h: At global scope: /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: error: 'OPAL_G7231_5k3' has not been declared /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: error: expected ',' or '...' before string constant /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:624: error: expected constructor, destructor, or type conversion before 'class' make[4]: *** [mod_h323.lo] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_h323-all] Error 1 make[1]: *** [mod_h323] Error 2 make: *** [mod_h323] Error 2 detail error log http://pastebin.freeswitch.org/18717 help to solve this issue is welcome and appreciated regards budi wibowo !DSPAM:4f6b1d7932768918913259! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/b7aef27a/attachment.html From kbdfck at gmail.com Thu Mar 22 17:16:00 2012 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Thu, 22 Mar 2012 18:16:00 +0400 Subject: [Freeswitch-users] DTMF passthrough delay / pass_rfc2833 problems In-Reply-To: References: Message-ID: Thanks Anthony, I will try this approach. 2012/3/7 Anthony Minessale > Add IGNORE_DTMF_DURATION to manual-rtp-bugs sofia profile param or > rtp_manual_rtp_bugs channel variable. > This should make FS react on receipt of the first packet in the dtmf > event. You'll still be subject to waiting for the next "first" event > before you get another. > > > On Wed, Mar 7, 2012 at 12:22 PM, Kristian Kielhofner > wrote: > > Dmitry, > > > > If your ATA devices aren't clamping the DTMF quickly enough you may > > have to just use inband. > > > > On Wed, Mar 7, 2012 at 4:29 AM, Dmitry Sytchev wrote: > >> Hi all > >> > >> I'd like to know is there a way to make freeswitch pass RFC2833 dtmf > right > >> after it receives first packets as in pass_rfc2833 mode, but still > recognize > >> DTMF for bind_meta_app or bind_digit_action? Seems when i enable > >> pass_rfc2833, bind_meta_app stops working. > >> > >> When we use ATA endpoints like Linksys PAP2T or SPA8000 without > >> pass_rfc2833, ATAs sends little piece of inband DTMF followed by RFC2833 > >> packets. While inband piece is immediately forwarded by FS, RFC2833 > packets > >> get relayed only after receiving end packets from endpoint, or at least > >> delayed, effectively making double DTMF on other side. > >> > >> We can't use no-media or proxy-media mode as we need to deal with > in-call > >> features activated by DTMF :( > >> > >> What can be done to solve this issues? > >> > >> > >> > >> -- > >> Best regards, > >> > >> Dmitry Sytchev, > >> IT Engineer > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Kristian Kielhofner > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Best regards, Dmitry Sytchev, IT Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/26c8e1d1/attachment-0001.html From genachka at gmail.com Wed Mar 21 02:50:53 2012 From: genachka at gmail.com (Gennady) Date: Tue, 20 Mar 2012 23:50:53 +0000 (UTC) Subject: [Freeswitch-users] Compiling Latest GIT References: <4F481410.1080203@privatedemail.net> Message-ID: Dome Charoenyost writes: > > apt-get install gawk > and then start first step I've installed in Ubuntu 64-bit 11.04 without issues before and am now having the exact same problem as described on a new Ubuntu 64-bit 11.10 server. I've tried to do a ./bootstrap.sh and ./configure and make -j1 as was suggested and gawk is installed and set up as alternative. But still same error. From msc at freeswitch.org Thu Mar 22 18:10:18 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Mar 2012 08:10:18 -0700 Subject: [Freeswitch-users] Myevents In-Reply-To: References: Message-ID: What is your socket command? If you do "full" then you'll get all events, otherwise you'll get the equivalent of "myevents." Check out the description here: http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Keywords -MC On Thu, Mar 22, 2012 at 1:02 AM, Gregor Nanger wrote: > One explanation... > > If I use in outbound connection event plain , I get only > subscribed events. But if I also use myevents command I get all events. > > So, does myevents command override event plain command? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/5a6c23e6/attachment.html From bdfoster at endigotech.com Thu Mar 22 18:19:50 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 22 Mar 2012 11:19:50 -0400 Subject: [Freeswitch-users] Compiling Latest GIT In-Reply-To: References: <4F481410.1080203@privatedemail.net> Message-ID: Gennady -j n isn't supported by the makefile, yet. I'd refrain from using it even if you're just throwing one thread at the cpu. -BDF On Tue, Mar 20, 2012 at 7:50 PM, Gennady wrote: > Dome Charoenyost writes: > > > > > apt-get install gawk > > and then start first step > > I've installed in Ubuntu 64-bit 11.04 without issues before and am now > having > the exact same problem as described on a new Ubuntu 64-bit 11.10 server. > > I've tried to do a ./bootstrap.sh and ./configure and make -j1 as was > suggested > and gawk is installed and set up as alternative. But still same error. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/add82e82/attachment.html From bdfoster at endigotech.com Thu Mar 22 18:22:17 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 22 Mar 2012 11:22:17 -0400 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after~30seconds In-Reply-To: References: <1363342dbcd.1079533506949741633.-8314378552953017400@zoho.com> <4F6970A2.000019.15156@FLIGHTPC> Message-ID: Yea I see another who's having reportedly the same issue. absolutely, how are you coming along? -BDF On Thu, Mar 22, 2012 at 5:41 AM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > Glad you got your problem sorted in the end Brian. > > Do I also remember seeing in this thread someone else with a similar issue? > > > On 21 March 2012 20:37, Brian Foster wrote: > >> I'm going to go ahead and mark this issue as solved. The solution was to >> do a make current. There was a bug in FreeSWITCH that was fixed before >> Anthony went on vacation. >> >> If anyone was on conference or if your name is Brian West, if there was a >> specific revision that this was fixed on please chime in for anyone else >> that may have this issue. >> >> Fwiw I actually went ahead and started with a clean install (due to >> deleting the source folder and also not knowing what all I messed with to >> try and solve this issue). After I did the install, both the external and >> internal profile were getting the local ipv4 address. To fix this, I went >> to conf/sip_profiles/external.xml and changed the ext-sip-ip and ext-rtp-ip >> from $${local_ip_v4} to stun:stun.freeswitch.org. After restarting the >> profile, the external profile was getting my external IP and my internal >> profile was getting the local IP. >> >> Thanks to everyone on the conference and on the thread for helping out. >> Brian, for repayment, I'll get you that pic you've always wanted lol. >> >> -BDF >> On Mar 21, 2012 8:43 AM, "Andrew Cassidy" < >> andrew at cassidywebservices.co.uk> wrote: >> >>> I think we also need to check those variables on the internal profile. >>> As said previously it looks like it's doing the tun on the internal profile >>> which is why the ack is being send to the wrong place. >>> >>> On 21 March 2012 06:09, Fernando - NextBilling IP Solutions < >>> fernandojdk at gmail.com> wrote: >>> >>>> What you have in your external profile on variables ext-rtp-ip end >>>> ext-sip-ip? >>>> ? >>>> ** >>>> >>>> * >>>> Best Regards, >>>> Fernando da Silva Santos >>>> NextBilling IP Solutions LTDA >>>> Phone: +55 21 2143-9000 >>>> MSN: fernandojdk at gmail.com >>>> www.nextbilling.com.br >>>> Rio de Janeiro, Brazil, BR >>>> Importante: >>>> Esta mensagem, incluindo todo seu conte?do, cont?m informa??es >>>> confidenciais, legalmente protegidas e destinadas a indiv?duo e prop?sito >>>> espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter >>>> sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o >>>> mais breve poss?vel. >>>> As informa??es contidas nesta mensagem e em seu conte?do s?o de >>>> responsabilidade de seu autor, n?o representando necessariamente id?ias, >>>> opini?es, pensamentos ou qualquer forma de posicionamento por parte da >>>> NextBilling IP Solutions. >>>> P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." >>>> * >>>> *-------Original Message-------* >>>> >>>> *From:* dingdong >>>> *Date:* 21/03/2012 02:57:02 >>>> *To:* FreeSWITCH Users Help >>>> *Subject:* Re: [Freeswitch-users] NAT issues - Outbound Call drops >>>> after~30seconds >>>> >>>> id like to know more bout wireshark and sip debugging,please include >>>> this on the conf call topics >>>> >>>> ---- On Tue, 20 Mar 2012 10:15:20 -0700 *Michael Collins < >>>> msc at freeswitch.org>* wrote ---- >>>> >>>> This is why I love Wireshark so much! Look at this purdee graph it >>>> makes: >>>> >>>> >>>> >>>> >>>> See all those 200 OK's that your FS is sending to the Grandstream? >>>> Guess what your GS is sending in response to those: NADA! If you look at >>>> the BYE that FS sends to the GS you'll even see the reason: >>>> >>>> SIP;cause=408;text=\"ACK Timeout\" >>>> >>>> FS never gets an ACK back from the GS. So the question is: why? I'm >>>> unfamiliar with the GS so I'll have to defer to those with more experience >>>> than I. However, I think you'll find that tcpdumps and analyzing w/ >>>> Wireshark is extremely helpful. (Open the pcap, click "Telephony > VoIP >>>> calls" and then a new dialog opens. In this case it shows two calls - >>>> meaning two call legs. Click "Select All" then click "Flow" and you'll get >>>> the cool graph. Click around and see what other stuff does. :) >>>> >>>> I'm thinking of doing a FreeSWITCH conference call presentation on the >>>> subject of collecting pcaps and doing Wireshark analysis. Let me know if >>>> you guys think that's a good presentation. >>>> >>>> -MC >>>> >>>> >>>> On Tue, Mar 20, 2012 at 10:00 AM, Brian Foster >>> > wrote: >>>> Andrew, >>>> >>>> root at homeserver:/usr/local/stund# ./client stunserver.org >>>> STUN client version 0.97 >>>> Primary: Independent Mapping, Independent Filter, preserves ports, will >>>> hairpin >>>> Return value is 0x000003 >>>> >>>> http://da1.endigovoip.com/dump.pcap >>>> >>>> Kristian, >>>> >>>> http://pastebin.freeswitch.org/18708 >>>> >>>> Michael, >>>> >>>> I did replace the IP's for security purposes, but now I've realized >>>> that it's needed and it's not really that big of a deal. I'll end up >>>> changing the Flowroute creds after this is fixed up. The prior siptrace is >>>> exactly one call (two legs). I don't think it's a carrier issue, as I've >>>> tried calling a buddy's server direct sip with the same issues. >>>> >>>> -BDF >>>> >>>> On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins wrote: >>>> We have scores of machines behind NAT talking to Flowroute with no >>>> problems, so there's got to be something potentially non-obvious but easy >>>> that needs to be set/unset. I noticed in the SIP trace that there are >>>> several calls. It's hard to know what's what. I think your best bet is a >>>> pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I >>>> also noticed that you redacted IP addrs - you won't be able to do this with >>>> a pcap. If security is an issue then I'd say get the pcap and let us know >>>> here on the list, then those who can have a look will email you privately >>>> and you can send the pcap file to them. >>>> >>>> -MC >>>> >>>> >>>> On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster >>> > wrote: >>>> Alright, so I admit... I'm a little rusty when it comes to NAT, etc. >>>> I've only set up FS so far on machines with no NAT, so this is sort of a >>>> new experience for me. >>>> >>>> I have a FreeSWITCH server located on the same local network as all of >>>> my phones here at the house. When I try to make a call to Flowroute, after >>>> about 30 seconds the call drops. It also does the exact same thing when I >>>> call a buddy's server directly via SIP. >>>> >>>> Here's a siptrace of the call (I didn't think that the actual FS log >>>> would be much help): >>>> http://pastebin.freeswitch.org/18697 >>>> >>>> ...and here's a paste of 'sofia status': >>>> http://pastebin.freeswitch.org/18698 >>>> >>>> ...and just for good measure, here's a paste of vars.xml: >>>> http://pastebin.freeswitch.org/18699 >>>> >>>> >>>> -- >>>> Brian D. Foster >>>> Endigo Computer LLC >>>> Email: bdfoster at endigotech.com >>>> Phone: 317-800-7876 >>>> Indianapolis, Indiana, USA >>>> >>>> This message contains confidential information and is intended for >>>> those listed in the "To:", "CC:", and/or "BCC:" fields of the message >>>> header. If you are not the intended recipient you are notified that >>>> disclosing, copying, distributing or taking any action in reliance on the >>>> contents of this information is strictly prohibited. E-mail transmission >>>> cannot be guaranteed to be secure or error-free as information could be >>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or >>>> contain viruses. The sender therefore does not accept liability for any >>>> errors or omissions in the contents of this message, which arise as a >>>> result of e-mail transmission. If verification is required please request a >>>> hard-copy version. >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> -- >>>> Brian D. Foster >>>> Endigo Computer LLC >>>> Email: bdfoster at endigotech.com >>>> Phone: 317-800-7876 >>>> Indianapolis, Indiana, USA >>>> >>>> This message contains confidential information and is intended for >>>> those listed in the "To:", "CC:", and/or "BCC:" fields of the message >>>> header. If you are not the intended recipient you are notified that >>>> disclosing, copying, distributing or taking any action in reliance on the >>>> contents of this information is strictly prohibited. E-mail transmission >>>> cannot be guaranteed to be secure or error-free as information could be >>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or >>>> contain viruses. The sender therefore does not accept liability for any >>>> errors or omissions in the contents of this message, which arise as a >>>> result of e-mail transmission. If verification is required please request a >>>> hard-copy version. >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Andrew Cassidy BSc (Hons) MBCS >>> Managing Director; Cassidy Web Services Ltd >>> T: 03300 100 960 F: 03300 100 961 >>> E: andrew at cassidywebservices.co.uk >>> W: www.cassidywebservices.co.uk >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Andrew Cassidy BSc (Hons) MBCS > Managing Director; Cassidy Web Services Ltd > T: 03300 100 960 F: 03300 100 961 > E: andrew at cassidywebservices.co.uk > W: www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/6a1df479/attachment-0001.html From Hector.Geraldino at ipsoft.com Thu Mar 22 18:38:45 2012 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Thu, 22 Mar 2012 11:38:45 -0400 Subject: [Freeswitch-users] Originate using inbound socket connection In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1507B744@cantor.std.visionutv.se> <6A6B4C284AD15042B429EB9D904544AD022D77CF5C@NY1-EXMB-01.ip-soft.net> Message-ID: <6A6B4C284AD15042B429EB9D904544AD022D77CFC9@NY1-EXMB-01.ip-soft.net> So the problem seems to be with the parameters you're using for the uuid_bridge command, not your implementation :) Can you do a '>show channels' from the fs cli and see what are the uuid's of the active channels in FS? Then use those uuids to perform the uuid_bridge and see what happens. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Lutz Sent: Wednesday, March 21, 2012 6:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Originate using inbound socket connection Nope, it's the same, it only says OK, but nothing else... 2012/3/21 Hector Geraldino : > What happens if you try to do the same operation from the fs_cli ? > > uuid_bridge legA_uuid legB_uuid > > Works on the console and not when you call it from your app? > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Lutz > Sent: Wednesday, March 21, 2012 5:20 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Originate using inbound socket connection > > Thanks, that worked. > > Now my next problem occurs... ;-) > > I am trying to bridge these calls to another allready established > session in the switch (in a Lua script). > > I can reach all data of the newly created session using my created > uuid which I specified in the originate. > > However, the uuid_bridge does not seem to work. (it says +OK and the > the uuid of the bleg) but everything stays silent and no message is > being wrtitten to console. > > When I check console, I do notice that the new session seems to use a > different uuid than I specified, (which seems strange as i can access > al sesssion data uusing uuid_ commands...) > > string cDialString = "{origination_uuid=" + legb_uuid + > ",origination_caller_id_number=" + thisAni + > ",origination_caller_id_name=" + thisAniName + "}sofia/external/" + > thisDestination + " &park()"; > > var eslEvent = eslDial.Api("originate", cDialString); > dispo = eslDial.Api("uuid_getvar", legb_uuid + " > endpoint_disposition").GetBody(); > while (eslDial.Api("uuid_exists", thisFrom).GetBody() == "true" && > eslDial.Api("uuid_exists", legb_uuid).GetBody() == "true" && dispo != > "ANSWER") > { > ? .. > ? blabla > } > > when I am connected I just do a "uuid_bridge legb_uuid" > > I get a response: > +OK > > My console says: > New Channel sofia/external/316....... at xxx.xxx.xxx.xxx[7cb13e5d-1dfe-483e-a8da-ec3a592700c7] > > and is a completely different uuid than my legb uuid i specified and > use to get to the data...... > > Any ideas? > > Thanks, > Mike > > 2012/3/21 Peter Olsson : >> You need to put the other end of the call somewhere >> >> The correct string is (example) "originate sofia/gateway/test/1002 &park()" >> >> This will call 1002 and then park the call. >> >> /Peter >> >> >> -----Ursprungligt meddelande----- >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz >> Skickat: den 21 mars 2012 14:49 >> Till: FreeSWITCH Users Help >> ?mne: [Freeswitch-users] Originate using inbound socket connection >> >> Hi Guys, >> >> I am working on a .net ?ESL host (inbound mode) and connect to the ESL using ESLConnection. >> Now I am trying to originate a call using the ESLConnection.Api() function. But it does't work as expected. >> >> It always returns: >> === >> Content-Type: api/response >> Content-Length: 125 >> Content-Length: 125 >> >> -USAGE: |&() >> [] [] [] [] [] === >> >> But I am pretty sure my string is right: >> >> {origination_caller_id_number=31341.....,origination_caller_id_name=31341......}sofia/external/31634258... at xxx.xxx.xxx.xxx >> >> I also tried with double {{ and }} but same result. Even if I take out the whole {} string and just use "sofia/external/etc.." I get the same message back. >> >> My code: >> >> ESLconnection eslDial = new ESLconnection("x.x.x.x", "8021", "x"); if (eslDial.Connected() == ESL_SUCCESS) { >> ? // Create a uuid used to identify the b-leg. >> ? string legb_uuid = eslDial.Api("create_uuid", "").GetBody(); >> ? string cDialString = "{{origination_uuid=" + legb_uuid + ",origination_caller_id_number=" + thisAni + ",origination_caller_id_name=" + thisAniName + "}}sofia/external/" + thisDestination; >> >> ? // Send the command >> ? var eslEvent = eslDial.Api("originate", cDialString); >> >> ? // Write the result to the console >> ? Console.WriteLine(eslEvent.Serialize(string.Empty)); >> ? return true; >> } >> >> Thanks for your help!, >> >> Regards, >> Michael Lutz >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> !DSPAM:4f69dabc32766734014540! >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pawel at voiceworks.pl Thu Mar 22 19:00:31 2012 From: pawel at voiceworks.pl (=?utf-8?Q?Pawe=C5=82_Pier=C5=9Bcionek?=) Date: Thu, 22 Mar 2012 17:00:31 +0100 Subject: [Freeswitch-users] Remote reason header in xml cdr Message-ID: <272D40BC-459B-4101-9A60-244C4D0B60A5@voiceworks.pl> Hi, I am originating new calls from local FS box via sofia to a remote SIP gw. Origination is directly via Socket or using LUA. Using fresh git version. Remote side responds ------------------------------------------------------------------------ recv 726 bytes from udp/[x]:15060 at 15:50:02.531445: ------------------------------------------------------------------------ SIP/2.0 503 Service Unavailable ... Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION" But local FS reports cause=41 into the CDR via mod_xml_cdr. Tried number of combinations of continue_on_fail, hangup_after_bridge, sip_ignore_remote_cause to see if it helps. My goal is to get the remote Q.850 cause into the local CDR. I cannot get/set session's variables if the session fails to connect so I cannot access sip_h_Reason and get it into the cdr :( Any ideas ? Urtho, From msc at freeswitch.org Thu Mar 22 19:01:58 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Mar 2012 09:01:58 -0700 Subject: [Freeswitch-users] FreeSWITCH Project - Call For Assistance Message-ID: Hello all! As you know, from time to time we come up with little tasks (and some not so little!) that members of the community can handle. I'd like to remind everyone of some the general things they can do to help out. I also have a few specific tasks that require some expertise in various areas. First, I'd like to have everyone check out this poston freeswitch.org. It links to a great article by Andy Lester, who has been doing FLOSS stuff for ages. He offers 14 ways that people can contribute to an open source project, most of which have little to do with actual programming. I highly recommend that you read it, regardless of how long you have been associated with FreeSWITCH or any other open source project. Even if you don't learn anything new you will have something specific to give to a newbie who asks. :) Next, I'd like to remind everyone that we can always use people who download the latest git and give it a test drive. Although we will be doing the 1.0/1.2 branching soon we will still need people to "make current" and try things out. As always, put your bug reports and feature requests in Jira. Note: the reporting bugs page of the wiki has information not only on collecting data (which helps you determine *if* it's really a bug) but also some tips on how to open a proper Jira ticket. Lastly, I'd like to remind everyone that we can always use help in verifying bug reports. It's not glamorous work but it is extremely helpful. Another area where people can help out is spreading the wealth of knowledge. We do this in three primary ways: - Answering questions on the mailing list - Helping people in the IRC channel - Updating the FreeSWITCH wiki It is part of my job to do all three of these, but as you can imagine one person can only accomplish so much. A number of community members have stepped up to help out, and to all of them I would like to say thank you. Also, I would like to remind everyone that I am available to assist those who wish to add to the wiki or who have questions about anything else with respect to answering questions from community members. Here are a few specific things that we talked about yesterday on the FreeSWITCH community conference call. If you have any knowledge or expertise in these areas I would like you to contact me off list and I can give you more information. - Wiki syntax highlighting - I installed a GeSHi-based syntax highlighteron our wiki but it is not working. Anyone with mediawiki or even just plain PHP skills could assist with troubleshooting. - FreeSWITCH debug log syntax highlighting - If anyone has any experience with creating syntax highlighting for any of the popular text editors (Emacs, VIM, Textmate, Notepad++, UltraEdit, etc.) then please let me know. I'd like to get syntax highlighting for at least one text editor in *nix and one in Windows. - Documenting new FreeSWITCH items - on the Feb 8 conference call we talked about a number of new features. Many of these need documentation on the wiki. The task here is to look at each new feature, search the wiki to see if it has been documented, and then to add documentation where needed I know this may seem like a lot of stuff but what I really would like to do is have several people each take one small piece and focus only on it. If we break these down into smaller tasks I think it will be easier to make progress. Please let me know where you would like to help out. If you're feeling adventurous you can even take on a task about which you know very little and then make it a learning project. (I do this all the time. :) Thanks again for all of your help. With the ClueCon season gearing up we are going to need you more than ever. I look forward to working with you in the coming weeks and months. -Michael S Collins -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/c7e71a3f/attachment.html From lazyvirus at gmx.com Thu Mar 22 19:23:34 2012 From: lazyvirus at gmx.com (Bzzz) Date: Thu, 22 Mar 2012 17:23:34 +0100 Subject: [Freeswitch-users] VM sound very choppy Message-ID: <20120322172334.2f0ee16a@anubis.defcon1> FS last git fusionpbx =========== Hi list, I'm testing FS w/ twinkle, put iLBC as first CODEC and don't follow far end CODEC preference. All extensions have proxy_media set. When I call the VM (*97) I get a very choppy audio. FS log says: [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16 at 8000hz 1 channels 30ms when twinkle says: ilbc I guess that the source of my problem, but how can I change that for FS to switch to iLBC? (I already modified CODECs names & orders into conf/vars.xml to set iLBX at 30i first, and twinkle is also set to 30ms) Jean-Yves -- Question authority. From robert.longfield at klinsight.com Thu Mar 22 19:27:54 2012 From: robert.longfield at klinsight.com (Robert Longfield) Date: Thu, 22 Mar 2012 12:27:54 -0400 Subject: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring In-Reply-To: References: <0488BB5D11C94E8285DD956672994480@KITPC003><80F4A0FF691F488F9BE7511BD0AA4BAC@KITPC003><9E00A259F85C4E2E9E4E0FF09A0CC03D@KITPC003><5B977376AA2547E6935E7E2792702134@KITPC003> Message-ID: Micheal, The tip about checking for a rouge copy of the ?group_dial_support? extension is exactly what the problem was. I haven?t gotten rid of default.xml and forgot to comment/remove the group_dial_support from the file. Also it seems I had to add ?sofia/internal/? in front of the extension. so became I wrongly assumed that anything in default.xml would be ignored or passed over if that rule existed elsewhere. -Rob From: Michael Collins Sent: Wednesday, March 21, 2012 8:11 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Okay, I have tried *really* hard and I cannot reproduce these symptoms. I've used your exact dialplan and it worked perfectly for me. I tried both group calling methods. (See default dialplan x2000 vs. x2001) I honestly think that this is a dialplan parsing issue. I suspect that you may have a rogue copy of the "group_dial_support" extension somewhere in your dialplan. The first thing I would do in your position is backup my configs, then clear out the /usr/local/freeswitch/conf directory tree. Run 'make samples' to get a fresh set of configs, then test with the stock x2000 and x2001 entries. FWIW, I've PB'd the changes I made to 2000 and 2001 and included a debug log of it working. See PB 18716, especially lines 163-165. Hope this helps... -MC On Wed, Mar 21, 2012 at 11:33 AM, Robert Longfield wrote: I think there was some confusion in the conf call and apparently all I was producing was static. When a caller is passed to extension 2001 which is our support group the extensions in the group all ring (extensions 1002 and 1003). At that point if the call is not answered the call is dropped or more recently it seems that FS is going to sleep and not dropping the call but giving only static. Here is a paste using the right one :) http://pastebin.freeswitch.org/18714 Thanks, -Rob From: Michael Collins Sent: Wednesday, March 21, 2012 2:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Rob, Check out this pastebin: http://pastebin.freeswitch.org/18712 Look at line #424. There's no more dialplan actions parsed after the bridge. Look in the extension named 'group_dial_support' and see what's in there. Confirm that you actually have something in there after the bridge. -MC On Tue, Mar 20, 2012 at 10:00 AM, Robert Longfield wrote: ugh, I can?t believe I forgot to include the pastebin http://pastebin.com/GYLvtDB3 The termination happens also when there is a single user in the call group. The call is transferred to the extension, that extension does not pickup and FS drops the call instead of the call going to VM. When you call any extension directly and the call is not answered you end up in that extensions VM like you should. -Rob From: Michael Collins Sent: Tuesday, March 20, 2012 12:22 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Malfunction! Need input! Get debug log of call from start to finish and put on pastebin. -MC On Tue, Mar 20, 2012 at 8:19 AM, Robert Longfield wrote: Thanks for the tip Brian, I tried using a loopback using the example in /dialplan/default.xml and I am still experiencing the same problem. I?ve tried a loopback that looks like: Only instead of dropping the call it seems to sleep... From: Brian Foster Sent: Monday, March 19, 2012 7:12 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml On Mar 19, 2012 4:43 PM, "Robert Longfield" wrote: I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM. My dialplan looks like: What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn?t work either. The message I see when Freeswitch hangs up is: Channel sofia/internal/sip:1002 at 72.38.184.18:39042 hanging up, cause: USER_BUSY The full output from cli can be seen here: http://pastebin.freeswitch.org/18696 I would like to get the call to transfer properly. Thanks -Robert ---------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/e9555256/attachment-0001.html From msc at freeswitch.org Thu Mar 22 19:53:10 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Mar 2012 09:53:10 -0700 Subject: [Freeswitch-users] VM sound very choppy In-Reply-To: <20120322172334.2f0ee16a@anubis.defcon1> References: <20120322172334.2f0ee16a@anubis.defcon1> Message-ID: No need to worry about "L16" codec - that's talking about signed linear 16 bit audio, i.e. uncompressed, raw audio. If you're talking directly to FreeSWITCH, for example when you call voicemail or call an IVR, then the FreeSWITCH side will definitely be "L16". However, if you are connecting through to another device, like another telephone, then you'll see codec negotiation. Remember: when audio passes through FreeSWITCH it always gets decoded (unless you're using proxy media, but don't do that) and re-encoded, so you'll always see references to L16 which is the "decoded" audio. The primary reason for decoding everything is that FreeSWITCH can then do fun stuff like eavesdropping, mixing audio, conference calls, etc. You can't (easily) mix encoded audio. So... your issue is most likely not with a codec mismatch but rather timing on the VM. I'll defer to those who know more about VMs and such. -MC On Thu, Mar 22, 2012 at 9:23 AM, Bzzz wrote: > FS last git > fusionpbx > =========== > > Hi list, > > I'm testing FS w/ twinkle, put iLBC as first CODEC and don't follow > far end CODEC preference. All extensions have proxy_media set. > > When I call the VM (*97) I get a very choppy audio. > > FS log says: [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16 at 8000hz1 channels 30ms > when twinkle says: ilbc > > I guess that the source of my problem, but how can I change that for > FS to switch to iLBC? (I already modified CODECs names & orders into > conf/vars.xml to set iLBX at 30i first, and twinkle is also set to 30ms) > > Jean-Yves > -- > Question authority. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/399b7427/attachment.html From msc at freeswitch.org Thu Mar 22 20:00:43 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Mar 2012 10:00:43 -0700 Subject: [Freeswitch-users] Call sent to group call terminates when group is busy instead of transferring In-Reply-To: References: <0488BB5D11C94E8285DD956672994480@KITPC003> <80F4A0FF691F488F9BE7511BD0AA4BAC@KITPC003> <9E00A259F85C4E2E9E4E0FF09A0CC03D@KITPC003> <5B977376AA2547E6935E7E2792702134@KITPC003> Message-ID: Aha! Thanks for reporting back. This is a good learning exercise for XML dialplan editing. Also, if 1000 is a registered user on your system you can use data="user/1000" on your bridge line. -MC On Thu, Mar 22, 2012 at 9:27 AM, Robert Longfield < robert.longfield at klinsight.com> wrote: > Micheal, > > The tip about checking for a rouge copy of the ?group_dial_support? > extension is exactly what the problem was. I haven?t gotten rid of > default.xml and forgot to comment/remove the group_dial_support from the > file. > > Also it seems I had to add ?sofia/internal/? in front of the extension. so > became application="bridge" data="sofia/internal/1000@$${domain}"/> > > I wrongly assumed that anything in default.xml would be ignored or passed > over if that rule existed elsewhere. > > -Rob > > *From:* Michael Collins > *Sent:* Wednesday, March 21, 2012 8:11 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Call sent to group call terminates when > group is busy instead of transferring > > Okay, I have tried *really* hard and I cannot reproduce these symptoms. > I've used your exact dialplan and it worked perfectly for me. I tried both > group calling methods. (See default dialplan x2000 vs. x2001) > > I honestly think that this is a dialplan parsing issue. I suspect that you > may have a rogue copy of the "group_dial_support" extension somewhere in > your dialplan. The first thing I would do in your position is backup my > configs, then clear out the /usr/local/freeswitch/conf directory tree. Run > 'make samples' to get a fresh set of configs, then test with the stock > x2000 and x2001 entries. FWIW, I've PB'd the changes I made to 2000 and > 2001 and included a debug log of it working. See PB 18716, especially lines > 163-165. > > Hope this helps... > > -MC > > On Wed, Mar 21, 2012 at 11:33 AM, Robert Longfield < > robert.longfield at klinsight.com> wrote: > >> I think there was some confusion in the conf call and apparently all I >> was producing was static. >> >> When a caller is passed to extension 2001 which is our support group the >> extensions in the group all ring (extensions 1002 and 1003). >> At that point if the call is not answered the call is dropped or more >> recently it seems that FS is going to sleep and not dropping the call but >> giving only static. >> >> Here is a paste using the right one :) >> >> http://pastebin.freeswitch.org/18714 >> >> Thanks, >> -Rob >> >> *From:* Michael Collins >> *Sent:* Wednesday, March 21, 2012 2:15 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates >> when group is busy instead of transferring >> >> Rob, >> >> Check out this pastebin: http://pastebin.freeswitch.org/18712 >> >> Look at line #424. There's no more dialplan actions parsed after the >> bridge. Look in the extension named 'group_dial_support' and see what's in >> there. Confirm that you actually have something in there after the bridge. >> >> -MC >> >> On Tue, Mar 20, 2012 at 10:00 AM, Robert Longfield < >> robert.longfield at klinsight.com> wrote: >> >>> ugh, I can?t believe I forgot to include the pastebin >>> >>> http://pastebin.com/GYLvtDB3 >>> >>> The termination happens also when there is a single user in the call >>> group. The call is transferred to the extension, that extension does not >>> pickup and FS drops the call instead of the call going to VM. When you call >>> any extension directly and the call is not answered you end up in that >>> extensions VM like you should. >>> >>> -Rob >>> >>> *From:* Michael Collins >>> *Sent:* Tuesday, March 20, 2012 12:22 PM >>> *To:* FreeSWITCH Users Help >>> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates >>> when group is busy instead of transferring >>> >>> Malfunction! Need input! >>> Get debug log of call from start to finish and put on pastebin. >>> -MC >>> >>> On Tue, Mar 20, 2012 at 8:19 AM, Robert Longfield < >>> robert.longfield at klinsight.com> wrote: >>> >>>> Thanks for the tip Brian, >>>> >>>> I tried using a loopback using the example in /dialplan/default.xml and >>>> I am still experiencing the same problem. >>>> >>>> I?ve tried a loopback that looks like: >>>> >>>> >>>> >>>> >>>> >>>> >>>> Only instead of dropping the call it seems to sleep... >>>> >>>> >>>> >>>> *From:* Brian Foster >>>> *Sent:* Monday, March 19, 2012 7:12 PM >>>> *To:* FreeSWITCH Users Help >>>> *Subject:* Re: [Freeswitch-users] Call sent to group call terminates >>>> when group is busy instead of transferring >>>> >>>> >>>> Try using loopback when you send the call to voicemail, also see the >>>> local extensions dialplan located in conf/dialplan/default.xml >>>> On Mar 19, 2012 4:43 PM, "Robert Longfield" < >>>> robert.longfield at klinsight.com> wrote: >>>> >>>>> I set up a group call for our support team in which all their >>>>> phones ring when someone needs to speak with them. If they are busy the >>>>> call should be transferred to a general extension which if not answered >>>>> then goes to that extensions VM. >>>>> >>>>> My dialplan looks like: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> What is happening is a caller selects the support option from the IVR, >>>>> ever phone in the support group rings, which is what should happen. If no >>>>> one picks up the call Freeswitch hangs up instead of transferring the call >>>>> to extension 1000. You can see that I also tried to send the call directly >>>>> to voicemail but that didn?t work either. >>>>> >>>>> The message I see when Freeswitch hangs up is: >>>>> >>>>> Channel sofia/internal/sip:1002 at 72.38.184.18:39042 hanging up, >>>>> cause: USER_BUSY >>>>> >>>>> The full output from cli can be seen here: >>>>> http://pastebin.freeswitch.org/18696 >>>>> >>>>> I would like to get the call to transfer properly. >>>>> >>>>> Thanks >>>>> -Robert >>>>> >>>>> ------------------------------ >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> ------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/7999686b/attachment-0001.html From lazyvirus at gmx.com Thu Mar 22 20:11:40 2012 From: lazyvirus at gmx.com (Bzzz) Date: Thu, 22 Mar 2012 18:11:40 +0100 Subject: [Freeswitch-users] VM sound very choppy In-Reply-To: References: <20120322172334.2f0ee16a@anubis.defcon1> Message-ID: <20120322181140.2550738e@anubis.defcon1> On Thu, 22 Mar 2012 09:53:10 -0700 Michael Collins wrote: > Remember: when audio passes through FreeSWITCH it always gets decoded > (unless you're using proxy media, but don't do that) Hmm, I was goofy on this one and didn't took notes about this choice, but I'm (almost) sure it was about ZRTP problems. I re-read the wiki about proxy_media and I changed "proxy_media" to "" into fusionpbx (possible choices are: "", "Bypass Media", "Bypass Media After Bridge" & "Proxy Media"), but even after a FS restart it's the same choppy sound (and logs):( > > So... your issue is most likely not with a codec mismatch but rather timing > on the VM. I'll defer to those who know more about VMs and such. In order to privilege iLBC, I modified conf/vars.xml as follow: JY -- From lazyvirus at gmx.com Thu Mar 22 20:19:41 2012 From: lazyvirus at gmx.com (Bzzz) Date: Thu, 22 Mar 2012 18:19:41 +0100 Subject: [Freeswitch-users] VM sound very choppy In-Reply-To: References: <20120322172334.2f0ee16a@anubis.defcon1> Message-ID: <20120322181941.27dcb43e@anubis.defcon1> On Thu, 22 Mar 2012 09:53:10 -0700 Michael Collins wrote: I'm really lost: I greped the log and *there is* a CODEC negotiation when FS receives '*97', but not when the VM answers!? -- From msc at freeswitch.org Thu Mar 22 20:21:58 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Mar 2012 10:21:58 -0700 Subject: [Freeswitch-users] VM sound very choppy In-Reply-To: <20120322181941.27dcb43e@anubis.defcon1> References: <20120322172334.2f0ee16a@anubis.defcon1> <20120322181941.27dcb43e@anubis.defcon1> Message-ID: pastebin.freeswitch.org please -MC On Thu, Mar 22, 2012 at 10:19 AM, Bzzz wrote: > On Thu, 22 Mar 2012 09:53:10 -0700 > Michael Collins wrote: > > I'm really lost: I greped the log and *there is* a CODEC negotiation > when FS receives '*97', but not when the VM answers!? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/8beeb38e/attachment.html From lazyvirus at gmx.com Thu Mar 22 20:28:38 2012 From: lazyvirus at gmx.com (Bzzz) Date: Thu, 22 Mar 2012 18:28:38 +0100 Subject: [Freeswitch-users] VM sound very choppy In-Reply-To: References: <20120322172334.2f0ee16a@anubis.defcon1> <20120322181941.27dcb43e@anubis.defcon1> Message-ID: <20120322182838.1fff224f@anubis.defcon1> On Thu, 22 Mar 2012 10:21:58 -0700 Michael Collins wrote: > pastebin.freeswitch.org please http://pastebin.freeswitch.org/18721 -- BOFH excuse #100: IRQ dropout From gregor at infomedia.si Thu Mar 22 21:31:16 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 22 Mar 2012 19:31:16 +0100 Subject: [Freeswitch-users] Myevents In-Reply-To: References: Message-ID: I have "full" Just want to confirm if this is by design or I missed something. I want to subscribe only to "myevents" plus only to events I want in "events plain ". But both commands don't work like this. If I use "events plain " I also get global events and not only My events... Is this by design? Maybe I should ommit "full" to get only events of this call? 2012/3/22 Michael Collins > What is your socket command? If you do "full" then you'll get all events, > otherwise you'll get the equivalent of "myevents." > > Check out the description here: > http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Keywords > > -MC > > > On Thu, Mar 22, 2012 at 1:02 AM, Gregor Nanger wrote: > >> One explanation... >> >> If I use in outbound connection event plain , I get only >> subscribed events. But if I also use myevents command I get all events. >> >> So, does myevents command override event plain command? >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/4cfc4106/attachment.html From bdfoster at endigotech.com Thu Mar 22 21:37:28 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 22 Mar 2012 14:37:28 -0400 Subject: [Freeswitch-users] FreeSWITCH Installer for Debian Message-ID: Alright, so I've managed to put together a BASH script to install FS on top of Debian (Squeeze). I took the information on the wiki, along with some best practices and threw them all together for an automated installer. I came up with this really because I'm 'lazy'. I'm on version 0.0.2, and it's not considered extremely safe due to almost no error checking. I'd like to add some of that in. Plans are in the works to do some command line switches, put some options in so that you can customize certain things about the FS install, as well as having a guided install. The guided install would really be for the beginners, and would do such things as setting the hostname, setting timezone data, basically build FreeSWITCH from an absolutely clean install. When Ken's Debian packages come out for FreeSWITCH, I plan on adding the ability to install from those packages as well as just installing from git head and (eventually) the 1.2 branch. I'd like to get some testers from the community and get some feedback on what they would like to see in this installer. It's still VERY basic, and that's a good thing! This way, I can add features from the get-go from users that are used to installing FS in a certain way. http://files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_latest.sh<< Current Version as of this writing is 0.0.2 To start, do this: cd wget files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_lastest.sh-O fs-debian-installer.sh chmod +x fs-debian-installer.sh ./fs-debian-installer.sh Let me know what you think! -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/d0bb349a/attachment.html From lazyvirus at gmx.com Thu Mar 22 21:51:03 2012 From: lazyvirus at gmx.com (Bzzz) Date: Thu, 22 Mar 2012 19:51:03 +0100 Subject: [Freeswitch-users] embedded hardware Message-ID: <20120322195103.146a357e@anubis.defcon1> Hi list, I hope this isn't too much off-topic. I'm almost sure this kind of hardware: http://versalogic.com/products/DS.asp?ProductID=207 could withstand ~ 100 simultaneous calls and more, but what if 10% to 25% are transcoded together (lets say from iLBC to G711)? With what RAM Qty? (512MB or 1GB) I'd like to build a small but powerful fit-anywhere-PBX, but I don't know anything about this kind of embedded card. Jean-Yves -- We are Pentium of Borg. Division is futile. You will be approximated. (seen in someone's .signature) From Hector.Geraldino at ipsoft.com Thu Mar 22 22:01:36 2012 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Thu, 22 Mar 2012 15:01:36 -0400 Subject: [Freeswitch-users] Myevents In-Reply-To: References: Message-ID: <6A6B4C284AD15042B429EB9D904544AD022D77D001@NY1-EXMB-01.ip-soft.net> Look at the description on the wiki: http://wiki.freeswitch.org/wiki/Mod_event_socket#Special_Case_-_.27myevents.27 What I understand from this entry is that, when you do an inbound socket connection to FS, if you use myevents uuid you'll receive ALL events for the specified uuid and the socket connection will be closed when the channel (call) is closed (dropped). There's a difference between the type of events you want to receive (which you're already filtering using 'event plain ') and for which channels you want to listen for those events. So, even if you do an 'event plain CHANNEL_ANSWER', you'll receive only the CHANNEL_ANSWER event but for all the calls answered in FreeSWITCH. If you only want to receive calls for an specific call (not 'global events' as you called), you can add a filter in the form: filter Unique-ID uuid By filtering events by uuid, you can control not only what events you want to be notified for, but also what channels (calls) you want to receive events for. myevents won't help you to do that. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gregor Nanger Sent: Thursday, March 22, 2012 2:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Myevents I have "full" Just want to confirm if this is by design or I missed something. I want to subscribe only to "myevents" plus only to events I want in "events plain ". But both commands don't work like this. If I use "events plain " I also get global events and not only My events... Is this by design? Maybe I should ommit "full" to get only events of this call? 2012/3/22 Michael Collins > What is your socket command? If you do "full" then you'll get all events, otherwise you'll get the equivalent of "myevents." Check out the description here: http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Keywords -MC On Thu, Mar 22, 2012 at 1:02 AM, Gregor Nanger > wrote: One explanation... If I use in outbound connection event plain , I get only subscribed events. But if I also use myevents command I get all events. So, does myevents command override event plain command? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/d6b0062f/attachment.html From robert.longfield at klinsight.com Thu Mar 22 22:16:05 2012 From: robert.longfield at klinsight.com (Robert Longfield) Date: Thu, 22 Mar 2012 15:16:05 -0400 Subject: [Freeswitch-users] Internal VM Audio is below acceptable limit In-Reply-To: <20120322181140.2550738e@anubis.defcon1> References: <20120322172334.2f0ee16a@anubis.defcon1> <20120322181140.2550738e@anubis.defcon1> Message-ID: <069571F839944F3B8EE9954A439CC1DB@KITPC003> I have FS up and running with the awesome help of this community. I have a few minor bugs to workout. Some of my own doing (music on hold) but one that I can't trackdown or figure out the cause is when someone internally calls an extension and tries to leave a voice mail the system says something to the effect that the audio was below acceptable limit, basically it was too short. If the person at the extension picks up there is a slight pause and then you can talk with no issues and the audio sounds great. If the same extension is called from an outside line and the call goes to VM there are no problems leaving a message. Pastebin of the log http://pastebin.freeswitch.org/18723 -Rob From gabe at gundy.org Thu Mar 22 22:26:43 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Thu, 22 Mar 2012 13:26:43 -0600 Subject: [Freeswitch-users] FreeSWITCH Installer for Debian In-Reply-To: References: Message-ID: On Thu, Mar 22, 2012 at 12:37 PM, Brian Foster wrote: > When Ken's Debian packages come out for ?FreeSWITCH, I plan on adding the > ability to install from those packages as well as just installing from git > head and (eventually) the 1.2 branch. Perhaps this can save someone, somewhere some effort: https://parseltone.org/browser/trunk/misc/build_freeswitch.sh Best, Gabe From peter.olsson at visionutveckling.se Thu Mar 22 22:26:56 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 22 Mar 2012 19:26:56 +0000 Subject: [Freeswitch-users] Internal VM Audio is below acceptable limit In-Reply-To: <069571F839944F3B8EE9954A439CC1DB@KITPC003> References: <20120322172334.2f0ee16a@anubis.defcon1> <20120322181140.2550738e@anubis.defcon1>, <069571F839944F3B8EE9954A439CC1DB@KITPC003> Message-ID: <1FFF97C269757C458224B7C895F35F1507C9B8@cantor.std.visionutv.se> Are you on real hardware, or on a virtual machine? /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Robert Longfield [robert.longfield at klinsight.com] Skickat: den 22 mars 2012 20:16 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Internal VM Audio is below acceptable limit I have FS up and running with the awesome help of this community. I have a few minor bugs to workout. Some of my own doing (music on hold) but one that I can't trackdown or figure out the cause is when someone internally calls an extension and tries to leave a voice mail the system says something to the effect that the audio was below acceptable limit, basically it was too short. If the person at the extension picks up there is a slight pause and then you can talk with no issues and the audio sounds great. If the same extension is called from an outside line and the call goes to VM there are no problems leaving a message. Pastebin of the log http://pastebin.freeswitch.org/18723 -Rob _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f6b791732761084654857! From robert.longfield at klinsight.com Thu Mar 22 22:40:30 2012 From: robert.longfield at klinsight.com (Robert Longfield) Date: Thu, 22 Mar 2012 15:40:30 -0400 Subject: [Freeswitch-users] Internal VM Audio is below acceptable limit In-Reply-To: <1FFF97C269757C458224B7C895F35F1507C9B8@cantor.std.visionutv.se> References: <20120322172334.2f0ee16a@anubis.defcon1><20120322181140.2550738e@anubis.defcon1>, <069571F839944F3B8EE9954A439CC1DB@KITPC003> <1FFF97C269757C458224B7C895F35F1507C9B8@cantor.std.visionutv.se> Message-ID: Hey Peter, Thank for the question. We have a mixture of softphones on computers and mobiles and handsets connected on a ATA box. It seems that the handsets can leave VM but the softphones cannot. I would never have thought of grabbing a handset to test this. For the office this won't be a problem for too long as we are moving to phones but we do have the option of working from home and those cases require softphones to work. -Rob -----Original Message----- From: Peter Olsson Sent: Thursday, March 22, 2012 3:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Internal VM Audio is below acceptable limit Are you on real hardware, or on a virtual machine? /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Robert Longfield [robert.longfield at klinsight.com] Skickat: den 22 mars 2012 20:16 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Internal VM Audio is below acceptable limit I have FS up and running with the awesome help of this community. I have a few minor bugs to workout. Some of my own doing (music on hold) but one that I can't trackdown or figure out the cause is when someone internally calls an extension and tries to leave a voice mail the system says something to the effect that the audio was below acceptable limit, basically it was too short. If the person at the extension picks up there is a slight pause and then you can talk with no issues and the audio sounds great. If the same extension is called from an outside line and the call goes to VM there are no problems leaving a message. Pastebin of the log http://pastebin.freeswitch.org/18723 -Rob _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f6b791732761084654857! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From bdfoster at endigotech.com Thu Mar 22 23:04:48 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 22 Mar 2012 16:04:48 -0400 Subject: [Freeswitch-users] Internal VM Audio is below acceptable limit In-Reply-To: References: <20120322172334.2f0ee16a@anubis.defcon1> <20120322181140.2550738e@anubis.defcon1> <069571F839944F3B8EE9954A439CC1DB@KITPC003> <1FFF97C269757C458224B7C895F35F1507C9B8@cantor.std.visionutv.se> Message-ID: Actually, I believe what he was asking about was the actual server, although the info you provided already does give us some insight into your setup. Is the freeswitch instance running on virtual or real hardware? -BDF On Mar 22, 2012 3:41 PM, "Robert Longfield" wrote: > Hey Peter, > > Thank for the question. We have a mixture of softphones on computers and > mobiles and handsets connected on a ATA box. It seems that the handsets can > leave VM but the softphones cannot. I would never have thought of grabbing > a > handset to test this. > For the office this won't be a problem for too long as we are moving to > phones but we do have the option of working from home and those cases > require softphones to work. > > -Rob > > > > -----Original Message----- > From: Peter Olsson > Sent: Thursday, March 22, 2012 3:26 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Internal VM Audio is below acceptable limit > > Are you on real hardware, or on a virtual machine? > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org > [freeswitch-users-bounces at lists.freeswitch.org] för Robert Longfield > [robert.longfield at klinsight.com] > Skickat: den 22 mars 2012 20:16 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] Internal VM Audio is below acceptable limit > > I have FS up and running with the awesome help of this community. I have a > few minor bugs to workout. Some of my own doing (music on hold) but one > that > I can't trackdown or figure out the cause is when someone internally calls > an extension and tries to leave a voice mail the system says something to > the effect that the audio was below acceptable limit, basically it was too > short. > If the person at the extension picks up there is a slight pause and then > you > can talk with no issues and the audio sounds great. > > If the same extension is called from an outside line and the call goes to > VM > there are no problems leaving a message. > > Pastebin of the log http://pastebin.freeswitch.org/18723 > > -Rob > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f6b791732761084654857! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/ad3a6c5c/attachment.html From gregor at infomedia.si Thu Mar 22 23:32:37 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 22 Mar 2012 21:32:37 +0100 Subject: [Freeswitch-users] Myevents In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD022D77D001@NY1-EXMB-01.ip-soft.net> References: <6A6B4C284AD15042B429EB9D904544AD022D77D001@NY1-EXMB-01.ip-soft.net> Message-ID: Thank you Hector! I think I got it. For what I want, I should use event plain + filter to uuid. What has better performance, filter or myevents? 2012/3/22 Hector Geraldino > Look at the description on the wiki:**** > > ** ** > > > http://wiki.freeswitch.org/wiki/Mod_event_socket#Special_Case_-_.27myevents.27 > **** > > ** ** > > What I understand from this entry is that, when you do an inbound socket > connection to FS, if you use myevents uuid you?ll receive ALL events for > the specified uuid and the socket connection will be closed when the > channel (call) is closed (dropped).**** > > ** ** > > There?s a difference between the type of events you want to receive (which > you?re already filtering using ?event plain ?) and for which > channels you want to listen for those events. So, even if you do an ?event > plain CHANNEL_ANSWER?, you?ll receive only the CHANNEL_ANSWER event but for > all the calls answered in FreeSWITCH. If you only want to receive calls for > an specific call (not ?global events? as you called), you can add a filter > in the form:**** > > ** ** > > filter Unique-ID uuid**** > > ** ** > > By filtering events by uuid, you can control not only what events you want > to be notified for, but also what channels (calls) you want to receive > events for. myevents won?t help you to do that.**** > > ** ** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor > Nanger > *Sent:* Thursday, March 22, 2012 2:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Myevents**** > > ** ** > > I have "full"**** > > ** ** > > Just want to confirm if this is by design or I missed something.**** > > ** ** > > I want to subscribe only to "myevents" plus only to events I want in > "events plain ". But both commands don't work like this. If I use > "events plain " I also get global events and not only My events... > **** > > ** ** > > Is this by design? Maybe I should ommit "full" to get only events of this > call?**** > > ** ** > > **** > > ** ** > > 2012/3/22 Michael Collins **** > > What is your socket command? If you do "full" then you'll get all events, > otherwise you'll get the equivalent of "myevents." > > Check out the description here: > http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Keywords > > -MC**** > > ** ** > > On Thu, Mar 22, 2012 at 1:02 AM, Gregor Nanger > wrote:**** > > One explanation...**** > > ** ** > > If I use in outbound connection event plain , I get only > subscribed events. But if I also use myevents command I get all events. ** > ** > > ** ** > > So, does myevents command override event plain command?**** > > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/2a5ab02f/attachment-0001.html From robert.longfield at klinsight.com Fri Mar 23 00:05:46 2012 From: robert.longfield at klinsight.com (Robert Longfield) Date: Thu, 22 Mar 2012 17:05:46 -0400 Subject: [Freeswitch-users] Internal VM Audio is below acceptable limit In-Reply-To: References: <20120322172334.2f0ee16a@anubis.defcon1><20120322181140.2550738e@anubis.defcon1><069571F839944F3B8EE9954A439CC1DB@KITPC003><1FFF97C269757C458224B7C895F35F1507C9B8@cantor.std.visionutv.se> Message-ID: Ah, FS is running on a VPS. From: Brian Foster Sent: Thursday, March 22, 2012 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Internal VM Audio is below acceptable limit Actually, I believe what he was asking about was the actual server, although the info you provided already does give us some insight into your setup. Is the freeswitch instance running on virtual or real hardware? -BDF On Mar 22, 2012 3:41 PM, "Robert Longfield" wrote: Hey Peter, Thank for the question. We have a mixture of softphones on computers and mobiles and handsets connected on a ATA box. It seems that the handsets can leave VM but the softphones cannot. I would never have thought of grabbing a handset to test this. For the office this won't be a problem for too long as we are moving to phones but we do have the option of working from home and those cases require softphones to work. -Rob -----Original Message----- From: Peter Olsson Sent: Thursday, March 22, 2012 3:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Internal VM Audio is below acceptable limit Are you on real hardware, or on a virtual machine? /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Robert Longfield [robert.longfield at klinsight.com] Skickat: den 22 mars 2012 20:16 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Internal VM Audio is below acceptable limit I have FS up and running with the awesome help of this community. I have a few minor bugs to workout. Some of my own doing (music on hold) but one that I can't trackdown or figure out the cause is when someone internally calls an extension and tries to leave a voice mail the system says something to the effect that the audio was below acceptable limit, basically it was too short. If the person at the extension picks up there is a slight pause and then you can talk with no issues and the audio sounds great. If the same extension is called from an outside line and the call goes to VM there are no problems leaving a message. Pastebin of the log http://pastebin.freeswitch.org/18723 -Rob _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f6b791732761084654857! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/074154b6/attachment.html From Hector.Geraldino at ipsoft.com Fri Mar 23 00:12:57 2012 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Thu, 22 Mar 2012 17:12:57 -0400 Subject: [Freeswitch-users] Myevents In-Reply-To: References: <6A6B4C284AD15042B429EB9D904544AD022D77D001@NY1-EXMB-01.ip-soft.net> Message-ID: <6A6B4C284AD15042B429EB9D904544AD022D77D020@NY1-EXMB-01.ip-soft.net> I think myevents is better suited for inbound socket connections, so just do a events plain + filter Unique-ID uuid when you want to filter events for your outbound channel. You shouldn't notice any differences in regards to performance, just be careful to register to events that you really want to be notified for. Enabling multiple event notifications in multiple channels will surely affect FreeSWITCH's performance. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gregor Nanger Sent: Thursday, March 22, 2012 4:33 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Myevents Thank you Hector! I think I got it. For what I want, I should use event plain + filter to uuid. What has better performance, filter or myevents? 2012/3/22 Hector Geraldino > Look at the description on the wiki: http://wiki.freeswitch.org/wiki/Mod_event_socket#Special_Case_-_.27myevents.27 What I understand from this entry is that, when you do an inbound socket connection to FS, if you use myevents uuid you'll receive ALL events for the specified uuid and the socket connection will be closed when the channel (call) is closed (dropped). There's a difference between the type of events you want to receive (which you're already filtering using 'event plain ') and for which channels you want to listen for those events. So, even if you do an 'event plain CHANNEL_ANSWER', you'll receive only the CHANNEL_ANSWER event but for all the calls answered in FreeSWITCH. If you only want to receive calls for an specific call (not 'global events' as you called), you can add a filter in the form: filter Unique-ID uuid By filtering events by uuid, you can control not only what events you want to be notified for, but also what channels (calls) you want to receive events for. myevents won't help you to do that. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gregor Nanger Sent: Thursday, March 22, 2012 2:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Myevents I have "full" Just want to confirm if this is by design or I missed something. I want to subscribe only to "myevents" plus only to events I want in "events plain ". But both commands don't work like this. If I use "events plain " I also get global events and not only My events... Is this by design? Maybe I should ommit "full" to get only events of this call? Error! Filename not specified. 2012/3/22 Michael Collins > What is your socket command? If you do "full" then you'll get all events, otherwise you'll get the equivalent of "myevents." Check out the description here: http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Keywords -MC On Thu, Mar 22, 2012 at 1:02 AM, Gregor Nanger > wrote: One explanation... If I use in outbound connection event plain , I get only subscribed events. But if I also use myevents command I get all events. So, does myevents command override event plain command? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/aaaf8f62/attachment-0001.html From garbytrash at gmail.com Fri Mar 23 00:44:11 2012 From: garbytrash at gmail.com (Zenny) Date: Thu, 22 Mar 2012 21:44:11 +0000 Subject: [Freeswitch-users] FreeSWITCH Installer for Debian In-Reply-To: References: Message-ID: Thanks Brian and Gabriel for the nice scripts. However Gabriel's seems more Debian-styled build. I prefer to go with that of Garbriel's ;-) Glad to find build_opensips.sh also in the same directory. Take care! /z On 3/22/12, Gabriel Gunderson wrote: > On Thu, Mar 22, 2012 at 12:37 PM, Brian Foster > wrote: >> When Ken's Debian packages come out for ?FreeSWITCH, I plan on adding the >> ability to install from those packages as well as just installing from git >> head and (eventually) the 1.2 branch. > > Perhaps this can save someone, somewhere some effort: > > https://parseltone.org/browser/trunk/misc/build_freeswitch.sh > > > Best, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From bdfoster at endigotech.com Fri Mar 23 00:56:53 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 22 Mar 2012 17:56:53 -0400 Subject: [Freeswitch-users] FreeSWITCH Installer for Debian In-Reply-To: References: Message-ID: As of now, building from source is still the "official" way to build/install FreeSWITCH. It will more than likely stay that way until the debian FreeSWITCH repo is stable, and that probably won't happen until 1.2 is official. When that time comes, you will be able to select how you want to install. This will be settable via a variable inside the script or (when the guided install gets implemented) you will be able to select how you want to install. There will always be a crowd that wants to stay on the bleeding edge, and it's my intention allow those users to do just that. -BDF On Mar 22, 2012 5:45 PM, "Zenny" wrote: > Thanks Brian and Gabriel for the nice scripts. > > However Gabriel's seems more Debian-styled build. I prefer to go with > that of Garbriel's ;-) > > Glad to find build_opensips.sh also in the same directory. > > Take care! > > /z > > On 3/22/12, Gabriel Gunderson wrote: > > On Thu, Mar 22, 2012 at 12:37 PM, Brian Foster > > wrote: > >> When Ken's Debian packages come out for FreeSWITCH, I plan on adding > the > >> ability to install from those packages as well as just installing from > git > >> head and (eventually) the 1.2 branch. > > > > Perhaps this can save someone, somewhere some effort: > > > > https://parseltone.org/browser/trunk/misc/build_freeswitch.sh > > > > > > Best, > > Gabe > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120322/18203866/attachment.html From gabe at gundy.org Fri Mar 23 01:17:18 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Thu, 22 Mar 2012 16:17:18 -0600 Subject: [Freeswitch-users] FreeSWITCH Installer for Debian In-Reply-To: References: Message-ID: On Thu, Mar 22, 2012 at 3:56 PM, Brian Foster wrote: > There will always be a crowd that wants to stay on the bleeding edge, and > it's my intention allow those users to do just that. I'm there with you Brian... I've been meaning to setup nightly builds. If I do, I'll let you know. Best, Gabe From gabe at gundy.org Fri Mar 23 01:21:10 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Thu, 22 Mar 2012 16:21:10 -0600 Subject: [Freeswitch-users] FreeSWITCH Installer for Debian In-Reply-To: References: Message-ID: On Thu, Mar 22, 2012 at 3:44 PM, Zenny wrote: > Glad to find build_opensips.sh also in the same directory. If you want to save the trouble of building, there are also some pre-built debs for both FreeSWITCH and OpenSIPS: https://parseltone.org/files/debs/ubuntu-11.4/ Best, Gabe From lazyvirus at gmx.com Fri Mar 23 02:44:23 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 23 Mar 2012 00:44:23 +0100 Subject: [Freeswitch-users] VM sound very choppy In-Reply-To: References: <20120322172334.2f0ee16a@anubis.defcon1> <20120322181941.27dcb43e@anubis.defcon1> Message-ID: <20120323004423.570bbeab@anubis.defcon1> On Thu, 22 Mar 2012 10:21:58 -0700 Michael Collins wrote: That the same thing when I put a call on hold: http://pastebin.freeswitch.org/18724 -- For some reason, this fortune reminds everyone of Marvin Zelkowitz. From lazyvirus at gmx.com Fri Mar 23 04:44:34 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 23 Mar 2012 02:44:34 +0100 Subject: [Freeswitch-users] VM sound very choppy In-Reply-To: <20120323004423.570bbeab@anubis.defcon1> References: <20120322172334.2f0ee16a@anubis.defcon1> <20120322181941.27dcb43e@anubis.defcon1> <20120323004423.570bbeab@anubis.defcon1> Message-ID: <20120323024434.7400cd13@anubis.defcon1> On Fri, 23 Mar 2012 00:44:23 +0100 Bzzz wrote: Hmm, weird: I deactivated "Proxy Media" into fusionpbx (so, I lost ZRTP as I don't have the SDK at this time), and now I've got a reinvite codec error: when I put a line on hold, moh's there but won't ever cut when taking the line back. http://pastebin.freeswitch.org/18725 JY -- From tech at tech-invent.ru Fri Mar 23 08:04:19 2012 From: tech at tech-invent.ru (Dmitry Golubenko) Date: Fri, 23 Mar 2012 12:04:19 +0700 Subject: [Freeswitch-users] mod h323 error In-Reply-To: <1FFF97C269757C458224B7C895F35F1507C35F@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1507C35F@cantor.std.visionutv.se> Message-ID: <4F6C0453.9060201@tech-invent.ru> 22.03.2012 20:15, Peter Olsson ?????: > It seems you might need to modify the Makefile, because of the errors > reported. > > Please read through this information first: > http://wiki.freeswitch.org/wiki/Mod_h323 > > I personally recommend not to use h323plus-trunk. I use h323-trunk because gnugk recommends use it http://www.gnugk.org/gnugk-manual-14.html#ss14.1 , and i need interoberability between gnugk and freeswitch so i prefer build with same version of h323plus. As of ptlib version, i used to 2.10.[1234] from svn http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/tags/v2_10_[1234] until recently (about 2 weeks ago) some change in v2_10_4 break build of gnugk'a h460presence.cxx so i switched to 2.10 branch http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/branches/v2_10 and build gnugk&fs succesfully To prevent hardcoding paths to mod_h323 Makefile maybe we need someone to change -I and -L flags in git to make use environment variables PTLIBDIR and OPENH323DIR ? > > /Peter > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *F?r *budi wibowo > *Skickat:* den 22 mars 2012 13:44 > *Till:* FreeSWITCH Users Help > *?mne:* [Freeswitch-users] mod h323 error > > hi > > i just tried to build mod h323 with ptlib 2.8.2 and h323plus-trunk, > FreeSWITCH Version 1.0.head (git-9d3401e 2012-03-19 20-06-36 -0500) > > when i make mod_323 i got this error > > usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: > error: 'OPAL_G7231_5k3' was not declared in this scope > > /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h: At > global scope: > > /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: > error: 'OPAL_G7231_5k3' has not been declared > > /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: > error: expected ',' or '...' before string constant > > /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:624: > error: expected constructor, destructor, or type conversion before 'class' > > make[4]: *** [mod_h323.lo] Error 1 > > make[3]: *** [all] Error 1 > > make[2]: *** [mod_h323-all] Error 1 > > make[1]: *** [mod_h323] Error 2 > > make: *** [mod_h323] Error 2 > > detail error log http://pastebin.freeswitch.org/18717 > > help to solve this issue is welcome and appreciated > > regards > > budi wibowo > > !DSPAM:4f6b1d7932768918913259! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From peter.olsson at visionutveckling.se Fri Mar 23 10:22:47 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 23 Mar 2012 07:22:47 +0000 Subject: [Freeswitch-users] Internal VM Audio is below acceptable limit Message-ID: <1FFF97C269757C458224B7C895F35F1507CB32@cantor.std.visionutv.se> Ok - that's the problem then :) If you want to make sure everything "just works" - install on real hardware. If you use the built in tools timer_test and time_test I'm pretty sure it will show you quite bad timing. It's not impossible to succeed with a virtual setup, but it will require lots of testing, and usually you need to be in control of the physical host, so it's possible to tweak some performance values for the virtual machine. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Robert Longfield Skickat: den 22 mars 2012 22:06 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Internal VM Audio is below acceptable limit Ah, FS is running on a VPS. From: Brian Foster Sent: Thursday, March 22, 2012 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Internal VM Audio is below acceptable limit Actually, I believe what he was asking about was the actual server, although the info you provided already does give us some insight into your setup. Is the freeswitch instance running on virtual or real hardware? -BDF On Mar 22, 2012 3:41 PM, "Robert Longfield" > wrote: Hey Peter, Thank for the question. We have a mixture of softphones on computers and mobiles and handsets connected on a ATA box. It seems that the handsets can leave VM but the softphones cannot. I would never have thought of grabbing a handset to test this. For the office this won't be a problem for too long as we are moving to phones but we do have the option of working from home and those cases require softphones to work. -Rob -----Original Message----- From: Peter Olsson Sent: Thursday, March 22, 2012 3:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Internal VM Audio is below acceptable limit Are you on real hardware, or on a virtual machine? /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Robert Longfield [robert.longfield at klinsight.com] Skickat: den 22 mars 2012 20:16 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Internal VM Audio is below acceptable limit I have FS up and running with the awesome help of this community. I have a few minor bugs to workout. Some of my own doing (music on hold) but one that I can't trackdown or figure out the cause is when someone internally calls an extension and tries to leave a voice mail the system says something to the effect that the audio was below acceptable limit, basically it was too short. If the person at the extension picks up there is a slight pause and then you can talk with no issues and the audio sounds great. If the same extension is called from an outside line and the call goes to VM there are no problems leaving a message. Pastebin of the log http://pastebin.freeswitch.org/18723 -Rob _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f6b937d32761551985203! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/d75a385d/attachment.html From peter.olsson at visionutveckling.se Fri Mar 23 10:24:30 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 23 Mar 2012 07:24:30 +0000 Subject: [Freeswitch-users] VM sound very choppy Message-ID: <1FFF97C269757C458224B7C895F35F1507CB41@cantor.std.visionutv.se> Another case of virtual machine setup maybe, or is this real hardware? /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Bzzz Skickat: den 23 mars 2012 00:44 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] VM sound very choppy On Thu, 22 Mar 2012 10:21:58 -0700 Michael Collins wrote: That the same thing when I put a call on hold: http://pastebin.freeswitch.org/18724 -- For some reason, this fortune reminds everyone of Marvin Zelkowitz. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f6bb86f32761437815508! From mytemike72 at gmail.com Fri Mar 23 13:47:04 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Fri, 23 Mar 2012 11:47:04 +0100 Subject: [Freeswitch-users] Originate using inbound socket connection In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD022D77CFC9@NY1-EXMB-01.ip-soft.net> References: <1FFF97C269757C458224B7C895F35F1507B744@cantor.std.visionutv.se> <6A6B4C284AD15042B429EB9D904544AD022D77CF5C@NY1-EXMB-01.ip-soft.net> <6A6B4C284AD15042B429EB9D904544AD022D77CFC9@NY1-EXMB-01.ip-soft.net> Message-ID: Thanks used the &park() and it worked!, except.... It seemed the bridge did not take place while my aleg is inside the lua script. I had to finish the script and the the calls where bridged (even the bridge command was allready send), the weird thing is that while I am still inside the lua script after the uuid_bridge, mys session:ready() immedeately becomes [false] while I am still inside the script. The bridge is now initiated on the non-lua side (ESL Connection). Would be an idea to move that to the Luascript using a session:bridge() however I have no session object of the bleg ... Only way to go there is calling the api to do a uuid_bridge as I know the uuid of the bleg, but by using the uuid_bridge in Lua instead of the session:bridge() would problably lead to the same.... Mike. 2012/3/22 Hector Geraldino : > So the problem seems to be with the parameters you're using for the uuid_bridge command, not your implementation :) > > Can you do a '>show channels' from the fs cli and see what are the uuid's of the active channels in FS? Then use those uuids to perform the uuid_bridge and see what happens. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Lutz > Sent: Wednesday, March 21, 2012 6:05 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Originate using inbound socket connection > > Nope, it's the same, it only says OK, but nothing else... > > > 2012/3/21 Hector Geraldino : >> What happens if you try to do the same operation from the fs_cli ? >> >> uuid_bridge legA_uuid legB_uuid >> >> Works on the console and not when you call it from your app? >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Lutz >> Sent: Wednesday, March 21, 2012 5:20 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Originate using inbound socket connection >> >> Thanks, that worked. >> >> Now my next problem occurs... ;-) >> >> I am trying to bridge these calls to another allready established >> session in the switch (in a Lua script). >> >> I can reach all data of the newly created session using my created >> uuid which I specified in the originate. >> >> However, the uuid_bridge does not seem to work. (it says +OK and the >> the uuid of the bleg) but everything stays silent and no message is >> being wrtitten to console. >> >> When I check console, I do notice that the new session seems to use a >> different uuid than I specified, (which seems strange as i can access >> al sesssion data uusing uuid_ commands...) >> >> string cDialString = "{origination_uuid=" + legb_uuid + >> ",origination_caller_id_number=" + thisAni + >> ",origination_caller_id_name=" + thisAniName + "}sofia/external/" + >> thisDestination + " &park()"; >> >> var eslEvent = eslDial.Api("originate", cDialString); >> dispo = eslDial.Api("uuid_getvar", legb_uuid + " >> endpoint_disposition").GetBody(); >> while (eslDial.Api("uuid_exists", thisFrom).GetBody() == "true" && >> eslDial.Api("uuid_exists", legb_uuid).GetBody() == "true" && dispo != >> "ANSWER") >> { >> ? .. >> ? blabla >> } >> >> when I am connected I just do a "uuid_bridge legb_uuid" >> >> I get a response: >> +OK >> >> My console says: >> New Channel sofia/external/316....... at xxx.xxx.xxx.xxx[7cb13e5d-1dfe-483e-a8da-ec3a592700c7] >> >> and is a completely different uuid than my legb uuid i specified and >> use to get to the data...... >> >> Any ideas? >> >> Thanks, >> Mike >> >> 2012/3/21 Peter Olsson : >>> You need to put the other end of the call somewhere >>> >>> The correct string is (example) "originate sofia/gateway/test/1002 &park()" >>> >>> This will call 1002 and then park the call. >>> >>> /Peter >>> >>> >>> -----Ursprungligt meddelande----- >>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz >>> Skickat: den 21 mars 2012 14:49 >>> Till: FreeSWITCH Users Help >>> ?mne: [Freeswitch-users] Originate using inbound socket connection >>> >>> Hi Guys, >>> >>> I am working on a .net ?ESL host (inbound mode) and connect to the ESL using ESLConnection. >>> Now I am trying to originate a call using the ESLConnection.Api() function. But it does't work as expected. >>> >>> It always returns: >>> === >>> Content-Type: api/response >>> Content-Length: 125 >>> Content-Length: 125 >>> >>> -USAGE: |&() >>> [] [] [] [] [] === >>> >>> But I am pretty sure my string is right: >>> >>> {origination_caller_id_number=31341.....,origination_caller_id_name=31341......}sofia/external/31634258... at xxx.xxx.xxx.xxx >>> >>> I also tried with double {{ and }} but same result. Even if I take out the whole {} string and just use "sofia/external/etc.." I get the same message back. >>> >>> My code: >>> >>> ESLconnection eslDial = new ESLconnection("x.x.x.x", "8021", "x"); if (eslDial.Connected() == ESL_SUCCESS) { >>> ? // Create a uuid used to identify the b-leg. >>> ? string legb_uuid = eslDial.Api("create_uuid", "").GetBody(); >>> ? string cDialString = "{{origination_uuid=" + legb_uuid + ",origination_caller_id_number=" + thisAni + ",origination_caller_id_name=" + thisAniName + "}}sofia/external/" + thisDestination; >>> >>> ? // Send the command >>> ? var eslEvent = eslDial.Api("originate", cDialString); >>> >>> ? // Write the result to the console >>> ? Console.WriteLine(eslEvent.Serialize(string.Empty)); >>> ? return true; >>> } >>> >>> Thanks for your help!, >>> >>> Regards, >>> Michael Lutz >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> !DSPAM:4f69dabc32766734014540! >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From peter.olsson at visionutveckling.se Fri Mar 23 14:07:12 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 23 Mar 2012 11:07:12 +0000 Subject: [Freeswitch-users] Originate using inbound socket connection Message-ID: <1FFF97C269757C458224B7C895F35F1507CD78@cantor.std.visionutv.se> If session:ready() becomes false, can't you just exit the script (as you should if it becomes false), and the bridge will be completed? /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz Skickat: den 23 mars 2012 11:47 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Originate using inbound socket connection Thanks used the &park() and it worked!, except.... It seemed the bridge did not take place while my aleg is inside the lua script. I had to finish the script and the the calls where bridged (even the bridge command was allready send), the weird thing is that while I am still inside the lua script after the uuid_bridge, mys session:ready() immedeately becomes [false] while I am still inside the script. The bridge is now initiated on the non-lua side (ESL Connection). Would be an idea to move that to the Luascript using a session:bridge() however I have no session object of the bleg ... Only way to go there is calling the api to do a uuid_bridge as I know the uuid of the bleg, but by using the uuid_bridge in Lua instead of the session:bridge() would problably lead to the same.... Mike. 2012/3/22 Hector Geraldino : > So the problem seems to be with the parameters you're using for the > uuid_bridge command, not your implementation :) > > Can you do a '>show channels' from the fs cli and see what are the uuid's of the active channels in FS? Then use those uuids to perform the uuid_bridge and see what happens. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Lutz > Sent: Wednesday, March 21, 2012 6:05 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Originate using inbound socket > connection > > Nope, it's the same, it only says OK, but nothing else... > > > 2012/3/21 Hector Geraldino : >> What happens if you try to do the same operation from the fs_cli ? >> >> uuid_bridge legA_uuid legB_uuid >> >> Works on the console and not when you call it from your app? >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Michael Lutz >> Sent: Wednesday, March 21, 2012 5:20 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Originate using inbound socket >> connection >> >> Thanks, that worked. >> >> Now my next problem occurs... ;-) >> >> I am trying to bridge these calls to another allready established >> session in the switch (in a Lua script). >> >> I can reach all data of the newly created session using my created >> uuid which I specified in the originate. >> >> However, the uuid_bridge does not seem to work. (it says +OK and the >> the uuid of the bleg) but everything stays silent and no message is >> being wrtitten to console. >> >> When I check console, I do notice that the new session seems to use a >> different uuid than I specified, (which seems strange as i can access >> al sesssion data uusing uuid_ commands...) >> >> string cDialString = "{origination_uuid=" + legb_uuid + >> ",origination_caller_id_number=" + thisAni + >> ",origination_caller_id_name=" + thisAniName + "}sofia/external/" + >> thisDestination + " &park()"; >> >> var eslEvent = eslDial.Api("originate", cDialString); dispo = >> eslDial.Api("uuid_getvar", legb_uuid + " >> endpoint_disposition").GetBody(); >> while (eslDial.Api("uuid_exists", thisFrom).GetBody() == "true" && >> eslDial.Api("uuid_exists", legb_uuid).GetBody() == "true" && dispo != >> "ANSWER") >> { >> ? .. >> ? blabla >> } >> >> when I am connected I just do a "uuid_bridge legb_uuid" >> >> I get a response: >> +OK >> >> My console says: >> New Channel >> sofia/external/316....... at xxx.xxx.xxx.xxx[7cb13e5d-1dfe-483e-a8da-ec3 >> a592700c7] >> >> and is a completely different uuid than my legb uuid i specified and >> use to get to the data...... >> >> Any ideas? >> >> Thanks, >> Mike >> >> 2012/3/21 Peter Olsson : >>> You need to put the other end of the call somewhere >>> >>> The correct string is (example) "originate sofia/gateway/test/1002 &park()" >>> >>> This will call 1002 and then park the call. >>> >>> /Peter >>> >>> >>> -----Ursprungligt meddelande----- >>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael >>> Lutz >>> Skickat: den 21 mars 2012 14:49 >>> Till: FreeSWITCH Users Help >>> ?mne: [Freeswitch-users] Originate using inbound socket connection >>> >>> Hi Guys, >>> >>> I am working on a .net ?ESL host (inbound mode) and connect to the ESL using ESLConnection. >>> Now I am trying to originate a call using the ESLConnection.Api() function. But it does't work as expected. >>> >>> It always returns: >>> === >>> Content-Type: api/response >>> Content-Length: 125 >>> Content-Length: 125 >>> >>> -USAGE: |&() >>> [] [] [] [] [] >>> === >>> >>> But I am pretty sure my string is right: >>> >>> {origination_caller_id_number=31341.....,origination_caller_id_name= >>> 31341......}sofia/external/31634258... at xxx.xxx.xxx.xxx >>> >>> I also tried with double {{ and }} but same result. Even if I take out the whole {} string and just use "sofia/external/etc.." I get the same message back. >>> >>> My code: >>> >>> ESLconnection eslDial = new ESLconnection("x.x.x.x", "8021", "x"); >>> if (eslDial.Connected() == ESL_SUCCESS) { >>> ? // Create a uuid used to identify the b-leg. >>> ? string legb_uuid = eslDial.Api("create_uuid", "").GetBody(); >>> ? string cDialString = "{{origination_uuid=" + legb_uuid + >>> ",origination_caller_id_number=" + thisAni + >>> ",origination_caller_id_name=" + thisAniName + "}}sofia/external/" + >>> thisDestination; >>> >>> ? // Send the command >>> ? var eslEvent = eslDial.Api("originate", cDialString); >>> >>> ? // Write the result to the console >>> ? Console.WriteLine(eslEvent.Serialize(string.Empty)); >>> ? return true; >>> } >>> >>> Thanks for your help!, >>> >>> Regards, >>> Michael Lutz >>> >>> ____________________________________________________________________ >>> _____ Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >>> sers >>> http://www.freeswitch.org >>> >>> >>> >>> >>> ____________________________________________________________________ >>> _____ Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >>> sers >>> http://www.freeswitch.org >> >> _____________________________________________________________________ >> ____ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> http://www.freeswitch.org >> >> _____________________________________________________________________ >> ____ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> http://www.freeswitch.org > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f6c547e32761029518556! From Daniel.Knaggs at realitysolutions.co.uk Fri Mar 23 14:19:09 2012 From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs) Date: Fri, 23 Mar 2012 11:19:09 +0000 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls Message-ID: Hello all, Got a bit of a strange one, we appear to be getting DTMF tones on incoming calls when the caller hasn?t even pressed any keys. It normally happens with 10 seconds or so after the call has been answered. Here is the log of it happening earlier: - 2012-03-23 10:38:16.852654 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF A (debug = 0) 2012-03-23 10:38:16.852654 [DEBUG] mod_freetdm.c:799 Queuing DTMF [A] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:16.915653 [DEBUG] switch_rtp.c:2420 Send start packet for [A] ts=49440 dur=160/160/2000 seq=46803 lw=49440 2012-03-23 10:38:16.936653 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=320/320/2000 seq=46804 lw=49600 2012-03-23 10:38:16.957652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=480/480/2000 seq=46805 lw=49760 2012-03-23 10:38:16.978652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=640/640/2000 seq=46806 lw=49920 2012-03-23 10:38:16.999652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=800/800/2000 seq=46807 lw=50080 2012-03-23 10:38:17.020651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=960/960/2000 seq=46808 lw=50240 2012-03-23 10:38:17.041651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1120/1120/2000 seq=46809 lw=50400 2012-03-23 10:38:17.062651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1280/1280/2000 seq=46810 lw=50560 2012-03-23 10:38:17.083650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1440/1440/2000 seq=46811 lw=50720 2012-03-23 10:38:17.104650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1600/1600/2000 seq=46812 lw=50880 2012-03-23 10:38:17.125650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1760/1760/2000 seq=46813 lw=51040 2012-03-23 10:38:17.146650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1920/1920/2000 seq=46814 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46815 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46816 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46817 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2271 Queue digit delay of 40ms 2012-03-23 10:38:18.070638 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF B (debug = 0) 2012-03-23 10:38:18.070638 [DEBUG] mod_freetdm.c:799 Queuing DTMF [B] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:18.133637 [DEBUG] switch_rtp.c:2420 Send start packet for [B] ts=59040 dur=160/160/2000 seq=46864 lw=59040 2012-03-23 10:38:18.154637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=320/320/2000 seq=46865 lw=59200 2012-03-23 10:38:18.175637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=480/480/2000 seq=46866 lw=59360 2012-03-23 10:38:18.196636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=640/640/2000 seq=46867 lw=59520 2012-03-23 10:38:18.217636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=800/800/2000 seq=46868 lw=59680 2012-03-23 10:38:18.238636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=960/960/2000 seq=46869 lw=59840 2012-03-23 10:38:18.259636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1120/1120/2000 seq=46870 lw=60000 2012-03-23 10:38:18.280636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1280/1280/2000 seq=46871 lw=60160 2012-03-23 10:38:18.301635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1440/1440/2000 seq=46872 lw=60320 2012-03-23 10:38:18.322635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1600/1600/2000 seq=46873 lw=60480 2012-03-23 10:38:18.343634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1760/1760/2000 seq=46874 lw=60640 2012-03-23 10:38:18.364634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1920/1920/2000 seq=46875 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46876 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46877 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46878 lw=60800 I?m very sure the caller isn?t dialling ?A? or ?B?! Running on ISDN via a TE121 card using E1 (euroisdn). Freeswitch version is ?FreeSWITCH Version 1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)?. Currently there is a ?bind_meta_app? in the config which binds a script on the B leg of the call which parks the call ? I haven?t tried turning this off yet to see if it?s this. Wondering if anyone has any ideas or has come across this before? Thanks in advance. [cid:imageacd695.PNG at 9e92e461.40aa4b96] Daniel Knaggs Software Developer Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE Tel: 01482 828000 / Fax: 01482 373100 Daniel.Knaggs at realitysolutions.co.uk www.realitysolutions.co.uk ________________________________ Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire [cid:image27a71e.PNG at c2da8488.4683bff1] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/5e6726e5/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: imageacd695.PNG Type: image/png Size: 22463 bytes Desc: imageacd695.PNG Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/5e6726e5/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: image27a71e.PNG Type: image/png Size: 69075 bytes Desc: image27a71e.PNG Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/5e6726e5/attachment-0003.png From miha at softnet.si Fri Mar 23 16:04:10 2012 From: miha at softnet.si (Miha) Date: Fri, 23 Mar 2012 14:04:10 +0100 Subject: [Freeswitch-users] Break&condition Message-ID: <4F6C74CA.2090404@softnet.si> Hi, In same extension a have multiple conditions. Problem is if the first condition is false, dialplan will go further as I have set on-true. How can I prevent that dialplan will go after break="on-true" on second condition and will not go looking condition inside condition. So if the variable mobilne is not set, in this dialplan FS will go looking to and reject call instead of goint to second condtion which is . I hope I make it clear:D Regards and thank you for your help! Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/9377023b/attachment.html From lazyvirus at gmx.com Fri Mar 23 16:41:43 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 23 Mar 2012 14:41:43 +0100 Subject: [Freeswitch-users] VM sound very choppy In-Reply-To: <1FFF97C269757C458224B7C895F35F1507CB41@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1507CB41@cantor.std.visionutv.se> Message-ID: <20120323144143.10bff4ad@anubis.defcon1> On Fri, 23 Mar 2012 07:24:30 +0000 Peter Olsson wrote: > Another case of virtual machine setup maybe, or is this real hardware? No no, real hardware. JY -- When in doubt, do it. It's much easier to apologize than to get permission. -- Grace Murray Hopper From Vladislav.Grishin at vts24.ru Fri Mar 23 17:02:25 2012 From: Vladislav.Grishin at vts24.ru (=?KOI8-R?Q?=22=E7=D2=C9=DB=C9=CE_=F7=2E=F3=2E=22?=) Date: Fri, 23 Mar 2012 18:02:25 +0400 Subject: [Freeswitch-users] How to set a variable for a call going from gateway? Message-ID: <4F6C8271.7020800@vts24.ru> After I read a http://wiki.freeswitch.com/wiki/Sofia.conf.xml#Variables I configured FS [root at freeswitch1 external]# more /usr/local/freeswitch/conf/sip_profiles/external/SMG-1016M.xml [root at freeswitch1 external]# I insert the info application into public context of dialplanbefore a transfer application? make call but don't see variables (SMG-1016M_inbound_var_name,SMG-1016M_outbound_var_name,SMG-1016M_both_var_name) in fs_cli. I want to see variable = into dialplan for each gateway in (single) external sip profile. How it to make? Vladislav Grishin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/67814c97/attachment.html From gregor at infomedia.si Fri Mar 23 17:04:41 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 23 Mar 2012 15:04:41 +0100 Subject: [Freeswitch-users] Myevents In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD022D77D020@NY1-EXMB-01.ip-soft.net> References: <6A6B4C284AD15042B429EB9D904544AD022D77D001@NY1-EXMB-01.ip-soft.net> <6A6B4C284AD15042B429EB9D904544AD022D77D020@NY1-EXMB-01.ip-soft.net> Message-ID: Thank you Hector! 2012/3/22 Hector Geraldino > I think myevents is better suited for inbound socket connections, so just > do a events plain + filter Unique-ID uuid when you want to filter > events for your outbound channel. **** > > ** ** > > You shouldn?t notice any differences in regards to performance, just be > careful to register to events that you really want to be notified for. > Enabling multiple event notifications in multiple channels will surely > affect FreeSWITCH?s performance.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor > Nanger > *Sent:* Thursday, March 22, 2012 4:33 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Myevents**** > > ** ** > > Thank you Hector!**** > > ** ** > > I think I got it.**** > > ** ** > > For what I want, I should use event plain + filter to uuid. **** > > ** ** > > What has better performance, filter or myevents?**** > > **** > > ** ** > > 2012/3/22 Hector Geraldino **** > > Look at the description on the wiki:**** > > **** > > > http://wiki.freeswitch.org/wiki/Mod_event_socket#Special_Case_-_.27myevents.27 > **** > > **** > > What I understand from this entry is that, when you do an inbound socket > connection to FS, if you use myevents uuid you?ll receive ALL events for > the specified uuid and the socket connection will be closed when the > channel (call) is closed (dropped).**** > > **** > > There?s a difference between the type of events you want to receive (which > you?re already filtering using ?event plain ?) and for which > channels you want to listen for those events. So, even if you do an ?event > plain CHANNEL_ANSWER?, you?ll receive only the CHANNEL_ANSWER event but for > all the calls answered in FreeSWITCH. If you only want to receive calls for > an specific call (not ?global events? as you called), you can add a filter > in the form:**** > > **** > > filter Unique-ID uuid**** > > **** > > By filtering events by uuid, you can control not only what events you want > to be notified for, but also what channels (calls) you want to receive > events for. myevents won?t help you to do that.**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor > Nanger > *Sent:* Thursday, March 22, 2012 2:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Myevents**** > > **** > > I have "full"**** > > **** > > Just want to confirm if this is by design or I missed something.**** > > **** > > I want to subscribe only to "myevents" plus only to events I want in > "events plain ". But both commands don't work like this. If I use > "events plain " I also get global events and not only My events... > **** > > **** > > Is this by design? Maybe I should ommit "full" to get only events of this > call?**** > > **** > > *Error! Filename not specified.***** > > **** > > 2012/3/22 Michael Collins **** > > What is your socket command? If you do "full" then you'll get all events, > otherwise you'll get the equivalent of "myevents." > > Check out the description here: > http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Keywords > > -MC**** > > **** > > On Thu, Mar 22, 2012 at 1:02 AM, Gregor Nanger > wrote:**** > > One explanation...**** > > **** > > If I use in outbound connection event plain , I get only > subscribed events. But if I also use myevents command I get all events. ** > ** > > **** > > So, does myevents command override event plain command?**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/94fd7e15/attachment-0001.html From freeswitch-list at puzzled.xs4all.nl Fri Mar 23 17:06:51 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Fri, 23 Mar 2012 15:06:51 +0100 Subject: [Freeswitch-users] mod h323 error In-Reply-To: <4F6C0453.9060201@tech-invent.ru> References: <1FFF97C269757C458224B7C895F35F1507C35F@cantor.std.visionutv.se> <4F6C0453.9060201@tech-invent.ru> Message-ID: <4F6C837B.5060508@puzzled.xs4all.nl> On 23-03-12 06:04, Dmitry Golubenko wrote: > 22.03.2012 20:15, Peter Olsson ?????: >> It seems you might need to modify the Makefile, because of the errors >> reported. >> >> Please read through this information first: >> http://wiki.freeswitch.org/wiki/Mod_h323 >> >> I personally recommend not to use h323plus-trunk. > I use h323-trunk because gnugk recommends use it > http://www.gnugk.org/gnugk-manual-14.html#ss14.1 , Afaik their recommendations ("Known Good" Combinations) are here: http://www.gnugk.org/compiling-gnugk.html And it does not say to use H323Plus from trunk. The highest versions to use is GnuGK 3.0 with H323Plus 1.24.0 and PTLib 2.10.1 (with the PDNS patch which you can find with Google). So why not try any of these recommendations? Regards, Patrick From gregor at infomedia.si Fri Mar 23 17:07:13 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 23 Mar 2012 15:07:13 +0100 Subject: [Freeswitch-users] DTMF detection Message-ID: One silly question. Is it possible to detect DTMF tones on leg A when call is not answered yet (it is in ringing state)? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/69efd9f2/attachment.html From peter.olsson at visionutveckling.se Fri Mar 23 17:12:49 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 23 Mar 2012 14:12:49 +0000 Subject: [Freeswitch-users] How to set a variable for a call going from gateway? In-Reply-To: <4F6C8271.7020800@vts24.ru> References: <4F6C8271.7020800@vts24.ru> Message-ID: <1FFF97C269757C458224B7C895F35F1507D004@cantor.std.visionutv.se> I don't think this works, at least not for unregistered gateways. You will probably need to do it like this (replace the example IP with the remote IP of the gateway); .... ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r "?????? ?.?." [Vladislav.Grishin at vts24.ru] Skickat: den 23 mars 2012 15:02 Cc: FreeSWITCH-users at lists.freeswitch.org ?mne: [Freeswitch-users] How to set a variable for a call going from gateway? After I read a http://wiki.freeswitch.com/wiki/Sofia.conf.xml#Variables I configured FS [root at freeswitch1 external]# more /usr/local/freeswitch/conf/sip_profiles/external/SMG-1016M.xml [root at freeswitch1 external]# I insert the info application into public context of dialplan before a transfer application? make call but don't see variables (SMG-1016M_inbound_var_name,SMG-1016M_outbound_var_name,SMG-1016M_both_var_name) in fs_cli. I want to see variable = into dialplan for each gateway in (single) external sip profile. How it to make? Vladislav Grishin !DSPAM:4f6c816232767275610573! From haloha201 at gmail.com Fri Mar 23 17:22:49 2012 From: haloha201 at gmail.com (haloha) Date: Fri, 23 Mar 2012 21:22:49 +0700 Subject: [Freeswitch-users] Need help on freeswitch conference Message-ID: Hi List i have problem with freeswitch conference example: 1. a company A with PSTN phone number 0822234562 - auto answer machine(IVR) and an extension of company A is from 1000 - 1990 2. a normal PSTN phone : 0833334512 i setup a freeswitch conference room with number is 50000, a moderator dial *2 to make out bound call my network topology: extension A and extension B ------> FS ------SP ------PSTN extension A is a moderator of conference room 50000 entension A calls 50000 and conference is created successfull so there are 2 cases : 1. extension A dial *2 and then enter 0833334512, 0833334512 is added to conference 50000 successfull and everything is fine 2 extension A dial *2 and enter 0822234562, 0822234562 is added to conference successfull The problem is when extension A dial 0822234562, the IVR is played which ask extension A to enter company A's extension, and extension A dials 1000 but it doesnt work because extension A, IVR are in the conference i do debug and see extension A, IVR in a conference and when extension A dial 1000 and see the DTMF event 1 0 0 0 in a conference so that to make it works i do the command line: conference dtmf <[member_id|all|last]> conference 50000 dtmf all 1000 is there a way in a case 2: to let extension A dials 1000 when IVR playing to reach company A's extension 1000 and then add extension A, 1000 into a conference 50000 Thank you Ha` From robert.longfield at klinsight.com Fri Mar 23 17:51:33 2012 From: robert.longfield at klinsight.com (Robert Longfield) Date: Fri, 23 Mar 2012 10:51:33 -0400 Subject: [Freeswitch-users] Internal VM Audio is below acceptable limit In-Reply-To: <1FFF97C269757C458224B7C895F35F1507CB32@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1507CB32@cantor.std.visionutv.se> Message-ID: <9A7FF78603394134BC0330FAE74C4C3F@KITPC003> Hey Peter, Very interesting, I certainly do not have access to the physical host to do any tweaking for performance values on the virtual machine. What bugs me though is if I pickup the handset attached to an ATA box I do not run into this problem with voicemail. If I had to guess I would have said that this is a problem with softphones and the way they handle audio. I do have an old PC kicking around that I can stick FS on. I?ll move my conf files over and run a test on a physical machine to see what happens. I did fight against putting FS on the VPS but lost. My concern was network latency not potential performance issues. -Rob From: Peter Olsson Sent: Friday, March 23, 2012 3:22 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] Internal VM Audio is below acceptable limit Ok ? that?s the problem then :) If you want to make sure everything ?just works? ? install on real hardware. If you use the built in tools timer_test and time_test I?m pretty sure it will show you quite bad timing. It?s not impossible to succeed with a virtual setup, but it will require lots of testing, and usually you need to be in control of the physical host, so it?s possible to tweak some performance values for the virtual machine. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Robert Longfield Skickat: den 22 mars 2012 22:06 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Internal VM Audio is below acceptable limit Ah, FS is running on a VPS. From: Brian Foster Sent: Thursday, March 22, 2012 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Internal VM Audio is below acceptable limit Actually, I believe what he was asking about was the actual server, although the info you provided already does give us some insight into your setup. Is the freeswitch instance running on virtual or real hardware? -BDF On Mar 22, 2012 3:41 PM, "Robert Longfield" wrote: Hey Peter, Thank for the question. We have a mixture of softphones on computers and mobiles and handsets connected on a ATA box. It seems that the handsets can leave VM but the softphones cannot. I would never have thought of grabbing a handset to test this. For the office this won't be a problem for too long as we are moving to phones but we do have the option of working from home and those cases require softphones to work. -Rob -----Original Message----- From: Peter Olsson Sent: Thursday, March 22, 2012 3:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Internal VM Audio is below acceptable limit Are you on real hardware, or on a virtual machine? /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Robert Longfield [robert.longfield at klinsight.com] Skickat: den 22 mars 2012 20:16 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Internal VM Audio is below acceptable limit I have FS up and running with the awesome help of this community. I have a few minor bugs to workout. Some of my own doing (music on hold) but one that I can't trackdown or figure out the cause is when someone internally calls an extension and tries to leave a voice mail the system says something to the effect that the audio was below acceptable limit, basically it was too short. If the person at the extension picks up there is a slight pause and then you can talk with no issues and the audio sounds great. If the same extension is called from an outside line and the call goes to VM there are no problems leaving a message. Pastebin of the log http://pastebin.freeswitch.org/18723 -Rob _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f6b937d32761551985203! -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/221c43cd/attachment-0001.html From miconda at gmail.com Fri Mar 23 18:02:12 2012 From: miconda at gmail.com (Daniel-Constantin Mierla) Date: Fri, 23 Mar 2012 16:02:12 +0100 Subject: [Freeswitch-users] ITSPA 2012 Award for Open Source VoIP Projects Message-ID: <4F6C9074.9010308@gmail.com> Hello, ITSPA UK has unveiled the winners of its 4th annual Awards, an event designed to celebrate innovation and best practice in the VoIP industry: * http://www.itspaawards.org.uk/ Open Source VoIP Projects won a special category this year, Members' Pick, for providing a real value to VoIP Industry. I had the chance to attend the event in London and I have been selected to pick up the award. I made a news on the website of the project I am mainly involved in (Kamailio) with more details: * http://www.kamailio.org/w/2012/03/itspa-awards-2012-open-source-voip-projects/ As you would expect, a complete voip platform usually involves several open source projects, for components such as load balancers, registrar, proxy, gateways or media servers, thus the decision of ITSPA for awarding to the group. FreeSwitch was a frequent presence in the VoIP systems of the ITSPA members I spoke to. I guess it is no surprise at all, you know it very well from the community. Therefore the award happened due to a significant contribution of FreeSwitch to IP Telephony industry as well. As another user of FreeSwitch project, I take the opportunity to thank again to the people behind the project. This time, special wishes to Brian West - fast and smooth recovery, come back quickly to full "ultra wide band" capacity! If anyone is looking for more insights (for news, blogs, personal curiosity) about the event, just drop me an email! Cheers, Daniel -- Daniel-Constantin Mierla Co-Founder Kamailio SIP Server - http://www.kamailio.org Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/ From a.avona at elios.net Fri Mar 23 12:47:28 2012 From: a.avona at elios.net (a.avona) Date: Fri, 23 Mar 2012 10:47:28 +0100 Subject: [Freeswitch-users] Grandstream 4104 Message-ID: <4F6C46B0.7080805@elios.net> Hi, all i have a GrandStream 4104 perfectly working (both incoming and outgoing calls work great) with an asterisk pbx. We are trying to change the asterisk with freeswitch, but i'm exeperiencing problems in configuring freeswitch for outgoing calls, incoming calls works well. this is my configuration and what i obtain in fs_cli consolle Can someone tell me where i'm wrong? in sip_profile/internal i created a file 00_to_pstn.xml this way in dialplan/default i created a file 00_to_pstn.xml this way _________________________________________________________________ 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:7559 IP 192.168.0.200 Rejected by acl "domains". Falling back to Digest auth. 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:7559 IP 192.168.0.200 Rejected by acl "domains". Falling back to Digest auth. 2012-03-23 10:38:43.937574 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1000 at 192.168.0.2 [fb33155c-74cb-11e1-9063-fb6a5f4599fb] 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:5526 Channel sofia/internal/1000 at 192.168.0.2 entering state [received][100] 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:5537 Remote SDP: v=0 o=- 935122583 0 IN IP4 192.168.0.200 s=SIPPER for PhonerLite c=IN IP4 192.168.0.200 t=0 0 m=audio 5062 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:2991 Set Codec sofia/internal/1000 at 192.168.0.2 PCMA/8000 20 ms 160 samples 64000 bits 2012-03-23 10:38:43.937574 [DEBUG] switch_core_codec.c:111 sofia/internal/1000 at 192.168.0.2 Original read codec set to PCMA:8 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4995 Set 2833 dtmf send/recv payload to 101 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:5749 (sofia/internal/1000 at 192.168.0.2) State Change CS_NEW -> CS_INIT 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_INIT 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1000 at 192.168.0.2) State INIT 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:85 sofia/internal/1000 at 192.168.0.2 SOFIA INIT 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:125 (sofia/internal/1000 at 192.168.0.2) State Change CS_INIT -> CS_ROUTING 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1000 at 192.168.0.2) State INIT going to sleep 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_ROUTING 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1886 (sofia/internal/1000 at 192.168.0.2) Callstate Change DOWN -> RINGING 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1000 at 192.168.0.2) State ROUTING 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:148 sofia/internal/1000 at 192.168.0.2 SOFIA ROUTING 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1000 at 192.168.0.2 Standard ROUTING 2012-03-23 10:38:43.937574 [INFO] mod_dialplan_xml.c:485 Processing 1000 <1000>->339XXXXXXX in context default Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unloop] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->tod_example] continue=true Dialplan: sofia/internal/1000 at 192.168.0.2 Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(open=true) Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->holiday_example] continue=true Dialplan: sofia/internal/1000 at 192.168.0.2 Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global-intercept] destination_number(339XXXXXXX) =~ /^886$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group-intercept] destination_number(339XXXXXXX) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [intercept-ext] destination_number(339XXXXXXX) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->redial] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [redial] destination_number(339XXXXXXX) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->global] continue=true Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/1000 at 192.168.0.2 Absolute Condition [global] Dialplan: sofia/internal/1000 at 192.168.0.2 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1000 at 192.168.0.2 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1000 at 192.168.0.2 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1000 at 192.168.0.2 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-2] destination_number(339XXXXXXX) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-1] destination_number(339XXXXXXX) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] destination_number(339XXXXXXX) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] destination_number(339XXXXXXX) =~ /^779$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->call_return] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call_return] destination_number(339XXXXXXX) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->del-group] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [del-group] destination_number(339XXXXXXX) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->add-group] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [add-group] destination_number(339XXXXXXX) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call-group-simo] destination_number(339XXXXXXX) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call-group-order] destination_number(339XXXXXXX) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [extension-intercom] destination_number(339XXXXXXX) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [Local_Extension] destination_number(339XXXXXXX) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->Local_Extension_Skinny] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [Local_Extension_Skinny] destination_number(339XXXXXXX) =~ /^(11[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->group_dial_sales] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group_dial_sales] destination_number(339XXXXXXX) =~ /^2000$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->group_dial_support] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group_dial_support] destination_number(339XXXXXXX) =~ /^2001$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->group_dial_billing] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group_dial_billing] destination_number(339XXXXXXX) =~ /^2002$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->operator] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [operator] destination_number(339XXXXXXX) =~ /^(operator|0)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->vmain] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [vmain] destination_number(339XXXXXXX) =~ /^vmain$|^4000$|^\*98$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->sip_uri] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [sip_uri] destination_number(339XXXXXXX) =~ /^sip:(.*)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->nb_conferences] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [nb_conferences] destination_number(339XXXXXXX) =~ /^(30\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->wb_conferences] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wb_conferences] destination_number(339XXXXXXX) =~ /^(31\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->uwb_conferences] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [uwb_conferences] destination_number(339XXXXXXX) =~ /^(32\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->cdquality_conferences] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [cdquality_conferences] destination_number(339XXXXXXX) =~ /^(33\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->freeswitch_public_conf_via_sip] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(339XXXXXXX) =~ /^9(888|8888|1616|3232)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->mad_boss_intercom] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [mad_boss_intercom] destination_number(339XXXXXXX) =~ /^0911$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->mad_boss_intercom] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [mad_boss_intercom] destination_number(339XXXXXXX) =~ /^0912$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->mad_boss] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [mad_boss] destination_number(339XXXXXXX) =~ /^0913$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ivr_demo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ivr_demo] destination_number(339XXXXXXX) =~ /^5000$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->dynamic_conference] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [dynamic_conference] destination_number(339XXXXXXX) =~ /^5001$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->rtp_multicast_page] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [rtp_multicast_page] destination_number(339XXXXXXX) =~ /^pagegroup$|^7243$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] destination_number(339XXXXXXX) =~ /^5900$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] destination_number(339XXXXXXX) =~ /^5901$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->valet_park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] destination_number(339XXXXXXX) =~ /^(6000)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->valet_park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] destination_number(339XXXXXXX) =~ /^(60\d[1-9])$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] destination_number(339XXXXXXX) =~ /park\+(\d+)/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] destination_number(339XXXXXXX) =~ /^parking$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] destination_number(339XXXXXXX) =~ /callpark/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] destination_number(339XXXXXXX) =~ /pickup/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->wait] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wait] destination_number(339XXXXXXX) =~ /^wait$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->fax_receive] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_receive] destination_number(339XXXXXXX) =~ /^9178$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->fax_transmit] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_transmit] destination_number(339XXXXXXX) =~ /^9179$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_180] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_180] destination_number(339XXXXXXX) =~ /^9180$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_183_uk_ring] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_183_uk_ring] destination_number(339XXXXXXX) =~ /^9181$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_183_music_ring] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_183_music_ring] destination_number(339XXXXXXX) =~ /^9182$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_post_answer_uk_ring] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(339XXXXXXX) =~ /^9183$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_post_answer_music] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_post_answer_music] destination_number(339XXXXXXX) =~ /^9184$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ClueCon] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ClueCon] destination_number(339XXXXXXX) =~ /^9191$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->show_info] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [show_info] destination_number(339XXXXXXX) =~ /^9192$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->video_record] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_record] destination_number(339XXXXXXX) =~ /^9193$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->video_playback] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_playback] destination_number(339XXXXXXX) =~ /^9194$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->delay_echo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [delay_echo] destination_number(339XXXXXXX) =~ /^9195$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->echo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [echo] destination_number(339XXXXXXX) =~ /^9196$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->milliwatt] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [milliwatt] destination_number(339XXXXXXX) =~ /^9197$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->tone_stream] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [tone_stream] destination_number(339XXXXXXX) =~ /^9198$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->zrtp_enrollement] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [zrtp_enrollement] destination_number(339XXXXXXX) =~ /^9787$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->hold_music] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [hold_music] destination_number(339XXXXXXX) =~ /^9664$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->from_pstn] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [from_pstn] destination_number(339XXXXXXX) =~ /^0000$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->101] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [101] destination_number(339XXXXXXX) =~ /^101$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->pizza_demo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [pizza_demo] destination_number(339XXXXXXX) =~ /^(pizza|74992)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->gxw4104-fxo-local] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [gxw4104-fxo-local] ${toll_allow}(domestic,international,local) =~ /local/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [gxw4104-fxo-local] destination_number(339XXXXXXX) =~ /^(\d{6,})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(effective_caller_id_number=0321234567) Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(effective_caller_id_name=ThisIsMyCompany) Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(ignore_early_media=ring_ready) Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(ringback=${us-ring}) Dialplan: sofia/internal/1000 at 192.168.0.2 Action bridge(sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/1000 at 192.168.0.2) State Change CS_ROUTING -> CS_EXECUTE 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1000 at 192.168.0.2) State ROUTING going to sleep 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_EXECUTE 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1000 at 192.168.0.2) State EXECUTE 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:241 sofia/internal/1000 at 192.168.0.2 SOFIA EXECUTE 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:192 sofia/internal/1000 at 192.168.0.2 Standard EXECUTE EXECUTE sofia/internal/1000 at 192.168.0.2 set(open=true) 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [open]=[true] EXECUTE sofia/internal/1000 at 192.168.0.2 hash(insert/192.168.0.2-spymap/1000/fb33155c-74cb-11e1-9063-fb6a5f4599fb) EXECUTE sofia/internal/1000 at 192.168.0.2 hash(insert/192.168.0.2-last_dial/1000/339XXXXXXX) EXECUTE sofia/internal/1000 at 192.168.0.2 hash(insert/192.168.0.2-last_dial/global/fb33155c-74cb-11e1-9063-fb6a5f4599fb) EXECUTE sofia/internal/1000 at 192.168.0.2 export(RFC2822_DATE=Fri, 23 Mar 2012 10:38:43 +0100) 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [RFC2822_DATE]=[Fri, 23 Mar 2012 10:38:43 +0100] EXECUTE sofia/internal/1000 at 192.168.0.2 set(effective_caller_id_number=0321234567) 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [effective_caller_id_number]=[0321234567] EXECUTE sofia/internal/1000 at 192.168.0.2 set(effective_caller_id_name=ThisIsMyCompany) 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [effective_caller_id_name]=[ThisIsMyCompany] EXECUTE sofia/internal/1000 at 192.168.0.2 set(ignore_early_media=ring_ready) 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [ignore_early_media]=[ring_ready] EXECUTE sofia/internal/1000 at 192.168.0.2 set(ringback=%(2000,4000,440,480)) 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [ringback]=[%(2000,4000,440,480)] EXECUTE sofia/internal/1000 at 192.168.0.2 bridge(sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1047 sofia/internal/1000 at 192.168.0.2 EXPORTING[export_vars] [RFC2822_DATE]=[Fri, 23 Mar 2012 10:38:43 +0100] to event 2012-03-23 10:38:43.937574 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-23 10:38:43.937574 [NOTICE] switch_channel.c:926 New Channel sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [fb34cf5a-74cb-11e1-9068-fb6a5f4599fb] 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:4691 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change CS_NEW -> CS_INIT 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State Change CS_INIT 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State INIT 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:85 sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 SOFIA INIT 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:125 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change CS_INIT -> CS_ROUTING 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State INIT going to sleep 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State Change CS_ROUTING 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1886 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Callstate Change DOWN -> RINGING 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State ROUTING 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:148 sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 SOFIA ROUTING 2012-03-23 10:38:43.957579 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State ROUTING going to sleep 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State Change CS_CONSUME_MEDIA 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State CONSUME_MEDIA 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State CONSUME_MEDIA going to sleep 2012-03-23 10:38:43.957579 [DEBUG] sofia.c:5526 Channel sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 entering state [calling][0] 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-23 10:38:43.957579 [DEBUG] sofia.c:5526 Channel sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 entering state [terminated][403] 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2848 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Callstate Change RINGING -> HANGUP 2012-03-23 10:38:43.957579 [NOTICE] sofia.c:6293 Hangup sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [CS_CONSUME_MEDIA] [CALL_REJECTED] 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [KILL] 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State Change CS_HANGUP 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State HANGUP 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:469 Channel sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 hanging up, cause: CALL_REJECTED 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:47 sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 Standard HANGUP, cause: CALL_REJECTED 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State HANGUP going to sleep 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change CS_HANGUP -> CS_REPORTING 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State Change CS_REPORTING 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State REPORTING 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:79 sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 Standard REPORTING, cause: CALL_REJECTED 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State REPORTING going to sleep 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change CS_REPORTING -> CS_DESTROY 2012-03-23 10:38:43.957579 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 21 [CALL_REJECTED] 2012-03-23 10:38:43.957579 [INFO] mod_dptools.c:2922 Originate Failed. Cause: CALL_REJECTED 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2848 (sofia/internal/1000 at 192.168.0.2) Callstate Change RINGING -> HANGUP 2012-03-23 10:38:43.957579 [NOTICE] mod_dptools.c:3041 Hangup sofia/internal/1000 at 192.168.0.2 [CS_EXECUTE] [CALL_REJECTED] 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/1000 at 192.168.0.2 [KILL] 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:2285 sofia/internal/1000 at 192.168.0.2 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1000 at 192.168.0.2) State EXECUTE going to sleep 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_HANGUP 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1000 at 192.168.0.2) State HANGUP 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:463 sofia/internal/1000 at 192.168.0.2 Overriding SIP cause 603 with 403 from the other leg 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1000 at 192.168.0.2 hanging up, cause: CALL_REJECTED 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:534 Responding to INVITE with: 403 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1000 at 192.168.0.2 Standard HANGUP, cause: CALL_REJECTED 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1000 at 192.168.0.2) State HANGUP going to sleep 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1000 at 192.168.0.2) State Change CS_HANGUP -> CS_REPORTING 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_REPORTING 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1000 at 192.168.0.2) State REPORTING 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1000 at 192.168.0.2 Standard REPORTING, cause: CALL_REJECTED 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1000 at 192.168.0.2) State REPORTING going to sleep 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1000 at 192.168.0.2) State Change CS_REPORTING -> CS_DESTROY 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1380 Session 13 (sofia/internal/1000 at 192.168.0.2) Locked, Waiting on external entities 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1398 Session 13 (sofia/internal/1000 at 192.168.0.2) Ended 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/1000 at 192.168.0.2 [CS_DESTROY] 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1000 at 192.168.0.2) Callstate Change HANGUP -> DOWN 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_DESTROY 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1000 at 192.168.0.2) State DESTROY 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:374 sofia/internal/1000 at 192.168.0.2 SOFIA DESTROY 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1000 at 192.168.0.2 Standard DESTROY 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1000 at 192.168.0.2) State DESTROY going to sleep 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1380 Session 14 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Locked, Waiting on external entities 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1398 Session 14 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Ended 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [CS_DESTROY] 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Callstate Change HANGUP -> DOWN 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State Change CS_DESTROY 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State DESTROY 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:374 sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 SOFIA DESTROY 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:86 sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 Standard DESTROY 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State DESTROY going to sleep freeswitch at internal> Thank's in advance Regards Accursio Avona From tpe at actimizer.com Fri Mar 23 13:42:58 2012 From: tpe at actimizer.com (Tor Petterson) Date: Fri, 23 Mar 2012 11:42:58 +0100 Subject: [Freeswitch-users] Large delay/latency when bridging SIP calls Message-ID: Hi I'm having problems with bridging sip calls in Freeswitch after the revision from February 27 "only flush on break when its a blocking situation part 1". I am using Freeswitch in a phone center application where I first call an agent, then call a lead and then bridge the two calls. If I use a version after Feb 27. it takes about 5 seconds after bridging before any sound gets through. Here is an anonymized version of my sofia profile: -- Tor Petterson tpe at actimizer.com Tobaksvejen 25, 2. tv. - 2860 S?borg Telephone: +45 39 55 05 32 www.actimizer.com From garbytrash at gmail.com Fri Mar 23 18:22:52 2012 From: garbytrash at gmail.com (Zenny) Date: Fri, 23 Mar 2012 15:22:52 +0000 Subject: [Freeswitch-users] ITSPA 2012 Award for Open Source VoIP Projects In-Reply-To: <4F6C9074.9010308@gmail.com> References: <4F6C9074.9010308@gmail.com> Message-ID: Congrats, Daniel and team for the award! On 3/23/12, Daniel-Constantin Mierla wrote: > Hello, > > ITSPA UK has unveiled the winners of its 4th annual Awards, an event > designed to celebrate innovation and best practice in the VoIP industry: > > * http://www.itspaawards.org.uk/ > > Open Source VoIP Projects won a special category this year, Members' > Pick, for providing a real value to VoIP Industry. > > I had the chance to attend the event in London and I have been selected > to pick up the award. I made a news on the website of the project I am > mainly involved in (Kamailio) with more details: > > * > http://www.kamailio.org/w/2012/03/itspa-awards-2012-open-source-voip-projects/ > > As you would expect, a complete voip platform usually involves several > open source projects, for components such as load balancers, registrar, > proxy, gateways or media servers, thus the decision of ITSPA for > awarding to the group. > > FreeSwitch was a frequent presence in the VoIP systems of the ITSPA > members I spoke to. I guess it is no surprise at all, you know it very > well from the community. Therefore the award happened due to a > significant contribution of FreeSwitch to IP Telephony industry as well. > > As another user of FreeSwitch project, I take the opportunity to thank > again to the people behind the project. This time, special wishes to > Brian West - fast and smooth recovery, come back quickly to full "ultra > wide band" capacity! > > If anyone is looking for more insights (for news, blogs, personal > curiosity) about the event, just drop me an email! > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > Co-Founder Kamailio SIP Server - http://www.kamailio.org > Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany > http://www.asipto.com/index.php/kamailio-advanced-training/ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Fri Mar 23 18:33:24 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Mar 2012 08:33:24 -0700 Subject: [Freeswitch-users] Break&condition In-Reply-To: <4F6C74CA.2090404@softnet.si> References: <4F6C74CA.2090404@softnet.si> Message-ID: Miha, It might help if you show us a bit more of your dialplan. It may be that you need to break some of these out into separate extensions. Also, what is your "big picture" application? What is the problem that you are attempting to solve? -MC On Fri, Mar 23, 2012 at 6:04 AM, Miha wrote: > Hi, > > In same extension a have multiple conditions. Problem is if the first > condition is false, dialplan will go further as I have set on-true. > How can I prevent that dialplan will go after break="on-true" on second > condition and will not go looking condition inside condition. > > So if the variable mobilne is not set, in this dialplan FS will go looking > to expression="^(051|041|031|030|040|070|071)(\d{6})|^(0038651|0038641|0038631|0038630|0038640|0038670|0038671)(\d{6})" > break="on-true"> and reject call instead of goint to second condtion which > is . > > I hope I make it clear:D > > > expression="^(051|041|031|030|040|070|071)(\d{6})|^(0038651|0038641|0038631|0038630|0038640|0038670|0038671)(\d{6})" > break="on-true"> > > > > > > > > > > > > > expression="^(090)(\d{4})|^(090)(\d{6})"> > > > > > Regards and thank you for your help! > > Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/cf468590/attachment.html From garbytrash at gmail.com Fri Mar 23 18:35:52 2012 From: garbytrash at gmail.com (Zenny) Date: Fri, 23 Mar 2012 15:35:52 +0000 Subject: [Freeswitch-users] ITSPA 2012 Award for Open Source VoIP Projects In-Reply-To: References: <4F6C9074.9010308@gmail.com> Message-ID: Oh, the headline was confusing. I thought that Kamailio received among the open source voip projects. But when I visited the ITSPA site, it is for all "Open Source Voip Products". I congratulate all who selflessly contributed to realizing what VoIP looks today including FreeSWITCH developers and others who made it possible with the components like (OpenSER/OpenSIPS, Siproxd, asterisk, yate, mediaproxy, radius, cdr-tools and scripts and frontends. Thanks for the great work!!! /z --- Support http://thehumanape.org On 3/23/12, Zenny wrote: > Congrats, Daniel and team for the award! > > On 3/23/12, Daniel-Constantin Mierla wrote: >> Hello, >> >> ITSPA UK has unveiled the winners of its 4th annual Awards, an event >> designed to celebrate innovation and best practice in the VoIP industry: >> >> * http://www.itspaawards.org.uk/ >> >> Open Source VoIP Projects won a special category this year, Members' >> Pick, for providing a real value to VoIP Industry. >> >> I had the chance to attend the event in London and I have been selected >> to pick up the award. I made a news on the website of the project I am >> mainly involved in (Kamailio) with more details: >> >> * >> http://www.kamailio.org/w/2012/03/itspa-awards-2012-open-source-voip-projects/ >> >> As you would expect, a complete voip platform usually involves several >> open source projects, for components such as load balancers, registrar, >> proxy, gateways or media servers, thus the decision of ITSPA for >> awarding to the group. >> >> FreeSwitch was a frequent presence in the VoIP systems of the ITSPA >> members I spoke to. I guess it is no surprise at all, you know it very >> well from the community. Therefore the award happened due to a >> significant contribution of FreeSwitch to IP Telephony industry as well. >> >> As another user of FreeSwitch project, I take the opportunity to thank >> again to the people behind the project. This time, special wishes to >> Brian West - fast and smooth recovery, come back quickly to full "ultra >> wide band" capacity! >> >> If anyone is looking for more insights (for news, blogs, personal >> curiosity) about the event, just drop me an email! >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> Co-Founder Kamailio SIP Server - http://www.kamailio.org >> Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany >> http://www.asipto.com/index.php/kamailio-advanced-training/ >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From miconda at gmail.com Fri Mar 23 19:03:05 2012 From: miconda at gmail.com (Daniel-Constantin Mierla) Date: Fri, 23 Mar 2012 17:03:05 +0100 Subject: [Freeswitch-users] ITSPA 2012 Award for Open Source VoIP Projects In-Reply-To: References: <4F6C9074.9010308@gmail.com> Message-ID: <4F6C9EB9.3040009@gmail.com> On 3/23/12 4:35 PM, Zenny wrote: > Oh, the headline was confusing. I thought that Kamailio received among > the open source voip projects. I hoped not to be confusing and it is indeed for all of the Open Source projects with relevant contributions to IP telephony. I explained a bit in my email the reason for awarding the group as a generic entity - blending together several open source application results in a more complete voip systems and it is very common practice (so many persons would have had difficulties to pick one since they use many). The news was propagated here because FS was really a frequent installation to ITSPA members' telephony platforms (again, among those I spoke to). Not the least, I am using it, I know many of its developers and their dedication to the project. I wouldn't have sent an email here about awards to other projects only - as you mentioned, there are other OS projects in this space, personally I tried to reflect from my direct interaction at the event, in the places I am involve in. Otherwise ITSPA made all public. In short, my email was to congratulate FreeSwitch project for being awarded by ITSPA UK. Cheers, Daniel -- Daniel-Constantin Mierla Co-Founder Kamailio SIP Server - http://www.kamailio.org Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/ From msc at freeswitch.org Fri Mar 23 19:55:44 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Mar 2012 09:55:44 -0700 Subject: [Freeswitch-users] FreeSWITCH Wiki Update: Syntax Highlighting Now Available! Message-ID: Hey all, Hat tip to David Knell for figuring out how to get the GeSHi syntax highlighting to work on wiki.freeswitch.org! If you'd like to have syntax highlighting on a snippet of code just do this: ...code... Languages include: c, perl, python, xml, lua, etc. Example: I added the basic information to this page on the wiki: http://wiki.freeswitch.org/wiki/Documentation_Guidelines#Wiki_Markup If you would like a non-glamorous but useful task to do: feel free to edit each code sample you see to have the tag with the appropriate language. Also, if there are any GeSHi warriors out there who could write a FreeSWITCH Log syntax highlighter module I'm sure that would be welcomed. :) Thanks again! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/dde286f5/attachment.html From msc at freeswitch.org Fri Mar 23 20:01:44 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Mar 2012 10:01:44 -0700 Subject: [Freeswitch-users] Myevents In-Reply-To: References: Message-ID: I'd recommend doing explicit filters. Filters are covered nicely in both FreeSWITCH books as well as the wiki. http://wiki.freeswitch.org/wiki/Mod_event_socket#filter -MC On Thu, Mar 22, 2012 at 11:31 AM, Gregor Nanger wrote: > I have "full" > > Just want to confirm if this is by design or I missed something. > > I want to subscribe only to "myevents" plus only to events I want in > "events plain ". But both commands don't work like this. If I use > "events plain " I also get global events and not only My events... > > Is this by design? Maybe I should ommit "full" to get only events of this > call? > > > > 2012/3/22 Michael Collins > >> What is your socket command? If you do "full" then you'll get all events, >> otherwise you'll get the equivalent of "myevents." >> >> Check out the description here: >> http://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Keywords >> >> -MC >> >> >> On Thu, Mar 22, 2012 at 1:02 AM, Gregor Nanger wrote: >> >>> One explanation... >>> >>> If I use in outbound connection event plain , I get only >>> subscribed events. But if I also use myevents command I get all events. >>> >>> So, does myevents command override event plain command? >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/90b6ad1f/attachment.html From william.suffill at gmail.com Fri Mar 23 20:01:54 2012 From: william.suffill at gmail.com (William Suffill) Date: Fri, 23 Mar 2012 13:01:54 -0400 Subject: [Freeswitch-users] FreeSWITCH Project - Call For Assistance In-Reply-To: References: Message-ID: Do we have some idea on how the highlighting on the pastebin is defined? Would it be possible to port that to various editors and other applications to be semi uniform? -- W From bdfoster at endigotech.com Fri Mar 23 20:07:42 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 23 Mar 2012 13:07:42 -0400 Subject: [Freeswitch-users] FreeSWITCH Installer for Debian In-Reply-To: References: Message-ID: Version 0.0.3 was released today: Changelog ================================================================================= 0.0.3 23/Mar/2012 1235 UCT Bug Fixes - Changed script shell from bash to sh for compatability Features Added -Made FS git address, FS user, FS group, installed packages, FS sounds install, FS MOH install, and base dir for source folder into variables - Option added to update/upgrade system (true/false variable) - Option to download/install init script Version 0.1.0 will be released sometime within the next week. -BDF On Thu, Mar 22, 2012 at 6:21 PM, Gabriel Gunderson wrote: > On Thu, Mar 22, 2012 at 3:44 PM, Zenny wrote: > > Glad to find build_opensips.sh also in the same directory. > > If you want to save the trouble of building, there are also some > pre-built debs for both FreeSWITCH and OpenSIPS: > > https://parseltone.org/files/debs/ubuntu-11.4/ > > > Best, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/a039eba6/attachment.html From bdfoster at endigotech.com Fri Mar 23 20:12:09 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 23 Mar 2012 13:12:09 -0400 Subject: [Freeswitch-users] FreeSWITCH Installer for Debian In-Reply-To: References: Message-ID: Whoops. The new version (and always the latest) can be downloaded from here: http://files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_latest.sh On Fri, Mar 23, 2012 at 1:07 PM, Brian Foster wrote: > Version 0.0.3 was released today: > > Changelog > > ================================================================================= > 0.0.3 23/Mar/2012 1235 UCT > > Bug Fixes > - Changed script shell from bash to sh for compatability > Features Added > -Made FS git address, FS user, FS group, installed packages, FS > sounds > install, FS MOH install, and base dir for source folder into > variables > - Option added to update/upgrade system (true/false variable) > - Option to download/install init script > > Version 0.1.0 will be released sometime within the next week. > > -BDF > > On Thu, Mar 22, 2012 at 6:21 PM, Gabriel Gunderson wrote: > >> On Thu, Mar 22, 2012 at 3:44 PM, Zenny wrote: >> > Glad to find build_opensips.sh also in the same directory. >> >> If you want to save the trouble of building, there are also some >> pre-built debs for both FreeSWITCH and OpenSIPS: >> >> https://parseltone.org/files/debs/ubuntu-11.4/ >> >> >> Best, >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/94d3ec50/attachment-0001.html From wstephen80 at gmail.com Fri Mar 23 20:31:32 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 23 Mar 2012 18:31:32 +0100 Subject: [Freeswitch-users] Strange "timer_test" result Message-ID: I have run a "timer_test" in a dedicated FS server and I see strange result: it's normal? Stephen freeswitch at internal> timer_test 20 40 Avg: 19.866ms Total Time: 795.880ms 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: samplecount after init: 1 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: samplecount after first step: 2 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 sleep 20 19568 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 sleep 20 38231 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 sleep 20 18847 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 sleep 20 13982 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 sleep 20 34793 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 sleep 20 2 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 sleep 20 23166 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 sleep 20 16957 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 sleep 20 17643 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 sleep 20 18786 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 sleep 20 25100 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 sleep 20 18552 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 sleep 20 18815 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 sleep 20 19464 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 sleep 20 22012 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 sleep 20 13980 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 sleep 20 19065 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 sleep 20 39585 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 sleep 20 2 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 sleep 20 26255 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 sleep 20 17872 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 sleep 20 32191 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 sleep 20 22634 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 sleep 20 15483 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 sleep 20 22813 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 sleep 20 17099 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 sleep 20 1 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 sleep 20 29108 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 sleep 20 11492 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 sleep 20 20855 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 sleep 20 32579 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 sleep 20 18173 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 sleep 20 22666 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 sleep 20 23792 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 sleep 20 26158 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 sleep 20 13080 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 sleep 20 24609 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 sleep 20 1 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 sleep 20 19413 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 sleep 20 19820 freeswitch at internal> status UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, 696 microseconds FreeSWITCH is ready 955611 session(s) since startup 2192 session(s) 0/50 6000 session(s) max min idle cpu 0.00/74.00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/9d97aad4/attachment.html From bdfoster at endigotech.com Fri Mar 23 20:32:41 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 23 Mar 2012 13:32:41 -0400 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: References: Message-ID: Please update to latest git. -BDF On Fri, Mar 23, 2012 at 7:19 AM, Daniel Knaggs < Daniel.Knaggs at realitysolutions.co.uk> wrote: > Hello all,**** > > ** ** > > Got a bit of a strange one, we appear to be getting DTMF tones on incoming > calls when the caller hasn?t even pressed any keys.**** > > ** ** > > It normally happens with 10 seconds or so after the call has been answered. > **** > > ** ** > > ** ** > > Here is the log of it happening earlier: -**** > > ** ** > > 2012-03-23 10:38:16.852654 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF > A (debug = 0)**** > > 2012-03-23 10:38:16.852654 [DEBUG] mod_freetdm.c:799 Queuing DTMF [A] in > channel FreeTDM/1:2/000 device 1:2**** > > 2012-03-23 10:38:16.915653 [DEBUG] switch_rtp.c:2420 Send start packet for > [A] ts=49440 dur=160/160/2000 seq=46803 lw=49440**** > > 2012-03-23 10:38:16.936653 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=320/320/2000 seq=46804 lw=49600**** > > 2012-03-23 10:38:16.957652 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=480/480/2000 seq=46805 lw=49760**** > > 2012-03-23 10:38:16.978652 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=640/640/2000 seq=46806 lw=49920**** > > 2012-03-23 10:38:16.999652 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=800/800/2000 seq=46807 lw=50080**** > > 2012-03-23 10:38:17.020651 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=960/960/2000 seq=46808 lw=50240**** > > 2012-03-23 10:38:17.041651 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=1120/1120/2000 seq=46809 lw=50400**** > > 2012-03-23 10:38:17.062651 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=1280/1280/2000 seq=46810 lw=50560**** > > 2012-03-23 10:38:17.083650 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=1440/1440/2000 seq=46811 lw=50720**** > > 2012-03-23 10:38:17.104650 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=1600/1600/2000 seq=46812 lw=50880**** > > 2012-03-23 10:38:17.125650 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=1760/1760/2000 seq=46813 lw=51040**** > > 2012-03-23 10:38:17.146650 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=1920/1920/2000 seq=46814 lw=51200**** > > 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for > [A] ts=49440 dur=2080/2080/2000 seq=46815 lw=51200**** > > 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for > [A] ts=49440 dur=2080/2080/2000 seq=46816 lw=51200**** > > 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for > [A] ts=49440 dur=2080/2080/2000 seq=46817 lw=51200**** > > 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2271 Queue digit delay of > 40ms**** > > 2012-03-23 10:38:18.070638 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF > B (debug = 0)**** > > 2012-03-23 10:38:18.070638 [DEBUG] mod_freetdm.c:799 Queuing DTMF [B] in > channel FreeTDM/1:2/000 device 1:2**** > > 2012-03-23 10:38:18.133637 [DEBUG] switch_rtp.c:2420 Send start packet for > [B] ts=59040 dur=160/160/2000 seq=46864 lw=59040**** > > 2012-03-23 10:38:18.154637 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=320/320/2000 seq=46865 lw=59200**** > > 2012-03-23 10:38:18.175637 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=480/480/2000 seq=46866 lw=59360**** > > 2012-03-23 10:38:18.196636 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=640/640/2000 seq=46867 lw=59520**** > > 2012-03-23 10:38:18.217636 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=800/800/2000 seq=46868 lw=59680**** > > 2012-03-23 10:38:18.238636 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=960/960/2000 seq=46869 lw=59840**** > > 2012-03-23 10:38:18.259636 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=1120/1120/2000 seq=46870 lw=60000**** > > 2012-03-23 10:38:18.280636 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=1280/1280/2000 seq=46871 lw=60160**** > > 2012-03-23 10:38:18.301635 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=1440/1440/2000 seq=46872 lw=60320**** > > 2012-03-23 10:38:18.322635 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=1600/1600/2000 seq=46873 lw=60480**** > > 2012-03-23 10:38:18.343634 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=1760/1760/2000 seq=46874 lw=60640**** > > 2012-03-23 10:38:18.364634 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=1920/1920/2000 seq=46875 lw=60800**** > > 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for > [B] ts=59040 dur=2080/2080/2000 seq=46876 lw=60800**** > > 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for > [B] ts=59040 dur=2080/2080/2000 seq=46877 lw=60800**** > > 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for > [B] ts=59040 dur=2080/2080/2000 seq=46878 lw=60800**** > > ** ** > > ** ** > > I?m very sure the caller isn?t dialling ?A? or ?B?!**** > > ** ** > > Running on ISDN via a TE121 card using E1 (euroisdn). Freeswitch version > is ?FreeSWITCH Version 1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)?.* > *** > > ** ** > > ** ** > > Currently there is a ?bind_meta_app? in the config which binds a script on > the B leg of the call which parks the call ? I haven?t tried turning this > off yet to see if it?s this.**** > > ** ** > > ** ** > > Wondering if anyone has any ideas or has come across this before?**** > > ** ** > > ** ** > > Thanks in advance.**** > > Daniel Knaggs > > Software Developer > Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston > upon Hull, East Yorkshire, HU7 0AE > Tel: 01482 828000 / Fax: 01482 373100 > Daniel.Knaggs at realitysolutions.co.uk > www.realitysolutions.co.uk > ------------------------------ > Sage Accredited Business Partner serving businesses in Yorkshire & > Lincolnshire > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/c6aad5f8/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 22463 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/c6aad5f8/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 69075 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/c6aad5f8/attachment-0003.png From bdfoster at endigotech.com Fri Mar 23 20:33:29 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 23 Mar 2012 13:33:29 -0400 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: Message-ID: Is this on virtualized or real hardware? -BDF On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde wrote: > I have run a "timer_test" in a dedicated FS server and I see strange > result: it's normal? > > Stephen > > > freeswitch at internal> timer_test 20 40 > Avg: 19.866ms Total Time: 795.880ms > > 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: > samplecount after init: 1 > 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: > samplecount after first step: 2 > 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 > sleep 20 19568 > 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 > sleep 20 38231 > 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 > sleep 20 18847 > 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 > sleep 20 13982 > 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 > sleep 20 34793 > 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 > sleep 20 2 > 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 > sleep 20 23166 > 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 > sleep 20 16957 > 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 > sleep 20 17643 > 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 > sleep 20 18786 > 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 > sleep 20 25100 > 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 > sleep 20 18552 > 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 > sleep 20 18815 > 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 > sleep 20 19464 > 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 > sleep 20 22012 > 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 > sleep 20 13980 > 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 > sleep 20 19065 > 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 > sleep 20 39585 > 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 > sleep 20 2 > 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 > sleep 20 26255 > 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 > sleep 20 17872 > 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 > sleep 20 32191 > 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 > sleep 20 22634 > 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 > sleep 20 15483 > 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 > sleep 20 22813 > 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 > sleep 20 17099 > 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 > sleep 20 1 > 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 > sleep 20 29108 > 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 > sleep 20 11492 > 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 > sleep 20 20855 > 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 > sleep 20 32579 > 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 > sleep 20 18173 > 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 > sleep 20 22666 > 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 > sleep 20 23792 > 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 > sleep 20 26158 > 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 > sleep 20 13080 > 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 > sleep 20 24609 > 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 > sleep 20 1 > 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 > sleep 20 19413 > 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 > sleep 20 19820 > freeswitch at internal> status > UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, > 696 microseconds > FreeSWITCH is ready > 955611 session(s) since startup > 2192 session(s) 0/50 > 6000 session(s) max > min idle cpu 0.00/74.00 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/b92a5139/attachment.html From bdfoster at endigotech.com Fri Mar 23 20:39:43 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 23 Mar 2012 13:39:43 -0400 Subject: [Freeswitch-users] Grandstream 4104 In-Reply-To: <4F6C46B0.7080805@elios.net> References: <4F6C46B0.7080805@elios.net> Message-ID: Accursio, Welcome to the FreeSWITCH Community! Can you get us a siptrace? fs_cli sofia global siptrace on ...then pastebin the failing call to http://pastebin.freeswitch.org Thanks! -BDF On Fri, Mar 23, 2012 at 5:47 AM, a.avona wrote: > Hi, all > i have a GrandStream 4104 perfectly working (both incoming and outgoing > calls work great) with an asterisk pbx. > We are trying to change the asterisk with freeswitch, but i'm > exeperiencing problems in configuring freeswitch for outgoing calls, > incoming calls works well. > this is my configuration and what i obtain in fs_cli consolle > > Can someone tell me where i'm wrong? > > in sip_profile/internal i created a file 00_to_pstn.xml this way > > > > > > > > > > > > > > > in dialplan/default i created a file 00_to_pstn.xml this way > > > > > data="effective_caller_id_number=0321234567"/> > data="effective_caller_id_name=ThisIsMyCompany"/> > > > data="sofia/internal/gxw4104-fxo1/$1 at 192.168.0.3:5060"/> > > > > > > > > _________________________________________________________________ > > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:7559 IP 192.168.0.200 > Rejected by acl "domains". Falling back to Digest auth. > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:7559 IP 192.168.0.200 > Rejected by acl "domains". Falling back to Digest auth. > 2012-03-23 10:38:43.937574 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/1000 at 192.168.0.2 [fb33155c-74cb-11e1-9063-fb6a5f4599fb] > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:5526 Channel > sofia/internal/1000 at 192.168.0.2 entering state [received][100] > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:5537 Remote SDP: > v=0 > o=- 935122583 0 IN IP4 192.168.0.200 > s=SIPPER for PhonerLite > c=IN IP4 192.168.0.200 > t=0 0 > m=audio 5062 RTP/AVP 8 3 0 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:2991 Set Codec > sofia/internal/1000 at 192.168.0.2 PCMA/8000 20 ms 160 samples 64000 bits > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_codec.c:111 > sofia/internal/1000 at 192.168.0.2 Original read codec set to PCMA:8 > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4995 Set 2833 dtmf > send/recv payload to 101 > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:5749 > (sofia/internal/1000 at 192.168.0.2) State Change CS_NEW -> CS_INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1000 at 192.168.0.2) State INIT > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:85 > sofia/internal/1000 at 192.168.0.2 SOFIA INIT > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:125 > (sofia/internal/1000 at 192.168.0.2) State Change CS_INIT -> CS_ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1000 at 192.168.0.2) State INIT going to sleep > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1886 > (sofia/internal/1000 at 192.168.0.2) Callstate Change DOWN -> RINGING > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1000 at 192.168.0.2) State ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:148 > sofia/internal/1000 at 192.168.0.2 SOFIA ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/1000 at 192.168.0.2 Standard ROUTING > 2012-03-23 10:38:43.937574 [INFO] mod_dialplan_xml.c:485 Processing 1000 > <1000>->339XXXXXXX in context default > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unloop] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->tod_example] > continue=true > Dialplan: sofia/internal/1000 at 192.168.0.2 Date/Time Match (PASS) > [tod_example] break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(open=true) > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->holiday_example] continue=true > Dialplan: sofia/internal/1000 at 192.168.0.2 Date/TimeMatch (FAIL) > [holiday_example] break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->global-intercept] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [global-intercept] destination_number(339XXXXXXX) =~ /^886$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group-intercept] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group-intercept] > destination_number(339XXXXXXX) =~ /^\*8$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->intercept-ext] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [intercept-ext] > destination_number(339XXXXXXX) =~ /^\*\*(\d+)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->redial] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [redial] > destination_number(339XXXXXXX) =~ /^(redial|870)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->global] > continue=true > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] > ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] > ${sip_has_crypto}() =~ > /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never > Dialplan: sofia/internal/1000 at 192.168.0.2 Absolute Condition [global] > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->snom-demo-2] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-2] > destination_number(339XXXXXXX) =~ /^9001$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->snom-demo-1] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-1] > destination_number(339XXXXXXX) =~ /^9000$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] > destination_number(339XXXXXXX) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] > destination_number(339XXXXXXX) =~ /^779$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->call_return] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call_return] > destination_number(339XXXXXXX) =~ /^\*69$|^869$|^lcr$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->del-group] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [del-group] > destination_number(339XXXXXXX) =~ /^80(\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->add-group] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [add-group] > destination_number(339XXXXXXX) =~ /^81(\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->call-group-simo] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call-group-simo] > destination_number(339XXXXXXX) =~ /^82(\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->call-group-order] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [call-group-order] destination_number(339XXXXXXX) =~ /^83(\d{2})$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->extension-intercom] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [extension-intercom] destination_number(339XXXXXXX) =~ > /^8(10[01][0-9])$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->Local_Extension] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [Local_Extension] > destination_number(339XXXXXXX) =~ /^(10[01][0-9])$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->Local_Extension_Skinny] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [Local_Extension_Skinny] destination_number(339XXXXXXX) =~ > /^(11[01][0-9])$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group_dial_sales] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [group_dial_sales] destination_number(339XXXXXXX) =~ /^2000$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group_dial_support] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [group_dial_support] destination_number(339XXXXXXX) =~ /^2001$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group_dial_billing] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [group_dial_billing] destination_number(339XXXXXXX) =~ /^2002$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->operator] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [operator] > destination_number(339XXXXXXX) =~ /^(operator|0)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->vmain] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [vmain] > destination_number(339XXXXXXX) =~ /^vmain$|^4000$|^\*98$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->sip_uri] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [sip_uri] > destination_number(339XXXXXXX) =~ /^sip:(.*)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->nb_conferences] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [nb_conferences] > destination_number(339XXXXXXX) =~ /^(30\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->wb_conferences] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wb_conferences] > destination_number(339XXXXXXX) =~ /^(31\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->uwb_conferences] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [uwb_conferences] > destination_number(339XXXXXXX) =~ /^(32\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->cdquality_conferences] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [cdquality_conferences] destination_number(339XXXXXXX) =~ /^(33\d{2})$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->freeswitch_public_conf_via_sip] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [freeswitch_public_conf_via_sip] destination_number(339XXXXXXX) =~ > /^9(888|8888|1616|3232)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->mad_boss_intercom] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [mad_boss_intercom] destination_number(339XXXXXXX) =~ /^0911$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->mad_boss_intercom] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [mad_boss_intercom] destination_number(339XXXXXXX) =~ /^0912$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->mad_boss] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [mad_boss] > destination_number(339XXXXXXX) =~ /^0913$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ivr_demo] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ivr_demo] > destination_number(339XXXXXXX) =~ /^5000$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->dynamic_conference] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [dynamic_conference] destination_number(339XXXXXXX) =~ /^5001$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->rtp_multicast_page] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [rtp_multicast_page] destination_number(339XXXXXXX) =~ > /^pagegroup$|^7243$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] > destination_number(339XXXXXXX) =~ /^5900$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] > destination_number(339XXXXXXX) =~ /^5901$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->valet_park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] > destination_number(339XXXXXXX) =~ /^(6000)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->valet_park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] > destination_number(339XXXXXXX) =~ /^(60\d[1-9])$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] > destination_number(339XXXXXXX) =~ /park\+(\d+)/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] > destination_number(339XXXXXXX) =~ /^parking$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] > destination_number(339XXXXXXX) =~ /callpark/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] > destination_number(339XXXXXXX) =~ /pickup/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->wait] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wait] > destination_number(339XXXXXXX) =~ /^wait$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->fax_receive] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_receive] > destination_number(339XXXXXXX) =~ /^9178$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->fax_transmit] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_transmit] > destination_number(339XXXXXXX) =~ /^9179$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_180] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_180] > destination_number(339XXXXXXX) =~ /^9180$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_183_uk_ring] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_183_uk_ring] destination_number(339XXXXXXX) =~ /^9181$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_183_music_ring] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_183_music_ring] destination_number(339XXXXXXX) =~ /^9182$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_post_answer_uk_ring] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_post_answer_uk_ring] destination_number(339XXXXXXX) =~ > /^9183$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_post_answer_music] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_post_answer_music] destination_number(339XXXXXXX) =~ /^9184$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ClueCon] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ClueCon] > destination_number(339XXXXXXX) =~ /^9191$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->show_info] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [show_info] > destination_number(339XXXXXXX) =~ /^9192$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->video_record] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_record] > destination_number(339XXXXXXX) =~ /^9193$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->video_playback] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_playback] > destination_number(339XXXXXXX) =~ /^9194$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->delay_echo] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [delay_echo] > destination_number(339XXXXXXX) =~ /^9195$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->echo] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [echo] > destination_number(339XXXXXXX) =~ /^9196$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->milliwatt] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [milliwatt] > destination_number(339XXXXXXX) =~ /^9197$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->tone_stream] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [tone_stream] > destination_number(339XXXXXXX) =~ /^9198$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->zrtp_enrollement] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [zrtp_enrollement] destination_number(339XXXXXXX) =~ /^9787$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->hold_music] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [hold_music] > destination_number(339XXXXXXX) =~ /^9664$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->from_pstn] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [from_pstn] > destination_number(339XXXXXXX) =~ /^0000$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->101] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [101] > destination_number(339XXXXXXX) =~ /^101$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->pizza_demo] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [pizza_demo] > destination_number(339XXXXXXX) =~ /^(pizza|74992)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->gxw4104-fxo-local] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) > [gxw4104-fxo-local] ${toll_allow}(domestic,international,local) =~ > /local/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) > [gxw4104-fxo-local] destination_number(339XXXXXXX) =~ /^(\d{6,})$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > set(effective_caller_id_number=0321234567) > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > set(effective_caller_id_name=ThisIsMyCompany) > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > set(ignore_early_media=ring_ready) > Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(ringback=${us-ring}) > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > bridge(sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/1000 at 192.168.0.2) State Change CS_ROUTING -> CS_EXECUTE > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1000 at 192.168.0.2) State ROUTING going to sleep > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_EXECUTE > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/1000 at 192.168.0.2) State EXECUTE > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:241 > sofia/internal/1000 at 192.168.0.2 SOFIA EXECUTE > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/1000 at 192.168.0.2 Standard EXECUTE > EXECUTE sofia/internal/1000 at 192.168.0.2 set(open=true) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET [open]=[true] > EXECUTE sofia/internal/1000 at 192.168.0.2 > hash(insert/192.168.0.2-spymap/1000/fb33155c-74cb-11e1-9063-fb6a5f4599fb) > EXECUTE sofia/internal/1000 at 192.168.0.2 > hash(insert/192.168.0.2-last_dial/1000/339XXXXXXX) > EXECUTE sofia/internal/1000 at 192.168.0.2 > > hash(insert/192.168.0.2-last_dial/global/fb33155c-74cb-11e1-9063-fb6a5f4599fb) > EXECUTE sofia/internal/1000 at 192.168.0.2 export(RFC2822_DATE=Fri, 23 Mar > 2012 10:38:43 +0100) > 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1093 EXPORT > (export_vars) [RFC2822_DATE]=[Fri, 23 Mar 2012 10:38:43 +0100] > EXECUTE sofia/internal/1000 at 192.168.0.2 > set(effective_caller_id_number=0321234567) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET > [effective_caller_id_number]=[0321234567] > EXECUTE sofia/internal/1000 at 192.168.0.2 > set(effective_caller_id_name=ThisIsMyCompany) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET > [effective_caller_id_name]=[ThisIsMyCompany] > EXECUTE sofia/internal/1000 at 192.168.0.2 set(ignore_early_media=ring_ready) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET [ignore_early_media]=[ring_ready] > EXECUTE sofia/internal/1000 at 192.168.0.2 set(ringback=%(2000,4000,440,480)) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET [ringback]=[%(2000,4000,440,480)] > EXECUTE sofia/internal/1000 at 192.168.0.2 > bridge(sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) > 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1047 > sofia/internal/1000 at 192.168.0.2 EXPORTING[export_vars] > [RFC2822_DATE]=[Fri, 23 Mar 2012 10:38:43 +0100] to event > 2012-03-23 10:38:43.937574 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2012-03-23 10:38:43.937574 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [fb34cf5a-74cb-11e1-9068-fb6a5f4599fb] > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:4691 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change > CS_NEW -> CS_INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State > Change CS_INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State INIT > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:85 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 SOFIA INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:125 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change > CS_INIT -> CS_ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State INIT > going to sleep > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State > Change CS_ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1886 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Callstate > Change DOWN -> RINGING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State ROUTING > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:148 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 SOFIA ROUTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change > CS_ROUTING -> CS_CONSUME_MEDIA > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State ROUTING > going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State > Change CS_CONSUME_MEDIA > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State > CONSUME_MEDIA > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State > CONSUME_MEDIA going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] sofia.c:5526 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 entering state > [calling][0] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] sofia.c:5526 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 entering state > [terminated][403] > 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2848 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Callstate > Change RINGING -> HANGUP > 2012-03-23 10:38:43.957579 [NOTICE] sofia.c:6293 Hangup > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [CS_CONSUME_MEDIA] [CALL_REJECTED] > 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [KILL] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State > Change CS_HANGUP > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State HANGUP > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 hanging up, > cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 Standard HANGUP, > cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State HANGUP > going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change > CS_HANGUP -> CS_REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State > Change CS_REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 Standard > REPORTING, cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State > REPORTING going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change > CS_REPORTING -> CS_DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_ivr_originate.c:3364 Originate > Resulted in Error Cause: 21 [CALL_REJECTED] > 2012-03-23 10:38:43.957579 [INFO] mod_dptools.c:2922 Originate Failed. > Cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2848 > (sofia/internal/1000 at 192.168.0.2) Callstate Change RINGING -> HANGUP > 2012-03-23 10:38:43.957579 [NOTICE] mod_dptools.c:3041 Hangup > sofia/internal/1000 at 192.168.0.2 [CS_EXECUTE] [CALL_REJECTED] > 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/1000 at 192.168.0.2 [KILL] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:2285 > sofia/internal/1000 at 192.168.0.2 skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/1000 at 192.168.0.2) State EXECUTE going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_HANGUP > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1000 at 192.168.0.2) State HANGUP > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:463 > sofia/internal/1000 at 192.168.0.2 Overriding SIP cause 603 with 403 from > the other leg > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/1000 at 192.168.0.2 hanging up, cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:534 Responding to INVITE > with: 403 > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/1000 at 192.168.0.2 Standard HANGUP, cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1000 at 192.168.0.2) State HANGUP going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/1000 at 192.168.0.2) State Change CS_HANGUP -> CS_REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1000 at 192.168.0.2) State REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/1000 at 192.168.0.2 Standard REPORTING, cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1000 at 192.168.0.2) State REPORTING going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/1000 at 192.168.0.2) State Change CS_REPORTING -> CS_DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1380 Session 13 > (sofia/internal/1000 at 192.168.0.2) Locked, Waiting on external entities > 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1398 Session > 13 (sofia/internal/1000 at 192.168.0.2) Ended > 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/1000 at 192.168.0.2 [CS_DESTROY] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/1000 at 192.168.0.2) Callstate Change HANGUP -> DOWN > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1000 at 192.168.0.2) State DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:374 > sofia/internal/1000 at 192.168.0.2 SOFIA DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/1000 at 192.168.0.2 Standard DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1000 at 192.168.0.2) State DESTROY going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1380 Session 14 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Locked, > Waiting on external entities > 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1398 Session > 14 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Ended > 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060[CS_DESTROY] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Callstate > Change HANGUP -> DOWN > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State > Change CS_DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:374 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 SOFIA DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 Standard DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State DESTROY > going to sleep > freeswitch at internal> > > Thank's in advance > Regards > Accursio Avona > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/e01b0ee3/attachment-0001.html From msc at freeswitch.org Fri Mar 23 20:42:38 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Mar 2012 10:42:38 -0700 Subject: [Freeswitch-users] FreeSWITCH Project - Call For Assistance In-Reply-To: References: Message-ID: AAMOF... http://www.freeswitch.org/sites/all/files/fslog.php.txt Turns out we use GeSHi for pastebin.freeswitch.org. :) Take it and run... -MC On Fri, Mar 23, 2012 at 10:01 AM, William Suffill wrote: > Do we have some idea on how the highlighting on the pastebin is > defined? Would it be possible to port that to various editors and > other applications to be semi uniform? > > -- W > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/5ef8d618/attachment.html From krice at freeswitch.org Fri Mar 23 21:36:07 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 23 Mar 2012 13:36:07 -0500 Subject: [Freeswitch-users] Major Patch Coming this weekend Message-ID: Hey Guys, In prep for 1.2 I?ll be merging a patch that makes some adjustments to the Makefile and configure scripts, this means you?ll have to rebootstrap. This hasn?t been done yet, but heads up its coming.... I?ll follow up to the list once that merge has happened as you?ll know you whats up when your make breaks K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/38c6f8ba/attachment.html From wstephen80 at gmail.com Fri Mar 23 21:46:18 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 23 Mar 2012 19:46:18 +0100 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: Message-ID: Real hardware, a dedicate server with 2 Xeon X5670 (a total 24 core each one with 12Mb cache at 2.93GHz) that is running at 20% - 25% of load. OS is CentOS 5.7 64bit and FS is (git-0626c89 2012-02-29 14-45-39 -0600) On Fri, Mar 23, 2012 at 6:33 PM, Brian Foster wrote: > Is this on virtualized or real hardware? > > -BDF > > On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde wrote: > >> I have run a "timer_test" in a dedicated FS server and I see strange >> result: it's normal? >> >> Stephen >> >> >> freeswitch at internal> timer_test 20 40 >> Avg: 19.866ms Total Time: 795.880ms >> >> 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: >> samplecount after init: 1 >> 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: >> samplecount after first step: 2 >> 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 >> sleep 20 19568 >> 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 >> sleep 20 38231 >> 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 >> sleep 20 18847 >> 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 >> sleep 20 13982 >> 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 >> sleep 20 34793 >> 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 >> sleep 20 2 >> 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 >> sleep 20 23166 >> 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 >> sleep 20 16957 >> 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 >> sleep 20 17643 >> 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 >> sleep 20 18786 >> 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 >> sleep 20 25100 >> 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 >> sleep 20 18552 >> 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 >> sleep 20 18815 >> 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 >> sleep 20 19464 >> 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 >> sleep 20 22012 >> 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 >> sleep 20 13980 >> 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 >> sleep 20 19065 >> 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 >> sleep 20 39585 >> 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 >> sleep 20 2 >> 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 >> sleep 20 26255 >> 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 >> sleep 20 17872 >> 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 >> sleep 20 32191 >> 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 >> sleep 20 22634 >> 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 >> sleep 20 15483 >> 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 >> sleep 20 22813 >> 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 >> sleep 20 17099 >> 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 >> sleep 20 1 >> 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 >> sleep 20 29108 >> 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 >> sleep 20 11492 >> 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 >> sleep 20 20855 >> 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 >> sleep 20 32579 >> 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 >> sleep 20 18173 >> 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 >> sleep 20 22666 >> 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 >> sleep 20 23792 >> 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 >> sleep 20 26158 >> 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 >> sleep 20 13080 >> 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 >> sleep 20 24609 >> 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 >> sleep 20 1 >> 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 >> sleep 20 19413 >> 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 >> sleep 20 19820 >> freeswitch at internal> status >> UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, >> 696 microseconds >> FreeSWITCH is ready >> 955611 session(s) since startup >> 2192 session(s) 0/50 >> 6000 session(s) max >> min idle cpu 0.00/74.00 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/7b3ed9c2/attachment.html From acrow at integrafin.co.uk Fri Mar 23 23:28:36 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 23 Mar 2012 20:28:36 +0000 Subject: [Freeswitch-users] Major Patch Coming this weekend In-Reply-To: References: Message-ID: <4F6CDCF4.50204@integrafin.co.uk> Ken, Any chances this will fix the "the Debian way" build problems (ie you get your .debs but quite a few modules, such as mod_pocketsphinx, are missing)? Or is that down to someone else? Don't mind switching to CentOS or the like but it is really nice to have debs compiled on one machine *just work* on all your others. Unless there is a way to produce RPMs with full dependency information too.. Cheers Alex On 23/03/12 18:36, Ken Rice wrote: > Hey Guys, > > In prep for 1.2 I'll be merging a patch that makes some adjustments to > the Makefile and configure scripts, this means you'll have to > rebootstrap. This hasn't been done yet, but heads up its coming.... > > I'll follow up to the list once that merge has happened as you'll know > you whats up when your make breaks > > K > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/b85ef563/attachment-0001.html From me at nybras.com Fri Mar 23 23:50:30 2012 From: me at nybras.com (Anderson Arboleya) Date: Fri, 23 Mar 2012 17:50:30 -0300 Subject: [Freeswitch-users] Gateway Configuration (asteriskexampleprovided) In-Reply-To: <4F6B2533.7030002@anew.com.ve> References: <4F6B02D4.000005.01040@FLIGHTPC> <4F6B0732.000008.01040@FLIGHTPC> <4F6B2533.7030002@anew.com.ve> Message-ID: Damn! I forgot to do a stupid telnet. Thank you for shaking me, I was assuming the mistake was mine because I'm so new to this. There was a problem with the provider's firewall but now it seems to be solved, when I do a "sofia status" the gateway appears REGED. Now I'm trying to properly route the calls in my dialplan to run my LuaScript. So far I have: /usr/local/freeswitch/conf/dialplan/public/testing.xml ======================================== ======================================== But when I call my number I got this log: ======================================== 2012-03-23 20:36:23.465456 [NOTICE] switch_channel.c:926 New Channel sofia/external/1135678377 at 189.84.133.130:5060[dae908e0-7527-11e1-8295-15ac0b75d081] 2012-03-23 20:36:23.465456 [NOTICE] sofia.c:5813 Hangup sofia/external/ 1135678377 at 189.84.133.130:5060 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2012-03-23 20:36:23.465456 [NOTICE] switch_core_session.c:1400 Session 8 (sofia/external/1135678377 at 189.84.133.130:5060) Ended 2012-03-23 20:36:23.465456 [NOTICE] switch_core_session.c:1402 Close Channel sofia/external/1135678377 at 189.84.133.130:5060 [CS_DESTROY] ======================================== When I tested the same Dialplan in the default context, calling from a SIP client without passing through the external gateway it worked fine and my lua script is hit, answering the call. What am I doing wrong? Thanks again. Anderson Arboleya On Thu, Mar 22, 2012 at 10:12 AM, Saugort Dario Garcia Tovar < dgarcia at anew.com.ve> wrote: > > Hi, > > Have you tried the basic first? > 1. Do a ping from your FS to your provider IP to check conectivity > 2. Do a telnet: telne PROVIDER_IP PORT and see if it open conection > 3. Have you a firewall installed? > 4. Have you FS behind a router doing NAT or it has network interface conected directly to internet? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/8a4aaca9/attachment.html From v.wend at hospital-berlin.de Sat Mar 24 00:32:59 2012 From: v.wend at hospital-berlin.de (Volker Wend) Date: Fri, 23 Mar 2012 22:32:59 +0100 Subject: [Freeswitch-users] Language Files Message-ID: Hi, where can I find german language soundfiles fort the voicemail module ? Br Volker -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/35d11803/attachment.html From krice at freeswitch.org Sat Mar 24 01:24:52 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 23 Mar 2012 17:24:52 -0500 Subject: [Freeswitch-users] Push possibly breaking things Message-ID: Ok that Push I meantioned is going to happen this evening... If you get breakage on make current or make after doing a pull, rebootstrap and reconfigure. K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/edf83935/attachment.html From bdfoster at endigotech.com Sat Mar 24 01:56:39 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 23 Mar 2012 18:56:39 -0400 Subject: [Freeswitch-users] Major Patch Coming this weekend In-Reply-To: <4F6CDCF4.50204@integrafin.co.uk> References: <4F6CDCF4.50204@integrafin.co.uk> Message-ID: Alex, It looks like this is when you are building from source. I don't think it has anything to do with the .deb's or any rpm's. As far as I know, the packaging stuff isn't stable yet (please correct me if I'm wrong, I'm not exactly in the loop). Please test them, but realize that they are incomplete. -BDF On Mar 23, 2012 4:30 PM, "Alex Crow" wrote: > Ken, > > Any chances this will fix the "the Debian way" build problems (ie you get > your .debs but quite a few modules, such as mod_pocketsphinx, are missing)? > > Or is that down to someone else? Don't mind switching to CentOS or the > like but it is really nice to have debs compiled on one machine *just work* > on all your others. Unless there is a way to produce RPMs with full > dependency information too.. > > Cheers > > Alex > > On 23/03/12 18:36, Ken Rice wrote: > > Hey Guys, > > In prep for 1.2 I?ll be merging a patch that makes some adjustments to the > Makefile and configure scripts, this means you?ll have to rebootstrap. This > hasn?t been done yet, but heads up its coming.... > > I?ll follow up to the list once that merge has happened as you?ll know you > whats up when your make breaks > > K > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/f8765a69/attachment.html From curriegrad2004 at gmail.com Sat Mar 24 02:00:49 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 23 Mar 2012 16:00:49 -0700 Subject: [Freeswitch-users] FreeSWITCH Installer for Debian In-Reply-To: References: Message-ID: One comment: building FS as root isn't something that's attractive nor appealing to me... On Fri, Mar 23, 2012 at 10:12 AM, Brian Foster wrote: > Whoops. > > The new version (and always the latest) can be downloaded from here: > http://files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_latest.sh > > > On Fri, Mar 23, 2012 at 1:07 PM, Brian Foster > wrote: >> >> Version 0.0.3 was released today: >> >> ?Changelog >> >> ?================================================================================= >> ?0.0.3 23/Mar/2012 1235 UCT >> >> ? ? ? ?Bug Fixes >> ? ? ? ?- Changed script shell from bash to sh for compatability >> ? ? ? ?Features Added >> ? ? ? ?-Made FS git address, FS user, FS group, ?installed packages, FS >> sounds >> ? ? ? ?install, FS MOH install, and base dir for source folder into >> variables >> ? ? ? ?- Option added to update/upgrade system (true/false variable) >> ? ? ? ?- Option to download/install init script >> >> Version 0.1.0 will be released sometime within the next week. >> >> -BDF >> >> On Thu, Mar 22, 2012 at 6:21 PM, Gabriel Gunderson wrote: >>> >>> On Thu, Mar 22, 2012 at 3:44 PM, Zenny wrote: >>> > Glad to find build_opensips.sh also in the same directory. >>> >>> If you want to save the trouble of building, there are also some >>> pre-built debs for both FreeSWITCH and OpenSIPS: >>> >>> https://parseltone.org/files/debs/ubuntu-11.4/ >>> >>> >>> Best, >>> Gabe >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From bdfoster at endigotech.com Sat Mar 24 02:05:57 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 23 Mar 2012 19:05:57 -0400 Subject: [Freeswitch-users] FreeSWITCH Installer for Debian In-Reply-To: References: Message-ID: A fix for that is in the works. -BDF On Mar 23, 2012 7:01 PM, "curriegrad2004" wrote: > One comment: building FS as root isn't something that's attractive nor > appealing to me... > > On Fri, Mar 23, 2012 at 10:12 AM, Brian Foster > wrote: > > Whoops. > > > > The new version (and always the latest) can be downloaded from here: > > > http://files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_latest.sh > > > > > > On Fri, Mar 23, 2012 at 1:07 PM, Brian Foster > > wrote: > >> > >> Version 0.0.3 was released today: > >> > >> Changelog > >> > >> > ================================================================================= > >> 0.0.3 23/Mar/2012 1235 UCT > >> > >> Bug Fixes > >> - Changed script shell from bash to sh for compatability > >> Features Added > >> -Made FS git address, FS user, FS group, installed packages, FS > >> sounds > >> install, FS MOH install, and base dir for source folder into > >> variables > >> - Option added to update/upgrade system (true/false variable) > >> - Option to download/install init script > >> > >> Version 0.1.0 will be released sometime within the next week. > >> > >> -BDF > >> > >> On Thu, Mar 22, 2012 at 6:21 PM, Gabriel Gunderson > wrote: > >>> > >>> On Thu, Mar 22, 2012 at 3:44 PM, Zenny wrote: > >>> > Glad to find build_opensips.sh also in the same directory. > >>> > >>> If you want to save the trouble of building, there are also some > >>> pre-built debs for both FreeSWITCH and OpenSIPS: > >>> > >>> https://parseltone.org/files/debs/ubuntu-11.4/ > >>> > >>> > >>> Best, > >>> Gabe > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Brian D. Foster > >> Endigo Computer LLC > >> Email: bdfoster at endigotech.com > >> Phone: 317-800-7876 > >> Indianapolis, Indiana, USA > >> > >> This message contains confidential information and is intended for those > >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. > If > >> you are not the intended recipient you are notified that disclosing, > >> copying, distributing or taking any action in reliance on the contents > of > >> this information is strictly prohibited. E-mail transmission cannot be > >> guaranteed to be secure or error-free as information could be > intercepted, > >> corrupted, lost, destroyed, arrive late or incomplete, or contain > viruses. > >> The sender therefore does not accept liability for any errors or > omissions > >> in the contents of this message, which arise as a result of e-mail > >> transmission. If verification is required please request a hard-copy > >> version. > >> > > > > > > > > -- > > Brian D. Foster > > Endigo Computer LLC > > Email: bdfoster at endigotech.com > > Phone: 317-800-7876 > > Indianapolis, Indiana, USA > > > > This message contains confidential information and is intended for those > > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. > If > > you are not the intended recipient you are notified that disclosing, > > copying, distributing or taking any action in reliance on the contents of > > this information is strictly prohibited. E-mail transmission cannot be > > guaranteed to be secure or error-free as information could be > intercepted, > > corrupted, lost, destroyed, arrive late or incomplete, or contain > viruses. > > The sender therefore does not accept liability for any errors or > omissions > > in the contents of this message, which arise as a result of e-mail > > transmission. If verification is required please request a hard-copy > > version. > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/ae780535/attachment.html From manavid at gmail.com Sat Mar 24 02:15:52 2012 From: manavid at gmail.com (Mohammad Amin Navid) Date: Fri, 23 Mar 2012 16:15:52 -0700 Subject: [Freeswitch-users] Major Patch Coming this weekend In-Reply-To: <4F6CDCF4.50204@integrafin.co.uk> References: <4F6CDCF4.50204@integrafin.co.uk> Message-ID: I use fpm https://github.com/jordansissel/fpm for building the deb packages and I have been very happy with it. On Mar 23, 2012, at 1:28 PM, Alex Crow wrote: > Ken, > > Any chances this will fix the "the Debian way" build problems (ie you get your .debs but quite a few modules, such as mod_pocketsphinx, are missing)? > > Or is that down to someone else? Don't mind switching to CentOS or the like but it is really nice to have debs compiled on one machine *just work* on all your others. Unless there is a way to produce RPMs with full dependency information too.. > > Cheers > > Alex > > On 23/03/12 18:36, Ken Rice wrote: >> >> Hey Guys, >> >> In prep for 1.2 I?ll be merging a patch that makes some adjustments to the Makefile and configure scripts, this means you?ll have to rebootstrap. This hasn?t been done yet, but heads up its coming.... >> >> I?ll follow up to the list once that merge has happened as you?ll know you whats up when your make breaks >> >> K >> -- >> This message has been scanned for viruses and >> dangerous content by MailScanner, and is >> believed to be clean. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/8117c90b/attachment.html From curriegrad2004 at gmail.com Sat Mar 24 02:22:59 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 23 Mar 2012 16:22:59 -0700 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: References: Message-ID: Have you checked for any faults on your TDM link on your side? On Fri, Mar 23, 2012 at 10:32 AM, Brian Foster wrote: > Please update to latest git. > > -BDF > > On Fri, Mar 23, 2012 at 7:19 AM, Daniel Knaggs < > Daniel.Knaggs at realitysolutions.co.uk> wrote: > >> Hello all,**** >> >> ** ** >> >> Got a bit of a strange one, we appear to be getting DTMF tones on >> incoming calls when the caller hasn?t even pressed any keys.**** >> >> ** ** >> >> It normally happens with 10 seconds or so after the call has been >> answered.**** >> >> ** ** >> >> ** ** >> >> Here is the log of it happening earlier: -**** >> >> ** ** >> >> 2012-03-23 10:38:16.852654 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing >> DTMF A (debug = 0)**** >> >> 2012-03-23 10:38:16.852654 [DEBUG] mod_freetdm.c:799 Queuing DTMF [A] in >> channel FreeTDM/1:2/000 device 1:2**** >> >> 2012-03-23 10:38:16.915653 [DEBUG] switch_rtp.c:2420 Send start packet >> for [A] ts=49440 dur=160/160/2000 seq=46803 lw=49440**** >> >> 2012-03-23 10:38:16.936653 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=320/320/2000 seq=46804 lw=49600**** >> >> 2012-03-23 10:38:16.957652 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=480/480/2000 seq=46805 lw=49760**** >> >> 2012-03-23 10:38:16.978652 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=640/640/2000 seq=46806 lw=49920**** >> >> 2012-03-23 10:38:16.999652 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=800/800/2000 seq=46807 lw=50080**** >> >> 2012-03-23 10:38:17.020651 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=960/960/2000 seq=46808 lw=50240**** >> >> 2012-03-23 10:38:17.041651 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=1120/1120/2000 seq=46809 lw=50400**** >> >> 2012-03-23 10:38:17.062651 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=1280/1280/2000 seq=46810 lw=50560**** >> >> 2012-03-23 10:38:17.083650 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=1440/1440/2000 seq=46811 lw=50720**** >> >> 2012-03-23 10:38:17.104650 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=1600/1600/2000 seq=46812 lw=50880**** >> >> 2012-03-23 10:38:17.125650 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=1760/1760/2000 seq=46813 lw=51040**** >> >> 2012-03-23 10:38:17.146650 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=1920/1920/2000 seq=46814 lw=51200**** >> >> 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for >> [A] ts=49440 dur=2080/2080/2000 seq=46815 lw=51200**** >> >> 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for >> [A] ts=49440 dur=2080/2080/2000 seq=46816 lw=51200**** >> >> 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for >> [A] ts=49440 dur=2080/2080/2000 seq=46817 lw=51200**** >> >> 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2271 Queue digit delay of >> 40ms**** >> >> 2012-03-23 10:38:18.070638 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing >> DTMF B (debug = 0)**** >> >> 2012-03-23 10:38:18.070638 [DEBUG] mod_freetdm.c:799 Queuing DTMF [B] in >> channel FreeTDM/1:2/000 device 1:2**** >> >> 2012-03-23 10:38:18.133637 [DEBUG] switch_rtp.c:2420 Send start packet >> for [B] ts=59040 dur=160/160/2000 seq=46864 lw=59040**** >> >> 2012-03-23 10:38:18.154637 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=320/320/2000 seq=46865 lw=59200**** >> >> 2012-03-23 10:38:18.175637 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=480/480/2000 seq=46866 lw=59360**** >> >> 2012-03-23 10:38:18.196636 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=640/640/2000 seq=46867 lw=59520**** >> >> 2012-03-23 10:38:18.217636 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=800/800/2000 seq=46868 lw=59680**** >> >> 2012-03-23 10:38:18.238636 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=960/960/2000 seq=46869 lw=59840**** >> >> 2012-03-23 10:38:18.259636 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=1120/1120/2000 seq=46870 lw=60000**** >> >> 2012-03-23 10:38:18.280636 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=1280/1280/2000 seq=46871 lw=60160**** >> >> 2012-03-23 10:38:18.301635 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=1440/1440/2000 seq=46872 lw=60320**** >> >> 2012-03-23 10:38:18.322635 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=1600/1600/2000 seq=46873 lw=60480**** >> >> 2012-03-23 10:38:18.343634 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=1760/1760/2000 seq=46874 lw=60640**** >> >> 2012-03-23 10:38:18.364634 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=1920/1920/2000 seq=46875 lw=60800**** >> >> 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for >> [B] ts=59040 dur=2080/2080/2000 seq=46876 lw=60800**** >> >> 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for >> [B] ts=59040 dur=2080/2080/2000 seq=46877 lw=60800**** >> >> 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for >> [B] ts=59040 dur=2080/2080/2000 seq=46878 lw=60800**** >> >> ** ** >> >> ** ** >> >> I?m very sure the caller isn?t dialling ?A? or ?B?!**** >> >> ** ** >> >> Running on ISDN via a TE121 card using E1 (euroisdn). Freeswitch version >> is ?FreeSWITCH Version 1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)?. >> **** >> >> ** ** >> >> ** ** >> >> Currently there is a ?bind_meta_app? in the config which binds a script >> on the B leg of the call which parks the call ? I haven?t tried turning >> this off yet to see if it?s this.**** >> >> ** ** >> >> ** ** >> >> Wondering if anyone has any ideas or has come across this before?**** >> >> ** ** >> >> ** ** >> >> Thanks in advance.**** >> >> Daniel Knaggs >> >> Software Developer >> Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, >> Kingston upon Hull, East Yorkshire, HU7 0AE >> Tel: 01482 828000 / Fax: 01482 373100 >> Daniel.Knaggs at realitysolutions.co.uk >> www.realitysolutions.co.uk >> ------------------------------ >> Sage Accredited Business Partner serving businesses in Yorkshire & >> Lincolnshire >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/87c0de43/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 22463 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/87c0de43/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 69075 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/87c0de43/attachment-0003.png From me at nybras.com Sat Mar 24 04:33:57 2012 From: me at nybras.com (Anderson Arboleya) Date: Fri, 23 Mar 2012 22:33:57 -0300 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION Message-ID: Hi, somebody knows what do I have to do in order to fix that hangup cause? Here's the complete log (with debug) of the inbound call. Spent my whole day trying to understand it.. without any luck. Thanks ============>>>>>>>>> 2012-03-24 01:18:37.865450 [NOTICE] switch_channel.c:926 New Channel sofia/external/1184049130 at 189.84.133.130:5060[4898b56c-754f-11e1-af7e-812af071ddc6] 2012-03-24 01:18:37.865450 [DEBUG] sofia.c:5532 Channel sofia/external/ 1184049130 at 189.84.133.130:5060 entering state [received][100] 2012-03-24 01:18:37.865450 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=VoipSwitch 6106 7106 IN IP4 189.84.133.130 s=VoipSIP i=Audio Session c=IN IP4 189.84.133.130 t=0 0 m=audio 6106 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[GSM:3:8000:20:13200] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-03-24 01:18:37.865450 [DEBUG] switch_channel.c:2848 (sofia/external/ 1184049130 at 189.84.133.130:5060) Callstate Change DOWN -> HANGUP 2012-03-24 01:18:37.865450 [NOTICE] sofia.c:5813 Hangup sofia/external/ 1184049130 at 189.84.133.130:5060 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2012-03-24 01:18:37.865450 [DEBUG] switch_channel.c:2871 Send signal sofia/external/1184049130 at 189.84.133.130:5060 [KILL] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change CS_HANGUP 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:602 (sofia/external/1184049130 at 189.84.133.130:5060) State HANGUP 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:469 Channel sofia/external/ 1184049130 at 189.84.133.130:5060 hanging up, cause: INCOMPATIBLE_DESTINATION 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:534 Responding to INVITE with: 488 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:47 sofia/external/1184049130 at 189.84.133.130:5060 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:602 (sofia/external/1184049130 at 189.84.133.130:5060) State HANGUP going to sleep 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:393 (sofia/external/1184049130 at 189.84.133.130:5060) State Change CS_HANGUP -> CS_REPORTING 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change CS_REPORTING 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:662 (sofia/external/1184049130 at 189.84.133.130:5060) State REPORTING 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:79 sofia/external/1184049130 at 189.84.133.130:5060 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:662 (sofia/external/1184049130 at 189.84.133.130:5060) State REPORTING going to sleep 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:387 (sofia/external/1184049130 at 189.84.133.130:5060) State Change CS_REPORTING -> CS_DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1382 Session 5 (sofia/external/1184049130 at 189.84.133.130:5060) Locked, Waiting on external entities 2012-03-24 01:18:37.865450 [NOTICE] switch_core_session.c:1400 Session 5 (sofia/external/1184049130 at 189.84.133.130:5060) Ended 2012-03-24 01:18:37.865450 [NOTICE] switch_core_session.c:1402 Close Channel sofia/external/1184049130 at 189.84.133.130:5060 [CS_DESTROY] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:491 (sofia/external/1184049130 at 189.84.133.130:5060) Callstate Change HANGUP -> DOWN 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:494 (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change CS_DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:504 (sofia/external/1184049130 at 189.84.133.130:5060) State DESTROY 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:374 sofia/external/ 1184049130 at 189.84.133.130:5060 SOFIA DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:86 sofia/external/1184049130 at 189.84.133.130:5060 Standard DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:504 (sofia/external/1184049130 at 189.84.133.130:5060) State DESTROY going to sleep <<<<<<<<<============ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/5bdfb647/attachment.html From fernandojdk at gmail.com Sat Mar 24 04:43:05 2012 From: fernandojdk at gmail.com (Fernando - NextBilling IP Solutions) Date: Fri, 23 Mar 2012 22:43:05 -0300 (Hora oficial do Brasil) Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION References: Message-ID: <4F6D26A9.000001.03216@FLIGHTPC> Its seem to be Codec. Your device sends codec G729 but sofia profile no have G729 enabled. Look at the comparisons: 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[GSM:3:8000:20:13200] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] Just enable codec G729 in vars.xml on param outgoing_codecs and global_codecs. ? Best regards, Importante: Esta mensagem, incluindo todo seu conte?do, cont?m informa??es confidenciais legalmente protegidas e destinadas a indiv?duo e prop?sito espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o mais breve poss?vel. As informa??es contidas nesta mensagem e em seu conte?do s?o de responsabilidade de seu autor, n?o representando necessariamente id?ias, opini?es, pensamentos ou qualquer forma de posicionamento por parte da NextBilling IP Solutions. P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." -------Original Message------- From: Anderson Arboleya Date: 23/03/2012 22:35:51 To: FreeSWITCH Users Help Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION Hi, somebody knows what do I have to do in order to fix that hangup cause? Here's the complete log (with debug) of the inbound call. Spent my whole day trying to understand it.. without any luck. Thanks ============>>>>>>>>> 2012-03-24 01:18:37.865450 [NOTICE] switch_channel.c:926 New Channel sofia/external/1184049130 at 189.84.133.130:5060 [4898b56c-754f-11e1-af7e-812af071ddc6] 2012-03-24 01:18:37.865450 [DEBUG] sofia.c:5532 Channel sofia/external/1184049130 at 189.84.133.130:5060 entering state [received][100] 2012-03-24 01:18:37.865450 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=VoipSwitch 6106 7106 IN IP4 189.84.133.130 s=VoipSIP i=Audio Session c=IN IP4 189.84.133.130 t=0 0 m=audio 6106 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[GSM:3:8000:20:13200] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-03-24 01:18:37.865450 [DEBUG] switch_channel.c:2848 (sofia/external/1184049130 at 189.84.133.130:5060) Callstate Change DOWN -> HANGUP 2012-03-24 01:18:37.865450 [NOTICE] sofia.c:5813 Hangup sofia/external/1184049130 at 189.84.133.130:5060 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2012-03-24 01:18:37.865450 [DEBUG] switch_channel.c:2871 Send signal sofia/external/1184049130 at 189.84.133.130:5060 [KILL] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change CS_HANGUP 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:602 (sofia/external/1184049130 at 189.84.133.130:5060) State HANGUP 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:469 Channel sofia/external/1184049130 at 189.84.133.130:5060 hanging up, cause: INCOMPATIBLE_DESTINATION 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:534 Responding to INVITE with: 488 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:47 sofia/external/1184049130 at 189.84.133.130:5060 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:602 (sofia/external/1184049130 at 189.84.133.130:5060) State HANGUP going to sleep 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:393 (sofia/external/1184049130 at 189.84.133.130:5060) State Change CS_HANGUP -> CS_REPORTING 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change CS_REPORTING 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:662 (sofia/external/1184049130 at 189.84.133.130:5060) State REPORTING 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:79 sofia/external/1184049130 at 189.84.133.130:5060 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:662 (sofia/external/1184049130 at 189.84.133.130:5060) State REPORTING going to sleep 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:387 (sofia/external/1184049130 at 189.84.133.130:5060) State Change CS_REPORTING -> CS_DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1382 Session 5 (sofia/external/1184049130 at 189.84.133.130:5060) Locked, Waiting on external entities 2012-03-24 01:18:37.865450 [NOTICE] switch_core_session.c:1400 Session 5 (sofia/external/1184049130 at 189.84.133.130:5060) Ended 2012-03-24 01:18:37.865450 [NOTICE] switch_core_session.c:1402 Close Channel sofia/external/1184049130 at 189.84.133.130:5060 [CS_DESTROY] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:491 (sofia/external/1184049130 at 189.84.133.130:5060) Callstate Change HANGUP -> DOWN 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:494 (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change CS_DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:504 (sofia/external/1184049130 at 189.84.133.130:5060) State DESTROY 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:374 sofia/external/1184049130 at 189.84.133.130:5060 SOFIA DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:86 sofia/external/1184049130 at 189.84.133.130:5060 Standard DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:504 (sofia/external/1184049130 at 189.84.133.130:5060) State DESTROY going to sleep <<<<<<<<<============ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/0aa2bd4f/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 20873 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/0aa2bd4f/attachment-0001.jpe From sos at sokhapkin.dyndns.org Sat Mar 24 04:45:49 2012 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 23 Mar 2012 21:45:49 -0400 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION In-Reply-To: References: Message-ID: <3944506.tN9pZ6Ijqo@sos> Install commercial g729 codec and the problem will go away. On Friday 23 March 2012 22:33:57 Anderson Arboleya wrote: > Hi, somebody knows what do I have to do in order to fix that hangup cause? > > Here's the complete log (with debug) of the inbound call. > > Spent my whole day trying to understand it.. without any luck. > > Thanks > > ============>>>>>>>>> > > 2012-03-24 01:18:37.865450 [NOTICE] switch_channel.c:926 New Channel > sofia/external/1184049130 at 189.84.133.130:5060[4898b56c-754f-11e1-af7e-812af0 > 71ddc6] 2012-03-24 01:18:37.865450 [DEBUG] sofia.c:5532 Channel > sofia/external/ 1184049130 at 189.84.133.130:5060 entering state > [received][100] > 2012-03-24 01:18:37.865450 [DEBUG] sofia.c:5543 Remote SDP: > v=0 > o=VoipSwitch 6106 7106 IN IP4 189.84.133.130 > s=VoipSIP > i=Audio Session > c=IN IP4 189.84.133.130 > t=0 0 > m=audio 6106 RTP/AVP 18 101 > a=rtpmap:18 G729/8000/1 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G722:9:8000:20:64000] > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000] > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[GSM:3:8000:20:13200] > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-03-24 01:18:37.865450 [DEBUG] switch_channel.c:2848 (sofia/external/ > 1184049130 at 189.84.133.130:5060) Callstate Change DOWN -> HANGUP > 2012-03-24 01:18:37.865450 [NOTICE] sofia.c:5813 Hangup sofia/external/ > 1184049130 at 189.84.133.130:5060 [CS_NEW] [INCOMPATIBLE_DESTINATION] > 2012-03-24 01:18:37.865450 [DEBUG] switch_channel.c:2871 Send signal > sofia/external/1184049130 at 189.84.133.130:5060 [KILL] > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal > sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change > CS_HANGUP > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:602 > (sofia/external/1184049130 at 189.84.133.130:5060) State HANGUP > 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:469 Channel sofia/external/ > 1184049130 at 189.84.133.130:5060 hanging up, cause: INCOMPATIBLE_DESTINATION > 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:534 Responding to INVITE > with: 488 > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:47 > sofia/external/1184049130 at 189.84.133.130:5060 Standard HANGUP, cause: > INCOMPATIBLE_DESTINATION > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:602 > (sofia/external/1184049130 at 189.84.133.130:5060) State HANGUP going to sleep > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:393 > (sofia/external/1184049130 at 189.84.133.130:5060) State Change CS_HANGUP -> > CS_REPORTING > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal > sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change > CS_REPORTING > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:662 > (sofia/external/1184049130 at 189.84.133.130:5060) State REPORTING > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:79 > sofia/external/1184049130 at 189.84.133.130:5060 Standard REPORTING, cause: > INCOMPATIBLE_DESTINATION > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:662 > (sofia/external/1184049130 at 189.84.133.130:5060) State REPORTING going to > sleep > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:387 > (sofia/external/1184049130 at 189.84.133.130:5060) State Change CS_REPORTING > -> CS_DESTROY > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal > sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1382 Session 5 > (sofia/external/1184049130 at 189.84.133.130:5060) Locked, Waiting on external > entities > 2012-03-24 01:18:37.865450 [NOTICE] switch_core_session.c:1400 Session 5 > (sofia/external/1184049130 at 189.84.133.130:5060) Ended > 2012-03-24 01:18:37.865450 [NOTICE] switch_core_session.c:1402 Close > Channel sofia/external/1184049130 at 189.84.133.130:5060 [CS_DESTROY] > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:491 > (sofia/external/1184049130 at 189.84.133.130:5060) Callstate Change HANGUP -> > DOWN > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:494 > (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change > CS_DESTROY > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:504 > (sofia/external/1184049130 at 189.84.133.130:5060) State DESTROY > 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:374 sofia/external/ > 1184049130 at 189.84.133.130:5060 SOFIA DESTROY > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:86 > sofia/external/1184049130 at 189.84.133.130:5060 Standard DESTROY > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:504 > (sofia/external/1184049130 at 189.84.133.130:5060) State DESTROY going to sleep > > <<<<<<<<<============ From brian at freeswitch.org Sat Mar 24 04:48:17 2012 From: brian at freeswitch.org (Brian West) Date: Fri, 23 Mar 2012 20:48:17 -0500 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION In-Reply-To: References: Message-ID: <6328884918316840960@unknownmsgid> You must now rescan the profile! Just adding it to vars.xml does nothing. sofia profile xxxx rescan (after reloadxml) /b Sent from my iPad On Mar 23, 2012, at 8:36 PM, Anderson Arboleya wrote: Hi, somebody knows what do I have to do in order to fix that hangup cause? Here's the complete log (with debug) of the inbound call. Spent my whole day trying to understand it.. without any luck. Thanks ============>>>>>>>>> 2012-03-24 01:18:37.865450 [NOTICE] switch_channel.c:926 New Channel sofia/external/1184049130 at 189.84.133.130:5060[4898b56c-754f-11e1-af7e-812af071ddc6] 2012-03-24 01:18:37.865450 [DEBUG] sofia.c:5532 Channel sofia/external/ 1184049130 at 189.84.133.130:5060 entering state [received][100] 2012-03-24 01:18:37.865450 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=VoipSwitch 6106 7106 IN IP4 189.84.133.130 s=VoipSIP i=Audio Session c=IN IP4 189.84.133.130 t=0 0 m=audio 6106 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[GSM:3:8000:20:13200] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-03-24 01:18:37.865450 [DEBUG] switch_channel.c:2848 (sofia/external/ 1184049130 at 189.84.133.130:5060) Callstate Change DOWN -> HANGUP 2012-03-24 01:18:37.865450 [NOTICE] sofia.c:5813 Hangup sofia/external/ 1184049130 at 189.84.133.130:5060 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2012-03-24 01:18:37.865450 [DEBUG] switch_channel.c:2871 Send signal sofia/external/1184049130 at 189.84.133.130:5060 [KILL] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change CS_HANGUP 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:602 (sofia/external/1184049130 at 189.84.133.130:5060) State HANGUP 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:469 Channel sofia/external/ 1184049130 at 189.84.133.130:5060 hanging up, cause: INCOMPATIBLE_DESTINATION 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:534 Responding to INVITE with: 488 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:47 sofia/external/1184049130 at 189.84.133.130:5060 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:602 (sofia/external/1184049130 at 189.84.133.130:5060) State HANGUP going to sleep 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:393 (sofia/external/1184049130 at 189.84.133.130:5060) State Change CS_HANGUP -> CS_REPORTING 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change CS_REPORTING 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:662 (sofia/external/1184049130 at 189.84.133.130:5060) State REPORTING 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:79 sofia/external/1184049130 at 189.84.133.130:5060 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:662 (sofia/external/1184049130 at 189.84.133.130:5060) State REPORTING going to sleep 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:387 (sofia/external/1184049130 at 189.84.133.130:5060) State Change CS_REPORTING -> CS_DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1382 Session 5 (sofia/external/1184049130 at 189.84.133.130:5060) Locked, Waiting on external entities 2012-03-24 01:18:37.865450 [NOTICE] switch_core_session.c:1400 Session 5 (sofia/external/1184049130 at 189.84.133.130:5060) Ended 2012-03-24 01:18:37.865450 [NOTICE] switch_core_session.c:1402 Close Channel sofia/external/1184049130 at 189.84.133.130:5060 [CS_DESTROY] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:491 (sofia/external/1184049130 at 189.84.133.130:5060) Callstate Change HANGUP -> DOWN 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:494 (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change CS_DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:504 (sofia/external/1184049130 at 189.84.133.130:5060) State DESTROY 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:374 sofia/external/ 1184049130 at 189.84.133.130:5060 SOFIA DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:86 sofia/external/1184049130 at 189.84.133.130:5060 Standard DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:504 (sofia/external/1184049130 at 189.84.133.130:5060) State DESTROY going to sleep <<<<<<<<<============ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/e7deed6c/attachment.html From bdfoster at endigotech.com Sat Mar 24 04:55:38 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 23 Mar 2012 21:55:38 -0400 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION In-Reply-To: <3944506.tN9pZ6Ijqo@sos> References: <3944506.tN9pZ6Ijqo@sos> Message-ID: On Mar 23, 2012 9:46 PM, "Sergey Okhapkin" wrote: > > Install commercial g729 codec and the problem will go away. No, if it just passes through FS there is no need for a license. If you are doing transcoding, then yes, a license is required. -BDF > > On Friday 23 March 2012 22:33:57 Anderson Arboleya wrote: > > Hi, somebody knows what do I have to do in order to fix that hangup cause? > > > > Here's the complete log (with debug) of the inbound call. > > > > Spent my whole day trying to understand it.. without any luck. > > > > Thanks > > > > ============>>>>>>>>> > > > > 2012-03-24 01:18:37.865450 [NOTICE] switch_channel.c:926 New Channel > > sofia/external/1184049130 at 189.84.133.130:5060 [4898b56c-754f-11e1-af7e-812af0 > > 71ddc6] 2012-03-24 01:18:37.865450 [DEBUG] sofia.c:5532 Channel > > sofia/external/ 1184049130 at 189.84.133.130:5060 entering state > > [received][100] > > 2012-03-24 01:18:37.865450 [DEBUG] sofia.c:5543 Remote SDP: > > v=0 > > o=VoipSwitch 6106 7106 IN IP4 189.84.133.130 > > s=VoipSIP > > i=Audio Session > > c=IN IP4 189.84.133.130 > > t=0 0 > > m=audio 6106 RTP/AVP 18 101 > > a=rtpmap:18 G729/8000/1 > > a=fmtp:18 annexb=no > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > > [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > > [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > > [G729:18:8000:20:8000]/[G722:9:8000:20:64000] > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > > [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000] > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > > [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > > [G729:18:8000:20:8000]/[GSM:3:8000:20:13200] > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > > [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > > [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > > [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > > [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > > [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > > [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > > send/recv payload to 101 > > 2012-03-24 01:18:37.865450 [DEBUG] switch_channel.c:2848 (sofia/external/ > > 1184049130 at 189.84.133.130:5060) Callstate Change DOWN -> HANGUP > > 2012-03-24 01:18:37.865450 [NOTICE] sofia.c:5813 Hangup sofia/external/ > > 1184049130 at 189.84.133.130:5060 [CS_NEW] [INCOMPATIBLE_DESTINATION] > > 2012-03-24 01:18:37.865450 [DEBUG] switch_channel.c:2871 Send signal > > sofia/external/1184049130 at 189.84.133.130:5060 [KILL] > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal > > sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:362 > > (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change > > CS_HANGUP > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:602 > > (sofia/external/1184049130 at 189.84.133.130:5060) State HANGUP > > 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:469 Channel sofia/external/ > > 1184049130 at 189.84.133.130:5060 hanging up, cause: INCOMPATIBLE_DESTINATION > > 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:534 Responding to INVITE > > with: 488 > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:47 > > sofia/external/1184049130 at 189.84.133.130:5060 Standard HANGUP, cause: > > INCOMPATIBLE_DESTINATION > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:602 > > (sofia/external/1184049130 at 189.84.133.130:5060) State HANGUP going to sleep > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:393 > > (sofia/external/1184049130 at 189.84.133.130:5060) State Change CS_HANGUP -> > > CS_REPORTING > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal > > sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:362 > > (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change > > CS_REPORTING > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:662 > > (sofia/external/1184049130 at 189.84.133.130:5060) State REPORTING > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:79 > > sofia/external/1184049130 at 189.84.133.130:5060 Standard REPORTING, cause: > > INCOMPATIBLE_DESTINATION > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:662 > > (sofia/external/1184049130 at 189.84.133.130:5060) State REPORTING going to > > sleep > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:387 > > (sofia/external/1184049130 at 189.84.133.130:5060) State Change CS_REPORTING > > -> CS_DESTROY > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal > > sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1382 Session 5 > > (sofia/external/1184049130 at 189.84.133.130:5060) Locked, Waiting on external > > entities > > 2012-03-24 01:18:37.865450 [NOTICE] switch_core_session.c:1400 Session 5 > > (sofia/external/1184049130 at 189.84.133.130:5060) Ended > > 2012-03-24 01:18:37.865450 [NOTICE] switch_core_session.c:1402 Close > > Channel sofia/external/1184049130 at 189.84.133.130:5060 [CS_DESTROY] > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:491 > > (sofia/external/1184049130 at 189.84.133.130:5060) Callstate Change HANGUP -> > > DOWN > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:494 > > (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change > > CS_DESTROY > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:504 > > (sofia/external/1184049130 at 189.84.133.130:5060) State DESTROY > > 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:374 sofia/external/ > > 1184049130 at 189.84.133.130:5060 SOFIA DESTROY > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:86 > > sofia/external/1184049130 at 189.84.133.130:5060 Standard DESTROY > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:504 > > (sofia/external/1184049130 at 189.84.133.130:5060) State DESTROY going to sleep > > > > <<<<<<<<<============ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120323/209cf28b/attachment-0001.html From anton.jugatsu at gmail.com Sat Mar 24 07:02:09 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Sat, 24 Mar 2012 08:02:09 +0400 Subject: [Freeswitch-users] Major Patch Coming this weekend In-Reply-To: References: <4F6CDCF4.50204@integrafin.co.uk> Message-ID: Thanks for the link to great app. 24.03.2012 3:16 ???????????? "Mohammad Amin Navid" ???????: > I use fpm https://github.com/jordansissel/fpm for building the deb > packages and I have been very happy with it. > > On Mar 23, 2012, at 1:28 PM, Alex Crow wrote: > > Ken, > > Any chances this will fix the "the Debian way" build problems (ie you get > your .debs but quite a few modules, such as mod_pocketsphinx, are missing)? > > Or is that down to someone else? Don't mind switching to CentOS or the > like but it is really nice to have debs compiled on one machine *just work* > on all your others. Unless there is a way to produce RPMs with full > dependency information too.. > > Cheers > > Alex > > On 23/03/12 18:36, Ken Rice wrote: > > Hey Guys, > > In prep for 1.2 I'll be merging a patch that makes some adjustments to the > Makefile and configure scripts, this means you'll have to rebootstrap. This > hasn't been done yet, but heads up its coming.... > > I'll follow up to the list once that merge has happened as you'll know you > whats up when your make breaks > > K > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/7feeece4/attachment.html From garbytrash at gmail.com Sat Mar 24 09:21:42 2012 From: garbytrash at gmail.com (Zenny) Date: Sat, 24 Mar 2012 06:21:42 +0000 Subject: [Freeswitch-users] FreeSWITCH Installer for Debian In-Reply-To: References: Message-ID: Brian: love to see your new releases, great script ;-) On 3/23/12, Brian Foster wrote: > A fix for that is in the works. > > -BDF > On Mar 23, 2012 7:01 PM, "curriegrad2004" wrote: > >> One comment: building FS as root isn't something that's attractive nor >> appealing to me... >> >> On Fri, Mar 23, 2012 at 10:12 AM, Brian Foster >> wrote: >> > Whoops. >> > >> > The new version (and always the latest) can be downloaded from here: >> > >> http://files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_latest.sh >> > >> > >> > On Fri, Mar 23, 2012 at 1:07 PM, Brian Foster >> > wrote: >> >> >> >> Version 0.0.3 was released today: >> >> >> >> Changelog >> >> >> >> >> >> ================================================================================= >> >> 0.0.3 23/Mar/2012 1235 UCT >> >> >> >> Bug Fixes >> >> - Changed script shell from bash to sh for compatability >> >> Features Added >> >> -Made FS git address, FS user, FS group, installed packages, FS >> >> sounds >> >> install, FS MOH install, and base dir for source folder into >> >> variables >> >> - Option added to update/upgrade system (true/false variable) >> >> - Option to download/install init script >> >> >> >> Version 0.1.0 will be released sometime within the next week. >> >> >> >> -BDF >> >> >> >> On Thu, Mar 22, 2012 at 6:21 PM, Gabriel Gunderson >> wrote: >> >>> >> >>> On Thu, Mar 22, 2012 at 3:44 PM, Zenny wrote: >> >>> > Glad to find build_opensips.sh also in the same directory. >> >>> >> >>> If you want to save the trouble of building, there are also some >> >>> pre-built debs for both FreeSWITCH and OpenSIPS: >> >>> >> >>> https://parseltone.org/files/debs/ubuntu-11.4/ >> >>> >> >>> >> >>> Best, >> >>> Gabe >> >>> >> >>> >> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> -- >> >> Brian D. Foster >> >> Endigo Computer LLC >> >> Email: bdfoster at endigotech.com >> >> Phone: 317-800-7876 >> >> Indianapolis, Indiana, USA >> >> >> >> This message contains confidential information and is intended for >> >> those >> >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. >> If >> >> you are not the intended recipient you are notified that disclosing, >> >> copying, distributing or taking any action in reliance on the contents >> of >> >> this information is strictly prohibited. E-mail transmission cannot be >> >> guaranteed to be secure or error-free as information could be >> intercepted, >> >> corrupted, lost, destroyed, arrive late or incomplete, or contain >> viruses. >> >> The sender therefore does not accept liability for any errors or >> omissions >> >> in the contents of this message, which arise as a result of e-mail >> >> transmission. If verification is required please request a hard-copy >> >> version. >> >> >> > >> > >> > >> > -- >> > Brian D. Foster >> > Endigo Computer LLC >> > Email: bdfoster at endigotech.com >> > Phone: 317-800-7876 >> > Indianapolis, Indiana, USA >> > >> > This message contains confidential information and is intended for those >> > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. >> If >> > you are not the intended recipient you are notified that disclosing, >> > copying, distributing or taking any action in reliance on the contents >> > of >> > this information is strictly prohibited. E-mail transmission cannot be >> > guaranteed to be secure or error-free as information could be >> intercepted, >> > corrupted, lost, destroyed, arrive late or incomplete, or contain >> viruses. >> > The sender therefore does not accept liability for any errors or >> omissions >> > in the contents of this message, which arise as a result of e-mail >> > transmission. If verification is required please request a hard-copy >> > version. >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From all.eforums at gmail.com Sat Mar 24 11:12:17 2012 From: all.eforums at gmail.com (A E [Gmail]) Date: Sat, 24 Mar 2012 04:12:17 -0400 Subject: [Freeswitch-users] skypopen scalability tests In-Reply-To: References: Message-ID: On Mon, May 2, 2011 at 8:50 PM, Seven Du wrote: > Perhaps there are two ways > > 1) each one interested in this list can config 20 different users on their > server, and then you can call 200 if we get 10 people. The other party just > need to create a dialplan and plan MOH. > > 2) 200 different clients can call your server at a certain time, with same > username, say Bob, is that reliable? > > -- > Seven Du > > That's a pretty good idea...assuming there's no limit on the number of calls that can be recd. by one Skype username. I'm assuming that the person registering as 'Bob' will need to run 200 instances of the Skype client registered as 'Bob' with the Skype network. Then, each person who wishes to participate in this trial can create 20 different users or (10 users if we can get 20 people) [I read somewhere that several users can be created using the Skype for Business online account manager]. They all call this user 'Bob' running on someone's relatively beefy machine and measure the performance and all sorts of quantities to get an idea of what can be handled and what happens in such a situation. Why did this thread die? > On Sunday, May 1, 2011 at 1:09 AM, Giovanni Maruzzelli wrote: > > Ooops, I was wrong in my previous mail (my memory no more what was > used to be :) ). > > You cannot have simultaneus calls from many instances of Bob to many > instances of Alice. It just does not works reliably. > > I was testing simultaneus calls from one client skypeusername (Bob) to > a server with many skypeusernames (Alice, Adam, etc). > > So, we're back in the situation I described (need to register many > different skypeusernames) with the added drawback that simultaneus > calls from one Skype client does not work well. > > Any other ideas? > -giovanni > > On Sat, Apr 30, 2011 at 7:01 PM, Giovanni Maruzzelli > wrote: > > On Sat, Apr 30, 2011 at 6:52 PM, Anton VG wrote: > > Any other ideas on how to proceed for scalability tests? > > > Hm, since 2.0.0.72 still allows simultaneous calls, you do not have to > register lots of usernames, just 2 of them for 2 PC's > than make 100 calls from 1 machine to another > > > I was going that way, couple years ago, but the Skype client was > scaling kind of badly for me. No more than a bunch of simultaneus > calls. > But at that time I was testing using a standard ALSA sound driver, and > Skype client for ALSA. > Maybe using Skype for OSS and skypopen.ko it will scale better. > > Good hint, at least I will check into it in the future. > > Other ideas? > > -giovanni > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/e80eeb41/attachment-0001.html From neilp at cs.stanford.edu Sat Mar 24 12:13:36 2012 From: neilp at cs.stanford.edu (Neil Patel) Date: Sat, 24 Mar 2012 14:43:36 +0530 Subject: [Freeswitch-users] no DTMF detection over VoIP Message-ID: Hi All, I have a basic IVR application in Lua connected to a PRI line. It currently is not responding to DTMF input given from any VoIP call (e.g. Skype). However, it accepts input from local mobile or landline calls just fine. I recently pulled latest from git and built; before that, DTMF detection from both were working. Is there something I need to configure to allow DTMF for VoIP? Thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/74fd6e9e/attachment.html From Daniel.Knaggs at realitysolutions.co.uk Sat Mar 24 15:27:47 2012 From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs) Date: Sat, 24 Mar 2012 12:27:47 +0000 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: References: Message-ID: They aren?t any faults that I can see. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 Sent: 23 March 2012 23:23 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls Have you checked for any faults on your TDM link on your side? On Fri, Mar 23, 2012 at 10:32 AM, Brian Foster > wrote: Please update to latest git. -BDF On Fri, Mar 23, 2012 at 7:19 AM, Daniel Knaggs > wrote: Hello all, Got a bit of a strange one, we appear to be getting DTMF tones on incoming calls when the caller hasn?t even pressed any keys. It normally happens with 10 seconds or so after the call has been answered. Here is the log of it happening earlier: - 2012-03-23 10:38:16.852654 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF A (debug = 0) 2012-03-23 10:38:16.852654 [DEBUG] mod_freetdm.c:799 Queuing DTMF [A] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:16.915653 [DEBUG] switch_rtp.c:2420 Send start packet for [A] ts=49440 dur=160/160/2000 seq=46803 lw=49440 2012-03-23 10:38:16.936653 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=320/320/2000 seq=46804 lw=49600 2012-03-23 10:38:16.957652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=480/480/2000 seq=46805 lw=49760 2012-03-23 10:38:16.978652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=640/640/2000 seq=46806 lw=49920 2012-03-23 10:38:16.999652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=800/800/2000 seq=46807 lw=50080 2012-03-23 10:38:17.020651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=960/960/2000 seq=46808 lw=50240 2012-03-23 10:38:17.041651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1120/1120/2000 seq=46809 lw=50400 2012-03-23 10:38:17.062651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1280/1280/2000 seq=46810 lw=50560 2012-03-23 10:38:17.083650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1440/1440/2000 seq=46811 lw=50720 2012-03-23 10:38:17.104650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1600/1600/2000 seq=46812 lw=50880 2012-03-23 10:38:17.125650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1760/1760/2000 seq=46813 lw=51040 2012-03-23 10:38:17.146650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1920/1920/2000 seq=46814 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46815 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46816 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46817 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2271 Queue digit delay of 40ms 2012-03-23 10:38:18.070638 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF B (debug = 0) 2012-03-23 10:38:18.070638 [DEBUG] mod_freetdm.c:799 Queuing DTMF [B] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:18.133637 [DEBUG] switch_rtp.c:2420 Send start packet for [B] ts=59040 dur=160/160/2000 seq=46864 lw=59040 2012-03-23 10:38:18.154637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=320/320/2000 seq=46865 lw=59200 2012-03-23 10:38:18.175637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=480/480/2000 seq=46866 lw=59360 2012-03-23 10:38:18.196636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=640/640/2000 seq=46867 lw=59520 2012-03-23 10:38:18.217636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=800/800/2000 seq=46868 lw=59680 2012-03-23 10:38:18.238636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=960/960/2000 seq=46869 lw=59840 2012-03-23 10:38:18.259636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1120/1120/2000 seq=46870 lw=60000 2012-03-23 10:38:18.280636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1280/1280/2000 seq=46871 lw=60160 2012-03-23 10:38:18.301635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1440/1440/2000 seq=46872 lw=60320 2012-03-23 10:38:18.322635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1600/1600/2000 seq=46873 lw=60480 2012-03-23 10:38:18.343634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1760/1760/2000 seq=46874 lw=60640 2012-03-23 10:38:18.364634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1920/1920/2000 seq=46875 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46876 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46877 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46878 lw=60800 I?m very sure the caller isn?t dialling ?A? or ?B?! Running on ISDN via a TE121 card using E1 (euroisdn). Freeswitch version is ?FreeSWITCH Version 1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)?. Currently there is a ?bind_meta_app? in the config which binds a script on the B leg of the call which parks the call ? I haven?t tried turning this off yet to see if it?s this. Wondering if anyone has any ideas or has come across this before? Thanks in advance. [Description: cid:imageacd695.PNG at 9e92e461.40aa4b96] Daniel Knaggs Software Developer Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE Tel: 01482 828000 / Fax: 01482 373100 Daniel.Knaggs at realitysolutions.co.uk www.realitysolutions.co.uk ________________________________ Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire [Description: cid:image27a71e.PNG at c2da8488.4683bff1] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/6623d31d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 22463 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/6623d31d/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 69075 bytes Desc: image002.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/6623d31d/attachment-0003.png From anita.hall at simmortel.com Sat Mar 24 16:52:02 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Sat, 24 Mar 2012 19:22:02 +0530 Subject: [Freeswitch-users] no DTMF detection over VoIP In-Reply-To: References: Message-ID: What is your config for DTMF in sip profile ? http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#dtmf-type Use tshark to generate a SIP trace and paste it on http://pastebin.freeswitch.org/ and give the link here. regards, Anita On Sat, Mar 24, 2012 at 2:43 PM, Neil Patel wrote: > Hi All, > > I have a basic IVR application in Lua connected to a PRI line. It > currently is not responding to DTMF input given from any VoIP call (e.g. > Skype). However, it accepts input from local mobile or landline calls just > fine. > > I recently pulled latest from git and built; before that, DTMF detection > from both were working. Is there something I need to configure to allow > DTMF for VoIP? > > Thanks, > Neil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/9635ae41/attachment.html From anita.hall at simmortel.com Sat Mar 24 18:22:12 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Sat, 24 Mar 2012 20:52:12 +0530 Subject: [Freeswitch-users] Fax on Sangoma Message-ID: Hi My client had been using an old torrenta cards which game problems with pretty much everything :) After evangelizing the cause of Sangoma for more than 3 months (yes, they are dumb), they are finally making the transition so I am pretty excited. The choice of card is A108D which does a whole bunch of DSP in the hardware. Now, the first hurdle that I have to make Sangoma jump across is getting incoming Fax over E1 right. We are using mod_spandsp, of course. So, here I will be needing a whole lot of help from the veterans of spandsp and Faxing :) I desperately need Sangoma to pass the Fax test or they will give me torrenta cards all over again! The primary cause of failures are - (49) The call dropped prematurely and (48) Disconnected after permitted retries. For example, in this case, can I conclude that the other end did not provide a Fax tone or is it something else? 2aeb5f7c-75b5-11e1-8f36-b3286880c45b EXECUTE FreeTDM/4:2/47615728 rxfax(/srv/fax/in/2aeb5f7c-75b5-11e1-8f36-b3286880c45b.tiff) 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] mod_spandsp_fax.c:1357 Raw read codec activation Success L16 20000 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] switch_core_codec.c:216 FreeTDM/4:2/47615728 Push codec L16:70 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] mod_spandsp_fax.c:1373 Raw write codec activation Success L16 2012-03-24 18:57:57.844876 [DEBUG] ftmod_wanpipe.c:965 [s4c2][4:2] First packet read stats: Rx queue len: 1, Rx queue size: 10 2012-03-24 18:57:57.904879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17 2012-03-24 18:57:57.924878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17 2012-03-24 18:57:58.104877 [DEBUG] ftmod_wanpipe.c:901 [s4c2][4:2] First packet write stats: Tx queue len: 1, Tx queue size: 5, Tx idle: 30 2012-03-24 18:57:58.444908 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17 2012-03-24 18:57:58.644907 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17 2012-03-24 18:57:58.664878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1 2012-03-24 18:57:58.764879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17 2012-03-24 18:57:58.764879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17 2012-03-24 18:57:58.784878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17 2012-03-24 18:57:58.804879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17 2012-03-24 18:57:59.244886 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17 2012-03-24 18:57:59.464887 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Retry number 1 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set rx type 0 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set tx type 4 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Sending ident 'Sangoma Fax Ident' 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Tx: CSI without final frame tag 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Tx: ff 03 40 74 6e 65 64 49 20 78 61 46 20 61 6d 6f 67 6e 61 53 20 20 20 2012-03-24 18:58:00.524877 [DEBUG] ftmod_libpri.c:1065 -- Hangup REQ on channel 4:1 2012-03-24 18:58:00.524877 [DEBUG] ftmod_libpri.c:1078 [s4c1][4:1] Changed state from UP to TERMINATING 2012-03-24 18:58:00.524877 [DEBUG] ftdm_state.c:511 [s4c1][4:1] Executing state processor for TERMINATING I will need some more hand-holding with logs later :) regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/4c659472/attachment.html From anton.jugatsu at gmail.com Sat Mar 24 18:54:00 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Sat, 24 Mar 2012 19:54:00 +0400 Subject: [Freeswitch-users] Fax on Sangoma In-Reply-To: References: Message-ID: Can you show dialplan snippet. 24.03.2012 19:24 ???????????? "Anita Hall" ???????: > Hi > > My client had been using an old torrenta cards which game problems with > pretty much everything :) After evangelizing the cause of Sangoma for more > than 3 months (yes, they are dumb), they are finally making the transition > so I am pretty excited. The choice of card is A108D which does a whole > bunch of DSP in the hardware. > > Now, the first hurdle that I have to make Sangoma jump across is getting > incoming Fax over E1 right. We are using mod_spandsp, of course. So, here I > will be needing a whole lot of help from the veterans of spandsp and Faxing > :) I desperately need Sangoma to pass the Fax test or they will give me > torrenta cards all over again! > > The primary cause of failures are - (49) The call dropped prematurely and > (48) Disconnected after permitted retries. > > For example, in this case, can I conclude that the other end did not > provide a Fax tone or is it something else? > > 2aeb5f7c-75b5-11e1-8f36-b3286880c45b EXECUTE FreeTDM/4:2/47615728 > rxfax(/srv/fax/in/2aeb5f7c-75b5-11e1-8f36-b3286880c45b.tiff) > 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] > mod_spandsp_fax.c:1357 Raw read codec activation Success L16 20000 > 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] > switch_core_codec.c:216 FreeTDM/4:2/47615728 Push codec L16:70 > 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] > mod_spandsp_fax.c:1373 Raw write codec activation Success L16 > 2012-03-24 18:57:57.844876 [DEBUG] ftmod_wanpipe.c:965 [s4c2][4:2] First > packet read stats: Rx queue len: 1, Rx queue size: 10 > 2012-03-24 18:57:57.904879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 17 > 2012-03-24 18:57:57.924878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 17 > 2012-03-24 18:57:58.104877 [DEBUG] ftmod_wanpipe.c:901 [s4c2][4:2] First > packet write stats: Tx queue len: 1, Tx queue size: 5, Tx idle: 30 > 2012-03-24 18:57:58.444908 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 17 > 2012-03-24 18:57:58.644907 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 17 > 2012-03-24 18:57:58.664878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 1 > 2012-03-24 18:57:58.764879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 17 > 2012-03-24 18:57:58.764879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 17 > 2012-03-24 18:57:58.784878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 17 > 2012-03-24 18:57:58.804879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 17 > 2012-03-24 18:57:59.244886 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 17 > 2012-03-24 18:57:59.464887 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 17 > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 T4 > expired in phase T30_PHASE_B_RX, state 17 > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Retry > number 1 > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 > Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set rx > type 0 > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set tx > type 4 > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Sending > ident 'Sangoma Fax Ident' > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Tx: > CSI without final frame tag > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Tx: ff > 03 40 74 6e 65 64 49 20 78 61 46 20 61 6d 6f 67 6e 61 53 20 20 20 > 2012-03-24 18:58:00.524877 [DEBUG] ftmod_libpri.c:1065 -- Hangup REQ on > channel 4:1 > 2012-03-24 18:58:00.524877 [DEBUG] ftmod_libpri.c:1078 [s4c1][4:1] Changed > state from UP to TERMINATING > 2012-03-24 18:58:00.524877 [DEBUG] ftdm_state.c:511 [s4c1][4:1] Executing > state processor for TERMINATING > > > I will need some more hand-holding with logs later :) > > > regards, > Anita > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/d20f608e/attachment-0001.html From neilp at cs.stanford.edu Sat Mar 24 21:18:54 2012 From: neilp at cs.stanford.edu (Neil Patel) Date: Sat, 24 Mar 2012 23:48:54 +0530 Subject: [Freeswitch-users] no DTMF detection over VoIP In-Reply-To: References: Message-ID: I have not modified the sip_profiles directory from standard. There is no variable set for dtmf-type in either internal.xml or external.xml. I did a trace using: sofia global siptrace on There wer no SIP packets logged. If the SIP call is going to a PRI profile (freetdm), will there still be SIP traffic? Thanks, Neil On Sat, Mar 24, 2012 at 7:22 PM, Anita Hall wrote: > What is your config for DTMF in sip profile ? > > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#dtmf-type > > Use tshark to generate a SIP trace and paste it on > http://pastebin.freeswitch.org/ and give the link here. > > regards, > Anita > > > > On Sat, Mar 24, 2012 at 2:43 PM, Neil Patel wrote: > >> Hi All, >> >> I have a basic IVR application in Lua connected to a PRI line. It >> currently is not responding to DTMF input given from any VoIP call (e.g. >> Skype). However, it accepts input from local mobile or landline calls just >> fine. >> >> I recently pulled latest from git and built; before that, DTMF detection >> from both were working. Is there something I need to configure to allow >> DTMF for VoIP? >> >> Thanks, >> Neil >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/1365288c/attachment.html From mario_fs at mgtech.com Sat Mar 24 22:24:37 2012 From: mario_fs at mgtech.com (Mario G) Date: Sat, 24 Mar 2012 12:24:37 -0700 Subject: [Freeswitch-users] Can user extension be set to use TCP? Message-ID: I have a need to set a softphone client to use TCP. I was not able to find anything on the wiki regarding TCP for a user extention parm. The only TCP references are for gateway setttings. Does anyone know if there is a parm for the user extension to be forced to use TCP? BTW, although I have TCP set in the client and it is working, however, FS is failing every night (all internal extension fail requiring a FS restart) which is why I am hoping a setting will work. Thanks, Mario G From me at nybras.com Sat Mar 24 22:25:32 2012 From: me at nybras.com (Anderson Arboleya) Date: Sat, 24 Mar 2012 16:25:32 -0300 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION In-Reply-To: References: <3944506.tN9pZ6Ijqo@sos> Message-ID: Hi everybody, thank you all so much! Finally my lua script was hit by the call, time to have some fun. :) Anderson Arboleya On Fri, Mar 23, 2012 at 10:55 PM, Brian Foster wrote: > > On Mar 23, 2012 9:46 PM, "Sergey Okhapkin" > wrote: > > > > Install commercial g729 codec and the problem will go away. > > No, if it just passes through FS there is no need for a license. If you > are doing transcoding, then yes, a license is required. > > -BDF > > > > > On Friday 23 March 2012 22:33:57 Anderson Arboleya wrote: > > > Hi, somebody knows what do I have to do in order to fix that hangup > cause? > > > > > > Here's the complete log (with debug) of the inbound call. > > > > > > Spent my whole day trying to understand it.. without any luck. > > > > > > Thanks > > > > > > ============>>>>>>>>> > > > > > > 2012-03-24 01:18:37.865450 [NOTICE] switch_channel.c:926 New Channel > > > sofia/external/1184049130 at 189.84.133.130:5060 > [4898b56c-754f-11e1-af7e-812af0 > > > 71ddc6] 2012-03-24 01:18:37.865450 [DEBUG] sofia.c:5532 Channel > > > sofia/external/ 1184049130 at 189.84.133.130:5060 entering state > > > [received][100] > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia.c:5543 Remote SDP: > > > v=0 > > > o=VoipSwitch 6106 7106 IN IP4 189.84.133.130 > > > s=VoipSIP > > > i=Audio Session > > > c=IN IP4 189.84.133.130 > > > t=0 0 > > > m=audio 6106 RTP/AVP 18 101 > > > a=rtpmap:18 G729/8000/1 > > > a=fmtp:18 annexb=no > > > a=rtpmap:101 telephone-event/8000 > > > a=fmtp:101 0-15 > > > > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec > Compare > > > [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec > Compare > > > [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec > Compare > > > [G729:18:8000:20:8000]/[G722:9:8000:20:64000] > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec > Compare > > > [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000] > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec > Compare > > > [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec > Compare > > > [G729:18:8000:20:8000]/[GSM:3:8000:20:13200] > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec > Compare > > > [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec > Compare > > > [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec > Compare > > > [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec > Compare > > > [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec > Compare > > > [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec > Compare > > > [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] > > > 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > > > send/recv payload to 101 > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_channel.c:2848 > (sofia/external/ > > > 1184049130 at 189.84.133.130:5060) Callstate Change DOWN -> HANGUP > > > 2012-03-24 01:18:37.865450 [NOTICE] sofia.c:5813 Hangup sofia/external/ > > > 1184049130 at 189.84.133.130:5060 [CS_NEW] [INCOMPATIBLE_DESTINATION] > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_channel.c:2871 Send signal > > > sofia/external/1184049130 at 189.84.133.130:5060 [KILL] > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send > signal > > > sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:362 > > > (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change > > > CS_HANGUP > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:602 > > > (sofia/external/1184049130 at 189.84.133.130:5060) State HANGUP > > > 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:469 Channel > sofia/external/ > > > 1184049130 at 189.84.133.130:5060 hanging up, cause: > INCOMPATIBLE_DESTINATION > > > 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:534 Responding to INVITE > > > with: 488 > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:47 > > > sofia/external/1184049130 at 189.84.133.130:5060 Standard HANGUP, cause: > > > INCOMPATIBLE_DESTINATION > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:602 > > > (sofia/external/1184049130 at 189.84.133.130:5060) State HANGUP going to > sleep > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:393 > > > (sofia/external/1184049130 at 189.84.133.130:5060) State Change > CS_HANGUP -> > > > CS_REPORTING > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send > signal > > > sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:362 > > > (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change > > > CS_REPORTING > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:662 > > > (sofia/external/1184049130 at 189.84.133.130:5060) State REPORTING > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:79 > > > sofia/external/1184049130 at 189.84.133.130:5060 Standard REPORTING, > cause: > > > INCOMPATIBLE_DESTINATION > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:662 > > > (sofia/external/1184049130 at 189.84.133.130:5060) State REPORTING going > to > > > sleep > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:387 > > > (sofia/external/1184049130 at 189.84.133.130:5060) State Change > CS_REPORTING > > > -> CS_DESTROY > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send > signal > > > sofia/external/1184049130 at 189.84.133.130:5060 [BREAK] > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1382 Session 5 > > > (sofia/external/1184049130 at 189.84.133.130:5060) Locked, Waiting on > external > > > entities > > > 2012-03-24 01:18:37.865450 [NOTICE] switch_core_session.c:1400 Session > 5 > > > (sofia/external/1184049130 at 189.84.133.130:5060) Ended > > > 2012-03-24 01:18:37.865450 [NOTICE] switch_core_session.c:1402 Close > > > Channel sofia/external/1184049130 at 189.84.133.130:5060 [CS_DESTROY] > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:491 > > > (sofia/external/1184049130 at 189.84.133.130:5060) Callstate Change > HANGUP -> > > > DOWN > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:494 > > > (sofia/external/1184049130 at 189.84.133.130:5060) Running State Change > > > CS_DESTROY > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:504 > > > (sofia/external/1184049130 at 189.84.133.130:5060) State DESTROY > > > 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:374 sofia/external/ > > > 1184049130 at 189.84.133.130:5060 SOFIA DESTROY > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:86 > > > sofia/external/1184049130 at 189.84.133.130:5060 Standard DESTROY > > > 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:504 > > > (sofia/external/1184049130 at 189.84.133.130:5060) State DESTROY going > to sleep > > > > > > <<<<<<<<<============ > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/7ba2b5fa/attachment-0001.html From mitch.capper at gmail.com Sat Mar 24 23:33:57 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Sat, 24 Mar 2012 13:33:57 -0700 Subject: [Freeswitch-users] Can user extension be set to use TCP? In-Reply-To: References: Message-ID: How a client connects to the server is a client specific setting. There is no extension variable for a user as its not something picked by the server. Setting TCP for the connection mode in the client is the correct way to do so, if your client is not connecting over TCP then its an issue with the client. ~Mitch From potxoka at gmail.com Sat Mar 24 23:50:00 2012 From: potxoka at gmail.com (Anto) Date: Sat, 24 Mar 2012 21:50:00 +0100 Subject: [Freeswitch-users] Codec negotiation with carriers In-Reply-To: References: Message-ID: Hello Can anyone show me your settings to compare with mine? I look for more traces, check my settings, can not understand the functioning of the profiles :-S. Searching the internet, I haven?t found examples (a supplier or system complete) of these configurations. thanks. Regards Anto 2012/3/17 Anto : > Hello > > I do not want to bother them, nor give me the solution. I would like > to learn how to configure it for myself (in fact one of the settings > works), but I do not understand the functioning correctly, for so not > to disturb in the future. > > I attached the various configurations and their sip traces, as well as > the logs (debug): > > First scenario ( Call isn?t established ) > http://pastebin.freeswitch.org/18685 > > Second scenario ( Call isn?t established ) > http://pastebin.freeswitch.org/18686 > > Third scenario ( Call isn?t established ) > http://pastebin.freeswitch.org/18687 > > Fourth scenario ( Call isn?t established ) > http://pastebin.freeswitch.org/18688 > > Fifth scenario ( Call established ) > http://pastebin.freeswitch.org/18689 > > Sixth scenario ( Fail transcoding ) > http://pastebin.freeswitch.org/18690 > > General settings: > > vars.xml > > ? ? ? ? data="global_codec_prefs=VB32,G7221,speex,PCMU,PCMA,BV16,G726-32,iLBC,GSM,G729,G723,AMR"/> > ? ? ? ? > ? ? ? ? data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,VB32,G7221,speex,BV16,G726-32,iLBC,GSM"/> > > ----------------------------------------------------------------------------------------------------------------------------------------------- > > dialplan\outbound.xml > > > ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? > > > > Thanks ! > > Regards > Anto > > 2012/3/16 Michael Collins : >> Well, this is a little better, however you don't have proper freeswitch logs >> on all these calls. For example, only the first call has freeswitch debug >> output. The other calls have sip traces, but not the first call. One call >> has what appears to be info-level output, but not debug-level output. >> >> I'd recommend that if you have this much information it might be good to put >> each call example in its own pastebin. Also, be sure to give a detailed >> description of what kind of call you are documenting. Some of your >> traces/debugs have no information explaining what the call is doing. Whether >> you are reporting a working or failed call, be sure to mention what kind of >> call it is. In the case of a failed call, be sure to mention what it is you >> are trying to do and what call result you expected to see. >> >> Thanks! >> >> -MC >> >> >> On Thu, Mar 15, 2012 at 2:22 PM, Anto wrote: >>> >>> Hi >>> >>> If, upload a file to trace and explanation to this address >>> http://pastebin.freeswitch.org/18599 >>> >>> I do not want disturb watching this ;-), I prefer to use a system to >>> understand, this scenario and for future projects. >>> >>> With everything I've read do not really understand what the codecs :-S >>> , but if I had been good to understand the rest of the operation of >>> FreeSWITCH (or so I think). Thanks ! >>> >>> Regards >>> Anto >>> >>> 2012/3/14 Michael Collins : >>> > Did you get sip traces and logs of working vs. non-working calls and put >>> > them on pastebin? Most likely there is an explanation but it will take >>> > some >>> > time and effort to figure it out. >>> > >>> > -MC >>> > >>> > >>> > On Wed, Mar 14, 2012 at 2:18 PM, Anto wrote: >>> >> >>> >> Hello >>> >> >>> >> I have searched previous messages in the list, I consulted the book of >>> >> FreeSWITCH (which I bought over a year), wiki and so on. I still do >>> >> not understand how and why in some cases I work. Also I downloaded >>> >> frontend to consult your code if there was something about this, but >>> >> still the same. I have several weeks with this question and I can not >>> >> find it. In the end I decided to spend the gateway to Asterisk, and >>> >> you at least understand its operation. Thank you very much to all :-) >>> >> >>> >> Best regards >>> >> Anto >>> >> >>> >> 2012/3/11 Anto : >>> >> > Hi >>> >> > >>> >> > I still do not find the solution and not really understanding, >>> >> > because >>> >> > it works:-S >>> >> > >>> >> > regards >>> >> > anto >>> >> > >>> >> > 2012/3/7 Anto : >>> >> >> Hello >>> >> >> >>> >> >> Attached file, with the traces of the different tests (with >>> >> >> different >>> >> >> configurations). >>> >> >> >>> >> >> http://pastebin.freeswitch.org/18599 >>> >> >> >>> >> >> I have read the url that you mentioned, the initial guide >>> >> >> FreeSWITCH, >>> >> >> that of mod_sofia, applications, etc.. I believe that most of the >>> >> >> wiki >>> >> >> (maybe when do not give the solution, read as much documentation is >>> >> >> worse idea :-S, lock me more). >>> >> >> >>> >> >> I made a configuration that works (I have not tested the audio), but >>> >> >> earlier (before I started "touch" the profiles) if I could talk to a >>> >> >> physical phone (several times). The problem is that I can not >>> >> >> understand why it works and sometimes not, and I would like to learn >>> >> >> :-). Not only do and forget, so I would like to learn and less >>> >> >> disturbing to the mail list and (maybe in the future) to help other >>> >> >> newbies like me :-). Thanks ! >>> >> >> >>> >> >> Best regards >>> >> >> Anto >>> >> >> >>> >> >> 2012/3/7 Michael Collins : >>> >> >>> You may want to read up on codec negotiation: >>> >> >>> http://wiki.freeswitch.org/wiki/Codec_negotiation >>> >> >>> >>> >> >>> There are different ways to handle codecs depending on your needs. >>> >> >>> I'd >>> >> >>> read >>> >> >>> that page first and then try out some of the suggestions. If you're >>> >> >>> still >>> >> >>> having trouble then I'd recommend getting SIP traces of the traffic >>> >> >>> and >>> >> >>> putting them on pastebin.freeswitch.org. The gang here is pretty >>> >> >>> good >>> >> >>> at >>> >> >>> looking over logs and helping with diagnosing problems. :) >>> >> >>> >>> >> >>> -MC >>> >> >>> >>> >> >>> On Tue, Mar 6, 2012 at 2:30 PM, Anto wrote: >>> >> >>>> >>> >> >>>> Hi >>> >> >>>> >>> >> >>>> I am following the steps in this direction >>> >> >>>> "http://wiki.freeswitch.org/wiki/SBC_Setup" and >>> >> >>>> >>> >> >>>> "http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice", >>> >> >>>> I reread the whole entire wiki (or so I lack), but do not quite >>> >> >>>> assimilate or finding the right formula to operate the bridge :-S. >>> >> >>>> >>> >> >>>> I captured traffic with ngrep, I enabled sip-trace, console >>> >> >>>> logconsole >>> >> >>>> 8, etc., but unless the transcoding error (only two of the >>> >> >>>> hundreds >>> >> >>>> of >>> >> >>>> combinations of settings that I have), I have not seen anything >>> >> >>>> "weird" :-S >>> >> >>>> >>> >> >>>> I have 3 suppliers, each with this codec: >>> >> >>>> >>> >> >>>> 1) ? ? ? ? ? 2) ? ? ? ? ? ? ?3) >>> >> >>>> G729 ? ? ? ?G729 ? ? ? ?G729 >>> >> >>>> G711u ? ? ?G711A ? ? ?G711A >>> >> >>>> G711A ? ? G711u ? ? ? G711u >>> >> >>>> ? ? ? ? ? ? ? ?G723 ? ? ? ? G723 >>> >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?G722 >>> >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?GSM >>> >> >>>> >>> >> >>>> I think I understand that when making an outside call, FreeSWITCH >>> >> >>>> follow these steps: >>> >> >>>> >>> >> >>>> USER -> ( ? Dialplan -> profile (internal) -> bridge (external) -> >>> >> >>>> profile (external) ? ) -> PROVIDER >>> >> >>>> >>> >> >>>> PROVIDER -> ( ? Dialplan -> profile (external) -> bridge >>> >> >>>> (internal) >>> >> >>>> -> >>> >> >>>> profile (internal) ?) -> USER >>> >> >>>> >>> >> >>>> right? >>> >> >>>> >>> >> >>>> Internal and external I set as follows (and not many changes have >>> >> >>>> done, and not remember it, because I've been testing days). If >>> >> >>>> outbound (outbound-codec-prefs) all codecs specified system does >>> >> >>>> not >>> >> >>>> handle the call, I have to specify these by hand. If active >>> >> >>>> inbound-proxy-media, not the caller. Some of the time I worked, >>> >> >>>> but >>> >> >>>> gave me an error that it can do transcoding G729 codec (I do >>> >> >>>> passthrough), but the proxy does not work half. >>> >> >>>> >>> >> >>>> If the outbound property (outbound-codec-prefs) all codecs >>> >> >>>> specified >>> >> >>>> system does not handle the call, I have to specify these by hand. >>> >> >>>> If >>> >> >>>> active inbound-proxy-media, not the caller. Some of the time I >>> >> >>>> worked, >>> >> >>>> but gave me an error that it can do transcoding G729 codec (I want >>> >> >>>> to >>> >> >>>> make passthrough), but the "proxy media" does not work. >>> >> >>>> >>> >> >>>> Basically, what I do is that local users can use all the codecs >>> >> >>>> allowed (iLBC, GSM, ...) and make an outside call, use the carrier >>> >> >>>> that will indicate the priority but the free codec. >>> >> >>>> >>> >> >>>> With this configuration works for me, but I would like to >>> >> >>>> understand >>> >> >>>> why so if it works and otherwise no. Coming to understand how to >>> >> >>>> configure properly and so as not to disturb the mail list ;-). >>> >> >>>> Thanks >>> >> >>>> ! >>> >> >>>> >>> >> >>>> Best regards >>> >> >>>> Anto >>> >> >>>> >>> >> >>>> vars.xml >>> >> >>>> >>> >> >>>> >> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> data="global_codec_prefs=iLBC,G7221,speex,PCMU,PCMA,BV16,G726-32,GSM,G729,G723,AMR"/> >>> >> >>>> >> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,iLBC,G7221,speex,BV16,G726-32,GSM"/> >>> >> >>>> >>> >> >>>> internal.xml >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> external.xml >>> >> >>>> >>> >> >>>> >>> >> >>>> >> >> >>>> value="$${carriers_codec_prefs}"/> >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> dialplan/outbound.xml >>> >> >>>> >>> >> >>>> >>> >> >>>> ? ? ? ? >>> >> >>>> ? ? ? ? ? ? ? ? >>> >> >>>> ? ? ? ? ? ? ? ? ?>> >> >>>> expression="^(\d+)$"> >>> >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? >>> >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? >>> >> >>>> ? ? ? ? ? ? ? ? ? ? ? ? >>> >> >>>> ? ? ? ? ? ? ? ? ? ? ? ?>> >> >>>> data="sofia/gateway/provider-2/$1"/> >>> >> >>>> ? ? ? ? ? ? ? ? ? >>> >> >>>> ? ? ? ? ? ? ? ? >>> >> >>>> ? ? ? ? >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> _________________________________________________________________________ >>> >> >>>> Professional FreeSWITCH Consulting Services: >>> >> >>>> consulting at freeswitch.org >>> >> >>>> http://www.freeswitchsolutions.com >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> >>> >> >>>> Official FreeSWITCH Sites >>> >> >>>> http://www.freeswitch.org >>> >> >>>> http://wiki.freeswitch.org >>> >> >>>> http://www.cluecon.com >>> >> >>>> >>> >> >>>> FreeSWITCH-users mailing list >>> >> >>>> FreeSWITCH-users at lists.freeswitch.org >>> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>>> >>> >> >>>> >>> >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>>> http://www.freeswitch.org >>> >> >>> >>> >> >>> >>> >> >>> >>> >> >>> >>> >> >>> >>> >> >>> _________________________________________________________________________ >>> >> >>> Professional FreeSWITCH Consulting Services: >>> >> >>> consulting at freeswitch.org >>> >> >>> http://www.freeswitchsolutions.com >>> >> >>> >>> >> >>> >>> >> >>> >>> >> >>> >>> >> >>> Official FreeSWITCH Sites >>> >> >>> http://www.freeswitch.org >>> >> >>> http://wiki.freeswitch.org >>> >> >>> http://www.cluecon.com >>> >> >>> >>> >> >>> FreeSWITCH-users mailing list >>> >> >>> FreeSWITCH-users at lists.freeswitch.org >>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >>> >> >>> >>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>> http://www.freeswitch.org >>> >> >>> >>> >> >>> >> >>> >> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://wiki.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> From mario_fs at mgtech.com Sun Mar 25 00:08:50 2012 From: mario_fs at mgtech.com (Mario G) Date: Sat, 24 Mar 2012 14:08:50 -0700 Subject: [Freeswitch-users] Can user extension be set to use TCP? In-Reply-To: References: Message-ID: <09CD56A1-2AAD-4CCA-9205-52CECBE69BC1@mgtech.com> Thanks, the client is working fine. It's just that since I added this new extension FS is failing all internal extensions in the middle of the night and not recovering. The hard phones are SPA960 Linksys/Cisco. This new extension is an iPad running SipPhone which requires TCP to run background. I'll be doing more testing to narrow things down. Today I am replacing the iPad app with Bria to see if that helps. Thanks again! Mario G On Mar 24, 2012, at 1:33 PM, Mitch Capper wrote: > How a client connects to the server is a client specific setting. > There is no extension variable for a user as its not something picked > by the server. Setting TCP for the connection mode in the client is > the correct way to do so, if your client is not connecting over TCP > then its an issue with the client. > > > ~Mitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Sun Mar 25 05:20:10 2012 From: msc at freeswitch.org (Michael Collins) Date: Sat, 24 Mar 2012 18:20:10 -0700 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: References: Message-ID: I'd do an audio recording of the call and open the resulting file in Audacity. It can do kind of a spectrum analysis to show you if there are any frequencies that might be fooling the dtmf detector. Also, I'm not familiar with using Digium cards with FreeTDM. (I have a Digium TE121 or similar but I don't happen to have any mobo's with the PCIe or whatever slot type it uses.) To Moises I'd ask: is there hardware DTMF detection in the Digium cards that you know of? Also, any known or suspected issues with false DTMF detection in the scenario mentioned by the OP? Gracias, MC On Sat, Mar 24, 2012 at 5:27 AM, Daniel Knaggs < Daniel.Knaggs at realitysolutions.co.uk> wrote: > They aren?t any faults that I can see.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of * > curriegrad2004 > *Sent:* 23 March 2012 23:23 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls**** > > ** ** > > Have you checked for any faults on your TDM link on your side?**** > > On Fri, Mar 23, 2012 at 10:32 AM, Brian Foster > wrote:**** > > Please update to latest git.**** > > ** ** > > -BDF**** > > On Fri, Mar 23, 2012 at 7:19 AM, Daniel Knaggs < > Daniel.Knaggs at realitysolutions.co.uk> wrote:**** > > Hello all,**** > > **** > > Got a bit of a strange one, we appear to be getting DTMF tones on incoming > calls when the caller hasn?t even pressed any keys.**** > > **** > > It normally happens with 10 seconds or so after the call has been answered. > **** > > **** > > **** > > Here is the log of it happening earlier: -**** > > **** > > 2012-03-23 10:38:16.852654 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF > A (debug = 0)**** > > 2012-03-23 10:38:16.852654 [DEBUG] mod_freetdm.c:799 Queuing DTMF [A] in > channel FreeTDM/1:2/000 device 1:2**** > > 2012-03-23 10:38:16.915653 [DEBUG] switch_rtp.c:2420 Send start packet for > [A] ts=49440 dur=160/160/2000 seq=46803 lw=49440**** > > 2012-03-23 10:38:16.936653 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=320/320/2000 seq=46804 lw=49600**** > > 2012-03-23 10:38:16.957652 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=480/480/2000 seq=46805 lw=49760**** > > 2012-03-23 10:38:16.978652 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=640/640/2000 seq=46806 lw=49920**** > > 2012-03-23 10:38:16.999652 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=800/800/2000 seq=46807 lw=50080**** > > 2012-03-23 10:38:17.020651 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=960/960/2000 seq=46808 lw=50240**** > > 2012-03-23 10:38:17.041651 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=1120/1120/2000 seq=46809 lw=50400**** > > 2012-03-23 10:38:17.062651 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=1280/1280/2000 seq=46810 lw=50560**** > > 2012-03-23 10:38:17.083650 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=1440/1440/2000 seq=46811 lw=50720**** > > 2012-03-23 10:38:17.104650 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=1600/1600/2000 seq=46812 lw=50880**** > > 2012-03-23 10:38:17.125650 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=1760/1760/2000 seq=46813 lw=51040**** > > 2012-03-23 10:38:17.146650 [DEBUG] switch_rtp.c:2323 Send middle packet > for [A] ts=49440 dur=1920/1920/2000 seq=46814 lw=51200**** > > 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for > [A] ts=49440 dur=2080/2080/2000 seq=46815 lw=51200**** > > 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for > [A] ts=49440 dur=2080/2080/2000 seq=46816 lw=51200**** > > 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for > [A] ts=49440 dur=2080/2080/2000 seq=46817 lw=51200**** > > 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2271 Queue digit delay of > 40ms**** > > 2012-03-23 10:38:18.070638 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF > B (debug = 0)**** > > 2012-03-23 10:38:18.070638 [DEBUG] mod_freetdm.c:799 Queuing DTMF [B] in > channel FreeTDM/1:2/000 device 1:2**** > > 2012-03-23 10:38:18.133637 [DEBUG] switch_rtp.c:2420 Send start packet for > [B] ts=59040 dur=160/160/2000 seq=46864 lw=59040**** > > 2012-03-23 10:38:18.154637 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=320/320/2000 seq=46865 lw=59200**** > > 2012-03-23 10:38:18.175637 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=480/480/2000 seq=46866 lw=59360**** > > 2012-03-23 10:38:18.196636 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=640/640/2000 seq=46867 lw=59520**** > > 2012-03-23 10:38:18.217636 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=800/800/2000 seq=46868 lw=59680**** > > 2012-03-23 10:38:18.238636 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=960/960/2000 seq=46869 lw=59840**** > > 2012-03-23 10:38:18.259636 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=1120/1120/2000 seq=46870 lw=60000**** > > 2012-03-23 10:38:18.280636 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=1280/1280/2000 seq=46871 lw=60160**** > > 2012-03-23 10:38:18.301635 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=1440/1440/2000 seq=46872 lw=60320**** > > 2012-03-23 10:38:18.322635 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=1600/1600/2000 seq=46873 lw=60480**** > > 2012-03-23 10:38:18.343634 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=1760/1760/2000 seq=46874 lw=60640**** > > 2012-03-23 10:38:18.364634 [DEBUG] switch_rtp.c:2323 Send middle packet > for [B] ts=59040 dur=1920/1920/2000 seq=46875 lw=60800**** > > 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for > [B] ts=59040 dur=2080/2080/2000 seq=46876 lw=60800**** > > 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for > [B] ts=59040 dur=2080/2080/2000 seq=46877 lw=60800**** > > 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for > [B] ts=59040 dur=2080/2080/2000 seq=46878 lw=60800**** > > **** > > **** > > I?m very sure the caller isn?t dialling ?A? or ?B?!**** > > **** > > Running on ISDN via a TE121 card using E1 (euroisdn). Freeswitch version > is ?FreeSWITCH Version 1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)?.* > *** > > **** > > **** > > Currently there is a ?bind_meta_app? in the config which binds a script on > the B leg of the call which parks the call ? I haven?t tried turning this > off yet to see if it?s this.**** > > **** > > **** > > Wondering if anyone has any ideas or has come across this before?**** > > **** > > **** > > Thanks in advance.**** > > [image: Description: cid:imageacd695.PNG at 9e92e461.40aa4b96]**** > > *Daniel Knaggs***** > > Software Developer**** > > > > **** > > Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon > Hull, East Yorkshire, HU7 0AE > Tel: 01482 828000 / Fax: 01482 373100 > Daniel.Knaggs at realitysolutions.co.uk > www.realitysolutions.co.uk **** > ------------------------------ > > Sage Accredited Business Partner serving businesses in Yorkshire & > Lincolnshire **** > > [image: Description: cid:image27a71e.PNG at c2da8488.4683bff1]**** > > **** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version.**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/428b9984/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 22463 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/428b9984/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 69075 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/428b9984/attachment-0003.png From msc at freeswitch.org Sun Mar 25 05:25:43 2012 From: msc at freeswitch.org (Michael Collins) Date: Sat, 24 Mar 2012 18:25:43 -0700 Subject: [Freeswitch-users] no DTMF detection over VoIP In-Reply-To: References: Message-ID: Neil, Howdy! Long time no chat. :) Regarding SIP/VoIP: A SIP call will not come in over a PRI circuit. SIP is VoIP and PRI is TDM - two totally different methods for carrying telephone calls. My guess is that you might have a bit of confusion about where and how calls are coming in. Let's start with the basics: Do you have a SIP/VoIP provider? If so, did they supply you with phone number(s) for dialing in to your system? Have you gone through the configuration process to make sure your FS box is properly communicating with the provider? If you need some help you can contact me off list. Of course, this means you'll have to take me to that Mexican food place the next time I'm in Frisco! :) -MC On Sat, Mar 24, 2012 at 11:18 AM, Neil Patel wrote: > I have not modified the sip_profiles directory from standard. There is no > variable set for dtmf-type in either internal.xml or external.xml. > > I did a trace using: > sofia global siptrace on > > There wer no SIP packets logged. If the SIP call is going to a PRI profile > (freetdm), will there still be SIP traffic? > > Thanks, > Neil > > > On Sat, Mar 24, 2012 at 7:22 PM, Anita Hall wrote: > >> What is your config for DTMF in sip profile ? >> >> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#dtmf-type >> >> Use tshark to generate a SIP trace and paste it on >> http://pastebin.freeswitch.org/ and give the link here. >> >> regards, >> Anita >> >> >> >> On Sat, Mar 24, 2012 at 2:43 PM, Neil Patel wrote: >> >>> Hi All, >>> >>> I have a basic IVR application in Lua connected to a PRI line. It >>> currently is not responding to DTMF input given from any VoIP call (e.g. >>> Skype). However, it accepts input from local mobile or landline calls just >>> fine. >>> >>> I recently pulled latest from git and built; before that, DTMF detection >>> from both were working. Is there something I need to configure to allow >>> DTMF for VoIP? >>> >>> Thanks, >>> Neil >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/7333e5a0/attachment.html From luis.daniel.lucio at gmail.com Sun Mar 25 05:32:48 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Sat, 24 Mar 2012 21:32:48 -0400 Subject: [Freeswitch-users] Packing for Mageia Message-ID: Helo, I'm trying to pack FS for Mageia Linux, how ever i've some problems understanding release numbers. As my readings, wiki says that always get latest git snapshot as it is stable. But i wonder to know how to label releases, i mean 1.0.0, or 1.0.7 or 1.1beta2 ? Can you help me on where i can get this number from git? Regards, LD From gabe at gundy.org Sun Mar 25 06:04:42 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 24 Mar 2012 20:04:42 -0600 Subject: [Freeswitch-users] Packing for Mageia In-Reply-To: References: Message-ID: On Sat, Mar 24, 2012 at 7:32 PM, Luis Daniel Lucio Quiroz wrote: > I'm trying to pack FS for Mageia Linux, how ever i've some problems > understanding release numbers. ?As my readings, wiki says that always > get latest git snapshot as it is stable. ?But i wonder to know how to > label releases, i mean 1.0.0, or 1.0.7 or 1.1beta2 ? First, thanks for your efforts to bring FreeSWITCH to Mageia Linux users. They'll love you for it ;) Currently, there is no tagged version that is worth packaging. For now, you should package the latest git and do some sort of time-based updates. If you wait a bit, that's going to change when they tag 1.2. See the mailing list for more info. Best, Gabe From gabe at gundy.org Sun Mar 25 06:08:29 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 24 Mar 2012 20:08:29 -0600 Subject: [Freeswitch-users] Large delay/latency when bridging SIP calls In-Reply-To: References: Message-ID: On Fri, Mar 23, 2012 at 4:42 AM, Tor Petterson wrote: > I am using Freeswitch in a phone center application where I first call > an agent, then call a lead and then bridge the two calls. > If I use a version after Feb 27. it takes about 5 seconds after > bridging before any sound gets through. > Here is an anonymized version of my sofia profile: I think it's fair to say that we're going to need logs before we can help. SIP traces etc. Best, Gabe From gabe at gundy.org Sun Mar 25 06:11:05 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 24 Mar 2012 20:11:05 -0600 Subject: [Freeswitch-users] DTMF detection In-Reply-To: References: Message-ID: On Fri, Mar 23, 2012 at 8:07 AM, Gregor Nanger wrote: > Is it possible to detect DTMF tones on leg A when call is not answered yet > (it is in ringing state)? There are a lot of variables here... how are you sending DTMF? When is the media being set up? Also, it seems like you might be able to test this yourself and let us know what you find. With some logs of what you're seeing, we might be able to tell you where to look for answers. Good luck! Best, Gabe From gabe at gundy.org Sun Mar 25 06:19:03 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 24 Mar 2012 20:19:03 -0600 Subject: [Freeswitch-users] embedded hardware In-Reply-To: <20120322195103.146a357e@anubis.defcon1> References: <20120322195103.146a357e@anubis.defcon1> Message-ID: On Thu, Mar 22, 2012 at 12:51 PM, Bzzz wrote: > I'm almost sure this kind of hardware: > http://versalogic.com/products/DS.asp?ProductID=207 > could withstand ~ 100 simultaneous calls and more, but > what if 10% to 25% are transcoded together (lets say > from iLBC to G711)? ? With what RAM Qty? (512MB or 1GB) > > I'd like to build a small but powerful fit-anywhere-PBX, > but I don't know anything about this kind of embedded card. We could all tell you what we think, but there are no hard-and-fast rules about how it will perform. There are *way* too many variables. I'm afraid there is only one rule you can trust... you try, you know :) Please consider adding your findings here: http://wiki.freeswitch.org/wiki/Real-world_results Good luck! Best, Gabe From gabe at gundy.org Sun Mar 25 06:20:49 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 24 Mar 2012 20:20:49 -0600 Subject: [Freeswitch-users] Mod_CDR and FIFO In-Reply-To: References: Message-ID: On Wed, Mar 21, 2012 at 12:44 PM, Vishal Kakkar wrote: > I thought it should be widely used scenario. Please help if i am doing > anything wrong. Can you pastebin some of those CDRs? Show exactly where you think they're messed up. Thanks, Gabe From brian at freeswitch.org Sun Mar 25 06:24:23 2012 From: brian at freeswitch.org (Brian West) Date: Sat, 24 Mar 2012 21:24:23 -0500 Subject: [Freeswitch-users] Mod_CDR and FIFO In-Reply-To: References: Message-ID: <-7633170388307147222@unknownmsgid> I'm going to guess its because it looks like the agent is calling the person in the queue, because technically that is what is taking place. Sent from my iPad On Mar 24, 2012, at 9:22 PM, Gabriel Gunderson wrote: > On Wed, Mar 21, 2012 at 12:44 PM, Vishal Kakkar wrote: >> I thought it should be widely used scenario. Please help if i am doing >> anything wrong. > > Can you pastebin some of those CDRs? Show exactly where you think > they're messed up. > > > Thanks, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gabe at gundy.org Sun Mar 25 06:24:41 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 24 Mar 2012 20:24:41 -0600 Subject: [Freeswitch-users] [OT] sflphone and tls In-Reply-To: <20120319222529.0206f70c@anubis.defcon1> References: <20120319222529.0206f70c@anubis.defcon1> Message-ID: On Mon, Mar 19, 2012 at 3:25 PM, Bzzz wrote: > Sorry for this off-topic, but I can't find any reliable doc on how > to configure sflphone to work w/ tls and freeswitch. Did you get this figured out? BTW, I didn't know about sflphone. Do you like it? I've been pretty happy with http://icanblink.com/ Gabe From lazyvirus at gmx.com Sun Mar 25 06:34:59 2012 From: lazyvirus at gmx.com (Bzzz) Date: Sun, 25 Mar 2012 04:34:59 +0200 Subject: [Freeswitch-users] embedded hardware In-Reply-To: References: <20120322195103.146a357e@anubis.defcon1> Message-ID: <20120325043459.0fcba539@anubis.defcon1> On Sat, 24 Mar 2012 20:19:03 -0600 Gabriel Gunderson wrote: > > We could all tell you what we think, but there are no hard-and-fast > rules about how it will perform. There are *way* too many variables. > I'm afraid there is only one rule you can trust... you try, you know > :) Yeah, after reading a lot, that was also my conclusion. > Please consider adding your findings here: > http://wiki.freeswitch.org/wiki/Real-world_results Thanks for the link. As my goal is being able to serve ~5-100 UAs w/ an unknown number of transcodings, I think I'll stick to my first idea: a Mini-ITX card w/ an Atom bi-core and 2GB RAM (4?) JY -- printk("ufs_read_super: fucking Sun blows me\n"); -- /usr/src/linux/fs/ufs/ufs_super.c From gabe at gundy.org Sun Mar 25 06:35:01 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 24 Mar 2012 20:35:01 -0600 Subject: [Freeswitch-users] Context problem with xml_curl directory In-Reply-To: <4F64E723.5090604@integrafin.co.uk> References: <4F64E723.5090604@integrafin.co.uk> Message-ID: On Sat, Mar 17, 2012 at 1:33 PM, Alex Crow wrote: > I have attached the PHP and a tcpdump trace in case anyone can see something > obviously incorrect. Make it easy on us by using plain text and putting it in pastebin. Boil it down to the simplest set of info. It doesn't really matter how you got the response (so no need for PHP). And it's not always convenient to read tcpdumps. Anyway, are you authing each call? I don't think it's enough to auth the SUA for registration and expect that those vars will be there. While I don't use that feature often, I'm pretty sure you have to auth each call for those variables to stick. Let us know what you find. Best, Gabe From gabe at gundy.org Sun Mar 25 06:41:45 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 24 Mar 2012 20:41:45 -0600 Subject: [Freeswitch-users] Strange gw behavior In-Reply-To: <531341332008393@web134.yandex.ru> References: <531341332008393@web134.yandex.ru> Message-ID: On Sat, Mar 17, 2012 at 12:19 PM, Serge S. Yuriev wrote: > When I register GW 'multifon' it breaks all outside calling - this GW is never used in DP as outbound BUT all calls trying to flow trough it with attributes from other GWs! Please take a little more time to explain what you're seeing. Try to be as clear as you can. We all want to help, but it's hard to take the time needed to find the problem in 800+ lines of logs when we don't really know what we're looking for. Also, since this is about GWs, consider posting your sofia configs and perhaps the dialplan snipped used to hit these GWs. Thanks! Best, Gabe From lazyvirus at gmx.com Sun Mar 25 06:47:21 2012 From: lazyvirus at gmx.com (Bzzz) Date: Sun, 25 Mar 2012 04:47:21 +0200 Subject: [Freeswitch-users] [OT] sflphone and tls In-Reply-To: References: <20120319222529.0206f70c@anubis.defcon1> Message-ID: <20120325044721.03ced2b7@anubis.defcon1> On Sat, 24 Mar 2012 20:24:41 -0600 Gabriel Gunderson wrote: > On Mon, Mar 19, 2012 at 3:25 PM, Bzzz wrote: > > Sorry for this off-topic, but I can't find any reliable doc on how > > to configure sflphone to work w/ tls and freeswitch. > > Did you get this figured out? No, I turned to jitsi (ex sip-communicator), which is very complete but has a big drawback: when online, it draws 100% of CPU on a celeron 2.0GHz, and 74% on an AthlonXP-2600+; that lead to choppy sound:( My favorite one is twinkle; TLS is not supported but ZRTP is, and it takes a ridiculous amount of CPU for a very good sound quality. > BTW, I didn't know about sflphone. Do you like it? I've been pretty > happy with http://icanblink.com/ I tested & rejected it: on both squeeze & sid it draws 100% CPU full time - too bad as it was looking interesting. JY -- Reality is a cop-out for people who can't handle drugs. From gabe at gundy.org Sun Mar 25 06:51:21 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 24 Mar 2012 20:51:21 -0600 Subject: [Freeswitch-users] [OT] sflphone and tls In-Reply-To: <20120325044721.03ced2b7@anubis.defcon1> References: <20120319222529.0206f70c@anubis.defcon1> <20120325044721.03ced2b7@anubis.defcon1> Message-ID: On Sat, Mar 24, 2012 at 8:47 PM, Bzzz wrote: >> BTW, I didn't know about sflphone. Do you like it? I've been pretty >> happy with http://icanblink.com/ > > I tested & rejected it: on both squeeze & sid it draws 100% CPU full > time - too bad as it was looking interesting. Strange... works great for me. gabe at work:~$ cat /etc/issue Ubuntu 11.10 \n \l Best, Gabe From lazyvirus at gmx.com Sun Mar 25 07:01:52 2012 From: lazyvirus at gmx.com (Bzzz) Date: Sun, 25 Mar 2012 05:01:52 +0200 Subject: [Freeswitch-users] [OT] sflphone and tls In-Reply-To: References: <20120319222529.0206f70c@anubis.defcon1> <20120325044721.03ced2b7@anubis.defcon1> Message-ID: <20120325050152.69ae1661@anubis.defcon1> On Sat, 24 Mar 2012 20:51:21 -0600 Gabriel Gunderson wrote: > > Strange... works great for me. > > gabe at work:~$ cat /etc/issue > Ubuntu 11.10 \n \l Yeah, I don't know where it can come from, and I can't find such issue on the web, except for older versions. JY -- From curriegrad2004 at gmail.com Sun Mar 25 07:10:34 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sat, 24 Mar 2012 20:10:34 -0700 Subject: [Freeswitch-users] no DTMF detection over VoIP In-Reply-To: References: Message-ID: If the call is coming from a TDM circuit, you'll have to do in-band DTMF detection in that situation. On Sat, Mar 24, 2012 at 6:25 PM, Michael Collins wrote: > Neil, > > Howdy! Long time no chat. :) > > Regarding SIP/VoIP: A SIP call will not come in over a PRI circuit. SIP is > VoIP and PRI is TDM - two totally different methods for carrying telephone > calls. My guess is that you might have a bit of confusion about where and > how calls are coming in. Let's start with the basics: Do you have a SIP/VoIP > provider? If so, did they supply you with phone number(s) for dialing in to > your system? Have you gone through the configuration process to make sure > your FS box is properly communicating with the provider? > > If you need some help you can contact me off list. Of course, this means > you'll have to take me to that Mexican food place the next time I'm in > Frisco! :) > > -MC > > > On Sat, Mar 24, 2012 at 11:18 AM, Neil Patel wrote: >> >> I have not modified the sip_profiles directory from standard. There is no >> variable set for dtmf-type in either internal.xml or external.xml. >> >> I did a trace using: >> sofia global siptrace on >> >> There wer no SIP packets logged. If the SIP call is going to a PRI profile >> (freetdm), will there still be SIP traffic? >> >> Thanks, >> Neil >> >> >> On Sat, Mar 24, 2012 at 7:22 PM, Anita Hall >> wrote: >>> >>> What is your config for DTMF in sip profile ? >>> >>> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#dtmf-type >>> >>> Use tshark to generate a SIP trace and paste it on >>> http://pastebin.freeswitch.org/ and give the link here. >>> >>> regards, >>> Anita >>> >>> >>> >>> On Sat, Mar 24, 2012 at 2:43 PM, Neil Patel >>> wrote: >>>> >>>> Hi All, >>>> >>>> I have a basic IVR application in Lua connected to a PRI line. It >>>> currently is not responding to DTMF input given from any VoIP call (e.g. >>>> Skype). However, it accepts input from local mobile or landline calls just >>>> fine. >>>> >>>> I recently pulled latest from git and built; before that, DTMF detection >>>> from both were working. Is there something I need to configure to allow DTMF >>>> for VoIP? >>>> >>>> Thanks, >>>> Neil >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From curriegrad2004 at gmail.com Sun Mar 25 07:14:05 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sat, 24 Mar 2012 20:14:05 -0700 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: References: Message-ID: Those detected tones could be something that the TDM provider didn't filter out for some odd reason. With E&M signalling on TDM links you may hear a squealing tone in the background. On Sat, Mar 24, 2012 at 6:20 PM, Michael Collins wrote: > I'd do an audio recording of the call and open the resulting file in > Audacity. It can do kind of a spectrum analysis to show you if there are > any frequencies that might be fooling the dtmf detector. Also, I'm not > familiar with using Digium cards with FreeTDM. (I have a Digium TE121 or > similar but I don't happen to have any mobo's with the PCIe or whatever > slot type it uses.) > > To Moises I'd ask: is there hardware DTMF detection in the Digium cards > that you know of? Also, any known or suspected issues with false DTMF > detection in the scenario mentioned by the OP? > > Gracias, > MC > > > On Sat, Mar 24, 2012 at 5:27 AM, Daniel Knaggs < > Daniel.Knaggs at realitysolutions.co.uk> wrote: > >> They aren?t any faults that I can see.**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of * >> curriegrad2004 >> *Sent:* 23 March 2012 23:23 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls**** >> >> ** ** >> >> Have you checked for any faults on your TDM link on your side?**** >> >> On Fri, Mar 23, 2012 at 10:32 AM, Brian Foster >> wrote:**** >> >> Please update to latest git.**** >> >> ** ** >> >> -BDF**** >> >> On Fri, Mar 23, 2012 at 7:19 AM, Daniel Knaggs < >> Daniel.Knaggs at realitysolutions.co.uk> wrote:**** >> >> Hello all,**** >> >> **** >> >> Got a bit of a strange one, we appear to be getting DTMF tones on >> incoming calls when the caller hasn?t even pressed any keys.**** >> >> **** >> >> It normally happens with 10 seconds or so after the call has been >> answered.**** >> >> **** >> >> **** >> >> Here is the log of it happening earlier: -**** >> >> **** >> >> 2012-03-23 10:38:16.852654 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing >> DTMF A (debug = 0)**** >> >> 2012-03-23 10:38:16.852654 [DEBUG] mod_freetdm.c:799 Queuing DTMF [A] in >> channel FreeTDM/1:2/000 device 1:2**** >> >> 2012-03-23 10:38:16.915653 [DEBUG] switch_rtp.c:2420 Send start packet >> for [A] ts=49440 dur=160/160/2000 seq=46803 lw=49440**** >> >> 2012-03-23 10:38:16.936653 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=320/320/2000 seq=46804 lw=49600**** >> >> 2012-03-23 10:38:16.957652 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=480/480/2000 seq=46805 lw=49760**** >> >> 2012-03-23 10:38:16.978652 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=640/640/2000 seq=46806 lw=49920**** >> >> 2012-03-23 10:38:16.999652 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=800/800/2000 seq=46807 lw=50080**** >> >> 2012-03-23 10:38:17.020651 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=960/960/2000 seq=46808 lw=50240**** >> >> 2012-03-23 10:38:17.041651 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=1120/1120/2000 seq=46809 lw=50400**** >> >> 2012-03-23 10:38:17.062651 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=1280/1280/2000 seq=46810 lw=50560**** >> >> 2012-03-23 10:38:17.083650 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=1440/1440/2000 seq=46811 lw=50720**** >> >> 2012-03-23 10:38:17.104650 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=1600/1600/2000 seq=46812 lw=50880**** >> >> 2012-03-23 10:38:17.125650 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=1760/1760/2000 seq=46813 lw=51040**** >> >> 2012-03-23 10:38:17.146650 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [A] ts=49440 dur=1920/1920/2000 seq=46814 lw=51200**** >> >> 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for >> [A] ts=49440 dur=2080/2080/2000 seq=46815 lw=51200**** >> >> 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for >> [A] ts=49440 dur=2080/2080/2000 seq=46816 lw=51200**** >> >> 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for >> [A] ts=49440 dur=2080/2080/2000 seq=46817 lw=51200**** >> >> 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2271 Queue digit delay of >> 40ms**** >> >> 2012-03-23 10:38:18.070638 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing >> DTMF B (debug = 0)**** >> >> 2012-03-23 10:38:18.070638 [DEBUG] mod_freetdm.c:799 Queuing DTMF [B] in >> channel FreeTDM/1:2/000 device 1:2**** >> >> 2012-03-23 10:38:18.133637 [DEBUG] switch_rtp.c:2420 Send start packet >> for [B] ts=59040 dur=160/160/2000 seq=46864 lw=59040**** >> >> 2012-03-23 10:38:18.154637 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=320/320/2000 seq=46865 lw=59200**** >> >> 2012-03-23 10:38:18.175637 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=480/480/2000 seq=46866 lw=59360**** >> >> 2012-03-23 10:38:18.196636 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=640/640/2000 seq=46867 lw=59520**** >> >> 2012-03-23 10:38:18.217636 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=800/800/2000 seq=46868 lw=59680**** >> >> 2012-03-23 10:38:18.238636 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=960/960/2000 seq=46869 lw=59840**** >> >> 2012-03-23 10:38:18.259636 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=1120/1120/2000 seq=46870 lw=60000**** >> >> 2012-03-23 10:38:18.280636 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=1280/1280/2000 seq=46871 lw=60160**** >> >> 2012-03-23 10:38:18.301635 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=1440/1440/2000 seq=46872 lw=60320**** >> >> 2012-03-23 10:38:18.322635 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=1600/1600/2000 seq=46873 lw=60480**** >> >> 2012-03-23 10:38:18.343634 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=1760/1760/2000 seq=46874 lw=60640**** >> >> 2012-03-23 10:38:18.364634 [DEBUG] switch_rtp.c:2323 Send middle packet >> for [B] ts=59040 dur=1920/1920/2000 seq=46875 lw=60800**** >> >> 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for >> [B] ts=59040 dur=2080/2080/2000 seq=46876 lw=60800**** >> >> 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for >> [B] ts=59040 dur=2080/2080/2000 seq=46877 lw=60800**** >> >> 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for >> [B] ts=59040 dur=2080/2080/2000 seq=46878 lw=60800**** >> >> **** >> >> **** >> >> I?m very sure the caller isn?t dialling ?A? or ?B?!**** >> >> **** >> >> Running on ISDN via a TE121 card using E1 (euroisdn). Freeswitch version >> is ?FreeSWITCH Version 1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)?. >> **** >> >> **** >> >> **** >> >> Currently there is a ?bind_meta_app? in the config which binds a script >> on the B leg of the call which parks the call ? I haven?t tried turning >> this off yet to see if it?s this.**** >> >> **** >> >> **** >> >> Wondering if anyone has any ideas or has come across this before?**** >> >> **** >> >> **** >> >> Thanks in advance.**** >> >> [image: Description: cid:imageacd695.PNG at 9e92e461.40aa4b96]**** >> >> *Daniel Knaggs***** >> >> Software Developer**** >> >> >> >> **** >> >> Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston >> upon Hull, East Yorkshire, HU7 0AE >> Tel: 01482 828000 / Fax: 01482 373100 >> Daniel.Knaggs at realitysolutions.co.uk >> www.realitysolutions.co.uk **** >> ------------------------------ >> >> Sage Accredited Business Partner serving businesses in Yorkshire & >> Lincolnshire **** >> >> [image: Description: cid:image27a71e.PNG at c2da8488.4683bff1]**** >> >> **** >> >> ** ** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> >> >> **** >> >> ** ** >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version.**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> ** ** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 69075 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120324/94b5eefd/attachment-0003.png From luis.daniel.lucio at gmail.com Sun Mar 25 07:30:15 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Sat, 24 Mar 2012 23:30:15 -0400 Subject: [Freeswitch-users] Packing for Mageia In-Reply-To: References: Message-ID: Le 24 mars 2012 22:04, Gabriel Gunderson a ?crit : > On Sat, Mar 24, 2012 at 7:32 PM, Luis Daniel Lucio Quiroz > wrote: >> I'm trying to pack FS for Mageia Linux, how ever i've some problems >> understanding release numbers. ?As my readings, wiki says that always >> get latest git snapshot as it is stable. ?But i wonder to know how to >> label releases, i mean 1.0.0, or 1.0.7 or 1.1beta2 ? > > First, thanks for your efforts to bring FreeSWITCH to Mageia Linux > users. They'll love you for it ;) > > Currently, there is no tagged version that is worth packaging. For > now, you should package the latest git and do some sort of time-based > updates. ?If you wait a bit, that's going to change when they tag 1.2. > ?See the mailing list for more info. > > > Best, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Thanks Gabriel, as i was reviewing GIT i relilize 2 things Changelog file says it is 1.0.7, but configure.in says 1.1.beta1 also in files.freeswitch.org i found rpms saying 1.1.beta2 dated march 7 So, according your recomendations what release do you recommend me? From brian at freeswitch.org Sun Mar 25 07:36:00 2012 From: brian at freeswitch.org (Brian West) Date: Sat, 24 Mar 2012 22:36:00 -0500 Subject: [Freeswitch-users] Packing for Mageia In-Reply-To: References: Message-ID: <8981939843652678481@unknownmsgid> Peas and carrots, peas and carrots? Stay tuned we will have such a thing soonish! Yours Truely, Lolly pop Sent from my eyePad On Mar 24, 2012, at 10:32 PM, Luis Daniel Lucio Quiroz wrote: > Changelog file says it is 1.0.7, but > configure.in says 1.1.beta1 > > also in files.freeswitch.org i found rpms saying 1.1.beta2 dated march 7 > > So, according your recomendations what release do you recommend me? From anton.jugatsu at gmail.com Sun Mar 25 08:39:48 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Sun, 25 Mar 2012 08:39:48 +0400 Subject: [Freeswitch-users] Grandstream 4104 In-Reply-To: <4F6C46B0.7080805@elios.net> References: <4F6C46B0.7080805@elios.net> Message-ID: First, after you paste all siptraces, try to initiate a call with originate command originate sofia/internal/XXXX at 192.168.0.3 9178 23.03.2012 19:13 ???????????? "a.avona" ???????: > Hi, all > i have a GrandStream 4104 perfectly working (both incoming and outgoing > calls work great) with an asterisk pbx. > We are trying to change the asterisk with freeswitch, but i'm > exeperiencing problems in configuring freeswitch for outgoing calls, > incoming calls works well. > this is my configuration and what i obtain in fs_cli consolle > > Can someone tell me where i'm wrong? > > in sip_profile/internal i created a file 00_to_pstn.xml this way > > > > > > > > > > > > > > > in dialplan/default i created a file 00_to_pstn.xml this way > > > > > data="effective_caller_id_number=0321234567"/> > data="effective_caller_id_name=ThisIsMyCompany"/> > > > data="sofia/internal/gxw4104-fxo1/$1 at 192.168.0.3:5060"/> > > > > > > > > _________________________________________________________________ > > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:7559 IP 192.168.0.200 > Rejected by acl "domains". Falling back to Digest auth. > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:7559 IP 192.168.0.200 > Rejected by acl "domains". Falling back to Digest auth. > 2012-03-23 10:38:43.937574 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/1000 at 192.168.0.2 [fb33155c-74cb-11e1-9063-fb6a5f4599fb] > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:5526 Channel > sofia/internal/1000 at 192.168.0.2 entering state [received][100] > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:5537 Remote SDP: > v=0 > o=- 935122583 0 IN IP4 192.168.0.200 > s=SIPPER for PhonerLite > c=IN IP4 192.168.0.200 > t=0 0 > m=audio 5062 RTP/AVP 8 3 0 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:2991 Set Codec > sofia/internal/1000 at 192.168.0.2 PCMA/8000 20 ms 160 samples 64000 bits > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_codec.c:111 > sofia/internal/1000 at 192.168.0.2 Original read codec set to PCMA:8 > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4995 Set 2833 dtmf > send/recv payload to 101 > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:5749 > (sofia/internal/1000 at 192.168.0.2) State Change CS_NEW -> CS_INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1000 at 192.168.0.2) State INIT > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:85 > sofia/internal/1000 at 192.168.0.2 SOFIA INIT > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:125 > (sofia/internal/1000 at 192.168.0.2) State Change CS_INIT -> CS_ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1000 at 192.168.0.2) State INIT going to sleep > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1886 > (sofia/internal/1000 at 192.168.0.2) Callstate Change DOWN -> RINGING > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1000 at 192.168.0.2) State ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:148 > sofia/internal/1000 at 192.168.0.2 SOFIA ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/1000 at 192.168.0.2 Standard ROUTING > 2012-03-23 10:38:43.937574 [INFO] mod_dialplan_xml.c:485 Processing 1000 > <1000>->339XXXXXXX in context default > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unloop] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->tod_example] > continue=true > Dialplan: sofia/internal/1000 at 192.168.0.2 Date/Time Match (PASS) > [tod_example] break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(open=true) > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->holiday_example] continue=true > Dialplan: sofia/internal/1000 at 192.168.0.2 Date/TimeMatch (FAIL) > [holiday_example] break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->global-intercept] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [global-intercept] destination_number(339XXXXXXX) =~ /^886$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group-intercept] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group-intercept] > destination_number(339XXXXXXX) =~ /^\*8$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->intercept-ext] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [intercept-ext] > destination_number(339XXXXXXX) =~ /^\*\*(\d+)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->redial] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [redial] > destination_number(339XXXXXXX) =~ /^(redial|870)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->global] > continue=true > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] > ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] > ${sip_has_crypto}() =~ > /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never > Dialplan: sofia/internal/1000 at 192.168.0.2 Absolute Condition [global] > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->snom-demo-2] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-2] > destination_number(339XXXXXXX) =~ /^9001$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->snom-demo-1] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-1] > destination_number(339XXXXXXX) =~ /^9000$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] > destination_number(339XXXXXXX) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] > destination_number(339XXXXXXX) =~ /^779$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->call_return] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call_return] > destination_number(339XXXXXXX) =~ /^\*69$|^869$|^lcr$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->del-group] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [del-group] > destination_number(339XXXXXXX) =~ /^80(\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->add-group] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [add-group] > destination_number(339XXXXXXX) =~ /^81(\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->call-group-simo] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call-group-simo] > destination_number(339XXXXXXX) =~ /^82(\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->call-group-order] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [call-group-order] destination_number(339XXXXXXX) =~ /^83(\d{2})$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->extension-intercom] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [extension-intercom] destination_number(339XXXXXXX) =~ > /^8(10[01][0-9])$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->Local_Extension] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [Local_Extension] > destination_number(339XXXXXXX) =~ /^(10[01][0-9])$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->Local_Extension_Skinny] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [Local_Extension_Skinny] destination_number(339XXXXXXX) =~ > /^(11[01][0-9])$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group_dial_sales] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [group_dial_sales] destination_number(339XXXXXXX) =~ /^2000$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group_dial_support] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [group_dial_support] destination_number(339XXXXXXX) =~ /^2001$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group_dial_billing] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [group_dial_billing] destination_number(339XXXXXXX) =~ /^2002$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->operator] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [operator] > destination_number(339XXXXXXX) =~ /^(operator|0)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->vmain] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [vmain] > destination_number(339XXXXXXX) =~ /^vmain$|^4000$|^\*98$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->sip_uri] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [sip_uri] > destination_number(339XXXXXXX) =~ /^sip:(.*)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->nb_conferences] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [nb_conferences] > destination_number(339XXXXXXX) =~ /^(30\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->wb_conferences] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wb_conferences] > destination_number(339XXXXXXX) =~ /^(31\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->uwb_conferences] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [uwb_conferences] > destination_number(339XXXXXXX) =~ /^(32\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->cdquality_conferences] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [cdquality_conferences] destination_number(339XXXXXXX) =~ /^(33\d{2})$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->freeswitch_public_conf_via_sip] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [freeswitch_public_conf_via_sip] destination_number(339XXXXXXX) =~ > /^9(888|8888|1616|3232)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->mad_boss_intercom] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [mad_boss_intercom] destination_number(339XXXXXXX) =~ /^0911$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->mad_boss_intercom] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [mad_boss_intercom] destination_number(339XXXXXXX) =~ /^0912$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->mad_boss] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [mad_boss] > destination_number(339XXXXXXX) =~ /^0913$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ivr_demo] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ivr_demo] > destination_number(339XXXXXXX) =~ /^5000$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->dynamic_conference] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [dynamic_conference] destination_number(339XXXXXXX) =~ /^5001$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->rtp_multicast_page] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [rtp_multicast_page] destination_number(339XXXXXXX) =~ > /^pagegroup$|^7243$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] > destination_number(339XXXXXXX) =~ /^5900$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] > destination_number(339XXXXXXX) =~ /^5901$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->valet_park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] > destination_number(339XXXXXXX) =~ /^(6000)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->valet_park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] > destination_number(339XXXXXXX) =~ /^(60\d[1-9])$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] > destination_number(339XXXXXXX) =~ /park\+(\d+)/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] > destination_number(339XXXXXXX) =~ /^parking$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] > destination_number(339XXXXXXX) =~ /callpark/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] > destination_number(339XXXXXXX) =~ /pickup/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->wait] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wait] > destination_number(339XXXXXXX) =~ /^wait$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->fax_receive] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_receive] > destination_number(339XXXXXXX) =~ /^9178$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->fax_transmit] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_transmit] > destination_number(339XXXXXXX) =~ /^9179$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_180] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_180] > destination_number(339XXXXXXX) =~ /^9180$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_183_uk_ring] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_183_uk_ring] destination_number(339XXXXXXX) =~ /^9181$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_183_music_ring] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_183_music_ring] destination_number(339XXXXXXX) =~ /^9182$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_post_answer_uk_ring] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_post_answer_uk_ring] destination_number(339XXXXXXX) =~ > /^9183$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_post_answer_music] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_post_answer_music] destination_number(339XXXXXXX) =~ /^9184$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ClueCon] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ClueCon] > destination_number(339XXXXXXX) =~ /^9191$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->show_info] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [show_info] > destination_number(339XXXXXXX) =~ /^9192$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->video_record] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_record] > destination_number(339XXXXXXX) =~ /^9193$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->video_playback] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_playback] > destination_number(339XXXXXXX) =~ /^9194$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->delay_echo] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [delay_echo] > destination_number(339XXXXXXX) =~ /^9195$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->echo] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [echo] > destination_number(339XXXXXXX) =~ /^9196$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->milliwatt] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [milliwatt] > destination_number(339XXXXXXX) =~ /^9197$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->tone_stream] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [tone_stream] > destination_number(339XXXXXXX) =~ /^9198$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->zrtp_enrollement] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [zrtp_enrollement] destination_number(339XXXXXXX) =~ /^9787$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->hold_music] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [hold_music] > destination_number(339XXXXXXX) =~ /^9664$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->from_pstn] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [from_pstn] > destination_number(339XXXXXXX) =~ /^0000$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->101] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [101] > destination_number(339XXXXXXX) =~ /^101$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->pizza_demo] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [pizza_demo] > destination_number(339XXXXXXX) =~ /^(pizza|74992)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->gxw4104-fxo-local] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) > [gxw4104-fxo-local] ${toll_allow}(domestic,international,local) =~ > /local/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) > [gxw4104-fxo-local] destination_number(339XXXXXXX) =~ /^(\d{6,})$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > set(effective_caller_id_number=0321234567) > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > set(effective_caller_id_name=ThisIsMyCompany) > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > set(ignore_early_media=ring_ready) > Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(ringback=${us-ring}) > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > bridge(sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/1000 at 192.168.0.2) State Change CS_ROUTING -> CS_EXECUTE > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1000 at 192.168.0.2) State ROUTING going to sleep > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_EXECUTE > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/1000 at 192.168.0.2) State EXECUTE > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:241 > sofia/internal/1000 at 192.168.0.2 SOFIA EXECUTE > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/1000 at 192.168.0.2 Standard EXECUTE > EXECUTE sofia/internal/1000 at 192.168.0.2 set(open=true) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET [open]=[true] > EXECUTE sofia/internal/1000 at 192.168.0.2 > hash(insert/192.168.0.2-spymap/1000/fb33155c-74cb-11e1-9063-fb6a5f4599fb) > EXECUTE sofia/internal/1000 at 192.168.0.2 > hash(insert/192.168.0.2-last_dial/1000/339XXXXXXX) > EXECUTE sofia/internal/1000 at 192.168.0.2 > > hash(insert/192.168.0.2-last_dial/global/fb33155c-74cb-11e1-9063-fb6a5f4599fb) > EXECUTE sofia/internal/1000 at 192.168.0.2 export(RFC2822_DATE=Fri, 23 Mar > 2012 10:38:43 +0100) > 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1093 EXPORT > (export_vars) [RFC2822_DATE]=[Fri, 23 Mar 2012 10:38:43 +0100] > EXECUTE sofia/internal/1000 at 192.168.0.2 > set(effective_caller_id_number=0321234567) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET > [effective_caller_id_number]=[0321234567] > EXECUTE sofia/internal/1000 at 192.168.0.2 > set(effective_caller_id_name=ThisIsMyCompany) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET > [effective_caller_id_name]=[ThisIsMyCompany] > EXECUTE sofia/internal/1000 at 192.168.0.2 set(ignore_early_media=ring_ready) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET [ignore_early_media]=[ring_ready] > EXECUTE sofia/internal/1000 at 192.168.0.2 set(ringback=%(2000,4000,440,480)) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET [ringback]=[%(2000,4000,440,480)] > EXECUTE sofia/internal/1000 at 192.168.0.2 > bridge(sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) > 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1047 > sofia/internal/1000 at 192.168.0.2 EXPORTING[export_vars] > [RFC2822_DATE]=[Fri, 23 Mar 2012 10:38:43 +0100] to event > 2012-03-23 10:38:43.937574 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2012-03-23 10:38:43.937574 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [fb34cf5a-74cb-11e1-9068-fb6a5f4599fb] > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:4691 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change > CS_NEW -> CS_INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State > Change CS_INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State INIT > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:85 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 SOFIA INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:125 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change > CS_INIT -> CS_ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State INIT > going to sleep > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State > Change CS_ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1886 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Callstate > Change DOWN -> RINGING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State ROUTING > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:148 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 SOFIA ROUTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change > CS_ROUTING -> CS_CONSUME_MEDIA > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State ROUTING > going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State > Change CS_CONSUME_MEDIA > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State > CONSUME_MEDIA > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State > CONSUME_MEDIA going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] sofia.c:5526 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 entering state > [calling][0] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] sofia.c:5526 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 entering state > [terminated][403] > 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2848 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Callstate > Change RINGING -> HANGUP > 2012-03-23 10:38:43.957579 [NOTICE] sofia.c:6293 Hangup > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [CS_CONSUME_MEDIA] [CALL_REJECTED] > 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [KILL] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State > Change CS_HANGUP > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State HANGUP > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 hanging up, > cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 Standard HANGUP, > cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State HANGUP > going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change > CS_HANGUP -> CS_REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State > Change CS_REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 Standard > REPORTING, cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State > REPORTING going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State Change > CS_REPORTING -> CS_DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_ivr_originate.c:3364 Originate > Resulted in Error Cause: 21 [CALL_REJECTED] > 2012-03-23 10:38:43.957579 [INFO] mod_dptools.c:2922 Originate Failed. > Cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2848 > (sofia/internal/1000 at 192.168.0.2) Callstate Change RINGING -> HANGUP > 2012-03-23 10:38:43.957579 [NOTICE] mod_dptools.c:3041 Hangup > sofia/internal/1000 at 192.168.0.2 [CS_EXECUTE] [CALL_REJECTED] > 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/1000 at 192.168.0.2 [KILL] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:2285 > sofia/internal/1000 at 192.168.0.2 skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/1000 at 192.168.0.2) State EXECUTE going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_HANGUP > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1000 at 192.168.0.2) State HANGUP > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:463 > sofia/internal/1000 at 192.168.0.2 Overriding SIP cause 603 with 403 from > the other leg > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/1000 at 192.168.0.2 hanging up, cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:534 Responding to INVITE > with: 403 > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/1000 at 192.168.0.2 Standard HANGUP, cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1000 at 192.168.0.2) State HANGUP going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/1000 at 192.168.0.2) State Change CS_HANGUP -> CS_REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1000 at 192.168.0.2) State REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/1000 at 192.168.0.2 Standard REPORTING, cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1000 at 192.168.0.2) State REPORTING going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/1000 at 192.168.0.2) State Change CS_REPORTING -> CS_DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1380 Session 13 > (sofia/internal/1000 at 192.168.0.2) Locked, Waiting on external entities > 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1398 Session > 13 (sofia/internal/1000 at 192.168.0.2) Ended > 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/1000 at 192.168.0.2 [CS_DESTROY] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/1000 at 192.168.0.2) Callstate Change HANGUP -> DOWN > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1000 at 192.168.0.2) State DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:374 > sofia/internal/1000 at 192.168.0.2 SOFIA DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/1000 at 192.168.0.2 Standard DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1000 at 192.168.0.2) State DESTROY going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1380 Session 14 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Locked, > Waiting on external entities > 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1398 Session > 14 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Ended > 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060[CS_DESTROY] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Callstate > Change HANGUP -> DOWN > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) Running State > Change CS_DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:374 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 SOFIA DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 Standard DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060) State DESTROY > going to sleep > freeswitch at internal> > > Thank's in advance > Regards > Accursio Avona > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120325/b8503608/attachment-0001.html From gerardo.barajas at gmail.com Sun Mar 25 11:29:39 2012 From: gerardo.barajas at gmail.com (Gerardo Barajas) Date: Sun, 25 Mar 2012 01:29:39 -0600 Subject: [Freeswitch-users] Fax on Sangoma In-Reply-To: References: Message-ID: I guess you can also contact Sangoma. They have an excellent support team! Saludos/Regards -- Ing. Gerardo Barajas Puente Ingenier?a | www.neocenter.com T:+52 (55) 8590-9000 x 7003 On Sat, Mar 24, 2012 at 9:22 AM, Anita Hall wrote: > Hi > > My client had been using an old torrenta cards which game problems with > pretty much everything :) After evangelizing the cause of Sangoma for more > than 3 months (yes, they are dumb), they are finally making the transition > so I am pretty excited. The choice of card is A108D which does a whole > bunch of DSP in the hardware. > > Now, the first hurdle that I have to make Sangoma jump across is getting > incoming Fax over E1 right. We are using mod_spandsp, of course. So, here I > will be needing a whole lot of help from the veterans of spandsp and Faxing > :) I desperately need Sangoma to pass the Fax test or they will give me > torrenta cards all over again! > > The primary cause of failures are - (49) The call dropped prematurely and > (48) Disconnected after permitted retries. > > For example, in this case, can I conclude that the other end did not > provide a Fax tone or is it something else? > > 2aeb5f7c-75b5-11e1-8f36-b3286880c45b EXECUTE FreeTDM/4:2/47615728 > rxfax(/srv/fax/in/2aeb5f7c-75b5-11e1-8f36-b3286880c45b.tiff) > 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] > mod_spandsp_fax.c:1357 Raw read codec activation Success L16 20000 > 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] > switch_core_codec.c:216 FreeTDM/4:2/47615728 Push codec L16:70 > 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] > mod_spandsp_fax.c:1373 Raw write codec activation Success L16 > 2012-03-24 18:57:57.844876 [DEBUG] ftmod_wanpipe.c:965 [s4c2][4:2] First > packet read stats: Rx queue len: 1, Rx queue size: 10 > 2012-03-24 18:57:57.904879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 17 > 2012-03-24 18:57:57.924878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 17 > 2012-03-24 18:57:58.104877 [DEBUG] ftmod_wanpipe.c:901 [s4c2][4:2] First > packet write stats: Tx queue len: 1, Tx queue size: 5, Tx idle: 30 > 2012-03-24 18:57:58.444908 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 17 > 2012-03-24 18:57:58.644907 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 17 > 2012-03-24 18:57:58.664878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 1 > 2012-03-24 18:57:58.764879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 17 > 2012-03-24 18:57:58.764879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 17 > 2012-03-24 18:57:58.784878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 17 > 2012-03-24 18:57:58.804879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 17 > 2012-03-24 18:57:59.244886 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 17 > 2012-03-24 18:57:59.464887 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 17 > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 T4 > expired in phase T30_PHASE_B_RX, state 17 > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Retry > number 1 > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 > Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set rx > type 0 > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set tx > type 4 > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Sending > ident 'Sangoma Fax Ident' > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Tx: > CSI without final frame tag > 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Tx: ff > 03 40 74 6e 65 64 49 20 78 61 46 20 61 6d 6f 67 6e 61 53 20 20 20 > 2012-03-24 18:58:00.524877 [DEBUG] ftmod_libpri.c:1065 -- Hangup REQ on > channel 4:1 > 2012-03-24 18:58:00.524877 [DEBUG] ftmod_libpri.c:1078 [s4c1][4:1] Changed > state from UP to TERMINATING > 2012-03-24 18:58:00.524877 [DEBUG] ftdm_state.c:511 [s4c1][4:1] Executing > state processor for TERMINATING > > > I will need some more hand-holding with logs later :) > > > regards, > Anita > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120325/a02bcda1/attachment.html From daggelinckxmichel at gmail.com Sun Mar 25 19:10:40 2012 From: daggelinckxmichel at gmail.com (Michel Daggelinckx) Date: Sun, 25 Mar 2012 17:10:40 +0200 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: <4F6F28F0.4090604@redsleeve.org> References: <4F6F28F0.4090604@redsleeve.org> Message-ID: <4F6F3570.4090608@gmail.com> Our favorite distro goes ARM. This gives a platform that feels like home and should have all our recepies work like a charm. Michel -------- Originele bericht -------- Onderwerp: [CentOS] CentOS for ARM Datum: Sun, 25 Mar 2012 15:17:20 +0100 Van: Gordan Bobic Antwoord-naar: CentOS mailing list Aan: centos-devel at centos.org, centos at centos.org Hi guys, I recently became aware that this is being worked on as of recently. A similar thing already exists, though. Those interested in an ARM port may also be interested in taking a look at RedSleeve Linux, which is an ARM port of the same upstream distribution as CentOS. You can look here for more info: http://www.redsleeve.org/about/ http://www.redsleeve.org/ In total 109 SRPMs had to be modified in order to get them to build and work on ARM. They are in SRPMS/changed directory on the RedSleeve mirror. (Note: About 5 of those 109 were changed in order to remove the upstream branding which isn't relevant to the CentOS effort as it is already taken care of. It is quite well known what those are.) I hope this is of interest to the CentOS ARM port efforts. Or, you could take RedSleeve for a spin, since that is already available. :) Gordan _______________________________________________ CentOS mailing list CentOS at centos.org http://lists.centos.org/mailman/listinfo/centos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120325/7b08b998/attachment.html From curriegrad2004 at gmail.com Sun Mar 25 20:17:18 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 25 Mar 2012 09:17:18 -0700 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: <4F6F3570.4090608@gmail.com> References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com> Message-ID: This sounds intresting. Let's see if they can get things going on there. On Sun, Mar 25, 2012 at 8:10 AM, Michel Daggelinckx < daggelinckxmichel at gmail.com> wrote: > Our favorite distro goes ARM. > This gives a platform that feels like home and should have all our > recepies work like a charm. > > Michel > > -------- Originele bericht -------- Onderwerp: [CentOS] CentOS for ARM Datum: > Sun, 25 Mar 2012 15:17:20 +0100 Van: Gordan Bobic Antwoord-naar: > CentOS mailing list Aan: > centos-devel at centos.org, centos at centos.org > > Hi guys, > > I recently became aware that this is being worked on as of recently. A > similar thing already exists, though. Those interested in an ARM port > may also be interested in taking a look at RedSleeve Linux, which is an > ARM port of the same upstream distribution as CentOS. > > You can look here for more info:http://www.redsleeve.org/about/http://www.redsleeve.org/ > > In total 109 SRPMs had to be modified in order to get them to build and > work on ARM. They are in SRPMS/changed directory on the RedSleeve > mirror. (Note: About 5 of those 109 were changed in order to remove the > upstream branding which isn't relevant to the CentOS effort as it is > already taken care of. It is quite well known what those are.) > > I hope this is of interest to the CentOS ARM port efforts. Or, you could > take RedSleeve for a spin, since that is already available. :) > > Gordan > _______________________________________________ > CentOS mailing listCentOS at centos.orghttp://lists.centos.org/mailman/listinfo/centos > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120325/238d8d7b/attachment.html From bdfoster at endigotech.com Sun Mar 25 21:00:04 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 25 Mar 2012 13:00:04 -0400 Subject: [Freeswitch-users] Fwd: FreeSWITCH Installer for Debian In-Reply-To: References: Message-ID: Version 0.0.4 was released today! This version adds some major error checking, so this should be a lot safer to run. I still need to do some logic so it could possibly fix an issue, then try again, but for now it just doesn't screw up your pretty new image ;-) The script also tells you what command it was on when it failed, and also gives the error number of that failed command (just in case it's not a 1, but some other number between 2-255 that actually means something). To get started, run to your terminal and type: wget http://goo.gl/kPSdg -O fs-debian-installer.sh chmod +x fs-debian-installer.sh ./fs-debian-installer.sh If there ends up being an issue and for some reason it doesn't install, please do this: rm -r /usr/local/src/freeswitch rm -r /usr/local/freeswitch script -a fs-installer.log ./fs-debian-installer.sh ... at the end of the script, press CRTL-D and it should give you a message saying it was saved. Then, send an email to fs-installer at endigotech.comwith the log attached. So you're probably wondering, why go through all of the effort to do a make && make install script? Well, this script is mostly designed for those who will be deploying FreeSWITCH on a constant or regular basis and want them to be uniform AND not have to look over them while FS is being installed. It can also be used by beginners, too! Anyway, please have a test and see what you think. If there's anything you'd change, let me know. I'm always open to ideas. -BDF On Sat, Mar 24, 2012 at 2:21 AM, Zenny wrote: > Brian: love to see your new releases, great script ;-) > > On 3/23/12, Brian Foster wrote: > > A fix for that is in the works. > > > > -BDF > > On Mar 23, 2012 7:01 PM, "curriegrad2004" > wrote: > > > >> One comment: building FS as root isn't something that's attractive nor > >> appealing to me... > >> > >> On Fri, Mar 23, 2012 at 10:12 AM, Brian Foster > > >> wrote: > >> > Whoops. > >> > > >> > The new version (and always the latest) can be downloaded from here: > >> > > >> > http://files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_latest.sh > >> > > >> > > >> > On Fri, Mar 23, 2012 at 1:07 PM, Brian Foster < > bdfoster at endigotech.com> > >> > wrote: > >> >> > >> >> Version 0.0.3 was released today: > >> >> > >> >> Changelog > >> >> > >> >> > >> > >> > ================================================================================= > >> >> 0.0.3 23/Mar/2012 1235 UCT > >> >> > >> >> Bug Fixes > >> >> - Changed script shell from bash to sh for compatability > >> >> Features Added > >> >> -Made FS git address, FS user, FS group, installed packages, > FS > >> >> sounds > >> >> install, FS MOH install, and base dir for source folder into > >> >> variables > >> >> - Option added to update/upgrade system (true/false variable) > >> >> - Option to download/install init script > >> >> > >> >> Version 0.1.0 will be released sometime within the next week. > >> >> > >> >> -BDF > >> >> > >> >> On Thu, Mar 22, 2012 at 6:21 PM, Gabriel Gunderson > >> wrote: > >> >>> > >> >>> On Thu, Mar 22, 2012 at 3:44 PM, Zenny > wrote: > >> >>> > Glad to find build_opensips.sh also in the same directory. > >> >>> > >> >>> If you want to save the trouble of building, there are also some > >> >>> pre-built debs for both FreeSWITCH and OpenSIPS: > >> >>> > >> >>> https://parseltone.org/files/debs/ubuntu-11.4/ > >> >>> > >> >>> > >> >>> Best, > >> >>> Gabe > >> >>> > >> >>> > >> > _________________________________________________________________________ > >> >>> Professional FreeSWITCH Consulting Services: > >> >>> consulting at freeswitch.org > >> >>> http://www.freeswitchsolutions.com > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> Official FreeSWITCH Sites > >> >>> http://www.freeswitch.org > >> >>> http://wiki.freeswitch.org > >> >>> http://www.cluecon.com > >> >>> > >> >>> FreeSWITCH-users mailing list > >> >>> FreeSWITCH-users at lists.freeswitch.org > >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>> http://www.freeswitch.org > >> >> > >> >> > >> >> > >> >> > >> >> -- > >> >> Brian D. Foster > >> >> Endigo Computer LLC > >> >> Email: bdfoster at endigotech.com > >> >> Phone: 317-800-7876 > >> >> Indianapolis, Indiana, USA > >> >> > >> >> This message contains confidential information and is intended for > >> >> those > >> >> listed in the "To:", "CC:", and/or "BCC:" fields of the message > header. > >> If > >> >> you are not the intended recipient you are notified that disclosing, > >> >> copying, distributing or taking any action in reliance on the > contents > >> of > >> >> this information is strictly prohibited. E-mail transmission cannot > be > >> >> guaranteed to be secure or error-free as information could be > >> intercepted, > >> >> corrupted, lost, destroyed, arrive late or incomplete, or contain > >> viruses. > >> >> The sender therefore does not accept liability for any errors or > >> omissions > >> >> in the contents of this message, which arise as a result of e-mail > >> >> transmission. If verification is required please request a hard-copy > >> >> version. > >> >> > >> > > >> > > >> > > >> > -- > >> > Brian D. Foster > >> > Endigo Computer LLC > >> > Email: bdfoster at endigotech.com > >> > Phone: 317-800-7876 > >> > Indianapolis, Indiana, USA > >> > > >> > This message contains confidential information and is intended for > those > >> > listed in the "To:", "CC:", and/or "BCC:" fields of the message > header. > >> If > >> > you are not the intended recipient you are notified that disclosing, > >> > copying, distributing or taking any action in reliance on the contents > >> > of > >> > this information is strictly prohibited. E-mail transmission cannot be > >> > guaranteed to be secure or error-free as information could be > >> intercepted, > >> > corrupted, lost, destroyed, arrive late or incomplete, or contain > >> viruses. > >> > The sender therefore does not accept liability for any errors or > >> omissions > >> > in the contents of this message, which arise as a result of e-mail > >> > transmission. If verification is required please request a hard-copy > >> > version. > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120325/41a146c8/attachment-0001.html From bdfoster at endigotech.com Mon Mar 26 00:47:10 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 25 Mar 2012 16:47:10 -0400 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: <4F6F3570.4090608@gmail.com> References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com> Message-ID: ...Debian works on ARM? :-) -BDF On Mar 25, 2012 11:13 AM, "Michel Daggelinckx" wrote: > Our favorite distro goes ARM. > This gives a platform that feels like home and should have all our > recepies work like a charm. > > Michel > > -------- Originele bericht -------- Onderwerp: [CentOS] CentOS for ARM Datum: > Sun, 25 Mar 2012 15:17:20 +0100 Van: Gordan Bobic Antwoord-naar: > CentOS mailing list Aan: > centos-devel at centos.org, centos at centos.org > > Hi guys, > > I recently became aware that this is being worked on as of recently. A > similar thing already exists, though. Those interested in an ARM port > may also be interested in taking a look at RedSleeve Linux, which is an > ARM port of the same upstream distribution as CentOS. > > You can look here for more info:http://www.redsleeve.org/about/http://www.redsleeve.org/ > > In total 109 SRPMs had to be modified in order to get them to build and > work on ARM. They are in SRPMS/changed directory on the RedSleeve > mirror. (Note: About 5 of those 109 were changed in order to remove the > upstream branding which isn't relevant to the CentOS effort as it is > already taken care of. It is quite well known what those are.) > > I hope this is of interest to the CentOS ARM port efforts. Or, you could > take RedSleeve for a spin, since that is already available. :) > > Gordan > _______________________________________________ > CentOS mailing listCentOS at centos.orghttp://lists.centos.org/mailman/listinfo/centos > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120325/cdffacaf/attachment.html From lazyvirus at gmx.com Mon Mar 26 00:56:20 2012 From: lazyvirus at gmx.com (Bzzz) Date: Sun, 25 Mar 2012 22:56:20 +0200 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com> Message-ID: <20120325225620.07ce66ea@anubis.defcon1> On Sun, 25 Mar 2012 16:47:10 -0400 Brian Foster wrote: > ...Debian works on ARM? :-) 'course: http://www.debian.org/ports/ -- Sniff sniff... Hey! Who farted? From daggelinckxmichel at gmail.com Mon Mar 26 00:57:42 2012 From: daggelinckxmichel at gmail.com (Michel Daggelinckx) Date: Sun, 25 Mar 2012 22:57:42 +0200 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com> Message-ID: <4F6F86C6.4030609@gmail.com> Op 25-03-12 22:47, Brian Foster schreef: > > ...Debian works on ARM? :-) > yup http://www.debian.org/ports/arm/ > > -BDF > > On Mar 25, 2012 11:13 AM, "Michel Daggelinckx" > > wrote: > > Our favorite distro goes ARM. > This gives a platform that feels like home and should have all our > recepies work like a charm. > > Michel > > -------- Originele bericht -------- > Onderwerp: [CentOS] CentOS for ARM > Datum: Sun, 25 Mar 2012 15:17:20 +0100 > Van: Gordan Bobic > > Antwoord-naar: CentOS mailing list > > Aan: centos-devel at centos.org , > centos at centos.org > > > > Hi guys, > > I recently became aware that this is being worked on as of recently. A > similar thing already exists, though. Those interested in an ARM port > may also be interested in taking a look at RedSleeve Linux, which is an > ARM port of the same upstream distribution as CentOS. > > You can look here for more info: > http://www.redsleeve.org/about/ > http://www.redsleeve.org/ > > In total 109 SRPMs had to be modified in order to get them to build and > work on ARM. They are in SRPMS/changed directory on the RedSleeve > mirror. (Note: About 5 of those 109 were changed in order to remove the > upstream branding which isn't relevant to the CentOS effort as it is > already taken care of. It is quite well known what those are.) > > I hope this is of interest to the CentOS ARM port efforts. Or, you could > take RedSleeve for a spin, since that is already available. :) > > Gordan > _______________________________________________ > CentOS mailing list > CentOS at centos.org > http://lists.centos.org/mailman/listinfo/centos > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120325/110e9831/attachment.html From bdfoster at endigotech.com Mon Mar 26 01:00:35 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 25 Mar 2012 17:00:35 -0400 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: <4F6F86C6.4030609@gmail.com> References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com> <4F6F86C6.4030609@gmail.com> Message-ID: Yea I know lol. But the point was I frigging hate CentOS lol -Brian On Mar 25, 2012 4:58 PM, "Michel Daggelinckx" wrote: > Op 25-03-12 22:47, Brian Foster schreef: > > ...Debian works on ARM? :-) > > yup > http://www.debian.org/ports/arm/ > > -BDF > On Mar 25, 2012 11:13 AM, "Michel Daggelinckx" < > daggelinckxmichel at gmail.com> wrote: > >> Our favorite distro goes ARM. >> This gives a platform that feels like home and should have all our >> recepies work like a charm. >> >> Michel >> >> -------- Originele bericht -------- Onderwerp: [CentOS] CentOS for ARM Datum: >> Sun, 25 Mar 2012 15:17:20 +0100 Van: Gordan Bobic Antwoord-naar: >> CentOS mailing list Aan: >> centos-devel at centos.org, centos at centos.org >> >> Hi guys, >> >> I recently became aware that this is being worked on as of recently. A >> similar thing already exists, though. Those interested in an ARM port >> may also be interested in taking a look at RedSleeve Linux, which is an >> ARM port of the same upstream distribution as CentOS. >> >> You can look here for more info:http://www.redsleeve.org/about/http://www.redsleeve.org/ >> >> In total 109 SRPMs had to be modified in order to get them to build and >> work on ARM. They are in SRPMS/changed directory on the RedSleeve >> mirror. (Note: About 5 of those 109 were changed in order to remove the >> upstream branding which isn't relevant to the CentOS effort as it is >> already taken care of. It is quite well known what those are.) >> >> I hope this is of interest to the CentOS ARM port efforts. Or, you could >> take RedSleeve for a spin, since that is already available. :) >> >> Gordan >> _______________________________________________ >> CentOS mailing listCentOS at centos.orghttp://lists.centos.org/mailman/listinfo/centos >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120325/8458be6e/attachment-0001.html From curriegrad2004 at gmail.com Mon Mar 26 02:46:29 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 25 Mar 2012 15:46:29 -0700 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com> <4F6F86C6.4030609@gmail.com> Message-ID: Unfortunately you'll find lots of CentOS boxes out there... On 2012-03-25 2:01 PM, "Brian Foster" wrote: > Yea I know lol. But the point was I frigging hate CentOS lol > > -Brian > On Mar 25, 2012 4:58 PM, "Michel Daggelinckx" > wrote: > >> Op 25-03-12 22:47, Brian Foster schreef: >> >> ...Debian works on ARM? :-) >> >> yup >> http://www.debian.org/ports/arm/ >> >> -BDF >> On Mar 25, 2012 11:13 AM, "Michel Daggelinckx" < >> daggelinckxmichel at gmail.com> wrote: >> >>> Our favorite distro goes ARM. >>> This gives a platform that feels like home and should have all our >>> recepies work like a charm. >>> >>> Michel >>> >>> -------- Originele bericht -------- Onderwerp: [CentOS] CentOS for ARM Datum: >>> Sun, 25 Mar 2012 15:17:20 +0100 Van: Gordan Bobic >>> Antwoord-naar: CentOS >>> mailing list Aan: >>> centos-devel at centos.org, centos at centos.org >>> >>> Hi guys, >>> >>> I recently became aware that this is being worked on as of recently. A >>> similar thing already exists, though. Those interested in an ARM port >>> may also be interested in taking a look at RedSleeve Linux, which is an >>> ARM port of the same upstream distribution as CentOS. >>> >>> You can look here for more info:http://www.redsleeve.org/about/http://www.redsleeve.org/ >>> >>> In total 109 SRPMs had to be modified in order to get them to build and >>> work on ARM. They are in SRPMS/changed directory on the RedSleeve >>> mirror. (Note: About 5 of those 109 were changed in order to remove the >>> upstream branding which isn't relevant to the CentOS effort as it is >>> already taken care of. It is quite well known what those are.) >>> >>> I hope this is of interest to the CentOS ARM port efforts. Or, you could >>> take RedSleeve for a spin, since that is already available. :) >>> >>> Gordan >>> _______________________________________________ >>> CentOS mailing listCentOS at centos.orghttp://lists.centos.org/mailman/listinfo/centos >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120325/e9449893/attachment.html From bdfoster at endigotech.com Mon Mar 26 03:59:07 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 25 Mar 2012 19:59:07 -0400 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com> <4F6F86C6.4030609@gmail.com> Message-ID: I know, I know... I'm the minority... :-) On Mar 25, 2012 6:47 PM, "curriegrad2004" wrote: > Unfortunately you'll find lots of CentOS boxes out there... > On 2012-03-25 2:01 PM, "Brian Foster" wrote: > >> Yea I know lol. But the point was I frigging hate CentOS lol >> >> -Brian >> On Mar 25, 2012 4:58 PM, "Michel Daggelinckx" < >> daggelinckxmichel at gmail.com> wrote: >> >>> Op 25-03-12 22:47, Brian Foster schreef: >>> >>> ...Debian works on ARM? :-) >>> >>> yup >>> http://www.debian.org/ports/arm/ >>> >>> -BDF >>> On Mar 25, 2012 11:13 AM, "Michel Daggelinckx" < >>> daggelinckxmichel at gmail.com> wrote: >>> >>>> Our favorite distro goes ARM. >>>> This gives a platform that feels like home and should have all our >>>> recepies work like a charm. >>>> >>>> Michel >>>> >>>> -------- Originele bericht -------- Onderwerp: [CentOS] CentOS for ARM Datum: >>>> Sun, 25 Mar 2012 15:17:20 +0100 Van: Gordan Bobic >>>> Antwoord-naar: CentOS >>>> mailing list Aan: >>>> centos-devel at centos.org, centos at centos.org >>>> >>>> Hi guys, >>>> >>>> I recently became aware that this is being worked on as of recently. A >>>> similar thing already exists, though. Those interested in an ARM port >>>> may also be interested in taking a look at RedSleeve Linux, which is an >>>> ARM port of the same upstream distribution as CentOS. >>>> >>>> You can look here for more info:http://www.redsleeve.org/about/http://www.redsleeve.org/ >>>> >>>> In total 109 SRPMs had to be modified in order to get them to build and >>>> work on ARM. They are in SRPMS/changed directory on the RedSleeve >>>> mirror. (Note: About 5 of those 109 were changed in order to remove the >>>> upstream branding which isn't relevant to the CentOS effort as it is >>>> already taken care of. It is quite well known what those are.) >>>> >>>> I hope this is of interest to the CentOS ARM port efforts. Or, you could >>>> take RedSleeve for a spin, since that is already available. :) >>>> >>>> Gordan >>>> _______________________________________________ >>>> CentOS mailing listCentOS at centos.orghttp://lists.centos.org/mailman/listinfo/centos >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120325/8d2f2042/attachment-0001.html From tknchris at gmail.com Mon Mar 26 02:02:54 2012 From: tknchris at gmail.com (chris) Date: Sun, 25 Mar 2012 18:02:54 -0400 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com> <4F6F86C6.4030609@gmail.com> Message-ID: +1, rpm hell :( also +1 to loving debian On Sun, Mar 25, 2012 at 5:00 PM, Brian Foster wrote: > Yea I know lol. But the point was I frigging hate CentOS lol > > -Brian > On Mar 25, 2012 4:58 PM, "Michel Daggelinckx" > wrote: > >> Op 25-03-12 22:47, Brian Foster schreef: >> >> ...Debian works on ARM? :-) >> >> yup >> http://www.debian.org/ports/arm/ >> >> -BDF >> On Mar 25, 2012 11:13 AM, "Michel Daggelinckx" < >> daggelinckxmichel at gmail.com> wrote: >> >>> Our favorite distro goes ARM. >>> This gives a platform that feels like home and should have all our >>> recepies work like a charm. >>> >>> Michel >>> >>> -------- Originele bericht -------- Onderwerp: [CentOS] CentOS for ARM Datum: >>> Sun, 25 Mar 2012 15:17:20 +0100 Van: Gordan Bobic >>> Antwoord-naar: CentOS >>> mailing list Aan: >>> centos-devel at centos.org, centos at centos.org >>> >>> Hi guys, >>> >>> I recently became aware that this is being worked on as of recently. A >>> similar thing already exists, though. Those interested in an ARM port >>> may also be interested in taking a look at RedSleeve Linux, which is an >>> ARM port of the same upstream distribution as CentOS. >>> >>> You can look here for more info:http://www.redsleeve.org/about/http://www.redsleeve.org/ >>> >>> In total 109 SRPMs had to be modified in order to get them to build and >>> work on ARM. They are in SRPMS/changed directory on the RedSleeve >>> mirror. (Note: About 5 of those 109 were changed in order to remove the >>> upstream branding which isn't relevant to the CentOS effort as it is >>> already taken care of. It is quite well known what those are.) >>> >>> I hope this is of interest to the CentOS ARM port efforts. Or, you could >>> take RedSleeve for a spin, since that is already available. :) >>> >>> Gordan >>> _______________________________________________ >>> CentOS mailing listCentOS at centos.orghttp://lists.centos.org/mailman/listinfo/centos >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120325/38f3f5c9/attachment-0001.html From tknchris at gmail.com Mon Mar 26 02:50:59 2012 From: tknchris at gmail.com (chris) Date: Sun, 25 Mar 2012 18:50:59 -0400 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com> <4F6F86C6.4030609@gmail.com> Message-ID: Sadly chris On Mar 25, 2012 6:47 PM, "curriegrad2004" wrote: > Unfortunately you'll find lots of CentOS boxes out there... > On 2012-03-25 2:01 PM, "Brian Foster" wrote: > >> Yea I know lol. But the point was I frigging hate CentOS lol >> >> -Brian >> On Mar 25, 2012 4:58 PM, "Michel Daggelinckx" < >> daggelinckxmichel at gmail.com> wrote: >> >>> Op 25-03-12 22:47, Brian Foster schreef: >>> >>> ...Debian works on ARM? :-) >>> >>> yup >>> http://www.debian.org/ports/arm/ >>> >>> -BDF >>> On Mar 25, 2012 11:13 AM, "Michel Daggelinckx" < >>> daggelinckxmichel at gmail.com> wrote: >>> >>>> Our favorite distro goes ARM. >>>> This gives a platform that feels like home and should have all our >>>> recepies work like a charm. >>>> >>>> Michel >>>> >>>> -------- Originele bericht -------- Onderwerp: [CentOS] CentOS for ARM Datum: >>>> Sun, 25 Mar 2012 15:17:20 +0100 Van: Gordan Bobic >>>> Antwoord-naar: CentOS >>>> mailing list Aan: >>>> centos-devel at centos.org, centos at centos.org >>>> >>>> Hi guys, >>>> >>>> I recently became aware that this is being worked on as of recently. A >>>> similar thing already exists, though. Those interested in an ARM port >>>> may also be interested in taking a look at RedSleeve Linux, which is an >>>> ARM port of the same upstream distribution as CentOS. >>>> >>>> You can look here for more info:http://www.redsleeve.org/about/http://www.redsleeve.org/ >>>> >>>> In total 109 SRPMs had to be modified in order to get them to build and >>>> work on ARM. They are in SRPMS/changed directory on the RedSleeve >>>> mirror. (Note: About 5 of those 109 were changed in order to remove the >>>> upstream branding which isn't relevant to the CentOS effort as it is >>>> already taken care of. It is quite well known what those are.) >>>> >>>> I hope this is of interest to the CentOS ARM port efforts. Or, you could >>>> take RedSleeve for a spin, since that is already available. :) >>>> >>>> Gordan >>>> _______________________________________________ >>>> CentOS mailing listCentOS at centos.orghttp://lists.centos.org/mailman/listinfo/centos >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120325/86b5ca4e/attachment-0001.html From lazyvirus at gmx.com Mon Mar 26 10:05:52 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 26 Mar 2012 08:05:52 +0200 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com> <4F6F86C6.4030609@gmail.com> Message-ID: <20120326080552.41b03468@anubis.defcon1> On Sun, 25 Mar 2012 18:02:54 -0400 chris wrote: > +1, rpm hell :( Why hell? Just wait 5-6 years and everything will be better: there will be just one rpm that'll install all softwares in one click:) -- From neilp at cs.stanford.edu Mon Mar 26 10:18:36 2012 From: neilp at cs.stanford.edu (Neil Patel) Date: Mon, 26 Mar 2012 11:48:36 +0530 Subject: [Freeswitch-users] no DTMF detection over VoIP In-Reply-To: References: Message-ID: I guess you do this by using start_dtmf? I tried adding it to my dialplan and also in my lua IVR script: session:execute("start_dtmf"); But no luck with either. On Sun, Mar 25, 2012 at 8:40 AM, curriegrad2004 wrote: > If the call is coming from a TDM circuit, you'll have to do in-band > DTMF detection in that situation. > > On Sat, Mar 24, 2012 at 6:25 PM, Michael Collins > wrote: > > Neil, > > > > Howdy! Long time no chat. :) > > > > Regarding SIP/VoIP: A SIP call will not come in over a PRI circuit. SIP > is > > VoIP and PRI is TDM - two totally different methods for carrying > telephone > > calls. My guess is that you might have a bit of confusion about where and > > how calls are coming in. Let's start with the basics: Do you have a > SIP/VoIP > > provider? If so, did they supply you with phone number(s) for dialing in > to > > your system? Have you gone through the configuration process to make sure > > your FS box is properly communicating with the provider? > > > > If you need some help you can contact me off list. Of course, this means > > you'll have to take me to that Mexican food place the next time I'm in > > Frisco! :) > > > > -MC > > > > > > On Sat, Mar 24, 2012 at 11:18 AM, Neil Patel > wrote: > >> > >> I have not modified the sip_profiles directory from standard. There is > no > >> variable set for dtmf-type in either internal.xml or external.xml. > >> > >> I did a trace using: > >> sofia global siptrace on > >> > >> There wer no SIP packets logged. If the SIP call is going to a PRI > profile > >> (freetdm), will there still be SIP traffic? > >> > >> Thanks, > >> Neil > >> > >> > >> On Sat, Mar 24, 2012 at 7:22 PM, Anita Hall > >> wrote: > >>> > >>> What is your config for DTMF in sip profile ? > >>> > >>> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#dtmf-type > >>> > >>> Use tshark to generate a SIP trace and paste it on > >>> http://pastebin.freeswitch.org/ and give the link here. > >>> > >>> regards, > >>> Anita > >>> > >>> > >>> > >>> On Sat, Mar 24, 2012 at 2:43 PM, Neil Patel > >>> wrote: > >>>> > >>>> Hi All, > >>>> > >>>> I have a basic IVR application in Lua connected to a PRI line. It > >>>> currently is not responding to DTMF input given from any VoIP call > (e.g. > >>>> Skype). However, it accepts input from local mobile or landline calls > just > >>>> fine. > >>>> > >>>> I recently pulled latest from git and built; before that, DTMF > detection > >>>> from both were working. Is there something I need to configure to > allow DTMF > >>>> for VoIP? > >>>> > >>>> Thanks, > >>>> Neil > >>>> > >>>> > >>>> > _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> http://www.cluecon.com > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/88e29517/attachment.html From miha at softnet.si Mon Mar 26 10:29:39 2012 From: miha at softnet.si (Miha) Date: Mon, 26 Mar 2012 08:29:39 +0200 Subject: [Freeswitch-users] Break&condition In-Reply-To: References: <4F6C74CA.2090404@softnet.si> Message-ID: <4F700CD3.2030904@softnet.si> Hi Michael, here is my dialplan: http://pastebin.freeswitch.org/18732 It is not perfect as I was witting it for the first time. I need to block certain destination numbers with toll_allow. I am trying to add this part: before it hits the main part of dialplan. So that I can block this destiantion numbers if condition is true. Thank you! Regards, Miha Regards, Miha On 3/23/2012 4:33 PM, Michael Collins wrote: > Miha, > > It might help if you show us a bit more of your dialplan. It may be > that you need to break some of these out into separate extensions. > Also, what is your "big picture" application? What is the problem that > you are attempting to solve? > > -MC > > > On Fri, Mar 23, 2012 at 6:04 AM, Miha > wrote: > > Hi, > > In same extension a have multiple conditions. Problem is if the > first condition is false, dialplan will go further as I have set > on-true. > How can I prevent that dialplan will go after break="on-true" on > second condition and will not go looking condition inside condition. > > So if the variable mobilne is not set, in this dialplan FS will go > looking to expression="^(051|041|031|030|040|070|071)(\d{6})|^(0038651|0038641|0038631|0038630|0038640|0038670|0038671)(\d{6})" > break="on-true"> and reject call instead of goint to second > condtion which is expression="tuje"/>. > > I hope I make it clear:D > > break="on-true" /> > expression="^(051|041|031|030|040|070|071)(\d{6})|^(0038651|0038641|0038631|0038630|0038640|0038670|0038671)(\d{6})" > break="on-true"> > > > > > > > > > > > > > expression="^(090)(\d{4})|^(090)(\d{6})"> > > > > > Regards and thank you for your help! > > Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/028bc144/attachment-0001.html From bdfoster at endigotech.com Mon Mar 26 10:30:45 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 26 Mar 2012 02:30:45 -0400 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: <20120326080552.41b03468@anubis.defcon1> References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com> <4F6F86C6.4030609@gmail.com> <20120326080552.41b03468@anubis.defcon1> Message-ID: Why wait 5-6 years when apt-get works just fine for me? -BDF On Mon, Mar 26, 2012 at 2:05 AM, Bzzz wrote: > On Sun, 25 Mar 2012 18:02:54 -0400 > chris wrote: > > > +1, rpm hell :( > > Why hell? > Just wait 5-6 years and everything will be better: > there will be just one rpm that'll install all softwares > in one click:) > > -- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/6373c83b/attachment.html From lazyvirus at gmx.com Mon Mar 26 10:40:44 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 26 Mar 2012 08:40:44 +0200 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com> <4F6F86C6.4030609@gmail.com> <20120326080552.41b03468@anubis.defcon1> Message-ID: <20120326084044.5a7f72bd@anubis.defcon1> On Mon, 26 Mar 2012 02:30:45 -0400 Brian Foster wrote: > Why wait 5-6 years when apt-get works just fine for me? Hu... as a long time Debianist, I was kidding. -- As long as we're going to reinvent the wheel again, we might as well try making it round this time. -- Mike Dennison From bdfoster at endigotech.com Mon Mar 26 11:11:52 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 26 Mar 2012 03:11:52 -0400 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: <20120326084044.5a7f72bd@anubis.defcon1> References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com> <4F6F86C6.4030609@gmail.com> <20120326080552.41b03468@anubis.defcon1> <20120326084044.5a7f72bd@anubis.defcon1> Message-ID: Well, I was half kidding ;-) Besides RPM hell, I've just had way too many issues with Red Hat stuff. I started on CentOS, moved to Fedora, half-way convinced myself I needed RHEL, then a friend introduced me to Debian. I've never looked back. I've turned into a major Debian advocate, I run Debian for any application and it hasn't failed me. Then again, sometimes it comes down to a comfort thing. Some that use CentOS and Fedora, etc came from RHEL whether it's because of work or otherwise. I advocate Debian, sure. However, sometimes it's more important to be on an OS without needing a bunch of re-learning. On Mon, Mar 26, 2012 at 2:40 AM, Bzzz wrote: > On Mon, 26 Mar 2012 02:30:45 -0400 > Brian Foster wrote: > > > Why wait 5-6 years when apt-get works just fine for me? > > Hu... as a long time Debianist, I was kidding. > > -- > As long as we're going to reinvent the wheel again, we might as well > try making it round this time. -- Mike Dennison > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/22528039/attachment.html From jdiaz at coinfru.com Mon Mar 26 11:24:08 2012 From: jdiaz at coinfru.com (Josue Diaz Cruz) Date: Mon, 26 Mar 2012 09:24:08 +0200 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com><4F6F86C6.4030609@gmail.com><20120326080552.41b03468@anubis.defcon1><20120326084044.5a7f72bd@anubis.defcon1> Message-ID: Brands are really good cause you can choose what you want. The taste of each one is something really personal. I am using Centos for some reasons. First as Brian told cause i start with redhat in linux. But in some cases it is not a question of taste. I was using Vyatta (debian based) for routing and colo services. Cause Vyatta is ussing debian we came to know that cause us problems when we have a big amount of l2tp tunnels. we could not grow. We use Centos and Suse and the problem disapeard. So, some times is taste, some times need. God bless you all. Josue Diaz Cruz Departamento Tecnico y Soporte jdiaz at coinfru.com C/ Balsicas 3 Alquerias | 30580 | Murcia www.coinfru.com Este e-mail contiene informaci?n confidencial, el contenido de la mismo se encuentra protegido por Ley. Cualquier persona distinta a su destinataria tiene prohibida su reproducci?n, uso, divulgaci?n o impresi?n total o parcial. Si ha recibido este mensaje por error, notifiquelo de inmediato al remitente borrando el mensaje original juntamente con sus ficheros anexos. Gracias. _____ De: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Brian Foster Enviado el: Monday, March 26, 2012 09:12 Para: FreeSWITCH Users Help Asunto: Re: [Freeswitch-users] CentOS for ARM Well, I was half kidding ;-) Besides RPM hell, I've just had way too many issues with Red Hat stuff. I started on CentOS, moved to Fedora, half-way convinced myself I needed RHEL, then a friend introduced me to Debian. I've never looked back. I've turned into a major Debian advocate, I run Debian for any application and it hasn't failed me. Then again, sometimes it comes down to a comfort thing. Some that use CentOS and Fedora, etc came from RHEL whether it's because of work or otherwise. I advocate Debian, sure. However, sometimes it's more important to be on an OS without needing a bunch of re-learning. On Mon, Mar 26, 2012 at 2:40 AM, Bzzz wrote: On Mon, 26 Mar 2012 02:30:45 -0400 Brian Foster wrote: > Why wait 5-6 years when apt-get works just fine for me? Hu... as a long time Debianist, I was kidding. -- As long as we're going to reinvent the wheel again, we might as well try making it round this time. -- Mike Dennison _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/7f9cd919/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4705 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/7f9cd919/attachment-0001.jpe From Daniel.Knaggs at realitysolutions.co.uk Mon Mar 26 11:43:45 2012 From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs) Date: Mon, 26 Mar 2012 07:43:45 +0000 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: References: Message-ID: Not a bad idea, Michael! I'll see if I can get that sorted. The DTMF tones only appear to be "lettered" ones, could I comment those out in the source code and rebuild (which files would I need to change)? Not the best workaround but should work. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 25 March 2012 02:20 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls I'd do an audio recording of the call and open the resulting file in Audacity. It can do kind of a spectrum analysis to show you if there are any frequencies that might be fooling the dtmf detector. Also, I'm not familiar with using Digium cards with FreeTDM. (I have a Digium TE121 or similar but I don't happen to have any mobo's with the PCIe or whatever slot type it uses.) To Moises I'd ask: is there hardware DTMF detection in the Digium cards that you know of? Also, any known or suspected issues with false DTMF detection in the scenario mentioned by the OP? Gracias, MC On Sat, Mar 24, 2012 at 5:27 AM, Daniel Knaggs > wrote: They aren't any faults that I can see. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 Sent: 23 March 2012 23:23 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls Have you checked for any faults on your TDM link on your side? On Fri, Mar 23, 2012 at 10:32 AM, Brian Foster > wrote: Please update to latest git. -BDF On Fri, Mar 23, 2012 at 7:19 AM, Daniel Knaggs > wrote: Hello all, Got a bit of a strange one, we appear to be getting DTMF tones on incoming calls when the caller hasn't even pressed any keys. It normally happens with 10 seconds or so after the call has been answered. Here is the log of it happening earlier: - 2012-03-23 10:38:16.852654 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF A (debug = 0) 2012-03-23 10:38:16.852654 [DEBUG] mod_freetdm.c:799 Queuing DTMF [A] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:16.915653 [DEBUG] switch_rtp.c:2420 Send start packet for [A] ts=49440 dur=160/160/2000 seq=46803 lw=49440 2012-03-23 10:38:16.936653 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=320/320/2000 seq=46804 lw=49600 2012-03-23 10:38:16.957652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=480/480/2000 seq=46805 lw=49760 2012-03-23 10:38:16.978652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=640/640/2000 seq=46806 lw=49920 2012-03-23 10:38:16.999652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=800/800/2000 seq=46807 lw=50080 2012-03-23 10:38:17.020651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=960/960/2000 seq=46808 lw=50240 2012-03-23 10:38:17.041651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1120/1120/2000 seq=46809 lw=50400 2012-03-23 10:38:17.062651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1280/1280/2000 seq=46810 lw=50560 2012-03-23 10:38:17.083650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1440/1440/2000 seq=46811 lw=50720 2012-03-23 10:38:17.104650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1600/1600/2000 seq=46812 lw=50880 2012-03-23 10:38:17.125650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1760/1760/2000 seq=46813 lw=51040 2012-03-23 10:38:17.146650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1920/1920/2000 seq=46814 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46815 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46816 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46817 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2271 Queue digit delay of 40ms 2012-03-23 10:38:18.070638 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF B (debug = 0) 2012-03-23 10:38:18.070638 [DEBUG] mod_freetdm.c:799 Queuing DTMF [B] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:18.133637 [DEBUG] switch_rtp.c:2420 Send start packet for [B] ts=59040 dur=160/160/2000 seq=46864 lw=59040 2012-03-23 10:38:18.154637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=320/320/2000 seq=46865 lw=59200 2012-03-23 10:38:18.175637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=480/480/2000 seq=46866 lw=59360 2012-03-23 10:38:18.196636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=640/640/2000 seq=46867 lw=59520 2012-03-23 10:38:18.217636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=800/800/2000 seq=46868 lw=59680 2012-03-23 10:38:18.238636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=960/960/2000 seq=46869 lw=59840 2012-03-23 10:38:18.259636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1120/1120/2000 seq=46870 lw=60000 2012-03-23 10:38:18.280636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1280/1280/2000 seq=46871 lw=60160 2012-03-23 10:38:18.301635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1440/1440/2000 seq=46872 lw=60320 2012-03-23 10:38:18.322635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1600/1600/2000 seq=46873 lw=60480 2012-03-23 10:38:18.343634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1760/1760/2000 seq=46874 lw=60640 2012-03-23 10:38:18.364634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1920/1920/2000 seq=46875 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46876 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46877 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46878 lw=60800 I'm very sure the caller isn't dialling "A" or "B"! Running on ISDN via a TE121 card using E1 (euroisdn). Freeswitch version is "FreeSWITCH Version 1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)". Currently there is a "bind_meta_app" in the config which binds a script on the B leg of the call which parks the call - I haven't tried turning this off yet to see if it's this. Wondering if anyone has any ideas or has come across this before? Thanks in advance. [Description: Description: cid:imageacd695.PNG at 9e92e461.40aa4b96] Daniel Knaggs Software Developer Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE Tel: 01482 828000 / Fax: 01482 373100 Daniel.Knaggs at realitysolutions.co.uk www.realitysolutions.co.uk ________________________________ Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire [Description: Description: cid:image27a71e.PNG at c2da8488.4683bff1] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/6afcd543/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 22463 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/6afcd543/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 69075 bytes Desc: image002.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/6afcd543/attachment-0003.png From Daniel.Knaggs at realitysolutions.co.uk Mon Mar 26 11:45:04 2012 From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs) Date: Mon, 26 Mar 2012 07:45:04 +0000 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: References: Message-ID: The quality of the calls are perfect, even when listening to silence over the ISDN it is perfect. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 Sent: 25 March 2012 04:14 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls Those detected tones could be something that the TDM provider didn't filter out for some odd reason. With E&M signalling on TDM links you may hear a squealing tone in the background. On Sat, Mar 24, 2012 at 6:20 PM, Michael Collins > wrote: I'd do an audio recording of the call and open the resulting file in Audacity. It can do kind of a spectrum analysis to show you if there are any frequencies that might be fooling the dtmf detector. Also, I'm not familiar with using Digium cards with FreeTDM. (I have a Digium TE121 or similar but I don't happen to have any mobo's with the PCIe or whatever slot type it uses.) To Moises I'd ask: is there hardware DTMF detection in the Digium cards that you know of? Also, any known or suspected issues with false DTMF detection in the scenario mentioned by the OP? Gracias, MC On Sat, Mar 24, 2012 at 5:27 AM, Daniel Knaggs > wrote: They aren?t any faults that I can see. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 Sent: 23 March 2012 23:23 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls Have you checked for any faults on your TDM link on your side? On Fri, Mar 23, 2012 at 10:32 AM, Brian Foster > wrote: Please update to latest git. -BDF On Fri, Mar 23, 2012 at 7:19 AM, Daniel Knaggs > wrote: Hello all, Got a bit of a strange one, we appear to be getting DTMF tones on incoming calls when the caller hasn?t even pressed any keys. It normally happens with 10 seconds or so after the call has been answered. Here is the log of it happening earlier: - 2012-03-23 10:38:16.852654 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF A (debug = 0) 2012-03-23 10:38:16.852654 [DEBUG] mod_freetdm.c:799 Queuing DTMF [A] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:16.915653 [DEBUG] switch_rtp.c:2420 Send start packet for [A] ts=49440 dur=160/160/2000 seq=46803 lw=49440 2012-03-23 10:38:16.936653 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=320/320/2000 seq=46804 lw=49600 2012-03-23 10:38:16.957652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=480/480/2000 seq=46805 lw=49760 2012-03-23 10:38:16.978652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=640/640/2000 seq=46806 lw=49920 2012-03-23 10:38:16.999652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=800/800/2000 seq=46807 lw=50080 2012-03-23 10:38:17.020651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=960/960/2000 seq=46808 lw=50240 2012-03-23 10:38:17.041651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1120/1120/2000 seq=46809 lw=50400 2012-03-23 10:38:17.062651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1280/1280/2000 seq=46810 lw=50560 2012-03-23 10:38:17.083650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1440/1440/2000 seq=46811 lw=50720 2012-03-23 10:38:17.104650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1600/1600/2000 seq=46812 lw=50880 2012-03-23 10:38:17.125650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1760/1760/2000 seq=46813 lw=51040 2012-03-23 10:38:17.146650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1920/1920/2000 seq=46814 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46815 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46816 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46817 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2271 Queue digit delay of 40ms 2012-03-23 10:38:18.070638 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF B (debug = 0) 2012-03-23 10:38:18.070638 [DEBUG] mod_freetdm.c:799 Queuing DTMF [B] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:18.133637 [DEBUG] switch_rtp.c:2420 Send start packet for [B] ts=59040 dur=160/160/2000 seq=46864 lw=59040 2012-03-23 10:38:18.154637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=320/320/2000 seq=46865 lw=59200 2012-03-23 10:38:18.175637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=480/480/2000 seq=46866 lw=59360 2012-03-23 10:38:18.196636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=640/640/2000 seq=46867 lw=59520 2012-03-23 10:38:18.217636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=800/800/2000 seq=46868 lw=59680 2012-03-23 10:38:18.238636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=960/960/2000 seq=46869 lw=59840 2012-03-23 10:38:18.259636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1120/1120/2000 seq=46870 lw=60000 2012-03-23 10:38:18.280636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1280/1280/2000 seq=46871 lw=60160 2012-03-23 10:38:18.301635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1440/1440/2000 seq=46872 lw=60320 2012-03-23 10:38:18.322635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1600/1600/2000 seq=46873 lw=60480 2012-03-23 10:38:18.343634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1760/1760/2000 seq=46874 lw=60640 2012-03-23 10:38:18.364634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1920/1920/2000 seq=46875 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46876 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46877 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46878 lw=60800 I?m very sure the caller isn?t dialling ?A? or ?B?! Running on ISDN via a TE121 card using E1 (euroisdn). Freeswitch version is ?FreeSWITCH Version 1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)?. Currently there is a ?bind_meta_app? in the config which binds a script on the B leg of the call which parks the call ? I haven?t tried turning this off yet to see if it?s this. Wondering if anyone has any ideas or has come across this before? Thanks in advance. [Description: cid:imageacd695.PNG at 9e92e461.40aa4b96] Daniel Knaggs Software Developer Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE Tel: 01482 828000 / Fax: 01482 373100 Daniel.Knaggs at realitysolutions.co.uk www.realitysolutions.co.uk ________________________________ Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire [Description: cid:image27a71e.PNG at c2da8488.4683bff1] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/3c67b789/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 22463 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/3c67b789/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 69075 bytes Desc: image002.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/3c67b789/attachment-0003.png From Daniel.Knaggs at realitysolutions.co.uk Mon Mar 26 11:47:20 2012 From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs) Date: Mon, 26 Mar 2012 07:47:20 +0000 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: References: Message-ID: This isn't going to cause any problems is it, Brian? I've seen a few posts on the list stating a few problems recently when upgrading. I must say that we are running in a production environment. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster Sent: 23 March 2012 17:33 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls Please update to latest git. -BDF On Fri, Mar 23, 2012 at 7:19 AM, Daniel Knaggs > wrote: Hello all, Got a bit of a strange one, we appear to be getting DTMF tones on incoming calls when the caller hasn't even pressed any keys. It normally happens with 10 seconds or so after the call has been answered. Here is the log of it happening earlier: - 2012-03-23 10:38:16.852654 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF A (debug = 0) 2012-03-23 10:38:16.852654 [DEBUG] mod_freetdm.c:799 Queuing DTMF [A] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:16.915653 [DEBUG] switch_rtp.c:2420 Send start packet for [A] ts=49440 dur=160/160/2000 seq=46803 lw=49440 2012-03-23 10:38:16.936653 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=320/320/2000 seq=46804 lw=49600 2012-03-23 10:38:16.957652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=480/480/2000 seq=46805 lw=49760 2012-03-23 10:38:16.978652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=640/640/2000 seq=46806 lw=49920 2012-03-23 10:38:16.999652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=800/800/2000 seq=46807 lw=50080 2012-03-23 10:38:17.020651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=960/960/2000 seq=46808 lw=50240 2012-03-23 10:38:17.041651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1120/1120/2000 seq=46809 lw=50400 2012-03-23 10:38:17.062651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1280/1280/2000 seq=46810 lw=50560 2012-03-23 10:38:17.083650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1440/1440/2000 seq=46811 lw=50720 2012-03-23 10:38:17.104650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1600/1600/2000 seq=46812 lw=50880 2012-03-23 10:38:17.125650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1760/1760/2000 seq=46813 lw=51040 2012-03-23 10:38:17.146650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1920/1920/2000 seq=46814 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46815 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46816 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46817 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2271 Queue digit delay of 40ms 2012-03-23 10:38:18.070638 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF B (debug = 0) 2012-03-23 10:38:18.070638 [DEBUG] mod_freetdm.c:799 Queuing DTMF [B] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:18.133637 [DEBUG] switch_rtp.c:2420 Send start packet for [B] ts=59040 dur=160/160/2000 seq=46864 lw=59040 2012-03-23 10:38:18.154637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=320/320/2000 seq=46865 lw=59200 2012-03-23 10:38:18.175637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=480/480/2000 seq=46866 lw=59360 2012-03-23 10:38:18.196636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=640/640/2000 seq=46867 lw=59520 2012-03-23 10:38:18.217636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=800/800/2000 seq=46868 lw=59680 2012-03-23 10:38:18.238636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=960/960/2000 seq=46869 lw=59840 2012-03-23 10:38:18.259636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1120/1120/2000 seq=46870 lw=60000 2012-03-23 10:38:18.280636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1280/1280/2000 seq=46871 lw=60160 2012-03-23 10:38:18.301635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1440/1440/2000 seq=46872 lw=60320 2012-03-23 10:38:18.322635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1600/1600/2000 seq=46873 lw=60480 2012-03-23 10:38:18.343634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1760/1760/2000 seq=46874 lw=60640 2012-03-23 10:38:18.364634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1920/1920/2000 seq=46875 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46876 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46877 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46878 lw=60800 I'm very sure the caller isn't dialling "A" or "B"! Running on ISDN via a TE121 card using E1 (euroisdn). Freeswitch version is "FreeSWITCH Version 1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)". Currently there is a "bind_meta_app" in the config which binds a script on the B leg of the call which parks the call - I haven't tried turning this off yet to see if it's this. Wondering if anyone has any ideas or has come across this before? Thanks in advance. [cid:image001.png at 01CD0B2D.0DE3D000] Daniel Knaggs Software Developer Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE Tel: 01482 828000 / Fax: 01482 373100 Daniel.Knaggs at realitysolutions.co.uk www.realitysolutions.co.uk ________________________________ Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire [cid:image002.png at 01CD0B2D.0DE3D000] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/ba20dd36/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 22463 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/ba20dd36/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 69075 bytes Desc: image002.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/ba20dd36/attachment-0003.png From miha at softnet.si Mon Mar 26 12:46:05 2012 From: miha at softnet.si (Miha) Date: Mon, 26 Mar 2012 10:46:05 +0200 Subject: [Freeswitch-users] Break&condition In-Reply-To: <4F700CD3.2030904@softnet.si> References: <4F6C74CA.2090404@softnet.si> <4F700CD3.2030904@softnet.si> Message-ID: <4F702CCD.8040309@softnet.si> Hi Michael, fortget it, I already put thim in extensions and it work. Regards, Miha On 3/26/2012 8:29 AM, Miha wrote: > Hi Michael, > > here is my dialplan: http://pastebin.freeswitch.org/18732 > > It is not perfect as I was witting it for the first time. > I need to block certain destination numbers with toll_allow. > > I am trying to add this part: > > > > expression="^(051|041|031|030|040|070|071)(\d{6})|^(0038651|0038641|0038631|0038630|0038640|0038670|0038671)(\d{6})" > break="on-true"> > > > > > > > > > > > > > expression="^(090)(\d{4})|^(090)(\d{6})"> > > > > before it hits the main part of dialplan. So that I can block this > destiantion numbers if condition is true. > > Thank you! > > Regards, > Miha > > Regards, > Miha > On 3/23/2012 4:33 PM, Michael Collins wrote: >> Miha, >> >> It might help if you show us a bit more of your dialplan. It may be >> that you need to break some of these out into separate extensions. >> Also, what is your "big picture" application? What is the problem >> that you are attempting to solve? >> >> -MC >> >> >> On Fri, Mar 23, 2012 at 6:04 AM, Miha > > wrote: >> >> Hi, >> >> In same extension a have multiple conditions. Problem is if the >> first condition is false, dialplan will go further as I have set >> on-true. >> How can I prevent that dialplan will go after break="on-true" on >> second condition and will not go looking condition inside condition. >> >> So if the variable mobilne is not set, in this dialplan FS will >> go looking to > expression="^(051|041|031|030|040|070|071)(\d{6})|^(0038651|0038641|0038631|0038630|0038640|0038670|0038671)(\d{6})" >> break="on-true"> and reject call instead of goint to second >> condtion which is > expression="tuje"/>. >> >> I hope I make it clear:D >> >> > break="on-true" /> >> > expression="^(051|041|031|030|040|070|071)(\d{6})|^(0038651|0038641|0038631|0038630|0038640|0038670|0038671)(\d{6})" >> break="on-true"> >> >> >> >> >> >> >> >> >> >> >> >> >> > expression="^(090)(\d{4})|^(090)(\d{6})"> >> >> >> >> >> Regards and thank you for your help! >> >> Miha >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/df6c4cc2/attachment.html From benkokakao at gmail.com Mon Mar 26 13:20:14 2012 From: benkokakao at gmail.com (Christian Benke) Date: Mon, 26 Mar 2012 11:20:14 +0200 Subject: [Freeswitch-users] FreeSWITCH Project - Call For Assistance In-Reply-To: References: Message-ID: > FreeSWITCH debug log syntax highlighting - If anyone has any experience with > creating syntax highlighting for any of the popular text editors (Emacs, > VIM, Textmate, Notepad++, UltraEdit, etc.) then please let me know. I'd like > to get syntax highlighting for at least one text editor in *nix and one in > Windows. For vim it's easy, there's a syntax highlighter by Michael Ricordeau(https://github.com/tamiel/Freeswitch_log_vim_syntax): git clone https://github.com/tamiel/Freeswitch_log_vim_syntax.git /etc/vim/syntax In vim: :set syntax=fslog Regards, Christian From Daniel.Knaggs at realitysolutions.co.uk Mon Mar 26 14:01:27 2012 From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs) Date: Mon, 26 Mar 2012 10:01:27 +0000 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: References: Message-ID: OK, call recording has been setup - waiting for it to happen now. Interestingly, before I issue the "record_session" application (and of course the "RECORD_*" variables) I had to execute "ring_ready" then "pre_answer" otherwise the caller gets silence (changing the order of those two commands results in silence). From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Daniel Knaggs Sent: 26 March 2012 08:44 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls Not a bad idea, Michael! I'll see if I can get that sorted. The DTMF tones only appear to be "lettered" ones, could I comment those out in the source code and rebuild (which files would I need to change)? Not the best workaround but should work. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 25 March 2012 02:20 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls I'd do an audio recording of the call and open the resulting file in Audacity. It can do kind of a spectrum analysis to show you if there are any frequencies that might be fooling the dtmf detector. Also, I'm not familiar with using Digium cards with FreeTDM. (I have a Digium TE121 or similar but I don't happen to have any mobo's with the PCIe or whatever slot type it uses.) To Moises I'd ask: is there hardware DTMF detection in the Digium cards that you know of? Also, any known or suspected issues with false DTMF detection in the scenario mentioned by the OP? Gracias, MC On Sat, Mar 24, 2012 at 5:27 AM, Daniel Knaggs > wrote: They aren't any faults that I can see. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 Sent: 23 March 2012 23:23 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls Have you checked for any faults on your TDM link on your side? On Fri, Mar 23, 2012 at 10:32 AM, Brian Foster > wrote: Please update to latest git. -BDF On Fri, Mar 23, 2012 at 7:19 AM, Daniel Knaggs > wrote: Hello all, Got a bit of a strange one, we appear to be getting DTMF tones on incoming calls when the caller hasn't even pressed any keys. It normally happens with 10 seconds or so after the call has been answered. Here is the log of it happening earlier: - 2012-03-23 10:38:16.852654 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF A (debug = 0) 2012-03-23 10:38:16.852654 [DEBUG] mod_freetdm.c:799 Queuing DTMF [A] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:16.915653 [DEBUG] switch_rtp.c:2420 Send start packet for [A] ts=49440 dur=160/160/2000 seq=46803 lw=49440 2012-03-23 10:38:16.936653 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=320/320/2000 seq=46804 lw=49600 2012-03-23 10:38:16.957652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=480/480/2000 seq=46805 lw=49760 2012-03-23 10:38:16.978652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=640/640/2000 seq=46806 lw=49920 2012-03-23 10:38:16.999652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=800/800/2000 seq=46807 lw=50080 2012-03-23 10:38:17.020651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=960/960/2000 seq=46808 lw=50240 2012-03-23 10:38:17.041651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1120/1120/2000 seq=46809 lw=50400 2012-03-23 10:38:17.062651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1280/1280/2000 seq=46810 lw=50560 2012-03-23 10:38:17.083650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1440/1440/2000 seq=46811 lw=50720 2012-03-23 10:38:17.104650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1600/1600/2000 seq=46812 lw=50880 2012-03-23 10:38:17.125650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1760/1760/2000 seq=46813 lw=51040 2012-03-23 10:38:17.146650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1920/1920/2000 seq=46814 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46815 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46816 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46817 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2271 Queue digit delay of 40ms 2012-03-23 10:38:18.070638 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF B (debug = 0) 2012-03-23 10:38:18.070638 [DEBUG] mod_freetdm.c:799 Queuing DTMF [B] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:18.133637 [DEBUG] switch_rtp.c:2420 Send start packet for [B] ts=59040 dur=160/160/2000 seq=46864 lw=59040 2012-03-23 10:38:18.154637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=320/320/2000 seq=46865 lw=59200 2012-03-23 10:38:18.175637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=480/480/2000 seq=46866 lw=59360 2012-03-23 10:38:18.196636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=640/640/2000 seq=46867 lw=59520 2012-03-23 10:38:18.217636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=800/800/2000 seq=46868 lw=59680 2012-03-23 10:38:18.238636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=960/960/2000 seq=46869 lw=59840 2012-03-23 10:38:18.259636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1120/1120/2000 seq=46870 lw=60000 2012-03-23 10:38:18.280636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1280/1280/2000 seq=46871 lw=60160 2012-03-23 10:38:18.301635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1440/1440/2000 seq=46872 lw=60320 2012-03-23 10:38:18.322635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1600/1600/2000 seq=46873 lw=60480 2012-03-23 10:38:18.343634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1760/1760/2000 seq=46874 lw=60640 2012-03-23 10:38:18.364634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1920/1920/2000 seq=46875 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46876 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46877 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46878 lw=60800 I'm very sure the caller isn't dialling "A" or "B"! Running on ISDN via a TE121 card using E1 (euroisdn). Freeswitch version is "FreeSWITCH Version 1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)". Currently there is a "bind_meta_app" in the config which binds a script on the B leg of the call which parks the call - I haven't tried turning this off yet to see if it's this. Wondering if anyone has any ideas or has come across this before? Thanks in advance. [Description: Description: Description: cid:imageacd695.PNG at 9e92e461.40aa4b96] Daniel Knaggs Software Developer Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE Tel: 01482 828000 / Fax: 01482 373100 Daniel.Knaggs at realitysolutions.co.uk www.realitysolutions.co.uk ________________________________ Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire [Description: Description: Description: cid:image27a71e.PNG at c2da8488.4683bff1] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/0100543a/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 22463 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/0100543a/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 69075 bytes Desc: image002.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/0100543a/attachment-0003.png From a.avona at elios.net Mon Mar 26 14:01:10 2012 From: a.avona at elios.net (a.avona) Date: Mon, 26 Mar 2012 12:01:10 +0200 Subject: [Freeswitch-users] Grandstream 4104 In-Reply-To: References: <4F6C46B0.7080805@elios.net> Message-ID: <4F703E66.5070607@elios.net> Hi, thank's for your answer i did as you said and if i digit originate sofia/internal/XXXX at 192.168.0.3 9178 outgoing calls work well. if i try to originate call from a client it doesn't work so i think the problem is in the default account configuration here is the siptraces Thank's for any suggestion reagards Accursio Avona recv 865 bytes from udp/[192.168.0.200]:5060 at 09:31:18.696255: ------------------------------------------------------------------------ INVITE sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport From: "1000" ;tag=645977894 To: Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 5 INVITE Contact: Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces, from-change User-Agent: SIPPER for PhonerLite P-Preferred-Identity: Content-Length: 260 v=0 o=- 162748822 0 IN IP4 192.168.0.200 s=SIPPER for PhonerLite c=IN IP4 192.168.0.200 t=0 0 m=audio 5062 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv ------------------------------------------------------------------------ send 379 bytes to udp/[192.168.0.200]:5060 at 09:31:18.696482: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport=5060 From: "1000" ;tag=645977894 To: Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 5 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:7559 IP 192.168.0.200 Rejected by acl "domains". Falling back to Digest auth. send 865 bytes to udp/[192.168.0.200]:5060 at 09:31:18.697335: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport=5060 From: "1000" ;tag=645977894 To: ;tag=pvaSD4jQ22N9D Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 5 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="192.168.0.2", nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 346 bytes from udp/[192.168.0.200]:5060 at 09:31:18.698468: ------------------------------------------------------------------------ ACK sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport From: "1000" ;tag=645977894 To: ;tag=pvaSD4jQ22N9D Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 5 ACK Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ recv 1137 bytes from udp/[192.168.0.200]:5060 at 09:31:18.699228: ------------------------------------------------------------------------ INVITE sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport From: "1000" ;tag=645977894 To: Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 6 INVITE Contact: Proxy-Authorization: Digest username="1000", realm="192.168.0.2", nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", uri="sip:339XXXXXXXX at 192.168.0.2", response="7378107bc277149e3b6ef00c1c766a71", algorithm=MD5, cnonce="234abcc436e2667097e7fe6eia53e8dd", qop=auth, nc=00000001 Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces, from-change User-Agent: SIPPER for PhonerLite P-Preferred-Identity: Content-Length: 260 v=0 o=- 162748822 0 IN IP4 192.168.0.200 s=SIPPER for PhonerLite c=IN IP4 192.168.0.200 t=0 0 m=audio 5062 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv ------------------------------------------------------------------------ send 379 bytes to udp/[192.168.0.200]:5060 at 09:31:18.699380: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport=5060 From: "1000" ;tag=645977894 To: Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 6 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:7559 IP 192.168.0.200 Rejected by acl "domains". Falling back to Digest auth. 2012-03-26 11:31:18.692558 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1000 at 192.168.0.2 [710da826-7726-11e1-bf72-e50b8b76e101] 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5526 Channel sofia/internal/1000 at 192.168.0.2 entering state [received][100] 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5537 Remote SDP: v=0 o=- 162748822 0 IN IP4 192.168.0.200 s=SIPPER for PhonerLite c=IN IP4 192.168.0.200 t=0 0 m=audio 5062 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:2991 Set Codec sofia/internal/1000 at 192.168.0.2 PCMA/8000 20 ms 160 samples 64000 bits 2012-03-26 11:31:18.692558 [DEBUG] switch_core_codec.c:111 sofia/internal/1000 at 192.168.0.2 Original read codec set to PCMA:8 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4995 Set 2833 dtmf send/recv payload to 101 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5749 (sofia/internal/1000 at 192.168.0.2) State Change CS_NEW -> CS_INIT 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_INIT 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1000 at 192.168.0.2) State INIT 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:85 sofia/internal/1000 at 192.168.0.2 SOFIA INIT 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:125 (sofia/internal/1000 at 192.168.0.2) State Change CS_INIT -> CS_ROUTING 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1000 at 192.168.0.2) State INIT going to sleep 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_ROUTING 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1886 (sofia/internal/1000 at 192.168.0.2) Callstate Change DOWN -> RINGING 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1000 at 192.168.0.2) State ROUTING 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:148 sofia/internal/1000 at 192.168.0.2 SOFIA ROUTING 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1000 at 192.168.0.2 Standard ROUTING 2012-03-26 11:31:18.692558 [INFO] mod_dialplan_xml.c:485 Processing 1000 <1000>->339XXXXXXXX in context default Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unloop] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->tod_example] continue=true Dialplan: sofia/internal/1000 at 192.168.0.2 Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(open=true) Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->holiday_example] continue=true Dialplan: sofia/internal/1000 at 192.168.0.2 Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global-intercept] destination_number(339XXXXXXXX) =~ /^886$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group-intercept] destination_number(339XXXXXXXX) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [intercept-ext] destination_number(339XXXXXXXX) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->redial] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [redial] destination_number(339XXXXXXXX) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->global] continue=true Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/1000 at 192.168.0.2 Absolute Condition [global] Dialplan: sofia/internal/1000 at 192.168.0.2 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1000 at 192.168.0.2 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1000 at 192.168.0.2 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1000 at 192.168.0.2 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-2] destination_number(339XXXXXXXX) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-1] destination_number(339XXXXXXXX) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] destination_number(339XXXXXXXX) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] destination_number(339XXXXXXXX) =~ /^779$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->call_return] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call_return] destination_number(339XXXXXXXX) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->del-group] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [del-group] destination_number(339XXXXXXXX) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->add-group] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [add-group] destination_number(339XXXXXXXX) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call-group-simo] destination_number(339XXXXXXXX) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call-group-order] destination_number(339XXXXXXXX) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [extension-intercom] destination_number(339XXXXXXXX) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [Local_Extension] destination_number(339XXXXXXXX) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->Local_Extension_Skinny] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [Local_Extension_Skinny] destination_number(339XXXXXXXX) =~ /^(11[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->group_dial_sales] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group_dial_sales] destination_number(339XXXXXXXX) =~ /^2000$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->group_dial_support] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group_dial_support] destination_number(339XXXXXXXX) =~ /^2001$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->group_dial_billing] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group_dial_billing] destination_number(339XXXXXXXX) =~ /^2002$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->operator] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [operator] destination_number(339XXXXXXXX) =~ /^(operator|0)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->vmain] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [vmain] destination_number(339XXXXXXXX) =~ /^vmain$|^4000$|^\*98$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->sip_uri] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [sip_uri] destination_number(339XXXXXXXX) =~ /^sip:(.*)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->nb_conferences] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [nb_conferences] destination_number(339XXXXXXXX) =~ /^(30\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->wb_conferences] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wb_conferences] destination_number(339XXXXXXXX) =~ /^(31\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->uwb_conferences] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [uwb_conferences] destination_number(339XXXXXXXX) =~ /^(32\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->cdquality_conferences] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [cdquality_conferences] destination_number(339XXXXXXXX) =~ /^(33\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->freeswitch_public_conf_via_sip] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(339XXXXXXXX) =~ /^9(888|8888|1616|3232)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->mad_boss_intercom] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [mad_boss_intercom] destination_number(339XXXXXXXX) =~ /^0911$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->mad_boss_intercom] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [mad_boss_intercom] destination_number(339XXXXXXXX) =~ /^0912$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->mad_boss] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [mad_boss] destination_number(339XXXXXXXX) =~ /^0913$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ivr_demo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ivr_demo] destination_number(339XXXXXXXX) =~ /^5000$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->dynamic_conference] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [dynamic_conference] destination_number(339XXXXXXXX) =~ /^5001$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->rtp_multicast_page] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [rtp_multicast_page] destination_number(339XXXXXXXX) =~ /^pagegroup$|^7243$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] destination_number(339XXXXXXXX) =~ /^5900$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] destination_number(339XXXXXXXX) =~ /^5901$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->valet_park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] destination_number(339XXXXXXXX) =~ /^(6000)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->valet_park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] destination_number(339XXXXXXXX) =~ /^(60\d[1-9])$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] destination_number(339XXXXXXXX) =~ /park\+(\d+)/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] destination_number(339XXXXXXXX) =~ /^parking$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] destination_number(339XXXXXXXX) =~ /callpark/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] destination_number(339XXXXXXXX) =~ /pickup/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->wait] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wait] destination_number(339XXXXXXXX) =~ /^wait$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->fax_receive] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_receive] destination_number(339XXXXXXXX) =~ /^9178$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->fax_transmit] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_transmit] destination_number(339XXXXXXXX) =~ /^9179$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_180] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_180] destination_number(339XXXXXXXX) =~ /^9180$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_183_uk_ring] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_183_uk_ring] destination_number(339XXXXXXXX) =~ /^9181$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_183_music_ring] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_183_music_ring] destination_number(339XXXXXXXX) =~ /^9182$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_post_answer_uk_ring] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(339XXXXXXXX) =~ /^9183$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_post_answer_music] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_post_answer_music] destination_number(339XXXXXXXX) =~ /^9184$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ClueCon] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ClueCon] destination_number(339XXXXXXXX) =~ /^9191$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->show_info] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [show_info] destination_number(339XXXXXXXX) =~ /^9192$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->video_record] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_record] destination_number(339XXXXXXXX) =~ /^9193$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->video_playback] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_playback] destination_number(339XXXXXXXX) =~ /^9194$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->delay_echo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [delay_echo] destination_number(339XXXXXXXX) =~ /^9195$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->echo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [echo] destination_number(339XXXXXXXX) =~ /^9196$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->milliwatt] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [milliwatt] destination_number(339XXXXXXXX) =~ /^9197$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->tone_stream] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [tone_stream] destination_number(339XXXXXXXX) =~ /^9198$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->zrtp_enrollement] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [zrtp_enrollement] destination_number(339XXXXXXXX) =~ /^9787$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->hold_music] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [hold_music] destination_number(339XXXXXXXX) =~ /^9664$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->from_pstn] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [from_pstn] destination_number(339XXXXXXXX) =~ /^0000$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->101] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [101] destination_number(339XXXXXXXX) =~ /^101$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->pizza_demo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [pizza_demo] destination_number(339XXXXXXXX) =~ /^(pizza|74992)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->gxw4104-fxo-local] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [gxw4104-fxo-local] ${toll_allow}(domestic,international,local) =~ /local/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [gxw4104-fxo-local] destination_number(339XXXXXXXX) =~ /^(\d{6,})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(effective_caller_id_number=0321234567) Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(effective_caller_id_name=ThisIsMyCompany) Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(ignore_early_media=ring_ready) Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(ringback=${us-ring}) Dialplan: sofia/internal/1000 at 192.168.0.2 Action bridge(sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/1000 at 192.168.0.2) State Change CS_ROUTING -> CS_EXECUTE 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1000 at 192.168.0.2) State ROUTING going to sleep 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_EXECUTE 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1000 at 192.168.0.2) State EXECUTE 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:241 sofia/internal/1000 at 192.168.0.2 SOFIA EXECUTE 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:192 sofia/internal/1000 at 192.168.0.2 Standard EXECUTE EXECUTE sofia/internal/1000 at 192.168.0.2 set(open=true) 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [open]=[true] EXECUTE sofia/internal/1000 at 192.168.0.2 hash(insert/192.168.0.2-spymap/1000/710da826-7726-11e1-bf72-e50b8b76e101) EXECUTE sofia/internal/1000 at 192.168.0.2 hash(insert/192.168.0.2-last_dial/1000/339XXXXXXXX) EXECUTE sofia/internal/1000 at 192.168.0.2 hash(insert/192.168.0.2-last_dial/global/710da826-7726-11e1-bf72-e50b8b76e101) EXECUTE sofia/internal/1000 at 192.168.0.2 export(RFC2822_DATE=Mon, 26 Mar 2012 11:31:18 +0200) 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [RFC2822_DATE]=[Mon, 26 Mar 2012 11:31:18 +0200] EXECUTE sofia/internal/1000 at 192.168.0.2 set(effective_caller_id_number=0321234567) 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [effective_caller_id_number]=[0321234567] EXECUTE sofia/internal/1000 at 192.168.0.2 set(effective_caller_id_name=ThisIsMyCompany) 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [effective_caller_id_name]=[ThisIsMyCompany] EXECUTE sofia/internal/1000 at 192.168.0.2 set(ignore_early_media=ring_ready) 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [ignore_early_media]=[ring_ready] EXECUTE sofia/internal/1000 at 192.168.0.2 set(ringback=%(2000,4000,440,480)) 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [ringback]=[%(2000,4000,440,480)] EXECUTE sofia/internal/1000 at 192.168.0.2 bridge(sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1047 sofia/internal/1000 at 192.168.0.2 EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Mar 2012 11:31:18 +0200] to event 2012-03-26 11:31:18.692558 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-26 11:31:18.692558 [NOTICE] switch_channel.c:926 New Channel sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [710f6738-7726-11e1-bf77-e50b8b76e101] 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:4691 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change CS_NEW -> CS_INIT 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State Change CS_INIT 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State INIT 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:85 sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA INIT send 1279 bytes to udp/[192.168.0.3]:5060 at 09:31:18.713245: ------------------------------------------------------------------------ INVITE sip:gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K Max-Forwards: 69 From: "ThisIsMyCompany" ;tag=reXaHtmyvm2eN To: Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a CSeq: 26041075 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 311 X-FS-Support: update_display,send_info P-Asserted-Identity: "ThisIsMyCompany" v=0 o=FreeSWITCH 1332736038 1332736039 IN IP4 192.168.0.2 s=FreeSWITCH c=IN IP4 192.168.0.2 t=0 0 m=audio 18240 RTP/AVP 8 98 99 9 0 3 101 13 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:125 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change CS_INIT -> CS_ROUTING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State INIT going to sleep 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State Change CS_ROUTING 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:1886 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate Change DOWN -> RINGING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State ROUTING 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:148 sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA ROUTING 2012-03-26 11:31:18.712698 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State ROUTING going to sleep 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State Change CS_CONSUME_MEDIA 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State CONSUME_MEDIA 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State CONSUME_MEDIA going to sleep 2012-03-26 11:31:18.712698 [DEBUG] sofia.c:5526 Channel sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 entering state [calling][0] recv 355 bytes from udp/[192.168.0.3]:5060 at 09:31:18.717462: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K From: "ThisIsMyCompany" ;tag=reXaHtmyvm2eN To: Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a CSeq: 26041075 INVITE User-Agent: Grandstream GXW4104 (HW 2.0, Ch:5) 1.3.4.9 Content-Length: 0 ------------------------------------------------------------------------ recv 367 bytes from udp/[192.168.0.3]:5060 at 09:31:18.717980: ------------------------------------------------------------------------ SIP/2.0 403 Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K From: "ThisIsMyCompany" ;tag=reXaHtmyvm2eN To: ;tag=3ga2B2jmSSe6H Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a CSeq: 26041075 INVITE User-Agent: Grandstream GXW4104 (HW 2.0, Ch:8) 1.3.4.9 Content-Length: 0 ------------------------------------------------------------------------ send 370 bytes to udp/[192.168.0.3]:5060 at 09:31:18.718065: ------------------------------------------------------------------------ ACK sip:gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K Max-Forwards: 69 From: "ThisIsMyCompany" ;tag=reXaHtmyvm2eN To: ;tag=3ga2B2jmSSe6H Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a CSeq: 26041075 ACK Content-Length: 0 ------------------------------------------------------------------------ 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] sofia.c:5526 Channel sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 entering state [terminated][403] 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2848 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate Change RINGING -> HANGUP 2012-03-26 11:31:18.712698 [NOTICE] sofia.c:6293 Hangup sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [CS_CONSUME_MEDIA] [CALL_REJECTED] 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [KILL] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State Change CS_HANGUP 2012-03-26 11:31:18.712698 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 21 [CALL_REJECTED] 2012-03-26 11:31:18.712698 [INFO] mod_dptools.c:2922 Originate Failed. Cause: CALL_REJECTED 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2848 (sofia/internal/1000 at 192.168.0.2) Callstate Change RINGING -> HANGUP 2012-03-26 11:31:18.712698 [NOTICE] mod_dptools.c:3041 Hangup sofia/internal/1000 at 192.168.0.2 [CS_EXECUTE] [CALL_REJECTED] 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/1000 at 192.168.0.2 [KILL] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:2285 sofia/internal/1000 at 192.168.0.2 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1000 at 192.168.0.2) State EXECUTE going to sleep 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_HANGUP 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1000 at 192.168.0.2) State HANGUP 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:463 sofia/internal/1000 at 192.168.0.2 Overriding SIP cause 603 with 403 from the other leg 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1000 at 192.168.0.2 hanging up, cause: CALL_REJECTED 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:534 Responding to INVITE with: 403 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1000 at 192.168.0.2 Standard HANGUP, cause: CALL_REJECTED 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1000 at 192.168.0.2) State HANGUP going to sleep 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1000 at 192.168.0.2) State Change CS_HANGUP -> CS_REPORTING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_REPORTING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1000 at 192.168.0.2) State REPORTING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1000 at 192.168.0.2 Standard REPORTING, cause: CALL_REJECTED 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1000 at 192.168.0.2) State REPORTING going to sleep 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1000 at 192.168.0.2) State Change CS_REPORTING -> CS_DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1380 Session 13 (sofia/internal/1000 at 192.168.0.2) Locked, Waiting on external entities 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1398 Session 13 (sofia/internal/1000 at 192.168.0.2) Ended 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/1000 at 192.168.0.2 [CS_DESTROY] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1000 at 192.168.0.2) Callstate Change HANGUP -> DOWN 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1000 at 192.168.0.2) State DESTROY 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:374 sofia/internal/1000 at 192.168.0.2 SOFIA DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1000 at 192.168.0.2 Standard DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1000 at 192.168.0.2) State DESTROY going to sleep send 833 bytes to udp/[192.168.0.200]:5060 at 09:31:18.720584: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport=5060 From: "1000" ;tag=645977894 To: ;tag=Q53HFZ3tZBcvS Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 6 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=21;text="CALL_REJECTED" Content-Length: 0 P-Asserted-Identity: "339XXXXXXXX" ------------------------------------------------------------------------ 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State HANGUP 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:469 Channel sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 hanging up, cause: CALL_REJECTED 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:47 sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard HANGUP, cause: CALL_REJECTED 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State HANGUP going to sleep 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change CS_HANGUP -> CS_REPORTING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State Change CS_REPORTING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State REPORTING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:79 sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard REPORTING, cause: CALL_REJECTED 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State REPORTING going to sleep recv 618 bytes from udp/[192.168.0.200]:5060 at 09:31:18.721757: ------------------------------------------------------------------------ ACK sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport From: "1000" ;tag=645977894 To: ;tag=Q53HFZ3tZBcvS Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 6 ACK Proxy-Authorization: Digest username="1000", realm="192.168.0.2", nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", uri="sip:339XXXXXXXX at 192.168.0.2", response="7378107bc277149e3b6ef00c1c766a71", algorithm=MD5, cnonce="234abcc436e2667097e7fe6eia53e8dd", qop=auth, nc=00000001 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change CS_REPORTING -> CS_DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1380 Session 14 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Locked, Waiting on external entities 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1398 Session 14 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Ended 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [CS_DESTROY] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate Change HANGUP -> DOWN 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State Change CS_DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State DESTROY 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:374 sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:86 sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State DESTROY going to sleep freeswitch at internal> Il 25/03/2012 06:39, Anton Kvashenkin ha scritto: > originate sofia/internal/XXXX at 192.168.0.3 9178 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/d101e370/attachment-0001.html From a.avona at elios.net Mon Mar 26 15:04:20 2012 From: a.avona at elios.net (a.avona) Date: Mon, 26 Mar 2012 13:04:20 +0200 Subject: [Freeswitch-users] Grandstream 4104 In-Reply-To: References: <4F6C46B0.7080805@elios.net> Message-ID: <4F704D34.9020504@elios.net> Hi, thank's for your answer i did as you said and if i digit originate sofia/internal/XXXX at 192.168.0.3 9178 outgoing calls work well. if i try to originate call from a client it doesn't work so i think the problem is in the default account configuration here is the siptraces Thank's for any suggestion reagards Accursio Avona recv 865 bytes from udp/[192.168.0.200]:5060 at 09:31:18.696255: ------------------------------------------------------------------------ INVITE sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport From: "1000" ;tag=645977894 To: Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 5 INVITE Contact: Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces, from-change User-Agent: SIPPER for PhonerLite P-Preferred-Identity: Content-Length: 260 v=0 o=- 162748822 0 IN IP4 192.168.0.200 s=SIPPER for PhonerLite c=IN IP4 192.168.0.200 t=0 0 m=audio 5062 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv ------------------------------------------------------------------------ send 379 bytes to udp/[192.168.0.200]:5060 at 09:31:18.696482: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport=5060 From: "1000" ;tag=645977894 To: Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 5 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:7559 IP 192.168.0.200 Rejected by acl "domains". Falling back to Digest auth. send 865 bytes to udp/[192.168.0.200]:5060 at 09:31:18.697335: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport=5060 From: "1000" ;tag=645977894 To: ;tag=pvaSD4jQ22N9D Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 5 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="192.168.0.2", nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 346 bytes from udp/[192.168.0.200]:5060 at 09:31:18.698468: ------------------------------------------------------------------------ ACK sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport From: "1000" ;tag=645977894 To: ;tag=pvaSD4jQ22N9D Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 5 ACK Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ recv 1137 bytes from udp/[192.168.0.200]:5060 at 09:31:18.699228: ------------------------------------------------------------------------ INVITE sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport From: "1000" ;tag=645977894 To: Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 6 INVITE Contact: Proxy-Authorization: Digest username="1000", realm="192.168.0.2", nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", uri="sip:339XXXXXXXX at 192.168.0.2", response="7378107bc277149e3b6ef00c1c766a71", algorithm=MD5, cnonce="234abcc436e2667097e7fe6eia53e8dd", qop=auth, nc=00000001 Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces, from-change User-Agent: SIPPER for PhonerLite P-Preferred-Identity: Content-Length: 260 v=0 o=- 162748822 0 IN IP4 192.168.0.200 s=SIPPER for PhonerLite c=IN IP4 192.168.0.200 t=0 0 m=audio 5062 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv ------------------------------------------------------------------------ send 379 bytes to udp/[192.168.0.200]:5060 at 09:31:18.699380: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport=5060 From: "1000" ;tag=645977894 To: Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 6 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:7559 IP 192.168.0.200 Rejected by acl "domains". Falling back to Digest auth. 2012-03-26 11:31:18.692558 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1000 at 192.168.0.2 [710da826-7726-11e1-bf72-e50b8b76e101] 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5526 Channel sofia/internal/1000 at 192.168.0.2 entering state [received][100] 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5537 Remote SDP: v=0 o=- 162748822 0 IN IP4 192.168.0.200 s=SIPPER for PhonerLite c=IN IP4 192.168.0.200 t=0 0 m=audio 5062 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:2991 Set Codec sofia/internal/1000 at 192.168.0.2 PCMA/8000 20 ms 160 samples 64000 bits 2012-03-26 11:31:18.692558 [DEBUG] switch_core_codec.c:111 sofia/internal/1000 at 192.168.0.2 Original read codec set to PCMA:8 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4995 Set 2833 dtmf send/recv payload to 101 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5749 (sofia/internal/1000 at 192.168.0.2) State Change CS_NEW -> CS_INIT 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_INIT 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1000 at 192.168.0.2) State INIT 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:85 sofia/internal/1000 at 192.168.0.2 SOFIA INIT 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:125 (sofia/internal/1000 at 192.168.0.2) State Change CS_INIT -> CS_ROUTING 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1000 at 192.168.0.2) State INIT going to sleep 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_ROUTING 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1886 (sofia/internal/1000 at 192.168.0.2) Callstate Change DOWN -> RINGING 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1000 at 192.168.0.2) State ROUTING 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:148 sofia/internal/1000 at 192.168.0.2 SOFIA ROUTING 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1000 at 192.168.0.2 Standard ROUTING 2012-03-26 11:31:18.692558 [INFO] mod_dialplan_xml.c:485 Processing 1000 <1000>->339XXXXXXXX in context default Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unloop] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->tod_example] continue=true Dialplan: sofia/internal/1000 at 192.168.0.2 Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(open=true) Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->holiday_example] continue=true Dialplan: sofia/internal/1000 at 192.168.0.2 Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global-intercept] destination_number(339XXXXXXXX) =~ /^886$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group-intercept] destination_number(339XXXXXXXX) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [intercept-ext] destination_number(339XXXXXXXX) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->redial] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [redial] destination_number(339XXXXXXXX) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->global] continue=true Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/1000 at 192.168.0.2 Absolute Condition [global] Dialplan: sofia/internal/1000 at 192.168.0.2 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1000 at 192.168.0.2 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1000 at 192.168.0.2 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1000 at 192.168.0.2 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-2] destination_number(339XXXXXXXX) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-1] destination_number(339XXXXXXXX) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] destination_number(339XXXXXXXX) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] destination_number(339XXXXXXXX) =~ /^779$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->call_return] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call_return] destination_number(339XXXXXXXX) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->del-group] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [del-group] destination_number(339XXXXXXXX) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->add-group] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [add-group] destination_number(339XXXXXXXX) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call-group-simo] destination_number(339XXXXXXXX) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call-group-order] destination_number(339XXXXXXXX) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [extension-intercom] destination_number(339XXXXXXXX) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [Local_Extension] destination_number(339XXXXXXXX) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->Local_Extension_Skinny] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [Local_Extension_Skinny] destination_number(339XXXXXXXX) =~ /^(11[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->group_dial_sales] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group_dial_sales] destination_number(339XXXXXXXX) =~ /^2000$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->group_dial_support] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group_dial_support] destination_number(339XXXXXXXX) =~ /^2001$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->group_dial_billing] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group_dial_billing] destination_number(339XXXXXXXX) =~ /^2002$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->operator] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [operator] destination_number(339XXXXXXXX) =~ /^(operator|0)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->vmain] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [vmain] destination_number(339XXXXXXXX) =~ /^vmain$|^4000$|^\*98$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->sip_uri] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [sip_uri] destination_number(339XXXXXXXX) =~ /^sip:(.*)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->nb_conferences] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [nb_conferences] destination_number(339XXXXXXXX) =~ /^(30\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->wb_conferences] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wb_conferences] destination_number(339XXXXXXXX) =~ /^(31\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->uwb_conferences] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [uwb_conferences] destination_number(339XXXXXXXX) =~ /^(32\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->cdquality_conferences] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [cdquality_conferences] destination_number(339XXXXXXXX) =~ /^(33\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->freeswitch_public_conf_via_sip] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(339XXXXXXXX) =~ /^9(888|8888|1616|3232)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->mad_boss_intercom] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [mad_boss_intercom] destination_number(339XXXXXXXX) =~ /^0911$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->mad_boss_intercom] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [mad_boss_intercom] destination_number(339XXXXXXXX) =~ /^0912$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->mad_boss] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [mad_boss] destination_number(339XXXXXXXX) =~ /^0913$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ivr_demo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ivr_demo] destination_number(339XXXXXXXX) =~ /^5000$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->dynamic_conference] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [dynamic_conference] destination_number(339XXXXXXXX) =~ /^5001$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->rtp_multicast_page] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [rtp_multicast_page] destination_number(339XXXXXXXX) =~ /^pagegroup$|^7243$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] destination_number(339XXXXXXXX) =~ /^5900$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] destination_number(339XXXXXXXX) =~ /^5901$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->valet_park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] destination_number(339XXXXXXXX) =~ /^(6000)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->valet_park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] destination_number(339XXXXXXXX) =~ /^(60\d[1-9])$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] destination_number(339XXXXXXXX) =~ /park\+(\d+)/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] destination_number(339XXXXXXXX) =~ /^parking$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] destination_number(339XXXXXXXX) =~ /callpark/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] destination_number(339XXXXXXXX) =~ /pickup/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->wait] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wait] destination_number(339XXXXXXXX) =~ /^wait$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->fax_receive] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_receive] destination_number(339XXXXXXXX) =~ /^9178$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->fax_transmit] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_transmit] destination_number(339XXXXXXXX) =~ /^9179$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_180] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_180] destination_number(339XXXXXXXX) =~ /^9180$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_183_uk_ring] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_183_uk_ring] destination_number(339XXXXXXXX) =~ /^9181$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_183_music_ring] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_183_music_ring] destination_number(339XXXXXXXX) =~ /^9182$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_post_answer_uk_ring] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(339XXXXXXXX) =~ /^9183$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_post_answer_music] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_post_answer_music] destination_number(339XXXXXXXX) =~ /^9184$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ClueCon] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ClueCon] destination_number(339XXXXXXXX) =~ /^9191$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->show_info] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [show_info] destination_number(339XXXXXXXX) =~ /^9192$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->video_record] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_record] destination_number(339XXXXXXXX) =~ /^9193$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->video_playback] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_playback] destination_number(339XXXXXXXX) =~ /^9194$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->delay_echo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [delay_echo] destination_number(339XXXXXXXX) =~ /^9195$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->echo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [echo] destination_number(339XXXXXXXX) =~ /^9196$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->milliwatt] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [milliwatt] destination_number(339XXXXXXXX) =~ /^9197$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->tone_stream] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [tone_stream] destination_number(339XXXXXXXX) =~ /^9198$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->zrtp_enrollement] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [zrtp_enrollement] destination_number(339XXXXXXXX) =~ /^9787$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->hold_music] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [hold_music] destination_number(339XXXXXXXX) =~ /^9664$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->from_pstn] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [from_pstn] destination_number(339XXXXXXXX) =~ /^0000$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->101] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [101] destination_number(339XXXXXXXX) =~ /^101$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->pizza_demo] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [pizza_demo] destination_number(339XXXXXXXX) =~ /^(pizza|74992)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->gxw4104-fxo-local] continue=false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [gxw4104-fxo-local] ${toll_allow}(domestic,international,local) =~ /local/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [gxw4104-fxo-local] destination_number(339XXXXXXXX) =~ /^(\d{6,})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(effective_caller_id_number=0321234567) Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(effective_caller_id_name=ThisIsMyCompany) Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(ignore_early_media=ring_ready) Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(ringback=${us-ring}) Dialplan: sofia/internal/1000 at 192.168.0.2 Action bridge(sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/1000 at 192.168.0.2) State Change CS_ROUTING -> CS_EXECUTE 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1000 at 192.168.0.2) State ROUTING going to sleep 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_EXECUTE 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1000 at 192.168.0.2) State EXECUTE 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:241 sofia/internal/1000 at 192.168.0.2 SOFIA EXECUTE 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:192 sofia/internal/1000 at 192.168.0.2 Standard EXECUTE EXECUTE sofia/internal/1000 at 192.168.0.2 set(open=true) 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [open]=[true] EXECUTE sofia/internal/1000 at 192.168.0.2 hash(insert/192.168.0.2-spymap/1000/710da826-7726-11e1-bf72-e50b8b76e101) EXECUTE sofia/internal/1000 at 192.168.0.2 hash(insert/192.168.0.2-last_dial/1000/339XXXXXXXX) EXECUTE sofia/internal/1000 at 192.168.0.2 hash(insert/192.168.0.2-last_dial/global/710da826-7726-11e1-bf72-e50b8b76e101) EXECUTE sofia/internal/1000 at 192.168.0.2 export(RFC2822_DATE=Mon, 26 Mar 2012 11:31:18 +0200) 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [RFC2822_DATE]=[Mon, 26 Mar 2012 11:31:18 +0200] EXECUTE sofia/internal/1000 at 192.168.0.2 set(effective_caller_id_number=0321234567) 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [effective_caller_id_number]=[0321234567] EXECUTE sofia/internal/1000 at 192.168.0.2 set(effective_caller_id_name=ThisIsMyCompany) 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [effective_caller_id_name]=[ThisIsMyCompany] EXECUTE sofia/internal/1000 at 192.168.0.2 set(ignore_early_media=ring_ready) 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [ignore_early_media]=[ring_ready] EXECUTE sofia/internal/1000 at 192.168.0.2 set(ringback=%(2000,4000,440,480)) 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 sofia/internal/1000 at 192.168.0.2 SET [ringback]=[%(2000,4000,440,480)] EXECUTE sofia/internal/1000 at 192.168.0.2 bridge(sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1047 sofia/internal/1000 at 192.168.0.2 EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Mar 2012 11:31:18 +0200] to event 2012-03-26 11:31:18.692558 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-26 11:31:18.692558 [NOTICE] switch_channel.c:926 New Channel sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [710f6738-7726-11e1-bf77-e50b8b76e101] 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:4691 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change CS_NEW -> CS_INIT 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State Change CS_INIT 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State INIT 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:85 sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA INIT send 1279 bytes to udp/[192.168.0.3]:5060 at 09:31:18.713245: ------------------------------------------------------------------------ INVITE sip:gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K Max-Forwards: 69 From: "ThisIsMyCompany" ;tag=reXaHtmyvm2eN To: Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a CSeq: 26041075 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 311 X-FS-Support: update_display,send_info P-Asserted-Identity: "ThisIsMyCompany" v=0 o=FreeSWITCH 1332736038 1332736039 IN IP4 192.168.0.2 s=FreeSWITCH c=IN IP4 192.168.0.2 t=0 0 m=audio 18240 RTP/AVP 8 98 99 9 0 3 101 13 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:125 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change CS_INIT -> CS_ROUTING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State INIT going to sleep 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State Change CS_ROUTING 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:1886 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate Change DOWN -> RINGING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State ROUTING 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:148 sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA ROUTING 2012-03-26 11:31:18.712698 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State ROUTING going to sleep 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State Change CS_CONSUME_MEDIA 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State CONSUME_MEDIA 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State CONSUME_MEDIA going to sleep 2012-03-26 11:31:18.712698 [DEBUG] sofia.c:5526 Channel sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 entering state [calling][0] recv 355 bytes from udp/[192.168.0.3]:5060 at 09:31:18.717462: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K From: "ThisIsMyCompany" ;tag=reXaHtmyvm2eN To: Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a CSeq: 26041075 INVITE User-Agent: Grandstream GXW4104 (HW 2.0, Ch:5) 1.3.4.9 Content-Length: 0 ------------------------------------------------------------------------ recv 367 bytes from udp/[192.168.0.3]:5060 at 09:31:18.717980: ------------------------------------------------------------------------ SIP/2.0 403 Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K From: "ThisIsMyCompany" ;tag=reXaHtmyvm2eN To: ;tag=3ga2B2jmSSe6H Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a CSeq: 26041075 INVITE User-Agent: Grandstream GXW4104 (HW 2.0, Ch:8) 1.3.4.9 Content-Length: 0 ------------------------------------------------------------------------ send 370 bytes to udp/[192.168.0.3]:5060 at 09:31:18.718065: ------------------------------------------------------------------------ ACK sip:gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K Max-Forwards: 69 From: "ThisIsMyCompany" ;tag=reXaHtmyvm2eN To: ;tag=3ga2B2jmSSe6H Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a CSeq: 26041075 ACK Content-Length: 0 ------------------------------------------------------------------------ 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] sofia.c:5526 Channel sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 entering state [terminated][403] 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2848 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate Change RINGING -> HANGUP 2012-03-26 11:31:18.712698 [NOTICE] sofia.c:6293 Hangup sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [CS_CONSUME_MEDIA] [CALL_REJECTED] 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [KILL] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State Change CS_HANGUP 2012-03-26 11:31:18.712698 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 21 [CALL_REJECTED] 2012-03-26 11:31:18.712698 [INFO] mod_dptools.c:2922 Originate Failed. Cause: CALL_REJECTED 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2848 (sofia/internal/1000 at 192.168.0.2) Callstate Change RINGING -> HANGUP 2012-03-26 11:31:18.712698 [NOTICE] mod_dptools.c:3041 Hangup sofia/internal/1000 at 192.168.0.2 [CS_EXECUTE] [CALL_REJECTED] 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/1000 at 192.168.0.2 [KILL] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:2285 sofia/internal/1000 at 192.168.0.2 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1000 at 192.168.0.2) State EXECUTE going to sleep 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_HANGUP 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1000 at 192.168.0.2) State HANGUP 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:463 sofia/internal/1000 at 192.168.0.2 Overriding SIP cause 603 with 403 from the other leg 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1000 at 192.168.0.2 hanging up, cause: CALL_REJECTED 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:534 Responding to INVITE with: 403 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1000 at 192.168.0.2 Standard HANGUP, cause: CALL_REJECTED 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1000 at 192.168.0.2) State HANGUP going to sleep 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1000 at 192.168.0.2) State Change CS_HANGUP -> CS_REPORTING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_REPORTING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1000 at 192.168.0.2) State REPORTING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1000 at 192.168.0.2 Standard REPORTING, cause: CALL_REJECTED 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1000 at 192.168.0.2) State REPORTING going to sleep 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1000 at 192.168.0.2) State Change CS_REPORTING -> CS_DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/1000 at 192.168.0.2 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1380 Session 13 (sofia/internal/1000 at 192.168.0.2) Locked, Waiting on external entities 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1398 Session 13 (sofia/internal/1000 at 192.168.0.2) Ended 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/1000 at 192.168.0.2 [CS_DESTROY] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1000 at 192.168.0.2) Callstate Change HANGUP -> DOWN 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1000 at 192.168.0.2) Running State Change CS_DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1000 at 192.168.0.2) State DESTROY 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:374 sofia/internal/1000 at 192.168.0.2 SOFIA DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1000 at 192.168.0.2 Standard DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1000 at 192.168.0.2) State DESTROY going to sleep send 833 bytes to udp/[192.168.0.200]:5060 at 09:31:18.720584: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport=5060 From: "1000" ;tag=645977894 To: ;tag=Q53HFZ3tZBcvS Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 6 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=21;text="CALL_REJECTED" Content-Length: 0 P-Asserted-Identity: "339XXXXXXXX" ------------------------------------------------------------------------ 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State HANGUP 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:469 Channel sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 hanging up, cause: CALL_REJECTED 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:47 sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard HANGUP, cause: CALL_REJECTED 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State HANGUP going to sleep 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change CS_HANGUP -> CS_REPORTING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State Change CS_REPORTING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State REPORTING 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:79 sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard REPORTING, cause: CALL_REJECTED 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State REPORTING going to sleep recv 618 bytes from udp/[192.168.0.200]:5060 at 09:31:18.721757: ------------------------------------------------------------------------ ACK sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport From: "1000" ;tag=645977894 To: ;tag=Q53HFZ3tZBcvS Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 CSeq: 6 ACK Proxy-Authorization: Digest username="1000", realm="192.168.0.2", nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", uri="sip:339XXXXXXXX at 192.168.0.2", response="7378107bc277149e3b6ef00c1c766a71", algorithm=MD5, cnonce="234abcc436e2667097e7fe6eia53e8dd", qop=auth, nc=00000001 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change CS_REPORTING -> CS_DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1380 Session 14 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Locked, Waiting on external entities 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1398 Session 14 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Ended 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [CS_DESTROY] 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate Change HANGUP -> DOWN 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State Change CS_DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State DESTROY 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:374 sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:86 sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard DESTROY 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State DESTROY going to sleep freeswitch at internal> Il 23/03/2012 18:39, Brian Foster ha scritto: > Accursio, > > Welcome to the FreeSWITCH Community! > > Can you get us a siptrace? > > fs_cli > sofia global siptrace on > > ...then pastebin the failing call to http://pastebin.freeswitch.org > > Thanks! > > -BDF > > On Fri, Mar 23, 2012 at 5:47 AM, a.avona > wrote: > > Hi, all > i have a GrandStream 4104 perfectly working (both incoming and > outgoing > calls work great) with an asterisk pbx. > We are trying to change the asterisk with freeswitch, but i'm > exeperiencing problems in configuring freeswitch for outgoing calls, > incoming calls works well. > this is my configuration and what i obtain in fs_cli consolle > > Can someone tell me where i'm wrong? > > in sip_profile/internal i created a file 00_to_pstn.xml this way > > > > > > > > > > > > > > > in dialplan/default i created a file 00_to_pstn.xml this way > > > > > data="effective_caller_id_number=0321234567"/> > data="effective_caller_id_name=ThisIsMyCompany"/> > > > data="sofia/internal/gxw4104-fxo1/$1 at 192.168.0.3:5060 > "/> > > > > > > > > _________________________________________________________________ > > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:7559 IP 192.168.0.200 > Rejected by acl "domains". Falling back to Digest auth. > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:7559 IP 192.168.0.200 > Rejected by acl "domains". Falling back to Digest auth. > 2012-03-23 10:38:43.937574 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/1000 at 192.168.0.2 > [fb33155c-74cb-11e1-9063-fb6a5f4599fb] > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:5526 Channel > sofia/internal/1000 at 192.168.0.2 entering > state [received][100] > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:5537 Remote SDP: > v=0 > o=- 935122583 0 IN IP4 192.168.0.200 > s=SIPPER for PhonerLite > c=IN IP4 192.168.0.200 > t=0 0 > m=audio 5062 RTP/AVP 8 3 0 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare > [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare > [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare > [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare > [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare > [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:2991 Set Codec > sofia/internal/1000 at 192.168.0.2 > PCMA/8000 20 ms 160 samples 64000 bits > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_codec.c:111 > sofia/internal/1000 at 192.168.0.2 Original > read codec set to PCMA:8 > 2012-03-23 10:38:43.937574 [DEBUG] sofia_glue.c:4995 Set 2833 dtmf > send/recv payload to 101 > 2012-03-23 10:38:43.937574 [DEBUG] sofia.c:5749 > (sofia/internal/1000 at 192.168.0.2 ) State > Change CS_NEW -> CS_INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 > [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2 ) > Running State Change CS_INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1000 at 192.168.0.2 ) State INIT > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:85 > sofia/internal/1000 at 192.168.0.2 SOFIA INIT > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:125 > (sofia/internal/1000 at 192.168.0.2 ) State > Change CS_INIT -> CS_ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 > [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1000 at 192.168.0.2 ) State > INIT going to sleep > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2 ) > Running State Change CS_ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1886 > (sofia/internal/1000 at 192.168.0.2 ) > Callstate Change DOWN -> RINGING > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1000 at 192.168.0.2 ) State > ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:148 > sofia/internal/1000 at 192.168.0.2 SOFIA > ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/1000 at 192.168.0.2 Standard > ROUTING > 2012-03-23 10:38:43.937574 [INFO] mod_dialplan_xml.c:485 > Processing 1000 > <1000>->339XXXXXXX in context default > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->unloop] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->tod_example] > continue=true > Dialplan: sofia/internal/1000 at 192.168.0.2 > Date/Time Match (PASS) > [tod_example] break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action set(open=true) > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->holiday_example] continue=true > Dialplan: sofia/internal/1000 at 192.168.0.2 > Date/TimeMatch (FAIL) > [holiday_example] break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->global-intercept] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [global-intercept] destination_number(339XXXXXXX) =~ /^886$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->group-intercept] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [group-intercept] > destination_number(339XXXXXXX) =~ /^\*8$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->intercept-ext] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [intercept-ext] > destination_number(339XXXXXXX) =~ /^\*\*(\d+)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->redial] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [redial] > destination_number(339XXXXXXX) =~ /^(redial|870)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->global] > continue=true > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [global] > ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [global] > ${sip_has_crypto}() =~ > /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never > Dialplan: sofia/internal/1000 at 192.168.0.2 > Absolute Condition [global] > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->snom-demo-2] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [snom-demo-2] > destination_number(339XXXXXXX) =~ /^9001$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->snom-demo-1] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [snom-demo-1] > destination_number(339XXXXXXX) =~ /^9000$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->eavesdrop] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [eavesdrop] > destination_number(339XXXXXXX) =~ /^88(\d{4})$|^\*0(.*)$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->eavesdrop] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [eavesdrop] > destination_number(339XXXXXXX) =~ /^779$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->call_return] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [call_return] > destination_number(339XXXXXXX) =~ /^\*69$|^869$|^lcr$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->del-group] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [del-group] > destination_number(339XXXXXXX) =~ /^80(\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->add-group] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [add-group] > destination_number(339XXXXXXX) =~ /^81(\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->call-group-simo] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [call-group-simo] > destination_number(339XXXXXXX) =~ /^82(\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->call-group-order] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [call-group-order] destination_number(339XXXXXXX) =~ /^83(\d{2})$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->extension-intercom] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [extension-intercom] destination_number(339XXXXXXX) =~ > /^8(10[01][0-9])$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->Local_Extension] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [Local_Extension] > destination_number(339XXXXXXX) =~ /^(10[01][0-9])$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->Local_Extension_Skinny] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [Local_Extension_Skinny] destination_number(339XXXXXXX) =~ > /^(11[01][0-9])$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->group_dial_sales] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [group_dial_sales] destination_number(339XXXXXXX) =~ /^2000$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->group_dial_support] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [group_dial_support] destination_number(339XXXXXXX) =~ /^2001$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->group_dial_billing] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [group_dial_billing] destination_number(339XXXXXXX) =~ /^2002$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->operator] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [operator] > destination_number(339XXXXXXX) =~ /^(operator|0)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->vmain] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [vmain] > destination_number(339XXXXXXX) =~ /^vmain$|^4000$|^\*98$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->sip_uri] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [sip_uri] > destination_number(339XXXXXXX) =~ /^sip:(.*)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->nb_conferences] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [nb_conferences] > destination_number(339XXXXXXX) =~ /^(30\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->wb_conferences] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [wb_conferences] > destination_number(339XXXXXXX) =~ /^(31\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->uwb_conferences] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [uwb_conferences] > destination_number(339XXXXXXX) =~ /^(32\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->cdquality_conferences] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [cdquality_conferences] destination_number(339XXXXXXX) =~ > /^(33\d{2})$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->freeswitch_public_conf_via_sip] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [freeswitch_public_conf_via_sip] destination_number(339XXXXXXX) =~ > /^9(888|8888|1616|3232)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->mad_boss_intercom] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [mad_boss_intercom] destination_number(339XXXXXXX) =~ /^0911$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->mad_boss_intercom] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [mad_boss_intercom] destination_number(339XXXXXXX) =~ /^0912$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->mad_boss] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [mad_boss] > destination_number(339XXXXXXX) =~ /^0913$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->ivr_demo] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [ivr_demo] > destination_number(339XXXXXXX) =~ /^5000$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->dynamic_conference] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [dynamic_conference] destination_number(339XXXXXXX) =~ /^5001$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->rtp_multicast_page] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [rtp_multicast_page] destination_number(339XXXXXXX) =~ > /^pagegroup$|^7243$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [park] > destination_number(339XXXXXXX) =~ /^5900$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->unpark] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [unpark] > destination_number(339XXXXXXX) =~ /^5901$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->valet_park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [valet_park] > destination_number(339XXXXXXX) =~ /^(6000)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->valet_park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [valet_park] > destination_number(339XXXXXXX) =~ /^(60\d[1-9])$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [park] > destination_number(339XXXXXXX) =~ /park\+(\d+)/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->unpark] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [unpark] > destination_number(339XXXXXXX) =~ /^parking$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->park] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [park] > destination_number(339XXXXXXX) =~ /callpark/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->unpark] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [unpark] > destination_number(339XXXXXXX) =~ /pickup/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->wait] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [wait] > destination_number(339XXXXXXX) =~ /^wait$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->fax_receive] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [fax_receive] > destination_number(339XXXXXXX) =~ /^9178$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->fax_transmit] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [fax_transmit] > destination_number(339XXXXXXX) =~ /^9179$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->ringback_180] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [ringback_180] > destination_number(339XXXXXXX) =~ /^9180$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->ringback_183_uk_ring] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [ringback_183_uk_ring] destination_number(339XXXXXXX) =~ /^9181$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->ringback_183_music_ring] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [ringback_183_music_ring] destination_number(339XXXXXXX) =~ /^9182$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->ringback_post_answer_uk_ring] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [ringback_post_answer_uk_ring] destination_number(339XXXXXXX) =~ > /^9183$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->ringback_post_answer_music] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [ringback_post_answer_music] destination_number(339XXXXXXX) =~ > /^9184$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->ClueCon] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [ClueCon] > destination_number(339XXXXXXX) =~ /^9191$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->show_info] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [show_info] > destination_number(339XXXXXXX) =~ /^9192$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->video_record] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [video_record] > destination_number(339XXXXXXX) =~ /^9193$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->video_playback] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [video_playback] > destination_number(339XXXXXXX) =~ /^9194$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->delay_echo] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [delay_echo] > destination_number(339XXXXXXX) =~ /^9195$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->echo] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [echo] > destination_number(339XXXXXXX) =~ /^9196$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->milliwatt] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [milliwatt] > destination_number(339XXXXXXX) =~ /^9197$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->tone_stream] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [tone_stream] > destination_number(339XXXXXXX) =~ /^9198$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->zrtp_enrollement] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [zrtp_enrollement] destination_number(339XXXXXXX) =~ /^9787$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->hold_music] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [hold_music] > destination_number(339XXXXXXX) =~ /^9664$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->from_pstn] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [from_pstn] > destination_number(339XXXXXXX) =~ /^0000$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->101] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [101] > destination_number(339XXXXXXX) =~ /^101$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->pizza_demo] > continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [pizza_demo] > destination_number(339XXXXXXX) =~ /^(pizza|74992)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->gxw4104-fxo-local] continue=false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (PASS) > [gxw4104-fxo-local] ${toll_allow}(domestic,international,local) =~ > /local/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (PASS) > [gxw4104-fxo-local] destination_number(339XXXXXXX) =~ /^(\d{6,})$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > set(effective_caller_id_number=0321234567) > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > set(effective_caller_id_name=ThisIsMyCompany) > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > set(ignore_early_media=ring_ready) > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action set(ringback=${us-ring}) > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > bridge(sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/1000 at 192.168.0.2 ) State > Change CS_ROUTING -> CS_EXECUTE > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 > [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1000 at 192.168.0.2 ) State > ROUTING going to sleep > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2 ) > Running State Change CS_EXECUTE > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/1000 at 192.168.0.2 ) State > EXECUTE > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:241 > sofia/internal/1000 at 192.168.0.2 SOFIA > EXECUTE > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/1000 at 192.168.0.2 Standard > EXECUTE > EXECUTE sofia/internal/1000 at 192.168.0.2 > set(open=true) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET > [open]=[true] > EXECUTE sofia/internal/1000 at 192.168.0.2 > hash(insert/192.168.0.2-spymap/1000/fb33155c-74cb-11e1-9063-fb6a5f4599fb) > EXECUTE sofia/internal/1000 at 192.168.0.2 > hash(insert/192.168.0.2-last_dial/1000/339XXXXXXX) > EXECUTE sofia/internal/1000 at 192.168.0.2 > hash(insert/192.168.0.2-last_dial/global/fb33155c-74cb-11e1-9063-fb6a5f4599fb) > EXECUTE sofia/internal/1000 at 192.168.0.2 > export(RFC2822_DATE=Fri, 23 Mar > 2012 10:38:43 +0100) > 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1093 EXPORT > (export_vars) [RFC2822_DATE]=[Fri, 23 Mar 2012 10:38:43 +0100] > EXECUTE sofia/internal/1000 at 192.168.0.2 > set(effective_caller_id_number=0321234567) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET > [effective_caller_id_number]=[0321234567] > EXECUTE sofia/internal/1000 at 192.168.0.2 > set(effective_caller_id_name=ThisIsMyCompany) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET > [effective_caller_id_name]=[ThisIsMyCompany] > EXECUTE sofia/internal/1000 at 192.168.0.2 > set(ignore_early_media=ring_ready) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET > [ignore_early_media]=[ring_ready] > EXECUTE sofia/internal/1000 at 192.168.0.2 > set(ringback=%(2000,4000,440,480)) > 2012-03-23 10:38:43.937574 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET > [ringback]=[%(2000,4000,440,480)] > EXECUTE sofia/internal/1000 at 192.168.0.2 > bridge(sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) > 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1047 > sofia/internal/1000 at 192.168.0.2 > EXPORTING[export_vars] > [RFC2822_DATE]=[Fri, 23 Mar 2012 10:38:43 +0100] to event > 2012-03-23 10:38:43.937574 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2012-03-23 10:38:43.937574 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > > [fb34cf5a-74cb-11e1-9068-fb6a5f4599fb] > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:4691 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State Change > CS_NEW -> CS_INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) Running State > Change CS_INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State INIT > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:85 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > SOFIA INIT > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:875 Send > signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] mod_sofia.c:125 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State Change > CS_INIT -> CS_ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [BREAK] > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State INIT > going to sleep > 2012-03-23 10:38:43.937574 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) Running State > Change CS_ROUTING > 2012-03-23 10:38:43.937574 [DEBUG] switch_channel.c:1886 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) Callstate > Change DOWN -> RINGING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State ROUTING > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:148 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > SOFIA ROUTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State Change > CS_ROUTING -> CS_CONSUME_MEDIA > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State ROUTING > going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) Running State > Change CS_CONSUME_MEDIA > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State > CONSUME_MEDIA > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State > CONSUME_MEDIA going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] sofia.c:5526 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > entering state > [calling][0] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:875 Send > signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:875 Send > signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:875 Send > signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] sofia.c:5526 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > entering state > [terminated][403] > 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2848 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) Callstate > Change RINGING -> HANGUP > 2012-03-23 10:38:43.957579 [NOTICE] sofia.c:6293 Hangup > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > > [CS_CONSUME_MEDIA] [CALL_REJECTED] > 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [KILL] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) Running State > Change CS_HANGUP > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State HANGUP > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > hanging up, > cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > Standard HANGUP, > cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State HANGUP > going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State Change > CS_HANGUP -> CS_REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) Running State > Change CS_REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > Standard > REPORTING, cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State > REPORTING going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State Change > CS_REPORTING -> CS_DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_ivr_originate.c:3364 > Originate > Resulted in Error Cause: 21 [CALL_REJECTED] > 2012-03-23 10:38:43.957579 [INFO] mod_dptools.c:2922 Originate Failed. > Cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2848 > (sofia/internal/1000 at 192.168.0.2 ) > Callstate Change RINGING -> HANGUP > 2012-03-23 10:38:43.957579 [NOTICE] mod_dptools.c:3041 Hangup > sofia/internal/1000 at 192.168.0.2 > [CS_EXECUTE] [CALL_REJECTED] > 2012-03-23 10:38:43.957579 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/1000 at 192.168.0.2 [KILL] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 > [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:2285 > sofia/internal/1000 at 192.168.0.2 skip > receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/1000 at 192.168.0.2 ) State > EXECUTE going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2 ) > Running State Change CS_HANGUP > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1000 at 192.168.0.2 ) State > HANGUP > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:463 > sofia/internal/1000 at 192.168.0.2 > Overriding SIP cause 603 with 403 from > the other leg > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/1000 at 192.168.0.2 hanging > up, cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:534 Responding to > INVITE > with: 403 > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/1000 at 192.168.0.2 Standard > HANGUP, cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1000 at 192.168.0.2 ) State > HANGUP going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/1000 at 192.168.0.2 ) State > Change CS_HANGUP -> CS_REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 > [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2 ) > Running State Change CS_REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1000 at 192.168.0.2 ) State > REPORTING > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/1000 at 192.168.0.2 Standard > REPORTING, cause: CALL_REJECTED > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1000 at 192.168.0.2 ) State > REPORTING going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/1000 at 192.168.0.2 ) State > Change CS_REPORTING -> CS_DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 > [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1380 > Session 13 > (sofia/internal/1000 at 192.168.0.2 ) > Locked, Waiting on external entities > 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1398 Session > 13 (sofia/internal/1000 at 192.168.0.2 ) Ended > 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/1000 at 192.168.0.2 > [CS_DESTROY] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/1000 at 192.168.0.2 ) > Callstate Change HANGUP -> DOWN > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/1000 at 192.168.0.2 ) > Running State Change CS_DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1000 at 192.168.0.2 ) State > DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:374 > sofia/internal/1000 at 192.168.0.2 SOFIA > DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/1000 at 192.168.0.2 Standard > DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1000 at 192.168.0.2 ) State > DESTROY going to sleep > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [BREAK] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_session.c:1380 > Session 14 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) Locked, > Waiting on external entities > 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1398 Session > 14 (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) Ended > 2012-03-23 10:38:43.957579 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > [CS_DESTROY] > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) Callstate > Change HANGUP -> DOWN > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) Running State > Change CS_DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] mod_sofia.c:374 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > SOFIA DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > Standard DESTROY > 2012-03-23 10:38:43.957579 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/gxw4104-fxo1/339XXXXXXX at 192.168.0.3:5060 > ) State DESTROY > going to sleep > freeswitch at internal> > > Thank's in advance > Regards > Accursio Avona > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for > those listed in the "To:", "CC:", and/or "BCC:" fields of the message > header. If you are not the intended recipient you are notified that > disclosing, copying, distributing or taking any action in reliance on > the contents of this information is strictly prohibited. E-mail > transmission cannot be guaranteed to be secure or error-free as > information could be intercepted, corrupted, lost, destroyed, arrive > late or incomplete, or contain viruses. The sender therefore does not > accept liability for any errors or omissions in the contents of this > message, which arise as a result of e-mail transmission. If > verification is required please request a hard-copy version. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/8d8fa7d7/attachment-0001.html From wojciech.kochanowski at nnv.pl Mon Mar 26 15:13:35 2012 From: wojciech.kochanowski at nnv.pl (Wojtek Kochanowski) Date: Mon, 26 Mar 2012 13:13:35 +0200 Subject: [Freeswitch-users] [mod cdr] Doesn't log call details Message-ID: Hello, I'm using mod_json_cdr to get logs from calls, but in certainly situations i don't get the second log. Usually I'm getting 2 logs: inbound and outbound. I'm always getting inbound log, but it's incomplete - for example I don't have information from which channel user have called. This info is in outbound log, but in about 50% calls I don't get it. Same situation with mod_xml. My calling scheme is rather untypical and it looks like: web browser -> rtmp -> freeswitch (freetdm) -> gsm When I'm using just FS to call (eg using Jitsi or other desktop client), everything is ok. FreeSWITCH Version 1.0.head (git-d2c9fb5 2012-02-06 14-12-22 -0600) root at fs# uname -a Linux fs 2.6.32-5-amd64 #1 SMP Mon Jan 16 16:22:28 UTC 2012 x86_64 GNU/Linux root at fs# cat /etc/issue Debian GNU/Linux 6.0 \n \l dialplan: http://pastebin.freeswitch.org/18733 json_cdr.conf: http://pastebin.freeswitch.org/18734 inbound log: http://pastebin.freeswitch.org/18735 example good outbound log: pastebin.freeswitch.org/18736 in "chan_name": "FreeTDM\/1:2\/513710737#" I can see which span and channel has been used. I don't know, is it just my configuration or bug? What else info do you need? Maybe it's my bad in dialplan configuration? Any ideas? Greet, Wojciech Kochanowski Junior System Administrator wojciech.kochanowski at nnv.pl NNV Sp. z o.o. ul. Wyszy?skiego 1 10-457 Olsztyn tel. +48 89 533 70 33 fax +48 89 533 03 78 www.nnv.pl S?d Rejonowy dla Miasta Olsztyna KRS 0000310781 NIP 739-369-92-87 REGON 280319486 Kapita? Zak?adowy 1 152 000 z?, wp?acony w ca?o?ci www.firmy.net W trosce o Tw?j biznes www.nieruchomosci-online.pl Najwi?cej aktualnych og?osze? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/80580536/attachment.html From gcd at i.ph Mon Mar 26 15:14:09 2012 From: gcd at i.ph (Nandy Dagondon) Date: Mon, 26 Mar 2012 19:14:09 +0800 Subject: [Freeswitch-users] Grandstream 4104 In-Reply-To: <4F703E66.5070607@elios.net> References: <4F6C46B0.7080805@elios.net> <4F703E66.5070607@elios.net> Message-ID: just a hint - did u try to use different port numbers when dialing out? if i remember right, the gateway assigns different port to every physical port i.e. 5060 for port1, 5062 for port2 and so on. so dialout on port1: on port2: i hope you'll make a progress. On Mon, Mar 26, 2012 at 6:01 PM, a.avona wrote: > Hi, thank's for your answer > i did as you said and if i digit > originate sofia/internal/XXXX at 192.168.0.3 9178 > outgoing calls work well. > > if i try to originate call from a client it doesn't work so i think the > problem is in the default account configuration > here is the siptraces > > > Thank's for any suggestion > reagards > Accursio Avona > > recv 865 bytes from udp/[192.168.0.200]:5060 at 09:31:18.696255: > > ------------------------------------------------------------------------ > > INVITE sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.0.200:5060 > ;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport > > From: "1000" ;tag=645977894 > > To: > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 5 INVITE > > Contact: > > Content-Type: application/sdp > > Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE > > Max-Forwards: 70 > > Supported: 100rel, replaces, from-change > > User-Agent: SIPPER for PhonerLite > > P-Preferred-Identity: > > Content-Length: 260 > > > v=0 > > o=- 162748822 0 IN IP4 192.168.0.200 > > s=SIPPER for PhonerLite > > c=IN IP4 192.168.0.200 > > t=0 0 > > m=audio 5062 RTP/AVP 8 3 0 101 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=sendrecv > > ------------------------------------------------------------------------ > > send 379 bytes to udp/[192.168.0.200]:5060 at 09:31:18.696482: > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 192.168.0.200:5060 > ;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport=5060 > > From: "1000" ;tag=645977894 > > To: > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 5 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 > -0600 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:7559 IP 192.168.0.200 Rejected > by acl "domains". Falling back to Digest auth. > > send 865 bytes to udp/[192.168.0.200]:5060 at 09:31:18.697335: > > ------------------------------------------------------------------------ > > SIP/2.0 407 Proxy Authentication Required > > Via: SIP/2.0/UDP 192.168.0.200:5060 > ;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport=5060 > > From: "1000" ;tag=645977894 > > To: > ;tag=pvaSD4jQ22N9D > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 5 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 > -0600 > > Accept: application/sdp > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > > Proxy-Authenticate: Digest realm="192.168.0.2", > nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", algorithm=MD5, qop="auth" > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > recv 346 bytes from udp/[192.168.0.200]:5060 at 09:31:18.698468: > > ------------------------------------------------------------------------ > > ACK sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.0.200:5060 > ;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport > > From: "1000" ;tag=645977894 > > To: > ;tag=pvaSD4jQ22N9D > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 5 ACK > > Max-Forwards: 70 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > recv 1137 bytes from udp/[192.168.0.200]:5060 at 09:31:18.699228: > > ------------------------------------------------------------------------ > > INVITE sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.0.200:5060 > ;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport > > From: "1000" ;tag=645977894 > > To: > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 6 INVITE > > Contact: > > Proxy-Authorization: Digest username="1000", realm="192.168.0.2", > nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", uri= > "sip:339XXXXXXXX at 192.168.0.2" , > response="7378107bc277149e3b6ef00c1c766a71", algorithm=MD5, > cnonce="234abcc436e2667097e7fe6eia53e8dd", qop=auth, nc=00000001 > > Content-Type: application/sdp > > Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE > > Max-Forwards: 70 > > Supported: 100rel, replaces, from-change > > User-Agent: SIPPER for PhonerLite > > P-Preferred-Identity: > > Content-Length: 260 > > > v=0 > > o=- 162748822 0 IN IP4 192.168.0.200 > > s=SIPPER for PhonerLite > > c=IN IP4 192.168.0.200 > > t=0 0 > > m=audio 5062 RTP/AVP 8 3 0 101 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=sendrecv > > ------------------------------------------------------------------------ > > send 379 bytes to udp/[192.168.0.200]:5060 at 09:31:18.699380: > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 192.168.0.200:5060 > ;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport=5060 > > From: "1000" ;tag=645977894 > > To: > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 6 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 > -0600 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:7559 IP 192.168.0.200 Rejected > by acl "domains". Falling back to Digest auth. > > 2012-03-26 11:31:18.692558 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/1000 at 192.168.0.2 [710da826-7726-11e1-bf72-e50b8b76e101] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5526 Channel > sofia/internal/1000 at 192.168.0.2 entering state [received][100] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5537 Remote SDP: > > v=0 > > o=- 162748822 0 IN IP4 192.168.0.200 > > s=SIPPER for PhonerLite > > c=IN IP4 192.168.0.200 > > t=0 0 > > m=audio 5062 RTP/AVP 8 3 0 101 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:2991 Set Codec > sofia/internal/1000 at 192.168.0.2 PCMA/8000 20 ms 160 samples 64000 bits > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_codec.c:111 > sofia/internal/1000 at 192.168.0.2 Original read codec set to PCMA:8 > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4995 Set 2833 dtmf > send/recv payload to 101 > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5749 ( > sofia/internal/1000 at 192.168.0.2) State Change CS_NEW -> CS_INIT > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/1000 at 192.168.0.2 [BREAK] > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 ( > sofia/internal/1000 at 192.168.0.2) Running State Change CS_INIT > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:401 ( > sofia/internal/1000 at 192.168.0.2) State INIT > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:85 > sofia/internal/1000 at 192.168.0.2 SOFIA INIT > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:125 ( > sofia/internal/1000 at 192.168.0.2) State Change CS_INIT -> CS_ROUTING > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/1000 at 192.168.0.2 [BREAK] > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:401 ( > sofia/internal/1000 at 192.168.0.2) State INIT going to sleep > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 ( > sofia/internal/1000 at 192.168.0.2) Running State Change CS_ROUTING > > 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1886 ( > sofia/internal/1000 at 192.168.0.2) Callstate Change DOWN -> RINGING > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:410 ( > sofia/internal/1000 at 192.168.0.2) State ROUTING > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:148 > sofia/internal/1000 at 192.168.0.2 SOFIA ROUTING > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/1000 at 192.168.0.2 Standard ROUTING > > 2012-03-26 11:31:18.692558 [INFO] mod_dialplan_xml.c:485 Processing 1000 > <1000>->339XXXXXXXX in context default > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unloop] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->tod_example] > continue=true > > Dialplan: sofia/internal/1000 at 192.168.0.2 Date/Time Match (PASS) > [tod_example] break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(open=true) > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->holiday_example] continue=true > > Dialplan: sofia/internal/1000 at 192.168.0.2 Date/TimeMatch (FAIL) > [holiday_example] break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->global-intercept] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global-intercept] > destination_number(339XXXXXXXX) =~ /^886$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group-intercept] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group-intercept] > destination_number(339XXXXXXXX) =~ /^\*8$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->intercept-ext] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [intercept-ext] > destination_number(339XXXXXXXX) =~ /^\*\*(\d+)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->redial] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [redial] > destination_number(339XXXXXXXX) =~ /^(redial|870)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->global] > continue=true > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] > ${call_debug}(false) =~ /^true$/ break=never > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] > ${sip_has_crypto}() =~ > /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never > > Dialplan: sofia/internal/1000 at 192.168.0.2 Absolute Condition [global] > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->snom-demo-2] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-2] > destination_number(339XXXXXXXX) =~ /^9001$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->snom-demo-1] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-1] > destination_number(339XXXXXXXX) =~ /^9000$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] > destination_number(339XXXXXXXX) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] > destination_number(339XXXXXXXX) =~ /^779$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->call_return] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call_return] > destination_number(339XXXXXXXX) =~ /^\*69$|^869$|^lcr$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->del-group] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [del-group] > destination_number(339XXXXXXXX) =~ /^80(\d{2})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->add-group] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [add-group] > destination_number(339XXXXXXXX) =~ /^81(\d{2})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->call-group-simo] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call-group-simo] > destination_number(339XXXXXXXX) =~ /^82(\d{2})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->call-group-order] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call-group-order] > destination_number(339XXXXXXXX) =~ /^83(\d{2})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->extension-intercom] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [extension-intercom] destination_number(339XXXXXXXX) =~ /^8(10[01][0-9])$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->Local_Extension] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [Local_Extension] > destination_number(339XXXXXXXX) =~ /^(10[01][0-9])$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->Local_Extension_Skinny] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [Local_Extension_Skinny] destination_number(339XXXXXXXX) =~ > /^(11[01][0-9])$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group_dial_sales] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group_dial_sales] > destination_number(339XXXXXXXX) =~ /^2000$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group_dial_support] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [group_dial_support] destination_number(339XXXXXXXX) =~ /^2001$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group_dial_billing] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [group_dial_billing] destination_number(339XXXXXXXX) =~ /^2002$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->operator] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [operator] > destination_number(339XXXXXXXX) =~ /^(operator|0)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->vmain] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [vmain] > destination_number(339XXXXXXXX) =~ /^vmain$|^4000$|^\*98$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->sip_uri] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [sip_uri] > destination_number(339XXXXXXXX) =~ /^sip:(.*)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->nb_conferences] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [nb_conferences] > destination_number(339XXXXXXXX) =~ /^(30\d{2})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->wb_conferences] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wb_conferences] > destination_number(339XXXXXXXX) =~ /^(31\d{2})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->uwb_conferences] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [uwb_conferences] > destination_number(339XXXXXXXX) =~ /^(32\d{2})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->cdquality_conferences] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [cdquality_conferences] destination_number(339XXXXXXXX) =~ /^(33\d{2})$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->freeswitch_public_conf_via_sip] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [freeswitch_public_conf_via_sip] destination_number(339XXXXXXXX) =~ > /^9(888|8888|1616|3232)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->mad_boss_intercom] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [mad_boss_intercom] destination_number(339XXXXXXXX) =~ /^0911$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->mad_boss_intercom] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [mad_boss_intercom] destination_number(339XXXXXXXX) =~ /^0912$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->mad_boss] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [mad_boss] > destination_number(339XXXXXXXX) =~ /^0913$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ivr_demo] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ivr_demo] > destination_number(339XXXXXXXX) =~ /^5000$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->dynamic_conference] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [dynamic_conference] destination_number(339XXXXXXXX) =~ /^5001$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->rtp_multicast_page] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [rtp_multicast_page] destination_number(339XXXXXXXX) =~ > /^pagegroup$|^7243$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] > destination_number(339XXXXXXXX) =~ /^5900$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] > destination_number(339XXXXXXXX) =~ /^5901$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->valet_park] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] > destination_number(339XXXXXXXX) =~ /^(6000)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->valet_park] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] > destination_number(339XXXXXXXX) =~ /^(60\d[1-9])$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] > destination_number(339XXXXXXXX) =~ /park\+(\d+)/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] > destination_number(339XXXXXXXX) =~ /^parking$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] > destination_number(339XXXXXXXX) =~ /callpark/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] > destination_number(339XXXXXXXX) =~ /pickup/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->wait] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wait] > destination_number(339XXXXXXXX) =~ /^wait$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->fax_receive] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_receive] > destination_number(339XXXXXXXX) =~ /^9178$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->fax_transmit] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_transmit] > destination_number(339XXXXXXXX) =~ /^9179$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ringback_180] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_180] > destination_number(339XXXXXXXX) =~ /^9180$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_183_uk_ring] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_183_uk_ring] destination_number(339XXXXXXXX) =~ /^9181$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_183_music_ring] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_183_music_ring] destination_number(339XXXXXXXX) =~ /^9182$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_post_answer_uk_ring] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_post_answer_uk_ring] destination_number(339XXXXXXXX) =~ /^9183$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_post_answer_music] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_post_answer_music] destination_number(339XXXXXXXX) =~ /^9184$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ClueCon] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ClueCon] > destination_number(339XXXXXXXX) =~ /^9191$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->show_info] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [show_info] > destination_number(339XXXXXXXX) =~ /^9192$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->video_record] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_record] > destination_number(339XXXXXXXX) =~ /^9193$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->video_playback] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_playback] > destination_number(339XXXXXXXX) =~ /^9194$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->delay_echo] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [delay_echo] > destination_number(339XXXXXXXX) =~ /^9195$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->echo] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [echo] > destination_number(339XXXXXXXX) =~ /^9196$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->milliwatt] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [milliwatt] > destination_number(339XXXXXXXX) =~ /^9197$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->tone_stream] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [tone_stream] > destination_number(339XXXXXXXX) =~ /^9198$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->zrtp_enrollement] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [zrtp_enrollement] > destination_number(339XXXXXXXX) =~ /^9787$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->hold_music] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [hold_music] > destination_number(339XXXXXXXX) =~ /^9664$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->from_pstn] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [from_pstn] > destination_number(339XXXXXXXX) =~ /^0000$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->101] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [101] > destination_number(339XXXXXXXX) =~ /^101$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->pizza_demo] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [pizza_demo] > destination_number(339XXXXXXXX) =~ /^(pizza|74992)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->gxw4104-fxo-local] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) > [gxw4104-fxo-local] ${toll_allow}(domestic,international,local) =~ /local/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) > [gxw4104-fxo-local] destination_number(339XXXXXXXX) =~ /^(\d{6,})$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > set(effective_caller_id_number=0321234567) > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > set(effective_caller_id_name=ThisIsMyCompany) > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > set(ignore_early_media=ring_ready) > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(ringback=${us-ring}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action bridge( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:154 ( > sofia/internal/1000 at 192.168.0.2) State Change CS_ROUTING -> CS_EXECUTE > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/1000 at 192.168.0.2 [BREAK] > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:410 ( > sofia/internal/1000 at 192.168.0.2) State ROUTING going to sleep > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 ( > sofia/internal/1000 at 192.168.0.2) Running State Change CS_EXECUTE > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:417 ( > sofia/internal/1000 at 192.168.0.2) State EXECUTE > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:241 > sofia/internal/1000 at 192.168.0.2 SOFIA EXECUTE > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/1000 at 192.168.0.2 Standard EXECUTE > > EXECUTE sofia/internal/1000 at 192.168.0.2 set(open=true) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET [open]=[true] > > EXECUTE sofia/internal/1000 at 192.168.0.2hash(insert/192.168.0.2-spymap/1000/710da826-7726-11e1-bf72-e50b8b76e101) > > EXECUTE sofia/internal/1000 at 192.168.0.2hash(insert/192.168.0.2-last_dial/1000/339XXXXXXXX) > > EXECUTE sofia/internal/1000 at 192.168.0.2hash(insert/192.168.0.2-last_dial/global/710da826-7726-11e1-bf72-e50b8b76e101) > > EXECUTE sofia/internal/1000 at 192.168.0.2 export(RFC2822_DATE=Mon, 26 Mar > 2012 11:31:18 +0200) > > 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1093 EXPORT > (export_vars) [RFC2822_DATE]=[Mon, 26 Mar 2012 11:31:18 +0200] > > EXECUTE sofia/internal/1000 at 192.168.0.2 set(effective_caller_id_number= > 0321234567) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET [effective_caller_id_number]= > [0321234567] > > EXECUTE sofia/internal/1000 at 192.168.0.2set(effective_caller_id_name=ThisIsMyCompany) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET > [effective_caller_id_name]=[ThisIsMyCompany] > > EXECUTE sofia/internal/1000 at 192.168.0.2 set(ignore_early_media=ring_ready) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET [ignore_early_media]=[ring_ready] > > EXECUTE sofia/internal/1000 at 192.168.0.2 set(ringback=%(2000,4000,440,480)) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET [ringback]=[%(2000,4000,440,480)] > > EXECUTE sofia/internal/1000 at 192.168.0.2 bridge( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) > > 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1047 > sofia/internal/1000 at 192.168.0.2 EXPORTING[export_vars] > [RFC2822_DATE]=[Mon, 26 Mar 2012 11:31:18 +0200] to event > > 2012-03-26 11:31:18.692558 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > > 2012-03-26 11:31:18.692558 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060[710f6738-7726-11e1-bf77-e50b8b76e101] > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:4691 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change > CS_NEW -> CS_INIT > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State > Change CS_INIT > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:401 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State INIT > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:85 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA INIT > > send 1279 bytes to udp/[192.168.0.3]:5060 at 09:31:18.713245: > > ------------------------------------------------------------------------ > > INVITE sip:gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K > > Max-Forwards: 69 > > From: "ThisIsMyCompany" > ;tag=reXaHtmyvm2eN > > To: > > Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a > > CSeq: 26041075 INVITE > > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 > -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > > Privacy: none > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 311 > > X-FS-Support: update_display,send_info > > P-Asserted-Identity: "ThisIsMyCompany" > > > v=0 > > o=FreeSWITCH 1332736038 1332736039 IN IP4 192.168.0.2 > > s=FreeSWITCH > > c=IN IP4 192.168.0.2 > > t=0 0 > > m=audio 18240 RTP/AVP 8 98 99 9 0 3 101 13 > > a=rtpmap:98 G7221/32000 > > a=fmtp:98 bitrate=48000 > > a=rtpmap:99 G7221/16000 > > a=fmtp:99 bitrate=32000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=ptime:20 > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:125 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change > CS_INIT -> CS_ROUTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:401 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State INIT > going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State > Change CS_ROUTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:1886 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate > Change DOWN -> RINGING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:410 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State ROUTING > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:148 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA ROUTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_ivr_originate.c:66 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change > CS_ROUTING -> CS_CONSUME_MEDIA > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:410 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State ROUTING > going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State > Change CS_CONSUME_MEDIA > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:429 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State > CONSUME_MEDIA > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:429 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State > CONSUME_MEDIA going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] sofia.c:5526 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 entering state > [calling][0] > > recv 355 bytes from udp/[192.168.0.3]:5060 at 09:31:18.717462: > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K > > From: "ThisIsMyCompany" > ;tag=reXaHtmyvm2eN > > To: > > Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a > > CSeq: 26041075 INVITE > > User-Agent: Grandstream GXW4104 (HW 2.0, Ch:5) 1.3.4.9 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > recv 367 bytes from udp/[192.168.0.3]:5060 at 09:31:18.717980: > > ------------------------------------------------------------------------ > > SIP/2.0 403 > > Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K > > From: "ThisIsMyCompany" > ;tag=reXaHtmyvm2eN > > To: > ;tag=3ga2B2jmSSe6H > > Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a > > CSeq: 26041075 INVITE > > User-Agent: Grandstream GXW4104 (HW 2.0, Ch:8) 1.3.4.9 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > send 370 bytes to udp/[192.168.0.3]:5060 at 09:31:18.718065: > > ------------------------------------------------------------------------ > > ACK sip:gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K > > Max-Forwards: 69 > > From: "ThisIsMyCompany" > ;tag=reXaHtmyvm2eN > > To: > ;tag=3ga2B2jmSSe6H > > Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a > > CSeq: 26041075 ACK > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] sofia.c:5526 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 entering state > [terminated][403] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2848 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate > Change RINGING -> HANGUP > > 2012-03-26 11:31:18.712698 [NOTICE] sofia.c:6293 Hangup > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060[CS_CONSUME_MEDIA] [CALL_REJECTED] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [KILL] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State > Change CS_HANGUP > > 2012-03-26 11:31:18.712698 [DEBUG] switch_ivr_originate.c:3364 Originate > Resulted in Error Cause: 21 [CALL_REJECTED] > > 2012-03-26 11:31:18.712698 [INFO] mod_dptools.c:2922 Originate Failed. > Cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2848 ( > sofia/internal/1000 at 192.168.0.2) Callstate Change RINGING -> HANGUP > > 2012-03-26 11:31:18.712698 [NOTICE] mod_dptools.c:3041 Hangup > sofia/internal/1000 at 192.168.0.2 [CS_EXECUTE] [CALL_REJECTED] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/1000 at 192.168.0.2 [KILL] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/1000 at 192.168.0.2 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:2285 > sofia/internal/1000 at 192.168.0.2 skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:417 ( > sofia/internal/1000 at 192.168.0.2) State EXECUTE going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 ( > sofia/internal/1000 at 192.168.0.2) Running State Change CS_HANGUP > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 ( > sofia/internal/1000 at 192.168.0.2) State HANGUP > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:463 > sofia/internal/1000 at 192.168.0.2 Overriding SIP cause 603 with 403 from > the other leg > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/1000 at 192.168.0.2 hanging up, cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:534 Responding to INVITE > with: 403 > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/1000 at 192.168.0.2 Standard HANGUP, cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 ( > sofia/internal/1000 at 192.168.0.2) State HANGUP going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:393 ( > sofia/internal/1000 at 192.168.0.2) State Change CS_HANGUP -> CS_REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/1000 at 192.168.0.2 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 ( > sofia/internal/1000 at 192.168.0.2) Running State Change CS_REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 ( > sofia/internal/1000 at 192.168.0.2) State REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/1000 at 192.168.0.2 Standard REPORTING, cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 ( > sofia/internal/1000 at 192.168.0.2) State REPORTING going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:387 ( > sofia/internal/1000 at 192.168.0.2) State Change CS_REPORTING -> CS_DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/1000 at 192.168.0.2 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1380 Session 13 ( > sofia/internal/1000 at 192.168.0.2) Locked, Waiting on external entities > > 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1398 Session 13 ( > sofia/internal/1000 at 192.168.0.2) Ended > > 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/1000 at 192.168.0.2 [CS_DESTROY] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:491 ( > sofia/internal/1000 at 192.168.0.2) Callstate Change HANGUP -> DOWN > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:494 ( > sofia/internal/1000 at 192.168.0.2) Running State Change CS_DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 ( > sofia/internal/1000 at 192.168.0.2) State DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:374 > sofia/internal/1000 at 192.168.0.2 SOFIA DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/1000 at 192.168.0.2 Standard DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 ( > sofia/internal/1000 at 192.168.0.2) State DESTROY going to sleep > > send 833 bytes to udp/[192.168.0.200]:5060 at 09:31:18.720584: > > ------------------------------------------------------------------------ > > SIP/2.0 403 Forbidden > > Via: SIP/2.0/UDP 192.168.0.200:5060 > ;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport=5060 > > From: "1000" ;tag=645977894 > > To: > ;tag=Q53HFZ3tZBcvS > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 6 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 > -0600 > > Accept: application/sdp > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > > Reason: Q.850;cause=21;text="CALL_REJECTED" > > Content-Length: 0 > > P-Asserted-Identity: "339XXXXXXXX" > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State HANGUP > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 hanging up, > cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard HANGUP, > cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State HANGUP > going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:393 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change > CS_HANGUP -> CS_REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State > Change CS_REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard > REPORTING, cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State REPORTING > going to sleep > > recv 618 bytes from udp/[192.168.0.200]:5060 at 09:31:18.721757: > > ------------------------------------------------------------------------ > > ACK sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.0.200:5060 > ;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport > > From: "1000" ;tag=645977894 > > To: > ;tag=Q53HFZ3tZBcvS > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 6 ACK > > Proxy-Authorization: Digest username="1000", realm="192.168.0.2", > nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", uri= > "sip:339XXXXXXXX at 192.168.0.2" , > response="7378107bc277149e3b6ef00c1c766a71", algorithm=MD5, > cnonce="234abcc436e2667097e7fe6eia53e8dd", qop=auth, nc=00000001 > > Max-Forwards: 70 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:387 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change > CS_REPORTING -> CS_DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1380 Session 14 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Locked, Waiting > on external entities > > 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1398 Session 14 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Ended > > 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060[CS_DESTROY] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:491 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate > Change HANGUP -> DOWN > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:494 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State > Change CS_DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:374 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 ( > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State DESTROY > going to sleep > > freeswitch at internal> > > > > Il 25/03/2012 06:39, Anton Kvashenkin ha scritto: > > originate sofia/internal/XXXX at 192.168.0.3 9178 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/3335794d/attachment-0001.html From a.avona at elios.net Mon Mar 26 16:27:24 2012 From: a.avona at elios.net (a.avona) Date: Mon, 26 Mar 2012 14:27:24 +0200 Subject: [Freeswitch-users] Grandstream 4104 In-Reply-To: References: <4F6C46B0.7080805@elios.net> <4F703E66.5070607@elios.net> Message-ID: <4F7060AC.9090903@elios.net> Hi Nandy, thank's for your suggestion but it didn't help me. I tryie all ports from 5060 to 5068 but it didn't work. if I try originate sofia/internal/XXXX at 192.168.0.3 :5062 9178 no calls are originated. any suggestion is very welcome. Il 26/03/2012 13:14, Nandy Dagondon ha scritto: > just a hint - did u try to use different port numbers when dialing > out? if i remember right, the gateway assigns different port to every > physical port i.e. 5060 for port1, 5062 for port2 and so on. > > so dialout on port1: data="sofia/internal/gxw4104-fxo1/$1 at 192.168.0.3:5060 > "/> > on port2: data="sofia/internal/gxw4104-fxo2/$1 at 192.168.0.3:5062 > "/> > > i hope you'll make a progress. > > On Mon, Mar 26, 2012 at 6:01 PM, a.avona > wrote: > > Hi, thank's for your answer > i did as you said and if i digit > originate sofia/internal/XXXX at 192.168.0.3 > 9178 > outgoing calls work well. > > if i try to originate call from a client it doesn't work so i > think the problem is in the default account configuration > here is the siptraces > > > Thank's for any suggestion > reagards > Accursio Avona > > recv 865 bytes from udp/[192.168.0.200]:5060 at 09:31:18.696255: > > ------------------------------------------------------------------------ > > INVITE sip:339XXXXXXXX at 192.168.0.2 > SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport > > From: "1000" > ;tag=645977894 > > To: > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > > CSeq: 5 INVITE > > Contact: > > > Content-Type: application/sdp > > Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, > UPDATE > > Max-Forwards: 70 > > Supported: 100rel, replaces, from-change > > User-Agent: SIPPER for PhonerLite > > P-Preferred-Identity: > > > Content-Length: 260 > > > v=0 > > o=- 162748822 0 IN IP4 192.168.0.200 > > s=SIPPER for PhonerLite > > c=IN IP4 192.168.0.200 > > t=0 0 > > m=audio 5062 RTP/AVP 8 3 0 101 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=sendrecv > > ------------------------------------------------------------------------ > > send 379 bytes to udp/[192.168.0.200]:5060 at 09:31:18.696482: > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport=5060 > > From: "1000" > ;tag=645977894 > > To: > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > > CSeq: 5 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 > 15-58-48 -0600 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:7559 IP 192.168.0.200 > Rejected by acl "domains". Falling back to Digest auth. > > send 865 bytes to udp/[192.168.0.200]:5060 at 09:31:18.697335: > > ------------------------------------------------------------------------ > > SIP/2.0 407 Proxy Authentication Required > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport=5060 > > From: "1000" > ;tag=645977894 > > To: > ;tag=pvaSD4jQ22N9D > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > > CSeq: 5 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 > 15-58-48 -0600 > > Accept: application/sdp > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, > sla, include-session-description, presence.winfo, message-summary, > refer > > Proxy-Authenticate: Digest realm="192.168.0.2", > nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", algorithm=MD5, > qop="auth" > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > recv 346 bytes from udp/[192.168.0.200]:5060 at 09:31:18.698468: > > ------------------------------------------------------------------------ > > ACK sip:339XXXXXXXX at 192.168.0.2 > SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport > > From: "1000" > ;tag=645977894 > > To: > ;tag=pvaSD4jQ22N9D > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > > CSeq: 5 ACK > > Max-Forwards: 70 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > recv 1137 bytes from udp/[192.168.0.200]:5060 at 09:31:18.699228: > > ------------------------------------------------------------------------ > > INVITE sip:339XXXXXXXX at 192.168.0.2 > SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport > > From: "1000" > ;tag=645977894 > > To: > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > > CSeq: 6 INVITE > > Contact: > > > Proxy-Authorization: Digest username="1000", realm="192.168.0.2", > nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", > uri="sip:339XXXXXXXX at 192.168.0.2" > , > response="7378107bc277149e3b6ef00c1c766a71", algorithm=MD5, > cnonce="234abcc436e2667097e7fe6eia53e8dd", qop=auth, nc=00000001 > > Content-Type: application/sdp > > Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, > UPDATE > > Max-Forwards: 70 > > Supported: 100rel, replaces, from-change > > User-Agent: SIPPER for PhonerLite > > P-Preferred-Identity: > > > Content-Length: 260 > > > v=0 > > o=- 162748822 0 IN IP4 192.168.0.200 > > s=SIPPER for PhonerLite > > c=IN IP4 192.168.0.200 > > t=0 0 > > m=audio 5062 RTP/AVP 8 3 0 101 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=sendrecv > > ------------------------------------------------------------------------ > > send 379 bytes to udp/[192.168.0.200]:5060 at 09:31:18.699380: > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport=5060 > > From: "1000" > ;tag=645977894 > > To: > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > > CSeq: 6 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 > 15-58-48 -0600 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:7559 IP 192.168.0.200 > Rejected by acl "domains". Falling back to Digest auth. > > 2012-03-26 11:31:18.692558 [NOTICE] switch_channel.c:926 New > Channel sofia/internal/1000 at 192.168.0.2 > > [710da826-7726-11e1-bf72-e50b8b76e101] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5526 Channel > sofia/internal/1000 at 192.168.0.2 > entering state > [received][100] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5537 Remote SDP: > > v=0 > > o=- 162748822 0 IN IP4 192.168.0.200 > > s=SIPPER for PhonerLite > > c=IN IP4 192.168.0.200 > > t=0 0 > > m=audio 5062 RTP/AVP 8 3 0 101 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:2991 Set Codec > sofia/internal/1000 at 192.168.0.2 > PCMA/8000 20 ms 160 > samples 64000 bits > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_codec.c:111 > sofia/internal/1000 at 192.168.0.2 > Original read codec set > to PCMA:8 > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4995 Set 2833 dtmf > send/recv payload to 101 > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5749 > (sofia/internal/1000 at 192.168.0.2 > ) State Change CS_NEW -> > CS_INIT > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 > [BREAK] > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2 > ) Running State Change CS_INIT > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1000 at 192.168.0.2 > ) State INIT > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:85 > sofia/internal/1000 at 192.168.0.2 > SOFIA INIT > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:125 > (sofia/internal/1000 at 192.168.0.2 > ) State Change CS_INIT -> > CS_ROUTING > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 > [BREAK] > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1000 at 192.168.0.2 > ) State INIT going to sleep > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2 > ) Running State Change > CS_ROUTING > > 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1886 > (sofia/internal/1000 at 192.168.0.2 > ) Callstate Change DOWN -> > RINGING > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1000 at 192.168.0.2 > ) State ROUTING > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:148 > sofia/internal/1000 at 192.168.0.2 > SOFIA ROUTING > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/1000 at 192.168.0.2 > Standard ROUTING > > 2012-03-26 11:31:18.692558 [INFO] mod_dialplan_xml.c:485 > Processing 1000 <1000>->339XXXXXXXX in context default > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->unloop] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->tod_example] continue=true > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Date/Time Match (PASS) > [tod_example] break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action set(open=true) > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->holiday_example] continue=true > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Date/TimeMatch (FAIL) > [holiday_example] break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->global-intercept] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [global-intercept] destination_number(339XXXXXXXX) =~ /^886$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->group-intercept] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [group-intercept] destination_number(339XXXXXXXX) =~ /^\*8$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->intercept-ext] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [intercept-ext] destination_number(339XXXXXXXX) =~ /^\*\*(\d+)$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->redial] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [redial] > destination_number(339XXXXXXXX) =~ /^(redial|870)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->global] > continue=true > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [global] > ${call_debug}(false) =~ /^true$/ break=never > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [global] > ${sip_has_crypto}() =~ > /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Absolute Condition [global] > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->snom-demo-2] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [snom-demo-2] destination_number(339XXXXXXXX) =~ /^9001$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->snom-demo-1] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [snom-demo-1] destination_number(339XXXXXXXX) =~ /^9000$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->eavesdrop] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [eavesdrop] > destination_number(339XXXXXXXX) =~ /^88(\d{4})$|^\*0(.*)$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->eavesdrop] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [eavesdrop] > destination_number(339XXXXXXXX) =~ /^779$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->call_return] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [call_return] destination_number(339XXXXXXXX) =~ > /^\*69$|^869$|^lcr$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->del-group] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [del-group] > destination_number(339XXXXXXXX) =~ /^80(\d{2})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->add-group] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [add-group] > destination_number(339XXXXXXXX) =~ /^81(\d{2})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->call-group-simo] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [call-group-simo] destination_number(339XXXXXXXX) =~ /^82(\d{2})$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->call-group-order] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [call-group-order] destination_number(339XXXXXXXX) =~ > /^83(\d{2})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->extension-intercom] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [extension-intercom] destination_number(339XXXXXXXX) =~ > /^8(10[01][0-9])$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->Local_Extension] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [Local_Extension] destination_number(339XXXXXXXX) =~ > /^(10[01][0-9])$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->Local_Extension_Skinny] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [Local_Extension_Skinny] destination_number(339XXXXXXXX) =~ > /^(11[01][0-9])$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->group_dial_sales] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [group_dial_sales] destination_number(339XXXXXXXX) =~ /^2000$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->group_dial_support] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [group_dial_support] destination_number(339XXXXXXXX) =~ /^2001$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->group_dial_billing] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [group_dial_billing] destination_number(339XXXXXXXX) =~ /^2002$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->operator] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [operator] > destination_number(339XXXXXXXX) =~ /^(operator|0)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->vmain] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [vmain] > destination_number(339XXXXXXXX) =~ /^vmain$|^4000$|^\*98$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->sip_uri] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [sip_uri] > destination_number(339XXXXXXXX) =~ /^sip:(.*)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->nb_conferences] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [nb_conferences] destination_number(339XXXXXXXX) =~ /^(30\d{2})$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->wb_conferences] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [wb_conferences] destination_number(339XXXXXXXX) =~ /^(31\d{2})$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->uwb_conferences] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [uwb_conferences] destination_number(339XXXXXXXX) =~ /^(32\d{2})$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->cdquality_conferences] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [cdquality_conferences] destination_number(339XXXXXXXX) =~ > /^(33\d{2})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->freeswitch_public_conf_via_sip] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [freeswitch_public_conf_via_sip] destination_number(339XXXXXXXX) > =~ /^9(888|8888|1616|3232)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->mad_boss_intercom] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [mad_boss_intercom] destination_number(339XXXXXXXX) =~ /^0911$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->mad_boss_intercom] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [mad_boss_intercom] destination_number(339XXXXXXXX) =~ /^0912$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->mad_boss] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [mad_boss] > destination_number(339XXXXXXXX) =~ /^0913$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->ivr_demo] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [ivr_demo] > destination_number(339XXXXXXXX) =~ /^5000$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->dynamic_conference] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [dynamic_conference] destination_number(339XXXXXXXX) =~ /^5001$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->rtp_multicast_page] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [rtp_multicast_page] destination_number(339XXXXXXXX) =~ > /^pagegroup$|^7243$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->park] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [park] > destination_number(339XXXXXXXX) =~ /^5900$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->unpark] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [unpark] > destination_number(339XXXXXXXX) =~ /^5901$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->valet_park] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [valet_park] > destination_number(339XXXXXXXX) =~ /^(6000)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->valet_park] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [valet_park] > destination_number(339XXXXXXXX) =~ /^(60\d[1-9])$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->park] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [park] > destination_number(339XXXXXXXX) =~ /park\+(\d+)/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->unpark] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [unpark] > destination_number(339XXXXXXXX) =~ /^parking$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->park] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [park] > destination_number(339XXXXXXXX) =~ /callpark/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->unpark] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [unpark] > destination_number(339XXXXXXXX) =~ /pickup/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->wait] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [wait] > destination_number(339XXXXXXXX) =~ /^wait$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->fax_receive] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [fax_receive] destination_number(339XXXXXXXX) =~ /^9178$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->fax_transmit] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [fax_transmit] destination_number(339XXXXXXXX) =~ /^9179$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->ringback_180] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [ringback_180] destination_number(339XXXXXXXX) =~ /^9180$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->ringback_183_uk_ring] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [ringback_183_uk_ring] destination_number(339XXXXXXXX) =~ /^9181$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->ringback_183_music_ring] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [ringback_183_music_ring] destination_number(339XXXXXXXX) =~ > /^9182$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->ringback_post_answer_uk_ring] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [ringback_post_answer_uk_ring] destination_number(339XXXXXXXX) =~ > /^9183$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->ringback_post_answer_music] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [ringback_post_answer_music] destination_number(339XXXXXXXX) =~ > /^9184$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->ClueCon] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [ClueCon] > destination_number(339XXXXXXXX) =~ /^9191$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->show_info] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [show_info] > destination_number(339XXXXXXXX) =~ /^9192$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->video_record] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [video_record] destination_number(339XXXXXXXX) =~ /^9193$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->video_playback] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [video_playback] destination_number(339XXXXXXXX) =~ /^9194$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->delay_echo] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [delay_echo] > destination_number(339XXXXXXXX) =~ /^9195$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->echo] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [echo] > destination_number(339XXXXXXXX) =~ /^9196$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->milliwatt] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [milliwatt] > destination_number(339XXXXXXXX) =~ /^9197$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->tone_stream] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [tone_stream] destination_number(339XXXXXXXX) =~ /^9198$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->zrtp_enrollement] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) > [zrtp_enrollement] destination_number(339XXXXXXXX) =~ /^9787$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->hold_music] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [hold_music] > destination_number(339XXXXXXXX) =~ /^9664$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->from_pstn] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [from_pstn] > destination_number(339XXXXXXXX) =~ /^0000$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing [default->101] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [101] > destination_number(339XXXXXXXX) =~ /^101$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->pizza_demo] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (FAIL) [pizza_demo] > destination_number(339XXXXXXXX) =~ /^(pizza|74992)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > parsing > [default->gxw4104-fxo-local] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (PASS) > [gxw4104-fxo-local] ${toll_allow}(domestic,international,local) =~ > /local/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Regex (PASS) > [gxw4104-fxo-local] destination_number(339XXXXXXXX) =~ > /^(\d{6,})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > set(effective_caller_id_number=0321234567 ) > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > set(effective_caller_id_name=ThisIsMyCompany) > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > set(ignore_early_media=ring_ready) > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > set(ringback=${us-ring}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 > Action > bridge(sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/1000 at 192.168.0.2 > ) State Change CS_ROUTING > -> CS_EXECUTE > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 > [BREAK] > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1000 at 192.168.0.2 > ) State ROUTING going to sleep > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2 > ) Running State Change > CS_EXECUTE > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/1000 at 192.168.0.2 > ) State EXECUTE > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:241 > sofia/internal/1000 at 192.168.0.2 > SOFIA EXECUTE > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/1000 at 192.168.0.2 > Standard EXECUTE > > EXECUTE sofia/internal/1000 at 192.168.0.2 > set(open=true) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 > SET [open]=[true] > > EXECUTE sofia/internal/1000 at 192.168.0.2 > > hash(insert/192.168.0.2-spymap/1000/710da826-7726-11e1-bf72-e50b8b76e101) > > EXECUTE sofia/internal/1000 at 192.168.0.2 > > hash(insert/192.168.0.2-last_dial/1000/339XXXXXXXX) > > EXECUTE sofia/internal/1000 at 192.168.0.2 > > hash(insert/192.168.0.2-last_dial/global/710da826-7726-11e1-bf72-e50b8b76e101) > > EXECUTE sofia/internal/1000 at 192.168.0.2 > export(RFC2822_DATE=Mon, > 26 Mar 2012 11:31:18 +0200) > > 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1093 EXPORT > (export_vars) [RFC2822_DATE]=[Mon, 26 Mar 2012 11:31:18 +0200] > > EXECUTE sofia/internal/1000 at 192.168.0.2 > > set(effective_caller_id_number=0321234567 ) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 > SET > [effective_caller_id_number]=[0321234567 ] > > EXECUTE sofia/internal/1000 at 192.168.0.2 > > set(effective_caller_id_name=ThisIsMyCompany) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 > SET > [effective_caller_id_name]=[ThisIsMyCompany] > > EXECUTE sofia/internal/1000 at 192.168.0.2 > > set(ignore_early_media=ring_ready) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 > SET > [ignore_early_media]=[ring_ready] > > EXECUTE sofia/internal/1000 at 192.168.0.2 > > set(ringback=%(2000,4000,440,480)) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 > SET > [ringback]=[%(2000,4000,440,480)] > > EXECUTE sofia/internal/1000 at 192.168.0.2 > > bridge(sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > > 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1047 > sofia/internal/1000 at 192.168.0.2 > EXPORTING[export_vars] > [RFC2822_DATE]=[Mon, 26 Mar 2012 11:31:18 +0200] to event > > 2012-03-26 11:31:18.692558 [DEBUG] switch_ivr_originate.c:1884 > Parsing global variables > > 2012-03-26 11:31:18.692558 [NOTICE] switch_channel.c:926 New > Channel sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > [710f6738-7726-11e1-bf77-e50b8b76e101] > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:4691 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State Change CS_NEW -> CS_INIT > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > Running State Change CS_INIT > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State INIT > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:85 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > SOFIA INIT > > send 1279 bytes to udp/[192.168.0.3]:5060 at 09:31:18.713245: > > ------------------------------------------------------------------------ > > INVITE sip:gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > SIP/2.0 > > Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K > > Max-Forwards: 69 > > From: "ThisIsMyCompany" > ;tag=reXaHtmyvm2eN > > To: > > > Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a > > CSeq: 26041075 INVITE > > Contact: > > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 > 15-58-48 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, > sla, include-session-description, presence.winfo, message-summary, > refer > > Privacy: none > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 311 > > X-FS-Support: update_display,send_info > > P-Asserted-Identity: "ThisIsMyCompany" > > > > v=0 > > o=FreeSWITCH 1332736038 1332736039 IN IP4 192.168.0.2 > > s=FreeSWITCH > > c=IN IP4 192.168.0.2 > > t=0 0 > > m=audio 18240 RTP/AVP 8 98 99 9 0 3 101 13 > > a=rtpmap:98 G7221/32000 > > a=fmtp:98 bitrate=48000 > > a=rtpmap:99 G7221/16000 > > a=fmtp:99 bitrate=32000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=ptime:20 > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:125 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State Change CS_INIT -> CS_ROUTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State INIT going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > Running State Change CS_ROUTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:1886 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > Callstate Change DOWN -> RINGING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State ROUTING > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:148 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > SOFIA ROUTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State Change CS_ROUTING -> CS_CONSUME_MEDIA > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State ROUTING going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > Running State Change CS_CONSUME_MEDIA > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State CONSUME_MEDIA > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State CONSUME_MEDIA going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] sofia.c:5526 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > entering state [calling][0] > > recv 355 bytes from udp/[192.168.0.3]:5060 at 09:31:18.717462: > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K > > From: "ThisIsMyCompany" > ;tag=reXaHtmyvm2eN > > To: > > > Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a > > CSeq: 26041075 INVITE > > User-Agent: Grandstream GXW4104 (HW 2.0, Ch:5) 1.3.4.9 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > recv 367 bytes from udp/[192.168.0.3]:5060 at 09:31:18.717980: > > ------------------------------------------------------------------------ > > SIP/2.0 403 > > Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K > > From: "ThisIsMyCompany" > ;tag=reXaHtmyvm2eN > > To: > ;tag=3ga2B2jmSSe6H > > Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a > > CSeq: 26041075 INVITE > > User-Agent: Grandstream GXW4104 (HW 2.0, Ch:8) 1.3.4.9 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > send 370 bytes to udp/[192.168.0.3]:5060 at 09:31:18.718065: > > ------------------------------------------------------------------------ > > ACK sip:gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > SIP/2.0 > > Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K > > Max-Forwards: 69 > > From: "ThisIsMyCompany" > ;tag=reXaHtmyvm2eN > > To: > ;tag=3ga2B2jmSSe6H > > Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a > > CSeq: 26041075 ACK > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] sofia.c:5526 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > entering state [terminated][403] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2848 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > Callstate Change RINGING -> HANGUP > > 2012-03-26 11:31:18.712698 [NOTICE] sofia.c:6293 Hangup > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > [CS_CONSUME_MEDIA] [CALL_REJECTED] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2871 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > [KILL] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > Running State Change CS_HANGUP > > 2012-03-26 11:31:18.712698 [DEBUG] switch_ivr_originate.c:3364 > Originate Resulted in Error Cause: 21 [CALL_REJECTED] > > 2012-03-26 11:31:18.712698 [INFO] mod_dptools.c:2922 Originate > Failed. Cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2848 > (sofia/internal/1000 at 192.168.0.2 > ) Callstate Change RINGING > -> HANGUP > > 2012-03-26 11:31:18.712698 [NOTICE] mod_dptools.c:3041 Hangup > sofia/internal/1000 at 192.168.0.2 > [CS_EXECUTE] [CALL_REJECTED] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2871 Send > signal sofia/internal/1000 at 192.168.0.2 > [KILL] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 > [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:2285 > sofia/internal/1000 at 192.168.0.2 > skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/1000 at 192.168.0.2 > ) State EXECUTE going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2 > ) Running State Change > CS_HANGUP > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1000 at 192.168.0.2 > ) State HANGUP > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:463 > sofia/internal/1000 at 192.168.0.2 > Overriding SIP cause 603 > with 403 from the other leg > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/1000 at 192.168.0.2 > hanging up, cause: > CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:534 Responding to > INVITE with: 403 > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/1000 at 192.168.0.2 > Standard HANGUP, cause: > CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1000 at 192.168.0.2 > ) State HANGUP going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/1000 at 192.168.0.2 > ) State Change CS_HANGUP > -> CS_REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 > [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2 > ) Running State Change > CS_REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1000 at 192.168.0.2 > ) State REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/1000 at 192.168.0.2 > Standard REPORTING, > cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1000 at 192.168.0.2 > ) State REPORTING going to > sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/1000 at 192.168.0.2 > ) State Change > CS_REPORTING -> CS_DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 > [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1380 > Session 13 (sofia/internal/1000 at 192.168.0.2 > ) Locked, Waiting on > external entities > > 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1398 > Session 13 (sofia/internal/1000 at 192.168.0.2 > ) Ended > > 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1400 > Close Channel sofia/internal/1000 at 192.168.0.2 > [CS_DESTROY] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/1000 at 192.168.0.2 > ) Callstate Change HANGUP > -> DOWN > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/1000 at 192.168.0.2 > ) Running State Change > CS_DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1000 at 192.168.0.2 > ) State DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:374 > sofia/internal/1000 at 192.168.0.2 > SOFIA DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/1000 at 192.168.0.2 > Standard DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1000 at 192.168.0.2 > ) State DESTROY going to sleep > > send 833 bytes to udp/[192.168.0.200]:5060 at 09:31:18.720584: > > ------------------------------------------------------------------------ > > SIP/2.0 403 Forbidden > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport=5060 > > From: "1000" > ;tag=645977894 > > To: > ;tag=Q53HFZ3tZBcvS > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > > CSeq: 6 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 > 15-58-48 -0600 > > Accept: application/sdp > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, > sla, include-session-description, presence.winfo, message-summary, > refer > > Reason: Q.850;cause=21;text="CALL_REJECTED" > > Content-Length: 0 > > P-Asserted-Identity: "339XXXXXXXX" > > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State HANGUP > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > hanging up, cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > Standard HANGUP, cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State HANGUP going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State Change CS_HANGUP -> CS_REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > Running State Change CS_REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > Standard REPORTING, cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State REPORTING going to sleep > > recv 618 bytes from udp/[192.168.0.200]:5060 at 09:31:18.721757: > > ------------------------------------------------------------------------ > > ACK sip:339XXXXXXXX at 192.168.0.2 > SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport > > From: "1000" > ;tag=645977894 > > To: > ;tag=Q53HFZ3tZBcvS > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > > CSeq: 6 ACK > > Proxy-Authorization: Digest username="1000", realm="192.168.0.2", > nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", > uri="sip:339XXXXXXXX at 192.168.0.2" > , > response="7378107bc277149e3b6ef00c1c766a71", algorithm=MD5, > cnonce="234abcc436e2667097e7fe6eia53e8dd", qop=auth, nc=00000001 > > Max-Forwards: 70 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State Change CS_REPORTING -> CS_DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1380 > Session 14 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > Locked, Waiting on external entities > > 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1398 > Session 14 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > Ended > > 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1400 > Close Channel > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > [CS_DESTROY] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > Callstate Change HANGUP -> DOWN > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > Running State Change CS_DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:374 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > SOFIA DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > > Standard DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > ) > State DESTROY going to sleep > > freeswitch at internal> > > > > > Il 25/03/2012 06:39, Anton Kvashenkin ha scritto: >> originate sofia/internal/XXXX at 192.168.0.3 >> 9178 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/64ddfd50/attachment-0001.html From gcd at i.ph Mon Mar 26 16:50:38 2012 From: gcd at i.ph (Nandy Dagondon) Date: Mon, 26 Mar 2012 20:50:38 +0800 Subject: [Freeswitch-users] Grandstream 4104 In-Reply-To: <4F7060AC.9090903@elios.net> References: <4F6C46B0.7080805@elios.net> <4F703E66.5070607@elios.net> <4F7060AC.9090903@elios.net> Message-ID: correction ... we should use sofia/gateway NOT sofia/internal to dialout to PSTN so dialout on port1: on port2: i hope it works this time. On Mon, Mar 26, 2012 at 8:27 PM, a.avona wrote: > Hi Nandy, > thank's for your suggestion but it didn't help me. > I tryie all ports from 5060 to 5068 but it didn't work. > > if I try originate sofia/internal/XXXX at 192.168.0.3:5062 9178 > no calls are originated. > > any suggestion is very welcome. > > > Il 26/03/2012 13:14, Nandy Dagondon ha scritto: > > just a hint - did u try to use different port numbers when dialing out? if > i remember right, the gateway assigns different port to every physical port > i.e. 5060 for port1, 5062 for port2 and so on. > > so dialout on port1: data="sofia/internal/gxw4104-fxo1/$1 at 192.168.0.3:5060"/> > on port2: data="sofia/internal/gxw4104-fxo2/$1 at 192.168.0.3:5062"/> > > > i hope you'll make a progress. > > On Mon, Mar 26, 2012 at 6:01 PM, a.avona wrote: > >> Hi, thank's for your answer >> i did as you said and if i digit >> originate sofia/internal/XXXX at 192.168.0.3 9178 >> outgoing calls work well. >> >> if i try to originate call from a client it doesn't work so i think the >> problem is in the default account configuration >> here is the siptraces >> >> >> Thank's for any suggestion >> reagards >> Accursio Avona >> >> recv 865 bytes from udp/[192.168.0.200]:5060 at 09:31:18.696255: >> >> ------------------------------------------------------------------------ >> >> INVITE sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 >> >> Via: SIP/2.0/UDP 192.168.0.200:5060 >> ;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport >> >> From: "1000" ;tag=645977894 >> >> To: >> >> Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 >> >> CSeq: 5 INVITE >> >> Contact: >> >> Content-Type: application/sdp >> >> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE >> >> Max-Forwards: 70 >> >> Supported: 100rel, replaces, from-change >> >> User-Agent: SIPPER for PhonerLite >> >> P-Preferred-Identity: >> >> Content-Length: 260 >> >> >> v=0 >> >> o=- 162748822 0 IN IP4 192.168.0.200 >> >> s=SIPPER for PhonerLite >> >> c=IN IP4 192.168.0.200 >> >> t=0 0 >> >> m=audio 5062 RTP/AVP 8 3 0 101 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:3 GSM/8000 >> >> a=rtpmap:0 PCMU/8000 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16 >> >> a=sendrecv >> >> ------------------------------------------------------------------------ >> >> send 379 bytes to udp/[192.168.0.200]:5060 at 09:31:18.696482: >> >> ------------------------------------------------------------------------ >> >> SIP/2.0 100 Trying >> >> Via: SIP/2.0/UDP 192.168.0.200:5060 >> ;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport=5060 >> >> From: "1000" ;tag=645977894 >> >> To: >> >> Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 >> >> CSeq: 5 INVITE >> >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 >> -0600 >> >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> >> 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:7559 IP 192.168.0.200 Rejected >> by acl "domains". Falling back to Digest auth. >> >> send 865 bytes to udp/[192.168.0.200]:5060 at 09:31:18.697335: >> >> ------------------------------------------------------------------------ >> >> SIP/2.0 407 Proxy Authentication Required >> >> Via: SIP/2.0/UDP 192.168.0.200:5060 >> ;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport=5060 >> >> From: "1000" ;tag=645977894 >> >> To: >> ;tag=pvaSD4jQ22N9D >> >> Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 >> >> CSeq: 5 INVITE >> >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 >> -0600 >> >> Accept: application/sdp >> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> >> Supported: timer, precondition, path, replaces >> >> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer >> >> Proxy-Authenticate: Digest realm="192.168.0.2", >> nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", algorithm=MD5, qop="auth" >> >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> >> recv 346 bytes from udp/[192.168.0.200]:5060 at 09:31:18.698468: >> >> ------------------------------------------------------------------------ >> >> ACK sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 >> >> Via: SIP/2.0/UDP 192.168.0.200:5060 >> ;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport >> >> From: "1000" ;tag=645977894 >> >> To: >> ;tag=pvaSD4jQ22N9D >> >> Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 >> >> CSeq: 5 ACK >> >> Max-Forwards: 70 >> >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> >> recv 1137 bytes from udp/[192.168.0.200]:5060 at 09:31:18.699228: >> >> ------------------------------------------------------------------------ >> >> INVITE sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 >> >> Via: SIP/2.0/UDP 192.168.0.200:5060 >> ;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport >> >> From: "1000" ;tag=645977894 >> >> To: >> >> Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 >> >> CSeq: 6 INVITE >> >> Contact: >> >> Proxy-Authorization: Digest username="1000", realm="192.168.0.2", >> nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", uri= >> "sip:339XXXXXXXX at 192.168.0.2" , >> response="7378107bc277149e3b6ef00c1c766a71", algorithm=MD5, >> cnonce="234abcc436e2667097e7fe6eia53e8dd", qop=auth, nc=00000001 >> >> Content-Type: application/sdp >> >> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE >> >> Max-Forwards: 70 >> >> Supported: 100rel, replaces, from-change >> >> User-Agent: SIPPER for PhonerLite >> >> P-Preferred-Identity: >> >> Content-Length: 260 >> >> >> v=0 >> >> o=- 162748822 0 IN IP4 192.168.0.200 >> >> s=SIPPER for PhonerLite >> >> c=IN IP4 192.168.0.200 >> >> t=0 0 >> >> m=audio 5062 RTP/AVP 8 3 0 101 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:3 GSM/8000 >> >> a=rtpmap:0 PCMU/8000 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16 >> >> a=sendrecv >> >> ------------------------------------------------------------------------ >> >> send 379 bytes to udp/[192.168.0.200]:5060 at 09:31:18.699380: >> >> ------------------------------------------------------------------------ >> >> SIP/2.0 100 Trying >> >> Via: SIP/2.0/UDP 192.168.0.200:5060 >> ;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport=5060 >> >> From: "1000" ;tag=645977894 >> >> To: >> >> Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 >> >> CSeq: 6 INVITE >> >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 >> -0600 >> >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> >> 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:7559 IP 192.168.0.200 Rejected >> by acl "domains". Falling back to Digest auth. >> >> 2012-03-26 11:31:18.692558 [NOTICE] switch_channel.c:926 New Channel >> sofia/internal/1000 at 192.168.0.2 [710da826-7726-11e1-bf72-e50b8b76e101] >> >> 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5526 Channel >> sofia/internal/1000 at 192.168.0.2 entering state [received][100] >> >> 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5537 Remote SDP: >> >> v=0 >> >> o=- 162748822 0 IN IP4 192.168.0.200 >> >> s=SIPPER for PhonerLite >> >> c=IN IP4 192.168.0.200 >> >> t=0 0 >> >> m=audio 5062 RTP/AVP 8 3 0 101 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:3 GSM/8000 >> >> a=rtpmap:0 PCMU/8000 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16 >> >> >> 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec >> Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] >> >> 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare >> [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] >> >> 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare >> [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] >> >> 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare >> [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] >> >> 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec Compare >> [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] >> >> 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:2991 Set Codec >> sofia/internal/1000 at 192.168.0.2 PCMA/8000 20 ms 160 samples 64000 bits >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_codec.c:111 >> sofia/internal/1000 at 192.168.0.2 Original read codec set to PCMA:8 >> >> 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4995 Set 2833 dtmf >> send/recv payload to 101 >> >> 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5749 ( >> sofia/internal/1000 at 192.168.0.2) State Change CS_NEW -> CS_INIT >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal >> sofia/internal/1000 at 192.168.0.2 [BREAK] >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 ( >> sofia/internal/1000 at 192.168.0.2) Running State Change CS_INIT >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:401 ( >> sofia/internal/1000 at 192.168.0.2) State INIT >> >> 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:85 >> sofia/internal/1000 at 192.168.0.2 SOFIA INIT >> >> 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:125 ( >> sofia/internal/1000 at 192.168.0.2) State Change CS_INIT -> CS_ROUTING >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal >> sofia/internal/1000 at 192.168.0.2 [BREAK] >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:401 ( >> sofia/internal/1000 at 192.168.0.2) State INIT going to sleep >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 ( >> sofia/internal/1000 at 192.168.0.2) Running State Change CS_ROUTING >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1886 ( >> sofia/internal/1000 at 192.168.0.2) Callstate Change DOWN -> RINGING >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:410 ( >> sofia/internal/1000 at 192.168.0.2) State ROUTING >> >> 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:148 >> sofia/internal/1000 at 192.168.0.2 SOFIA ROUTING >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:104 >> sofia/internal/1000 at 192.168.0.2 Standard ROUTING >> >> 2012-03-26 11:31:18.692558 [INFO] mod_dialplan_xml.c:485 Processing 1000 >> <1000>->339XXXXXXXX in context default >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unloop] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unloop] >> ${unroll_loops}(true) =~ /^true$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unloop] >> ${sip_looped_call}() =~ /^true$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->tod_example] >> continue=true >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Date/Time Match (PASS) >> [tod_example] break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(open=true) >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->holiday_example] continue=true >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Date/TimeMatch (FAIL) >> [holiday_example] break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->global-intercept] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [global-intercept] destination_number(339XXXXXXXX) =~ /^886$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->group-intercept] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [group-intercept] >> destination_number(339XXXXXXXX) =~ /^\*8$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->intercept-ext] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [intercept-ext] >> destination_number(339XXXXXXXX) =~ /^\*\*(\d+)$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->redial] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [redial] >> destination_number(339XXXXXXXX) =~ /^(redial|870)$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->global] >> continue=true >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] >> ${call_debug}(false) =~ /^true$/ break=never >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] >> ${sip_has_crypto}() =~ >> /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Absolute Condition [global] >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Action >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Action >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Action >> hash(insert/${domain_name}-last_dial/global/${uuid}) >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Action >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->snom-demo-2] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-2] >> destination_number(339XXXXXXXX) =~ /^9001$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->snom-demo-1] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-1] >> destination_number(339XXXXXXXX) =~ /^9000$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] >> destination_number(339XXXXXXXX) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] >> destination_number(339XXXXXXXX) =~ /^779$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->call_return] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call_return] >> destination_number(339XXXXXXXX) =~ /^\*69$|^869$|^lcr$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->del-group] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [del-group] >> destination_number(339XXXXXXXX) =~ /^80(\d{2})$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->add-group] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [add-group] >> destination_number(339XXXXXXXX) =~ /^81(\d{2})$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->call-group-simo] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call-group-simo] >> destination_number(339XXXXXXXX) =~ /^82(\d{2})$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->call-group-order] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [call-group-order] destination_number(339XXXXXXXX) =~ /^83(\d{2})$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->extension-intercom] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [extension-intercom] destination_number(339XXXXXXXX) =~ /^8(10[01][0-9])$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->Local_Extension] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [Local_Extension] >> destination_number(339XXXXXXXX) =~ /^(10[01][0-9])$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->Local_Extension_Skinny] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [Local_Extension_Skinny] destination_number(339XXXXXXXX) =~ >> /^(11[01][0-9])$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->group_dial_sales] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [group_dial_sales] destination_number(339XXXXXXXX) =~ /^2000$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->group_dial_support] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [group_dial_support] destination_number(339XXXXXXXX) =~ /^2001$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->group_dial_billing] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [group_dial_billing] destination_number(339XXXXXXXX) =~ /^2002$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->operator] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [operator] >> destination_number(339XXXXXXXX) =~ /^(operator|0)$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->vmain] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [vmain] >> destination_number(339XXXXXXXX) =~ /^vmain$|^4000$|^\*98$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->sip_uri] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [sip_uri] >> destination_number(339XXXXXXXX) =~ /^sip:(.*)$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->nb_conferences] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [nb_conferences] >> destination_number(339XXXXXXXX) =~ /^(30\d{2})$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->wb_conferences] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wb_conferences] >> destination_number(339XXXXXXXX) =~ /^(31\d{2})$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->uwb_conferences] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [uwb_conferences] >> destination_number(339XXXXXXXX) =~ /^(32\d{2})$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->cdquality_conferences] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [cdquality_conferences] destination_number(339XXXXXXXX) =~ /^(33\d{2})$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->freeswitch_public_conf_via_sip] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [freeswitch_public_conf_via_sip] destination_number(339XXXXXXXX) =~ >> /^9(888|8888|1616|3232)$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->mad_boss_intercom] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [mad_boss_intercom] destination_number(339XXXXXXXX) =~ /^0911$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->mad_boss_intercom] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [mad_boss_intercom] destination_number(339XXXXXXXX) =~ /^0912$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->mad_boss] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [mad_boss] >> destination_number(339XXXXXXXX) =~ /^0913$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ivr_demo] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ivr_demo] >> destination_number(339XXXXXXXX) =~ /^5000$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->dynamic_conference] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [dynamic_conference] destination_number(339XXXXXXXX) =~ /^5001$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->rtp_multicast_page] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [rtp_multicast_page] destination_number(339XXXXXXXX) =~ >> /^pagegroup$|^7243$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] >> destination_number(339XXXXXXXX) =~ /^5900$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] >> destination_number(339XXXXXXXX) =~ /^5901$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->valet_park] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] >> destination_number(339XXXXXXXX) =~ /^(6000)$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->valet_park] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] >> destination_number(339XXXXXXXX) =~ /^(60\d[1-9])$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] >> source(mod_sofia) =~ /mod_sofia/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] >> destination_number(339XXXXXXXX) =~ /park\+(\d+)/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] >> source(mod_sofia) =~ /mod_sofia/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] >> destination_number(339XXXXXXXX) =~ /^parking$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] >> source(mod_sofia) =~ /mod_sofia/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] >> destination_number(339XXXXXXXX) =~ /callpark/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] >> source(mod_sofia) =~ /mod_sofia/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] >> destination_number(339XXXXXXXX) =~ /pickup/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->wait] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wait] >> destination_number(339XXXXXXXX) =~ /^wait$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->fax_receive] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_receive] >> destination_number(339XXXXXXXX) =~ /^9178$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->fax_transmit] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_transmit] >> destination_number(339XXXXXXXX) =~ /^9179$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->ringback_180] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_180] >> destination_number(339XXXXXXXX) =~ /^9180$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->ringback_183_uk_ring] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [ringback_183_uk_ring] destination_number(339XXXXXXXX) =~ /^9181$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->ringback_183_music_ring] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [ringback_183_music_ring] destination_number(339XXXXXXXX) =~ /^9182$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->ringback_post_answer_uk_ring] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [ringback_post_answer_uk_ring] destination_number(339XXXXXXXX) =~ /^9183$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->ringback_post_answer_music] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [ringback_post_answer_music] destination_number(339XXXXXXXX) =~ /^9184$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ClueCon] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ClueCon] >> destination_number(339XXXXXXXX) =~ /^9191$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->show_info] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [show_info] >> destination_number(339XXXXXXXX) =~ /^9192$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->video_record] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_record] >> destination_number(339XXXXXXXX) =~ /^9193$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->video_playback] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_playback] >> destination_number(339XXXXXXXX) =~ /^9194$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->delay_echo] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [delay_echo] >> destination_number(339XXXXXXXX) =~ /^9195$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->echo] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [echo] >> destination_number(339XXXXXXXX) =~ /^9196$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->milliwatt] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [milliwatt] >> destination_number(339XXXXXXXX) =~ /^9197$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->tone_stream] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [tone_stream] >> destination_number(339XXXXXXXX) =~ /^9198$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->zrtp_enrollement] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) >> [zrtp_enrollement] destination_number(339XXXXXXXX) =~ /^9787$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->hold_music] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [hold_music] >> destination_number(339XXXXXXXX) =~ /^9664$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->from_pstn] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [from_pstn] >> destination_number(339XXXXXXXX) =~ /^0000$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->101] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [101] >> destination_number(339XXXXXXXX) =~ /^101$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->pizza_demo] >> continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [pizza_demo] >> destination_number(339XXXXXXXX) =~ /^(pizza|74992)$/ break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 parsing >> [default->gxw4104-fxo-local] continue=false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) >> [gxw4104-fxo-local] ${toll_allow}(domestic,international,local) =~ /local/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) >> [gxw4104-fxo-local] destination_number(339XXXXXXXX) =~ /^(\d{6,})$/ >> break=on-false >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Action >> set(effective_caller_id_number=0321234567) >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Action >> set(effective_caller_id_name=ThisIsMyCompany) >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Action >> set(ignore_early_media=ring_ready) >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(ringback=${us-ring}) >> >> Dialplan: sofia/internal/1000 at 192.168.0.2 Action bridge( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:154 ( >> sofia/internal/1000 at 192.168.0.2) State Change CS_ROUTING -> CS_EXECUTE >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal >> sofia/internal/1000 at 192.168.0.2 [BREAK] >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:410 ( >> sofia/internal/1000 at 192.168.0.2) State ROUTING going to sleep >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 ( >> sofia/internal/1000 at 192.168.0.2) Running State Change CS_EXECUTE >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:417 ( >> sofia/internal/1000 at 192.168.0.2) State EXECUTE >> >> 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:241 >> sofia/internal/1000 at 192.168.0.2 SOFIA EXECUTE >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:192 >> sofia/internal/1000 at 192.168.0.2 Standard EXECUTE >> >> EXECUTE sofia/internal/1000 at 192.168.0.2 set(open=true) >> >> 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 >> sofia/internal/1000 at 192.168.0.2 SET [open]=[true] >> >> EXECUTE sofia/internal/1000 at 192.168.0.2hash(insert/192.168.0.2-spymap/1000/710da826-7726-11e1-bf72-e50b8b76e101) >> >> EXECUTE sofia/internal/1000 at 192.168.0.2hash(insert/192.168.0.2-last_dial/1000/339XXXXXXXX) >> >> EXECUTE sofia/internal/1000 at 192.168.0.2hash(insert/192.168.0.2-last_dial/global/710da826-7726-11e1-bf72-e50b8b76e101) >> >> EXECUTE sofia/internal/1000 at 192.168.0.2 export(RFC2822_DATE=Mon, 26 Mar >> 2012 11:31:18 +0200) >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1093 EXPORT >> (export_vars) [RFC2822_DATE]=[Mon, 26 Mar 2012 11:31:18 +0200] >> >> EXECUTE sofia/internal/1000 at 192.168.0.2 set(effective_caller_id_number= >> 0321234567) >> >> 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 >> sofia/internal/1000 at 192.168.0.2 SET [effective_caller_id_number]= >> [0321234567] >> >> EXECUTE sofia/internal/1000 at 192.168.0.2set(effective_caller_id_name=ThisIsMyCompany) >> >> 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 >> sofia/internal/1000 at 192.168.0.2 SET >> [effective_caller_id_name]=[ThisIsMyCompany] >> >> EXECUTE sofia/internal/1000 at 192.168.0.2set(ignore_early_media=ring_ready) >> >> 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 >> sofia/internal/1000 at 192.168.0.2 SET [ignore_early_media]=[ring_ready] >> >> EXECUTE sofia/internal/1000 at 192.168.0.2set(ringback=%(2000,4000,440,480)) >> >> 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 >> sofia/internal/1000 at 192.168.0.2 SET [ringback]=[%(2000,4000,440,480)] >> >> EXECUTE sofia/internal/1000 at 192.168.0.2 bridge( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1047 >> sofia/internal/1000 at 192.168.0.2 EXPORTING[export_vars] >> [RFC2822_DATE]=[Mon, 26 Mar 2012 11:31:18 +0200] to event >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_ivr_originate.c:1884 Parsing >> global variables >> >> 2012-03-26 11:31:18.692558 [NOTICE] switch_channel.c:926 New Channel >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060[710f6738-7726-11e1-bf77-e50b8b76e101] >> >> 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:4691 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change >> CS_NEW -> CS_INIT >> >> 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send signal >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State >> Change CS_INIT >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:401 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State INIT >> >> 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:85 >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA INIT >> >> send 1279 bytes to udp/[192.168.0.3]:5060 at 09:31:18.713245: >> >> ------------------------------------------------------------------------ >> >> INVITE sip:gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SIP/2.0 >> >> Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K >> >> Max-Forwards: 69 >> >> From: "ThisIsMyCompany" >> ;tag=reXaHtmyvm2eN >> >> To: >> >> Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a >> >> CSeq: 26041075 INVITE >> >> Contact: >> >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 >> -0600 >> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> >> Supported: timer, precondition, path, replaces >> >> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer >> >> Privacy: none >> >> Content-Type: application/sdp >> >> Content-Disposition: session >> >> Content-Length: 311 >> >> X-FS-Support: update_display,send_info >> >> P-Asserted-Identity: "ThisIsMyCompany" >> >> >> v=0 >> >> o=FreeSWITCH 1332736038 1332736039 IN IP4 192.168.0.2 >> >> s=FreeSWITCH >> >> c=IN IP4 192.168.0.2 >> >> t=0 0 >> >> m=audio 18240 RTP/AVP 8 98 99 9 0 3 101 13 >> >> a=rtpmap:98 G7221/32000 >> >> a=fmtp:98 bitrate=48000 >> >> a=rtpmap:99 G7221/16000 >> >> a=fmtp:99 bitrate=32000 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16 >> >> a=ptime:20 >> >> ------------------------------------------------------------------------ >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] >> >> 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:125 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change >> CS_INIT -> CS_ROUTING >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:401 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State INIT >> going to sleep >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State >> Change CS_ROUTING >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:1886 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate >> Change DOWN -> RINGING >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:410 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State ROUTING >> >> 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:148 >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA ROUTING >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_ivr_originate.c:66 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change >> CS_ROUTING -> CS_CONSUME_MEDIA >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:410 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State ROUTING >> going to sleep >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State >> Change CS_CONSUME_MEDIA >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:429 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State >> CONSUME_MEDIA >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:429 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State >> CONSUME_MEDIA going to sleep >> >> 2012-03-26 11:31:18.712698 [DEBUG] sofia.c:5526 Channel >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 entering state >> [calling][0] >> >> recv 355 bytes from udp/[192.168.0.3]:5060 at 09:31:18.717462: >> >> ------------------------------------------------------------------------ >> >> SIP/2.0 100 Trying >> >> Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K >> >> From: "ThisIsMyCompany" >> ;tag=reXaHtmyvm2eN >> >> To: >> >> Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a >> >> CSeq: 26041075 INVITE >> >> User-Agent: Grandstream GXW4104 (HW 2.0, Ch:5) 1.3.4.9 >> >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> >> recv 367 bytes from udp/[192.168.0.3]:5060 at 09:31:18.717980: >> >> ------------------------------------------------------------------------ >> >> SIP/2.0 403 >> >> Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K >> >> From: "ThisIsMyCompany" >> ;tag=reXaHtmyvm2eN >> >> To: >> ;tag=3ga2B2jmSSe6H >> >> Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a >> >> CSeq: 26041075 INVITE >> >> User-Agent: Grandstream GXW4104 (HW 2.0, Ch:8) 1.3.4.9 >> >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> >> send 370 bytes to udp/[192.168.0.3]:5060 at 09:31:18.718065: >> >> ------------------------------------------------------------------------ >> >> ACK sip:gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SIP/2.0 >> >> Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K >> >> Max-Forwards: 69 >> >> From: "ThisIsMyCompany" >> ;tag=reXaHtmyvm2eN >> >> To: >> ;tag=3ga2B2jmSSe6H >> >> Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a >> >> CSeq: 26041075 ACK >> >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send signal >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] >> >> 2012-03-26 11:31:18.712698 [DEBUG] sofia.c:5526 Channel >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 entering state >> [terminated][403] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2848 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate >> Change RINGING -> HANGUP >> >> 2012-03-26 11:31:18.712698 [NOTICE] sofia.c:6293 Hangup >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060[CS_CONSUME_MEDIA] [CALL_REJECTED] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2871 Send signal >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [KILL] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State >> Change CS_HANGUP >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_ivr_originate.c:3364 Originate >> Resulted in Error Cause: 21 [CALL_REJECTED] >> >> 2012-03-26 11:31:18.712698 [INFO] mod_dptools.c:2922 Originate Failed. >> Cause: CALL_REJECTED >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2848 ( >> sofia/internal/1000 at 192.168.0.2) Callstate Change RINGING -> HANGUP >> >> 2012-03-26 11:31:18.712698 [NOTICE] mod_dptools.c:3041 Hangup >> sofia/internal/1000 at 192.168.0.2 [CS_EXECUTE] [CALL_REJECTED] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2871 Send signal >> sofia/internal/1000 at 192.168.0.2 [KILL] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal >> sofia/internal/1000 at 192.168.0.2 [BREAK] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:2285 >> sofia/internal/1000 at 192.168.0.2 skip receive message >> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:417 ( >> sofia/internal/1000 at 192.168.0.2) State EXECUTE going to sleep >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 ( >> sofia/internal/1000 at 192.168.0.2) Running State Change CS_HANGUP >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 ( >> sofia/internal/1000 at 192.168.0.2) State HANGUP >> >> 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:463 >> sofia/internal/1000 at 192.168.0.2 Overriding SIP cause 603 with 403 from >> the other leg >> >> 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:469 Channel >> sofia/internal/1000 at 192.168.0.2 hanging up, cause: CALL_REJECTED >> >> 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:534 Responding to INVITE >> with: 403 >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:47 >> sofia/internal/1000 at 192.168.0.2 Standard HANGUP, cause: CALL_REJECTED >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 ( >> sofia/internal/1000 at 192.168.0.2) State HANGUP going to sleep >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:393 ( >> sofia/internal/1000 at 192.168.0.2) State Change CS_HANGUP -> CS_REPORTING >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal >> sofia/internal/1000 at 192.168.0.2 [BREAK] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 ( >> sofia/internal/1000 at 192.168.0.2) Running State Change CS_REPORTING >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 ( >> sofia/internal/1000 at 192.168.0.2) State REPORTING >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:79 >> sofia/internal/1000 at 192.168.0.2 Standard REPORTING, cause: CALL_REJECTED >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 ( >> sofia/internal/1000 at 192.168.0.2) State REPORTING going to sleep >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:387 ( >> sofia/internal/1000 at 192.168.0.2) State Change CS_REPORTING -> CS_DESTROY >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal >> sofia/internal/1000 at 192.168.0.2 [BREAK] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1380 Session 13 ( >> sofia/internal/1000 at 192.168.0.2) Locked, Waiting on external entities >> >> 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1398 Session 13 >> (sofia/internal/1000 at 192.168.0.2) Ended >> >> 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1400 Close >> Channel sofia/internal/1000 at 192.168.0.2 [CS_DESTROY] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:491 ( >> sofia/internal/1000 at 192.168.0.2) Callstate Change HANGUP -> DOWN >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:494 ( >> sofia/internal/1000 at 192.168.0.2) Running State Change CS_DESTROY >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 ( >> sofia/internal/1000 at 192.168.0.2) State DESTROY >> >> 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:374 >> sofia/internal/1000 at 192.168.0.2 SOFIA DESTROY >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:86 >> sofia/internal/1000 at 192.168.0.2 Standard DESTROY >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 ( >> sofia/internal/1000 at 192.168.0.2) State DESTROY going to sleep >> >> send 833 bytes to udp/[192.168.0.200]:5060 at 09:31:18.720584: >> >> ------------------------------------------------------------------------ >> >> SIP/2.0 403 Forbidden >> >> Via: SIP/2.0/UDP 192.168.0.200:5060 >> ;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport=5060 >> >> From: "1000" ;tag=645977894 >> >> To: >> ;tag=Q53HFZ3tZBcvS >> >> Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 >> >> CSeq: 6 INVITE >> >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 15-58-48 >> -0600 >> >> Accept: application/sdp >> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> >> Supported: timer, precondition, path, replaces >> >> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer >> >> Reason: Q.850;cause=21;text="CALL_REJECTED" >> >> Content-Length: 0 >> >> P-Asserted-Identity: "339XXXXXXXX" >> >> >> ------------------------------------------------------------------------ >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State HANGUP >> >> 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:469 Channel >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 hanging up, >> cause: CALL_REJECTED >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:47 >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard >> HANGUP, cause: CALL_REJECTED >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State HANGUP >> going to sleep >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:393 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change >> CS_HANGUP -> CS_REPORTING >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State >> Change CS_REPORTING >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State REPORTING >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:79 >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard >> REPORTING, cause: CALL_REJECTED >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State >> REPORTING going to sleep >> >> recv 618 bytes from udp/[192.168.0.200]:5060 at 09:31:18.721757: >> >> ------------------------------------------------------------------------ >> >> ACK sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 >> >> Via: SIP/2.0/UDP 192.168.0.200:5060 >> ;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport >> >> From: "1000" ;tag=645977894 >> >> To: >> ;tag=Q53HFZ3tZBcvS >> >> Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 >> >> CSeq: 6 ACK >> >> Proxy-Authorization: Digest username="1000", realm="192.168.0.2", >> nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", uri= >> "sip:339XXXXXXXX at 192.168.0.2" , >> response="7378107bc277149e3b6ef00c1c766a71", algorithm=MD5, >> cnonce="234abcc436e2667097e7fe6eia53e8dd", qop=auth, nc=00000001 >> >> Max-Forwards: 70 >> >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:387 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State Change >> CS_REPORTING -> CS_DESTROY >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send signal >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1380 Session 14 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Locked, >> Waiting on external entities >> >> 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1398 Session 14 >> (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Ended >> >> 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1400 Close >> Channel sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060[CS_DESTROY] >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:491 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate >> Change HANGUP -> DOWN >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:494 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running State >> Change CS_DESTROY >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State DESTROY >> >> 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:374 >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA DESTROY >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:86 >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard DESTROY >> >> 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 ( >> sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State DESTROY >> going to sleep >> >> freeswitch at internal> >> >> >> >> Il 25/03/2012 06:39, Anton Kvashenkin ha scritto: >> >> originate sofia/internal/XXXX at 192.168.0.3 9178 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/85c85e18/attachment-0001.html From lazyvirus at gmx.com Mon Mar 26 17:02:58 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 26 Mar 2012 15:02:58 +0200 Subject: [Freeswitch-users] CentOS for ARM In-Reply-To: References: <4F6F28F0.4090604@redsleeve.org> <4F6F3570.4090608@gmail.com> <4F6F86C6.4030609@gmail.com> <20120326080552.41b03468@anubis.defcon1> <20120326084044.5a7f72bd@anubis.defcon1> Message-ID: <20120326150258.53dc0887@anubis.defcon1> On Mon, 26 Mar 2012 03:11:52 -0400 Brian Foster wrote: Well, I also started w/ RH, but just started: my graphic card wasn't recognized and I was stuck at 1/2 installation. Then a friend of mine show me Debian:) -- From a.avona at elios.net Mon Mar 26 17:05:15 2012 From: a.avona at elios.net (a.avona) Date: Mon, 26 Mar 2012 15:05:15 +0200 Subject: [Freeswitch-users] Grandstream 4104 SOLVED In-Reply-To: <4F703E66.5070607@elios.net> References: <4F6C46B0.7080805@elios.net> <4F703E66.5070607@elios.net> Message-ID: <4F70698B.2090006@elios.net> Thank's a lot to everyone who tried to help me. I removed the configuration file for the gateway (after all fs doesn't register to the gxw4104) and substitute in my dialplan the line with the line and this seems to work Regards Accursio Avona Il 26/03/2012 12:01, a.avona ha scritto: > Hi, thank's for your answer > i did as you said and if i digit > originate sofia/internal/XXXX at 192.168.0.3 9178 > outgoing calls work well. > > if i try to originate call from a client it doesn't work so i think > the problem is in the default account configuration > here is the siptraces > > > Thank's for any suggestion > reagards > Accursio Avona > > recv 865 bytes from udp/[192.168.0.200]:5060 at 09:31:18.696255: > > ------------------------------------------------------------------------ > > INVITE sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport > > From: "1000" ;tag=645977894 > > To: > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 5 INVITE > > Contact: > > Content-Type: application/sdp > > Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE > > Max-Forwards: 70 > > Supported: 100rel, replaces, from-change > > User-Agent: SIPPER for PhonerLite > > P-Preferred-Identity: > > Content-Length: 260 > > > v=0 > > o=- 162748822 0 IN IP4 192.168.0.200 > > s=SIPPER for PhonerLite > > c=IN IP4 192.168.0.200 > > t=0 0 > > m=audio 5062 RTP/AVP 8 3 0 101 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=sendrecv > > ------------------------------------------------------------------------ > > send 379 bytes to udp/[192.168.0.200]:5060 at 09:31:18.696482: > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport=5060 > > From: "1000" ;tag=645977894 > > To: > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 5 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 > 15-58-48 -0600 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:7559 IP 192.168.0.200 > Rejected by acl "domains". Falling back to Digest auth. > > send 865 bytes to udp/[192.168.0.200]:5060 at 09:31:18.697335: > > ------------------------------------------------------------------------ > > SIP/2.0 407 Proxy Authentication Required > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport=5060 > > From: "1000" ;tag=645977894 > > To: ;tag=pvaSD4jQ22N9D > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 5 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 > 15-58-48 -0600 > > Accept: application/sdp > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, > sla, include-session-description, presence.winfo, message-summary, refer > > Proxy-Authenticate: Digest realm="192.168.0.2", > nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", algorithm=MD5, qop="auth" > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > recv 346 bytes from udp/[192.168.0.200]:5060 at 09:31:18.698468: > > ------------------------------------------------------------------------ > > ACK sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195f9001e68586d6b;rport > > From: "1000" ;tag=645977894 > > To: ;tag=pvaSD4jQ22N9D > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 5 ACK > > Max-Forwards: 70 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > recv 1137 bytes from udp/[192.168.0.200]:5060 at 09:31:18.699228: > > ------------------------------------------------------------------------ > > INVITE sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport > > From: "1000" ;tag=645977894 > > To: > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 6 INVITE > > Contact: > > Proxy-Authorization: Digest username="1000", realm="192.168.0.2", > nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", > uri="sip:339XXXXXXXX at 192.168.0.2", > response="7378107bc277149e3b6ef00c1c766a71", algorithm=MD5, > cnonce="234abcc436e2667097e7fe6eia53e8dd", qop=auth, nc=00000001 > > Content-Type: application/sdp > > Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE > > Max-Forwards: 70 > > Supported: 100rel, replaces, from-change > > User-Agent: SIPPER for PhonerLite > > P-Preferred-Identity: > > Content-Length: 260 > > > v=0 > > o=- 162748822 0 IN IP4 192.168.0.200 > > s=SIPPER for PhonerLite > > c=IN IP4 192.168.0.200 > > t=0 0 > > m=audio 5062 RTP/AVP 8 3 0 101 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=sendrecv > > ------------------------------------------------------------------------ > > send 379 bytes to udp/[192.168.0.200]:5060 at 09:31:18.699380: > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport=5060 > > From: "1000" ;tag=645977894 > > To: > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 6 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 > 15-58-48 -0600 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:7559 IP 192.168.0.200 > Rejected by acl "domains". Falling back to Digest auth. > > 2012-03-26 11:31:18.692558 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/1000 at 192.168.0.2 [710da826-7726-11e1-bf72-e50b8b76e101] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5526 Channel > sofia/internal/1000 at 192.168.0.2 entering state [received][100] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5537 Remote SDP: > > v=0 > > o=- 162748822 0 IN IP4 192.168.0.200 > > s=SIPPER for PhonerLite > > c=IN IP4 192.168.0.200 > > t=0 0 > > m=audio 5062 RTP/AVP 8 3 0 101 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4874 Audio Codec > Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:2991 Set Codec > sofia/internal/1000 at 192.168.0.2 PCMA/8000 20 ms 160 samples 64000 bits > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_codec.c:111 > sofia/internal/1000 at 192.168.0.2 Original read codec set to PCMA:8 > > 2012-03-26 11:31:18.692558 [DEBUG] sofia_glue.c:4995 Set 2833 dtmf > send/recv payload to 101 > > 2012-03-26 11:31:18.692558 [DEBUG] sofia.c:5749 > (sofia/internal/1000 at 192.168.0.2) State Change CS_NEW -> CS_INIT > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_INIT > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1000 at 192.168.0.2) State INIT > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:85 > sofia/internal/1000 at 192.168.0.2 SOFIA INIT > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:125 > (sofia/internal/1000 at 192.168.0.2) State Change CS_INIT -> CS_ROUTING > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1000 at 192.168.0.2) State INIT going to sleep > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_ROUTING > > 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1886 > (sofia/internal/1000 at 192.168.0.2) Callstate Change DOWN -> RINGING > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1000 at 192.168.0.2) State ROUTING > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:148 > sofia/internal/1000 at 192.168.0.2 SOFIA ROUTING > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/1000 at 192.168.0.2 Standard ROUTING > > 2012-03-26 11:31:18.692558 [INFO] mod_dialplan_xml.c:485 Processing > 1000 <1000>->339XXXXXXXX in context default > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unloop] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->tod_example] continue=true > > Dialplan: sofia/internal/1000 at 192.168.0.2 Date/Time Match (PASS) > [tod_example] break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(open=true) > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->holiday_example] continue=true > > Dialplan: sofia/internal/1000 at 192.168.0.2 Date/TimeMatch (FAIL) > [holiday_example] break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->global-intercept] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [global-intercept] destination_number(339XXXXXXXX) =~ /^886$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group-intercept] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [group-intercept] destination_number(339XXXXXXXX) =~ /^\*8$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->intercept-ext] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [intercept-ext] > destination_number(339XXXXXXXX) =~ /^\*\*(\d+)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->redial] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [redial] > destination_number(339XXXXXXXX) =~ /^(redial|870)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->global] > continue=true > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] > ${call_debug}(false) =~ /^true$/ break=never > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [global] > ${sip_has_crypto}() =~ > /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never > > Dialplan: sofia/internal/1000 at 192.168.0.2 Absolute Condition [global] > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->snom-demo-2] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-2] > destination_number(339XXXXXXXX) =~ /^9001$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->snom-demo-1] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [snom-demo-1] > destination_number(339XXXXXXXX) =~ /^9000$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] > destination_number(339XXXXXXXX) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->eavesdrop] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [eavesdrop] > destination_number(339XXXXXXXX) =~ /^779$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->call_return] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [call_return] > destination_number(339XXXXXXXX) =~ /^\*69$|^869$|^lcr$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->del-group] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [del-group] > destination_number(339XXXXXXXX) =~ /^80(\d{2})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->add-group] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [add-group] > destination_number(339XXXXXXXX) =~ /^81(\d{2})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->call-group-simo] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [call-group-simo] destination_number(339XXXXXXXX) =~ /^82(\d{2})$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->call-group-order] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [call-group-order] destination_number(339XXXXXXXX) =~ /^83(\d{2})$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->extension-intercom] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [extension-intercom] destination_number(339XXXXXXXX) =~ > /^8(10[01][0-9])$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->Local_Extension] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [Local_Extension] destination_number(339XXXXXXXX) =~ /^(10[01][0-9])$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->Local_Extension_Skinny] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [Local_Extension_Skinny] destination_number(339XXXXXXXX) =~ > /^(11[01][0-9])$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group_dial_sales] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [group_dial_sales] destination_number(339XXXXXXXX) =~ /^2000$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group_dial_support] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [group_dial_support] destination_number(339XXXXXXXX) =~ /^2001$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->group_dial_billing] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [group_dial_billing] destination_number(339XXXXXXXX) =~ /^2002$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->operator] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [operator] > destination_number(339XXXXXXXX) =~ /^(operator|0)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->vmain] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [vmain] > destination_number(339XXXXXXXX) =~ /^vmain$|^4000$|^\*98$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->sip_uri] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [sip_uri] > destination_number(339XXXXXXXX) =~ /^sip:(.*)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->nb_conferences] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [nb_conferences] destination_number(339XXXXXXXX) =~ /^(30\d{2})$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->wb_conferences] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [wb_conferences] destination_number(339XXXXXXXX) =~ /^(31\d{2})$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->uwb_conferences] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [uwb_conferences] destination_number(339XXXXXXXX) =~ /^(32\d{2})$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->cdquality_conferences] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [cdquality_conferences] destination_number(339XXXXXXXX) =~ > /^(33\d{2})$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->freeswitch_public_conf_via_sip] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [freeswitch_public_conf_via_sip] destination_number(339XXXXXXXX) =~ > /^9(888|8888|1616|3232)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->mad_boss_intercom] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [mad_boss_intercom] destination_number(339XXXXXXXX) =~ /^0911$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->mad_boss_intercom] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [mad_boss_intercom] destination_number(339XXXXXXXX) =~ /^0912$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->mad_boss] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [mad_boss] > destination_number(339XXXXXXXX) =~ /^0913$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ivr_demo] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ivr_demo] > destination_number(339XXXXXXXX) =~ /^5000$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->dynamic_conference] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [dynamic_conference] destination_number(339XXXXXXXX) =~ /^5001$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->rtp_multicast_page] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [rtp_multicast_page] destination_number(339XXXXXXXX) =~ > /^pagegroup$|^7243$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] > destination_number(339XXXXXXXX) =~ /^5900$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] > destination_number(339XXXXXXXX) =~ /^5901$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->valet_park] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] > destination_number(339XXXXXXXX) =~ /^(6000)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->valet_park] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [valet_park] > destination_number(339XXXXXXXX) =~ /^(60\d[1-9])$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] > destination_number(339XXXXXXXX) =~ /park\+(\d+)/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] > destination_number(339XXXXXXXX) =~ /^parking$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->park] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [park] > source(mod_sofia) =~ /mod_sofia/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [park] > destination_number(339XXXXXXXX) =~ /callpark/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->unpark] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) [unpark] > source(mod_sofia) =~ /mod_sofia/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [unpark] > destination_number(339XXXXXXXX) =~ /pickup/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->wait] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [wait] > destination_number(339XXXXXXXX) =~ /^wait$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->fax_receive] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_receive] > destination_number(339XXXXXXXX) =~ /^9178$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->fax_transmit] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [fax_transmit] > destination_number(339XXXXXXXX) =~ /^9179$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_180] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ringback_180] > destination_number(339XXXXXXXX) =~ /^9180$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_183_uk_ring] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_183_uk_ring] destination_number(339XXXXXXXX) =~ /^9181$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_183_music_ring] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_183_music_ring] destination_number(339XXXXXXXX) =~ /^9182$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_post_answer_uk_ring] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_post_answer_uk_ring] destination_number(339XXXXXXXX) =~ > /^9183$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->ringback_post_answer_music] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [ringback_post_answer_music] destination_number(339XXXXXXXX) =~ > /^9184$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->ClueCon] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [ClueCon] > destination_number(339XXXXXXXX) =~ /^9191$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->show_info] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [show_info] > destination_number(339XXXXXXXX) =~ /^9192$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->video_record] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [video_record] > destination_number(339XXXXXXXX) =~ /^9193$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->video_playback] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [video_playback] destination_number(339XXXXXXXX) =~ /^9194$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->delay_echo] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [delay_echo] > destination_number(339XXXXXXXX) =~ /^9195$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->echo] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [echo] > destination_number(339XXXXXXXX) =~ /^9196$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->milliwatt] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [milliwatt] > destination_number(339XXXXXXXX) =~ /^9197$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->tone_stream] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [tone_stream] > destination_number(339XXXXXXXX) =~ /^9198$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->zrtp_enrollement] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) > [zrtp_enrollement] destination_number(339XXXXXXXX) =~ /^9787$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->hold_music] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [hold_music] > destination_number(339XXXXXXXX) =~ /^9664$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->from_pstn] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [from_pstn] > destination_number(339XXXXXXXX) =~ /^0000$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing [default->101] > continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [101] > destination_number(339XXXXXXXX) =~ /^101$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->pizza_demo] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (FAIL) [pizza_demo] > destination_number(339XXXXXXXX) =~ /^(pizza|74992)$/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 parsing > [default->gxw4104-fxo-local] continue=false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) > [gxw4104-fxo-local] ${toll_allow}(domestic,international,local) =~ > /local/ break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Regex (PASS) > [gxw4104-fxo-local] destination_number(339XXXXXXXX) =~ /^(\d{6,})$/ > break=on-false > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > set(effective_caller_id_number=0321234567) > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > set(effective_caller_id_name=ThisIsMyCompany) > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > set(ignore_early_media=ring_ready) > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action set(ringback=${us-ring}) > > Dialplan: sofia/internal/1000 at 192.168.0.2 Action > bridge(sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/1000 at 192.168.0.2) State Change CS_ROUTING -> CS_EXECUTE > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1000 at 192.168.0.2) State ROUTING going to sleep > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_EXECUTE > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/1000 at 192.168.0.2) State EXECUTE > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:241 > sofia/internal/1000 at 192.168.0.2 SOFIA EXECUTE > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/1000 at 192.168.0.2 Standard EXECUTE > > EXECUTE sofia/internal/1000 at 192.168.0.2 set(open=true) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET [open]=[true] > > EXECUTE sofia/internal/1000 at 192.168.0.2 > hash(insert/192.168.0.2-spymap/1000/710da826-7726-11e1-bf72-e50b8b76e101) > > EXECUTE sofia/internal/1000 at 192.168.0.2 > hash(insert/192.168.0.2-last_dial/1000/339XXXXXXXX) > > EXECUTE sofia/internal/1000 at 192.168.0.2 > hash(insert/192.168.0.2-last_dial/global/710da826-7726-11e1-bf72-e50b8b76e101) > > EXECUTE sofia/internal/1000 at 192.168.0.2 export(RFC2822_DATE=Mon, 26 > Mar 2012 11:31:18 +0200) > > 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1093 EXPORT > (export_vars) [RFC2822_DATE]=[Mon, 26 Mar 2012 11:31:18 +0200] > > EXECUTE sofia/internal/1000 at 192.168.0.2 > set(effective_caller_id_number=0321234567) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET > [effective_caller_id_number]=[0321234567] > > EXECUTE sofia/internal/1000 at 192.168.0.2 > set(effective_caller_id_name=ThisIsMyCompany) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET > [effective_caller_id_name]=[ThisIsMyCompany] > > EXECUTE sofia/internal/1000 at 192.168.0.2 set(ignore_early_media=ring_ready) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET [ignore_early_media]=[ring_ready] > > EXECUTE sofia/internal/1000 at 192.168.0.2 set(ringback=%(2000,4000,440,480)) > > 2012-03-26 11:31:18.692558 [DEBUG] mod_dptools.c:1281 > sofia/internal/1000 at 192.168.0.2 SET [ringback]=[%(2000,4000,440,480)] > > EXECUTE sofia/internal/1000 at 192.168.0.2 > bridge(sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) > > 2012-03-26 11:31:18.692558 [DEBUG] switch_channel.c:1047 > sofia/internal/1000 at 192.168.0.2 EXPORTING[export_vars] > [RFC2822_DATE]=[Mon, 26 Mar 2012 11:31:18 +0200] to event > > 2012-03-26 11:31:18.692558 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > > 2012-03-26 11:31:18.692558 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > [710f6738-7726-11e1-bf77-e50b8b76e101] > > 2012-03-26 11:31:18.692558 [DEBUG] mod_sofia.c:4691 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State > Change CS_NEW -> CS_INIT > > 2012-03-26 11:31:18.692558 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running > State Change CS_INIT > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State INIT > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:85 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA INIT > > send 1279 bytes to udp/[192.168.0.3]:5060 at 09:31:18.713245: > > ------------------------------------------------------------------------ > > INVITE sip:gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K > > Max-Forwards: 69 > > From: "ThisIsMyCompany" ;tag=reXaHtmyvm2eN > > To: > > Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a > > CSeq: 26041075 INVITE > > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 > 15-58-48 -0600 > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, > sla, include-session-description, presence.winfo, message-summary, refer > > Privacy: none > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 311 > > X-FS-Support: update_display,send_info > > P-Asserted-Identity: "ThisIsMyCompany" > > > v=0 > > o=FreeSWITCH 1332736038 1332736039 IN IP4 192.168.0.2 > > s=FreeSWITCH > > c=IN IP4 192.168.0.2 > > t=0 0 > > m=audio 18240 RTP/AVP 8 98 99 9 0 3 101 13 > > a=rtpmap:98 G7221/32000 > > a=fmtp:98 bitrate=48000 > > a=rtpmap:99 G7221/16000 > > a=fmtp:99 bitrate=32000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=ptime:20 > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:125 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State > Change CS_INIT -> CS_ROUTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State INIT > going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running > State Change CS_ROUTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:1886 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate > Change DOWN -> RINGING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State ROUTING > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:148 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA ROUTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State > Change CS_ROUTING -> CS_CONSUME_MEDIA > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State > ROUTING going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running > State Change CS_CONSUME_MEDIA > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State > CONSUME_MEDIA > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State > CONSUME_MEDIA going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] sofia.c:5526 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 entering > state [calling][0] > > recv 355 bytes from udp/[192.168.0.3]:5060 at 09:31:18.717462: > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K > > From: "ThisIsMyCompany" ;tag=reXaHtmyvm2eN > > To: > > Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a > > CSeq: 26041075 INVITE > > User-Agent: Grandstream GXW4104 (HW 2.0, Ch:5) 1.3.4.9 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > recv 367 bytes from udp/[192.168.0.3]:5060 at 09:31:18.717980: > > ------------------------------------------------------------------------ > > SIP/2.0 403 > > Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K > > From: "ThisIsMyCompany" ;tag=reXaHtmyvm2eN > > To: ;tag=3ga2B2jmSSe6H > > Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a > > CSeq: 26041075 INVITE > > User-Agent: Grandstream GXW4104 (HW 2.0, Ch:8) 1.3.4.9 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > send 370 bytes to udp/[192.168.0.3]:5060 at 09:31:18.718065: > > ------------------------------------------------------------------------ > > ACK sip:gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bK7Ume46yDFSK0K > > Max-Forwards: 69 > > From: "ThisIsMyCompany" ;tag=reXaHtmyvm2eN > > To: ;tag=3ga2B2jmSSe6H > > Call-ID: 486dc943-f1c9-122f-5cb5-0800276bec7a > > CSeq: 26041075 ACK > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:875 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] sofia.c:5526 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 entering > state [terminated][403] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2848 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate > Change RINGING -> HANGUP > > 2012-03-26 11:31:18.712698 [NOTICE] sofia.c:6293 Hangup > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > [CS_CONSUME_MEDIA] [CALL_REJECTED] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [KILL] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running > State Change CS_HANGUP > > 2012-03-26 11:31:18.712698 [DEBUG] switch_ivr_originate.c:3364 > Originate Resulted in Error Cause: 21 [CALL_REJECTED] > > 2012-03-26 11:31:18.712698 [INFO] mod_dptools.c:2922 Originate Failed. > Cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2848 > (sofia/internal/1000 at 192.168.0.2) Callstate Change RINGING -> HANGUP > > 2012-03-26 11:31:18.712698 [NOTICE] mod_dptools.c:3041 Hangup > sofia/internal/1000 at 192.168.0.2 [CS_EXECUTE] [CALL_REJECTED] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/1000 at 192.168.0.2 [KILL] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:2285 > sofia/internal/1000 at 192.168.0.2 skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/1000 at 192.168.0.2) State EXECUTE going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_HANGUP > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1000 at 192.168.0.2) State HANGUP > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:463 > sofia/internal/1000 at 192.168.0.2 Overriding SIP cause 603 with 403 from > the other leg > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/1000 at 192.168.0.2 hanging up, cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:534 Responding to > INVITE with: 403 > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/1000 at 192.168.0.2 Standard HANGUP, cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1000 at 192.168.0.2) State HANGUP going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/1000 at 192.168.0.2) State Change CS_HANGUP -> CS_REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1000 at 192.168.0.2) State REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/1000 at 192.168.0.2 Standard REPORTING, cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1000 at 192.168.0.2) State REPORTING going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/1000 at 192.168.0.2) State Change CS_REPORTING -> CS_DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/1000 at 192.168.0.2 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1380 Session > 13 (sofia/internal/1000 at 192.168.0.2) Locked, Waiting on external entities > > 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1398 Session > 13 (sofia/internal/1000 at 192.168.0.2) Ended > > 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/1000 at 192.168.0.2 [CS_DESTROY] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/1000 at 192.168.0.2) Callstate Change HANGUP -> DOWN > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/1000 at 192.168.0.2) Running State Change CS_DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1000 at 192.168.0.2) State DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:374 > sofia/internal/1000 at 192.168.0.2 SOFIA DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/1000 at 192.168.0.2 Standard DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1000 at 192.168.0.2) State DESTROY going to sleep > > send 833 bytes to udp/[192.168.0.200]:5060 at 09:31:18.720584: > > ------------------------------------------------------------------------ > > SIP/2.0 403 Forbidden > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport=5060 > > From: "1000" ;tag=645977894 > > To: ;tag=Q53HFZ3tZBcvS > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 6 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1aa9103 2012-03-01 > 15-58-48 -0600 > > Accept: application/sdp > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, > sla, include-session-description, presence.winfo, message-summary, refer > > Reason: Q.850;cause=21;text="CALL_REJECTED" > > Content-Length: 0 > > P-Asserted-Identity: "339XXXXXXXX" > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State HANGUP > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:469 Channel > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 hanging up, > cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard > HANGUP, cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State > HANGUP going to sleep > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State > Change CS_HANGUP -> CS_REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running > State Change CS_REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State REPORTING > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard > REPORTING, cause: CALL_REJECTED > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State > REPORTING going to sleep > > recv 618 bytes from udp/[192.168.0.200]:5060 at 09:31:18.721757: > > ------------------------------------------------------------------------ > > ACK sip:339XXXXXXXX at 192.168.0.2 SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.0.200:5060;branch=z9hG4bK00be621f9475e11195fa001e68586d6b;rport > > From: "1000" ;tag=645977894 > > To: ;tag=Q53HFZ3tZBcvS > > Call-ID: 00BE621F-9475-E111-95F8-001E68586D6B at 192.168.0.200 > > CSeq: 6 ACK > > Proxy-Authorization: Digest username="1000", realm="192.168.0.2", > nonce="710d05a6-7726-11e1-bf71-e50b8b76e101", > uri="sip:339XXXXXXXX at 192.168.0.2", > response="7378107bc277149e3b6ef00c1c766a71", algorithm=MD5, > cnonce="234abcc436e2667097e7fe6eia53e8dd", qop=auth, nc=00000001 > > Max-Forwards: 70 > > Content-Length: 0 > > > ------------------------------------------------------------------------ > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State > Change CS_REPORTING -> CS_DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 [BREAK] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_session.c:1380 Session > 14 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Locked, > Waiting on external entities > > 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1398 Session > 14 (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Ended > > 2012-03-26 11:31:18.712698 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 > [CS_DESTROY] > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Callstate > Change HANGUP -> DOWN > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) Running > State Change CS_DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] mod_sofia.c:374 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 SOFIA DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060 Standard DESTROY > > 2012-03-26 11:31:18.712698 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/gxw4104-fxo1/339XXXXXXXX at 192.168.0.3:5060) State > DESTROY going to sleep > > freeswitch at internal> > > > > > Il 25/03/2012 06:39, Anton Kvashenkin ha scritto: >> originate sofia/internal/XXXX at 192.168.0.3 9178 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/6edd13da/attachment-0001.html From nasida at live.ru Mon Mar 26 17:10:16 2012 From: nasida at live.ru (Yuriy Nasida) Date: Mon, 26 Mar 2012 17:10:16 +0400 Subject: [Freeswitch-users] VoiceChanger for FS Message-ID: Hello guys, If anybody know any similar implementation of VoiceChanger (or something like this) for FS ? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/a54eba2e/attachment.html From avi at avimarcus.net Mon Mar 26 17:17:08 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 26 Mar 2012 15:17:08 +0200 Subject: [Freeswitch-users] VoiceChanger for FS In-Reply-To: References: Message-ID: Take a look at: http://wiki.freeswitch.org/wiki/Mod_soundtouch Not sure how well maintained this is, though. -Avi 2012/3/26 Yuriy Nasida > Hello guys, > > If anybody know any similar implementation of VoiceChanger (or something > like this) for FS ? > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/77239e61/attachment.html From andrew at cassidywebservices.co.uk Mon Mar 26 17:22:19 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 26 Mar 2012 14:22:19 +0100 Subject: [Freeswitch-users] mod_callcenter field 'system' Message-ID: Hi guys, Just been looking at my agents table for mod_callcenter. There is a field called 'system' which has been set to 'single_box'. Anyone know what this field is for? -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/8dcea740/attachment.html From tpe at actimizer.com Mon Mar 26 17:40:14 2012 From: tpe at actimizer.com (Tor Petterson) Date: Mon, 26 Mar 2012 15:40:14 +0200 Subject: [Freeswitch-users] Large delay/latency when bridging SIP calls In-Reply-To: References: Message-ID: 2012/3/25 Gabriel Gunderson : > I think it's fair to say that we're going to need logs before we can > help. SIP traces etc. I have uploaded a log file and a SIP trace to pastebin: http://pastebin.freeswitch.org/18740 and http://pastebin.freeswitch.org/18741 I have found that if I comment out the else statement in switch_rtp.c line 3105 to 3107 the problem goes away. -- Tor Petterson tpe at actimizer.com Tobaksvejen 25, 2. tv. - 2860 S?borg Telephone: +45 39 55 05 32 www.actimizer.com From wstephen80 at gmail.com Mon Mar 26 17:52:06 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Mon, 26 Mar 2012 15:52:06 +0200 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: Message-ID: Can this issue affect the voice quality? On Fri, Mar 23, 2012 at 7:46 PM, Stephen Wilde wrote: > Real hardware, a dedicate server with 2 Xeon X5670 (a total 24 core each > one with 12Mb cache at 2.93GHz) that is running at 20% - 25% of load. OS is > CentOS 5.7 64bit and FS is (git-0626c89 2012-02-29 14-45-39 -0600) > > > On Fri, Mar 23, 2012 at 6:33 PM, Brian Foster wrote: > >> Is this on virtualized or real hardware? >> >> -BDF >> >> On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde wrote: >> >>> I have run a "timer_test" in a dedicated FS server and I see strange >>> result: it's normal? >>> >>> Stephen >>> >>> >>> freeswitch at internal> timer_test 20 40 >>> Avg: 19.866ms Total Time: 795.880ms >>> >>> 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: >>> samplecount after init: 1 >>> 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: >>> samplecount after first step: 2 >>> 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 >>> sleep 20 19568 >>> 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 >>> sleep 20 38231 >>> 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 >>> sleep 20 18847 >>> 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 >>> sleep 20 13982 >>> 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 >>> sleep 20 34793 >>> 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 >>> sleep 20 2 >>> 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 >>> sleep 20 23166 >>> 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 >>> sleep 20 16957 >>> 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 >>> sleep 20 17643 >>> 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 >>> sleep 20 18786 >>> 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 >>> sleep 20 25100 >>> 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 >>> sleep 20 18552 >>> 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 >>> sleep 20 18815 >>> 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 >>> sleep 20 19464 >>> 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 >>> sleep 20 22012 >>> 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 >>> sleep 20 13980 >>> 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 >>> sleep 20 19065 >>> 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 >>> sleep 20 39585 >>> 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 >>> sleep 20 2 >>> 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 >>> sleep 20 26255 >>> 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 >>> sleep 20 17872 >>> 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 >>> sleep 20 32191 >>> 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 >>> sleep 20 22634 >>> 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 >>> sleep 20 15483 >>> 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 >>> sleep 20 22813 >>> 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 >>> sleep 20 17099 >>> 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 >>> sleep 20 1 >>> 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 >>> sleep 20 29108 >>> 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 >>> sleep 20 11492 >>> 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 >>> sleep 20 20855 >>> 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 >>> sleep 20 32579 >>> 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 >>> sleep 20 18173 >>> 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 >>> sleep 20 22666 >>> 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 >>> sleep 20 23792 >>> 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 >>> sleep 20 26158 >>> 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 >>> sleep 20 13080 >>> 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 >>> sleep 20 24609 >>> 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 >>> sleep 20 1 >>> 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 >>> sleep 20 19413 >>> 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 >>> sleep 20 19820 >>> freeswitch at internal> status >>> UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, >>> 696 microseconds >>> FreeSWITCH is ready >>> 955611 session(s) since startup >>> 2192 session(s) 0/50 >>> 6000 session(s) max >>> min idle cpu 0.00/74.00 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/60bbc778/attachment-0001.html From me at nevian.org Mon Mar 26 17:57:48 2012 From: me at nevian.org (Serge S. Yuriev) Date: Mon, 26 Mar 2012 17:57:48 +0400 Subject: [Freeswitch-users] Strange gw behavior In-Reply-To: References: <531341332008393@web134.yandex.ru> Message-ID: <842241332770268@web97.yandex.ru> Hi, Thank you for attention! 25.03.2012, 06:41, "Gabriel Gunderson" : > On Sat, Mar 17, 2012 at 12:19 PM, Serge S. Yuriev wrote: > >> ?When I register GW 'multifon' it breaks all outside calling - this GW is never used in DP as outbound BUT all calls trying to flow trough it with attributes from other GWs! > > Please take a little more time to explain what you're seeing. Try to > be as clear as you can. We all want to help, but it's hard to take the > time needed to find the problem in 800+ lines of logs when we don't > really know what we're looking for. Sorry for my bad language, I'll try rephrase in simple steps Scenaio 1, "Multifon" GW disabled All works as expected 1. Client INVITE 2. Server TRYING 3. Server does LCR 4. Server INVITE to selected GW (from 3) Scenaio 2, "Multifon" GW enabled Call fails 1. Client INVITE 2. Server TRYING 3. Server does LCR 4. Server INVITE to Multifon GW (which is NOT in LCR) instead real GW but uses parameters from chosen > Also, since this is about GWs, consider posting your sofia configs and > perhaps the dialplan snipped used to hit these GWs. GW config http://pastebin.freeswitch.org/18738 Dialplan snippet http://pastebin.freeswitch.org/18739 -- wbr, Serge From peter.olsson at visionutveckling.se Mon Mar 26 18:08:42 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 26 Mar 2012 14:08:42 +0000 Subject: [Freeswitch-users] Strange "timer_test" result Message-ID: <1FFF97C269757C458224B7C895F35F1507ECDA@cantor.std.visionutv.se> Yes, most likely. It's very strange though, that a machine like this gives so poor timing results. The first thing I would do is to upgrade to latest GIT HEAD. A patch was commited about two weeks ago, that does the calculation for timer_test more properly (so time for logging is not calculated). Have you tried to do the same test with no load at all? /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen Wilde Skickat: den 26 mars 2012 15:52 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Strange "timer_test" result Can this issue affect the voice quality? On Fri, Mar 23, 2012 at 7:46 PM, Stephen Wilde > wrote: Real hardware, a dedicate server with 2 Xeon X5670 (a total 24 core each one with 12Mb cache at 2.93GHz) that is running at 20% - 25% of load. OS is CentOS 5.7 64bit and FS is (git-0626c89 2012-02-29 14-45-39 -0600) On Fri, Mar 23, 2012 at 6:33 PM, Brian Foster > wrote: Is this on virtualized or real hardware? -BDF On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde > wrote: I have run a "timer_test" in a dedicated FS server and I see strange result: it's normal? Stephen freeswitch at internal> timer_test 20 40 Avg: 19.866ms Total Time: 795.880ms 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: samplecount after init: 1 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: samplecount after first step: 2 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 sleep 20 19568 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 sleep 20 38231 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 sleep 20 18847 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 sleep 20 13982 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 sleep 20 34793 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 sleep 20 2 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 sleep 20 23166 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 sleep 20 16957 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 sleep 20 17643 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 sleep 20 18786 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 sleep 20 25100 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 sleep 20 18552 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 sleep 20 18815 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 sleep 20 19464 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 sleep 20 22012 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 sleep 20 13980 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 sleep 20 19065 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 sleep 20 39585 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 sleep 20 2 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 sleep 20 26255 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 sleep 20 17872 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 sleep 20 32191 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 sleep 20 22634 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 sleep 20 15483 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 sleep 20 22813 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 sleep 20 17099 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 sleep 20 1 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 sleep 20 29108 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 sleep 20 11492 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 sleep 20 20855 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 sleep 20 32579 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 sleep 20 18173 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 sleep 20 22666 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 sleep 20 23792 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 sleep 20 26158 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 sleep 20 13080 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 sleep 20 24609 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 sleep 20 1 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 sleep 20 19413 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 sleep 20 19820 freeswitch at internal> status UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, 696 microseconds FreeSWITCH is ready 955611 session(s) since startup 2192 session(s) 0/50 6000 session(s) max min idle cpu 0.00/74.00 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f7073a332761138918569! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/0866e7c6/attachment-0001.html From wstephen80 at gmail.com Mon Mar 26 18:19:45 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Mon, 26 Mar 2012 16:19:45 +0200 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: <1FFF97C269757C458224B7C895F35F1507ECDA@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1507ECDA@cantor.std.visionutv.se> Message-ID: Thank you Peter for your reply! I have already tried the upgrade to latest git, now I'm on FreeSWITCH Version 1.1.beta1 (git-c31a799 2012-03-24 14-11-49 -0700) and the result is the same. I have tried with few load (during night) and in this case the timer_test is perfect. Any suggestion? On Mon, Mar 26, 2012 at 4:08 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Yes, most likely.**** > > ** ** > > It?s very strange though, that a machine like this gives so poor timing > results.**** > > ** ** > > The first thing I would do is to upgrade to latest GIT HEAD. A patch was > commited about two weeks ago, that does the calculation for timer_test more > properly (so time for logging is not calculated).**** > > ** ** > > Have you tried to do the same test with no load at all?**** > > ** ** > > /Peter**** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Stephen Wilde > *Skickat:* den 26 mars 2012 15:52 > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] Strange "timer_test" result**** > > ** ** > > Can this issue affect the voice quality?**** > > ** ** > > On Fri, Mar 23, 2012 at 7:46 PM, Stephen Wilde > wrote:**** > > Real hardware, a dedicate server with 2 Xeon X5670 (a total 24 core each > one with 12Mb cache at 2.93GHz) that is running at 20% - 25% of load. OS is > CentOS 5.7 64bit and FS is (git-0626c89 2012-02-29 14-45-39 -0600)**** > > ** ** > > On Fri, Mar 23, 2012 at 6:33 PM, Brian Foster > wrote:**** > > Is this on virtualized or real hardware?**** > > ** ** > > -BDF**** > > On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde > wrote:**** > > I have run a "timer_test" in a dedicated FS server and I see strange > result: it's normal?**** > > ** ** > > Stephen**** > > ** ** > > ** ** > > freeswitch at internal> timer_test 20 40**** > > Avg: 19.866ms Total Time: 795.880ms**** > > ** ** > > 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: > samplecount after init: 1**** > > 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: > samplecount after first step: 2**** > > 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 > sleep 20 19568**** > > 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 > sleep 20 38231**** > > 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 > sleep 20 18847**** > > 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 > sleep 20 13982**** > > 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 > sleep 20 34793**** > > 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 > sleep 20 2**** > > 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 > sleep 20 23166**** > > 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 > sleep 20 16957**** > > 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 > sleep 20 17643**** > > 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 > sleep 20 18786**** > > 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 > sleep 20 25100**** > > 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 > sleep 20 18552**** > > 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 > sleep 20 18815**** > > 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 > sleep 20 19464**** > > 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 > sleep 20 22012**** > > 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 > sleep 20 13980**** > > 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 > sleep 20 19065**** > > 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 > sleep 20 39585**** > > 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 > sleep 20 2**** > > 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 > sleep 20 26255**** > > 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 > sleep 20 17872**** > > 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 > sleep 20 32191**** > > 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 > sleep 20 22634**** > > 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 > sleep 20 15483**** > > 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 > sleep 20 22813**** > > 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 > sleep 20 17099**** > > 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 > sleep 20 1**** > > 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 > sleep 20 29108**** > > 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 > sleep 20 11492**** > > 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 > sleep 20 20855**** > > 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 > sleep 20 32579**** > > 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 > sleep 20 18173**** > > 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 > sleep 20 22666**** > > 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 > sleep 20 23792**** > > 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 > sleep 20 26158**** > > 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 > sleep 20 13080**** > > 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 > sleep 20 24609**** > > 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 > sleep 20 1**** > > 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 > sleep 20 19413**** > > 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 > sleep 20 19820**** > > freeswitch at internal> status**** > > UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, > 696 microseconds**** > > FreeSWITCH is ready**** > > 955611 session(s) since startup**** > > 2192 session(s) 0/50**** > > 6000 session(s) max**** > > min idle cpu 0.00/74.00**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version.**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > ** ** > > !DSPAM:4f7073a332761138918569! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/6e4250c1/attachment.html From peter.olsson at visionutveckling.se Mon Mar 26 18:31:55 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 26 Mar 2012 14:31:55 +0000 Subject: [Freeswitch-users] Strange "timer_test" result Message-ID: <1FFF97C269757C458224B7C895F35F1507ED07@cantor.std.visionutv.se> You could try to enable the "old" 1 ms-timer. It will be a little less efficient (a little more CPU load), and I'm not sure it helps, but it's worth a try. In switch.conf.xml (under autoload_configs), add this: - within the settings-tags. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen Wilde Skickat: den 26 mars 2012 16:20 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Strange "timer_test" result Thank you Peter for your reply! I have already tried the upgrade to latest git, now I'm on FreeSWITCH Version 1.1.beta1 (git-c31a799 2012-03-24 14-11-49 -0700) and the result is the same. I have tried with few load (during night) and in this case the timer_test is perfect. Any suggestion? On Mon, Mar 26, 2012 at 4:08 PM, Peter Olsson > wrote: Yes, most likely. It's very strange though, that a machine like this gives so poor timing results. The first thing I would do is to upgrade to latest GIT HEAD. A patch was commited about two weeks ago, that does the calculation for timer_test more properly (so time for logging is not calculated). Have you tried to do the same test with no load at all? /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen Wilde Skickat: den 26 mars 2012 15:52 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Strange "timer_test" result Can this issue affect the voice quality? On Fri, Mar 23, 2012 at 7:46 PM, Stephen Wilde > wrote: Real hardware, a dedicate server with 2 Xeon X5670 (a total 24 core each one with 12Mb cache at 2.93GHz) that is running at 20% - 25% of load. OS is CentOS 5.7 64bit and FS is (git-0626c89 2012-02-29 14-45-39 -0600) On Fri, Mar 23, 2012 at 6:33 PM, Brian Foster > wrote: Is this on virtualized or real hardware? -BDF On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde > wrote: I have run a "timer_test" in a dedicated FS server and I see strange result: it's normal? Stephen freeswitch at internal> timer_test 20 40 Avg: 19.866ms Total Time: 795.880ms 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: samplecount after init: 1 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: samplecount after first step: 2 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 sleep 20 19568 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 sleep 20 38231 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 sleep 20 18847 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 sleep 20 13982 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 sleep 20 34793 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 sleep 20 2 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 sleep 20 23166 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 sleep 20 16957 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 sleep 20 17643 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 sleep 20 18786 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 sleep 20 25100 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 sleep 20 18552 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 sleep 20 18815 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 sleep 20 19464 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 sleep 20 22012 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 sleep 20 13980 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 sleep 20 19065 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 sleep 20 39585 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 sleep 20 2 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 sleep 20 26255 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 sleep 20 17872 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 sleep 20 32191 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 sleep 20 22634 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 sleep 20 15483 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 sleep 20 22813 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 sleep 20 17099 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 sleep 20 1 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 sleep 20 29108 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 sleep 20 11492 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 sleep 20 20855 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 sleep 20 32579 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 sleep 20 18173 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 sleep 20 22666 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 sleep 20 23792 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 sleep 20 26158 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 sleep 20 13080 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 sleep 20 24609 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 sleep 20 1 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 sleep 20 19413 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 sleep 20 19820 freeswitch at internal> status UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, 696 microseconds FreeSWITCH is ready 955611 session(s) since startup 2192 session(s) 0/50 6000 session(s) max min idle cpu 0.00/74.00 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f7079c632761636018988! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/86a3559f/attachment-0001.html From mitch.capper at gmail.com Mon Mar 26 18:35:09 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Mon, 26 Mar 2012 07:35:09 -0700 Subject: [Freeswitch-users] VoiceChanger for FS In-Reply-To: References: Message-ID: Actually soundtouch is maintained just fine and is works quite well:) There is also a more playful example with ladspa and mixing it with a generic autotune type example. ~Mitch From olli_aro at yahoo.co.uk Mon Mar 26 12:04:34 2012 From: olli_aro at yahoo.co.uk (Olli Aro) Date: Mon, 26 Mar 2012 09:04:34 +0100 Subject: [Freeswitch-users] Binary download with curl Message-ID: Hi all, Been trying to compile Freeswitch on Windows in order to get in the XML curl module, however it is not looking easy (76 more fatal errors to figure out...) Before I start that task I thought it is always worth asking :) Would anyone happen to have a download link to already compiled Windows 32-bit binary for close to recent version of Freeswitch with the XML curl module? Regards, Olli -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/d8704bc5/attachment.html From faisal.rehman22 at hotmail.com Mon Mar 26 14:40:34 2012 From: faisal.rehman22 at hotmail.com (Faisal Rehman) Date: Mon, 26 Mar 2012 15:40:34 +0500 Subject: [Freeswitch-users] 200 OK should have "Session-Expires: 1800; refresher=uas" Message-ID: Hi Everyone, One of our customers wants to have Require: TimerSession-Expires: 1800;refresher=uas in 200 OK. It seems that enable timer and expires can be set by sofia.xml or profile.xml but how to set refresher. Can anybody explain please? Regards, Faisal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/ed158c06/attachment.html From wstephen80 at gmail.com Mon Mar 26 18:40:15 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Mon, 26 Mar 2012 16:40:15 +0200 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: <1FFF97C269757C458224B7C895F35F1507ED07@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1507ED07@cantor.std.visionutv.se> Message-ID: Ok, I'll try with this parameter, I think I have to do a Freeswitch restart to make this change... Stephen On Mon, Mar 26, 2012 at 4:31 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > You could try to enable the ?old? 1 ms-timer. It will be a little less > efficient (a little more CPU load), and I?m not sure it helps, but it?s > worth a try.**** > > ** ** > > In switch.conf.xml (under autoload_configs), add this: name="1ms-timer" value="true"/> - within the settings-tags.**** > > ** ** > > /Peter**** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Stephen Wilde > *Skickat:* den 26 mars 2012 16:20 > > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] Strange "timer_test" result**** > > ** ** > > Thank you Peter for your reply!**** > > I have already tried the upgrade to latest git, now I'm on FreeSWITCH > Version 1.1.beta1 (git-c31a799 2012-03-24 14-11-49 -0700) and the result is > the same.**** > > I have tried with few load (during night) and in this case the timer_test > is perfect.**** > > Any suggestion?**** > > ** ** > > ** ** > > On Mon, Mar 26, 2012 at 4:08 PM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote:**** > > Yes, most likely.**** > > **** > > It?s very strange though, that a machine like this gives so poor timing > results.**** > > **** > > The first thing I would do is to upgrade to latest GIT HEAD. A patch was > commited about two weeks ago, that does the calculation for timer_test more > properly (so time for logging is not calculated).**** > > **** > > Have you tried to do the same test with no load at all?**** > > **** > > /Peter**** > > **** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Stephen Wilde > *Skickat:* den 26 mars 2012 15:52 > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] Strange "timer_test" result**** > > **** > > Can this issue affect the voice quality?**** > > **** > > On Fri, Mar 23, 2012 at 7:46 PM, Stephen Wilde > wrote:**** > > Real hardware, a dedicate server with 2 Xeon X5670 (a total 24 core each > one with 12Mb cache at 2.93GHz) that is running at 20% - 25% of load. OS is > CentOS 5.7 64bit and FS is (git-0626c89 2012-02-29 14-45-39 -0600)**** > > **** > > On Fri, Mar 23, 2012 at 6:33 PM, Brian Foster > wrote:**** > > Is this on virtualized or real hardware?**** > > **** > > -BDF**** > > On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde > wrote:**** > > I have run a "timer_test" in a dedicated FS server and I see strange > result: it's normal?**** > > **** > > Stephen**** > > **** > > **** > > freeswitch at internal> timer_test 20 40**** > > Avg: 19.866ms Total Time: 795.880ms**** > > **** > > 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: > samplecount after init: 1**** > > 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: > samplecount after first step: 2**** > > 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 > sleep 20 19568**** > > 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 > sleep 20 38231**** > > 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 > sleep 20 18847**** > > 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 > sleep 20 13982**** > > 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 > sleep 20 34793**** > > 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 > sleep 20 2**** > > 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 > sleep 20 23166**** > > 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 > sleep 20 16957**** > > 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 > sleep 20 17643**** > > 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 > sleep 20 18786**** > > 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 > sleep 20 25100**** > > 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 > sleep 20 18552**** > > 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 > sleep 20 18815**** > > 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 > sleep 20 19464**** > > 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 > sleep 20 22012**** > > 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 > sleep 20 13980**** > > 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 > sleep 20 19065**** > > 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 > sleep 20 39585**** > > 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 > sleep 20 2**** > > 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 > sleep 20 26255**** > > 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 > sleep 20 17872**** > > 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 > sleep 20 32191**** > > 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 > sleep 20 22634**** > > 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 > sleep 20 15483**** > > 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 > sleep 20 22813**** > > 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 > sleep 20 17099**** > > 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 > sleep 20 1**** > > 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 > sleep 20 29108**** > > 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 > sleep 20 11492**** > > 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 > sleep 20 20855**** > > 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 > sleep 20 32579**** > > 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 > sleep 20 18173**** > > 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 > sleep 20 22666**** > > 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 > sleep 20 23792**** > > 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 > sleep 20 26158**** > > 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 > sleep 20 13080**** > > 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 > sleep 20 24609**** > > 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 > sleep 20 1**** > > 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 > sleep 20 19413**** > > 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 > sleep 20 19820**** > > freeswitch at internal> status**** > > UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, > 696 microseconds**** > > FreeSWITCH is ready**** > > 955611 session(s) since startup**** > > 2192 session(s) 0/50**** > > 6000 session(s) max**** > > min idle cpu 0.00/74.00**** > > **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > **** > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version.**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > **** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > !DSPAM:4f7079c632761636018988! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/2424113a/attachment-0001.html From msc at freeswitch.org Mon Mar 26 18:42:21 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Mar 2012 07:42:21 -0700 Subject: [Freeswitch-users] VoiceChanger for FS In-Reply-To: References: Message-ID: 2012/3/26 Yuriy Nasida > Hello guys, > > If anybody know any similar implementation of VoiceChanger (or something > like this) for FS ? > > Thanks. > > Like Mitch said, you definitely want to check out mod_ladspa. As the name implies it uses LADPSA which means you'll need to be running FS on Linux. I believe mod_ladspa comes with a config for something like "autotune" or "autotalent" or something like that. In any case, it's a great place to start. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/7fa22b4f/attachment.html From randy.andrade at gmail.com Mon Mar 26 18:55:24 2012 From: randy.andrade at gmail.com (Randy Andrade) Date: Mon, 26 Mar 2012 10:55:24 -0400 Subject: [Freeswitch-users] Is it possible to git pull an older version? Message-ID: Hey all, I know this is a little bit off topic, but I figured I'd ask here, since FreeSWITCH is what I'm "git pull"ing.. I've run into a situation where I'm unable to successfully compile the latest versions of FreeSWITCH, and I'm thinking it's because I've been running an older version (FreeSWITCH Version 1.0.head (git-6f4c4ea 2011-09-20 15-22-09 +0400)) for quite some time. I'm not altogether unhappy with the way it's running, but I also don't like being in the situation I'm currently in, where I'm unable to update my installation, and since I've updated my sources, I'm also unable to compile any additional modules for my existing version. Just wondering if anyone knows how to do a git pull but specify an older date / version and revert your sources back to that. I've done a little bit of digging (and will continue to do so), but so far haven't come up with any syntax that does this successfully. Sorry for asking such a n00b question, but I don't use git regularly, actually I don't use it for anything except FreeSWITCH. BTW: My installation is running on Debian 6.0.4 Linux, installed following all the guides in all the default locations. Randy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/c8aea642/attachment.html From wstephen80 at gmail.com Mon Mar 26 19:02:41 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Mon, 26 Mar 2012 17:02:41 +0200 Subject: [Freeswitch-users] Is it possible to git pull an older version? In-Reply-To: References: Message-ID: If you have done a git pull, you can do git checkout [version] where [version] is the commit hash you want to use (you can specify the first chars, i.e. your current commit hash is 6f4c4ea) On Mon, Mar 26, 2012 at 4:55 PM, Randy Andrade wrote: > Hey all, I know this is a little bit off topic, but I figured I'd ask > here, since FreeSWITCH is what I'm "git pull"ing.. > > I've run into a situation where I'm unable to successfully compile the > latest versions of FreeSWITCH, and I'm thinking it's because I've been > running an older version (FreeSWITCH Version 1.0.head (git-6f4c4ea > 2011-09-20 15-22-09 +0400)) for quite some time. I'm not altogether unhappy > with the way it's running, but I also don't like being in the situation I'm > currently in, where I'm unable to update my installation, and since I've > updated my sources, I'm also unable to compile any additional modules for > my existing version. > > Just wondering if anyone knows how to do a git pull but specify an older > date / version and revert your sources back to that. I've done a little bit > of digging (and will continue to do so), but so far haven't come up with > any syntax that does this successfully. > > Sorry for asking such a n00b question, but I don't use git regularly, > actually I don't use it for anything except FreeSWITCH. > > BTW: My installation is running on Debian 6.0.4 Linux, installed following > all the guides in all the default locations. > > Randy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/95884ae1/attachment.html From mitch.capper at gmail.com Mon Mar 26 19:04:13 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Mon, 26 Mar 2012 08:04:13 -0700 Subject: [Freeswitch-users] Is it possible to git pull an older version? In-Reply-To: References: Message-ID: Yes you can, I have been working on some tips for how to use git bisect to actually track down an exact situation like this. Its not yet complete but ill paste what I have so far: git bisect start [bad_commit] [good_commit] -- [path1] [path2] ... or git bisect start [bad_commit] [HEAD~10] - within the last 10 it broke then compile / test etc/ the tell git: git bisect bad or git bisect good then it will move on to the next git bisect reset when done git bisect skip if you cant test a specific revision Auto have it test each one if you have a script: git bisect run [my_script] [arguments] Note that the script (my_script in the above example) should exit with code 0 if the current source code is good, and exit with a code between 1 and 127 (inclusive), except 125, if the current source code is bad. So for you, you would want to do: git bisect start HEAD HEAD~100 lets say if you knew 100 commits ago was fine. Then try compiling and use git bisect bad if it works or git bisect good if it doesn't. This will track down where it stopped working for you so not only you know but we will have an easy time fixing it (probably) once you can tell us the exact commit. If you want to go back to a specific commit you can use git log to see the commits too rather than just jumping back a fixed number. ~Mitch From anthony.minessale at gmail.com Mon Mar 26 19:13:56 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 26 Mar 2012 10:13:56 -0500 Subject: [Freeswitch-users] 200 OK should have "Session-Expires: 1800; refresher=uas" In-Reply-To: References: Message-ID: We don't currently support setting the refresher it always tries to make the opposite side of the call be the refresher to combat NAT. On Mon, Mar 26, 2012 at 5:40 AM, Faisal Rehman wrote: > Hi Everyone, > > One of our customers wants to have > > Require: Timer > Session-Expires: 1800;refresher=uas > > in?200 OK. > > It seems that enable timer and expires can be set by sofia.xml or > profile.xml but how to set refresher. Can anybody explain please? > > > Regards, > > Faisal > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anthony.minessale at gmail.com Mon Mar 26 19:53:16 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 26 Mar 2012 10:53:16 -0500 Subject: [Freeswitch-users] Large delay/latency when bridging SIP calls In-Reply-To: References: Message-ID: please try HEAD and please report bugs on JIRA not here going forward. On Mon, Mar 26, 2012 at 8:40 AM, Tor Petterson wrote: > 2012/3/25 Gabriel Gunderson : >> I think it's fair to say that we're going to need logs before we can >> help. SIP traces etc. > I have uploaded a log file and a SIP trace to pastebin: > http://pastebin.freeswitch.org/18740 and http://pastebin.freeswitch.org/18741 > > I have found that if I comment out the else statement in switch_rtp.c > line 3105 to 3107 the problem goes away. > > > -- > Tor Petterson > > tpe at actimizer.com > > Tobaksvejen 25, 2. tv. - 2860 S?borg > Telephone: +45 39 55 05 32 > www.actimizer.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From freeswitch-users at digitaldan.com Mon Mar 26 19:56:56 2012 From: freeswitch-users at digitaldan.com (Dan) Date: Mon, 26 Mar 2012 09:56:56 -0600 (MDT) Subject: [Freeswitch-users] Problem getting TALK and NOTALK events In-Reply-To: <64256728-687c-408c-9c98-b3a045864462@radio> Message-ID: Anyone have an idea or should I open a bug for this? Thanks! ----- Original Message ----- From: "Dan" To: "FreeSWITCH Users Help" Sent: Friday, March 16, 2012 7:25:36 AM Subject: Re: [Freeswitch-users] Problem getting TALK and NOTALK events Thanks for the response, I went ahead and subscribed to all events, but am still not getting these events. I have also enabled in my sip profile, not sure if this is redundant or not with the dial plan directives. If VAD is indeed enabled but not working, is there something else I should try? Thanks. ----- Original Message ----- From: "Michael Collins" To: "FreeSWITCH Users Help" Sent: Thursday, March 15, 2012 5:29:21 PM Subject: Re: [Freeswitch-users] Problem getting TALK and NOTALK events Can you confirm if the TALK/NOTALK events are even firing? Try listening to all events and then sifting through the barrage of data to see if those events are present. If they are then you know you just have a filtering issue. If they are not present then you know there's an issue with VAD or something like that. -MC On Thu, Mar 15, 2012 at 8:59 AM, Dan < freeswitch-users at digitaldan.com > wrote: Hi, I'm having some issues getting TALK / NOTALK events to fire on an incoming stream to Freeswitch. In my ESL application I am subscribing to RECORD_START RECORD_STOP TALK NOTALK, Below is the dial plan I am using: I can see that VAD is enabled: 2012-03-15 09:49:25.523832 [DEBUG] switch_rtp.c:4130 Activate VAD codec PCMU 20ms 2012-03-15 09:49:25.523832 [DEBUG] sofia_glue.c:3353 AUDIO RTP Engage VAD for sofia/external/ 3035551212 at 10.10.10.1 ( in out ) In my ESL app I get the RECORD_START and RECORD_STOP but not the talk events. I'm on git version "2c52f23 2012-02-18 08:37:47 -0600", Any ideas? Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/96f0d9fb/attachment-0001.html From wojciech.kochanowski at nnv.pl Mon Mar 26 20:11:42 2012 From: wojciech.kochanowski at nnv.pl (Wojtek Kochanowski) Date: Mon, 26 Mar 2012 18:11:42 +0200 Subject: [Freeswitch-users] mod_cdr doesn't log call details Message-ID: Hello, I'm using mod_json_cdr to get logs from calls, but in certainly situations i don't get the second log. Usually I'm getting 2 logs: inbound and outbound. I'm always getting inbound log, but it's incomplete - for example I don't have information from which channel user have called. This info is in outbound log, but in about 50% calls I don't get it. Same situation with mod_xml. My calling scheme is rather untypical and it looks like: web browser -> rtmp -> freeswitch (freetdm) -> gsm When I'm using just FS to call (eg using Jitsi or other desktop client), everything is ok. FreeSWITCH Version 1.0.head (git-d2c9fb5 2012-02-06 14-12-22 -0600) root at fs# uname -a Linux fs 2.6.32-5-amd64 #1 SMP Mon Jan 16 16:22:28 UTC 2012 x86_64 GNU/Linux root at fs# cat /etc/issue Debian GNU/Linux 6.0 \n \l dialplan: http://pastebin.freeswitch.org/18733 json_cdr.conf: http://pastebin.freeswitch.org/18734 inbound log: http://pastebin.freeswitch.org/18735 example good outbound log: pastebin.freeswitch.org/18736 in "chan_name": "FreeTDM\/1:2\/513710737#" I can see which span and channel has been used. I don't know, is it just my configuration or bug? What else info do you need? Maybe it's my bad in dialplan configuration? Any ideas? Pozdrawiam, Wojciech Kochanowski Junior System Administrator wojciech.kochanowski at nnv.pl NNV Sp. z o.o. ul. Wyszy?skiego 1 10-457 Olsztyn tel. +48 89 533 70 33 fax +48 89 533 03 78 www.nnv.pl S?d Rejonowy dla Miasta Olsztyna KRS 0000310781 NIP 739-369-92-87 REGON 280319486 Kapita? Zak?adowy 1 152 000 z?, wp?acony w ca?o?ci www.firmy.net W trosce o Tw?j biznes www.nieruchomosci-online.pl Najwi?cej aktualnych og?osze? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/5b0f5b2f/attachment.html From luis.daniel.lucio at gmail.com Mon Mar 26 20:45:10 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Mon, 26 Mar 2012 12:45:10 -0400 Subject: [Freeswitch-users] Why does FS is not ussin my libs? Message-ID: Helo, while compiling FS, i realize that for some reason FS is not using my ldns lib, and it triess to use its own. My RPM pagackes for ldns are these: rpm -ql lib64ldns1-1.6.12-1.mga2 /usr/lib64/libldns.so.1 /usr/lib64/libldns.so.1.6.12 /usr/share/doc/lib64ldns1 /usr/share/doc/lib64ldns1/LICENSE /usr/share/doc/lib64ldns1/README rpm -ql lib64ldns-devel-1.6.12-1.mga2 /usr/bin/ldns-config /usr/include/ldns /usr/include/ldns/buffer.h /usr/include/ldns/common.h /usr/include/ldns/dname.h /usr/include/ldns/dnssec.h /usr/include/ldns/dnssec_sign.h /usr/include/ldns/dnssec_verify.h /usr/include/ldns/dnssec_zone.h /usr/include/ldns/error.h /usr/include/ldns/higher.h /usr/include/ldns/host2str.h /usr/include/ldns/host2wire.h /usr/include/ldns/keys.h /usr/include/ldns/ldns.h /usr/include/ldns/net.h /usr/include/ldns/packet.h /usr/include/ldns/parse.h /usr/include/ldns/rbtree.h /usr/include/ldns/rdata.h /usr/include/ldns/resolver.h /usr/include/ldns/rr.h /usr/include/ldns/rr_functions.h /usr/include/ldns/sha1.h /usr/include/ldns/sha2.h /usr/include/ldns/str2host.h /usr/include/ldns/tsig.h /usr/include/ldns/update.h /usr/include/ldns/util.h /usr/include/ldns/wire2host.h /usr/include/ldns/zone.h /usr/lib64/libldns.so /usr/share/doc/lib64ldns-devel /usr/share/doc/lib64ldns-devel/Changelog /usr/share/doc/lib64ldns-devel/README /usr/share/doc/lib64ldns-devel/doc /usr/share/doc/lib64ldns-devel/doc/API-header.xml /usr/share/doc/lib64ldns-devel/doc/API.xml /usr/share/doc/lib64ldns-devel/doc/CodingStyle /usr/share/doc/lib64ldns-devel/doc/TODO /usr/share/doc/lib64ldns-devel/doc/design.dox /usr/share/doc/lib64ldns-devel/doc/dns-lib-implementations /usr/share/doc/lib64ldns-devel/doc/function_manpages /usr/share/doc/lib64ldns-devel/doc/header.html /usr/share/doc/lib64ldns-devel/doc/images /usr/share/doc/lib64ldns-devel/doc/images/LogoInGradientBar2-y100.png /usr/share/doc/lib64ldns-devel/doc/images/libdnsoverview.png /usr/share/doc/lib64ldns-devel/doc/images/libdnsoverview.svg /usr/share/doc/lib64ldns-devel/doc/libdns.css /usr/share/doc/lib64ldns-devel/doc/tutorial1_mx.dox /usr/share/doc/lib64ldns-devel/doc/tutorial2_zone.dox /usr/share/doc/lib64ldns-devel/doc/tutorial3_signzone.dox /usr/share/man/man3/ldns_bget_token.3.xz /usr/share/man/man3/ldns_bgetc.3.xz /usr/share/man/man3/ldns_bskipcs.3.xz /usr/share/man/man3/ldns_buffer.3.xz /usr/share/man/man3/ldns_buffer2pkt_wire.3.xz /usr/share/man/man3/ldns_buffer_at.3.xz /usr/share/man/man3/ldns_buffer_available.3.xz /usr/share/man/man3/ldns_buffer_available_at.3.xz /usr/share/man/man3/ldns_buffer_begin.3.xz /usr/share/man/man3/ldns_buffer_capacity.3.xz /usr/share/man/man3/ldns_buffer_clear.3.xz /usr/share/man/man3/ldns_buffer_current.3.xz /usr/share/man/man3/ldns_buffer_end.3.xz /usr/share/man/man3/ldns_buffer_export.3.xz /usr/share/man/man3/ldns_buffer_flip.3.xz /usr/share/man/man3/ldns_buffer_free.3.xz /usr/share/man/man3/ldns_buffer_limit.3.xz /usr/share/man/man3/ldns_buffer_new.3.xz /usr/share/man/man3/ldns_buffer_new_frm_data.3.xz /usr/share/man/man3/ldns_buffer_position.3.xz /usr/share/man/man3/ldns_buffer_printf.3.xz /usr/share/man/man3/ldns_buffer_read.3.xz /usr/share/man/man3/ldns_buffer_read_at.3.xz /usr/share/man/man3/ldns_buffer_read_u16.3.xz /usr/share/man/man3/ldns_buffer_read_u16_at.3.xz /usr/share/man/man3/ldns_buffer_read_u32.3.xz /usr/share/man/man3/ldns_buffer_read_u32_at.3.xz /usr/share/man/man3/ldns_buffer_read_u8.3.xz /usr/share/man/man3/ldns_buffer_read_u8_at.3.xz /usr/share/man/man3/ldns_buffer_remaining.3.xz /usr/share/man/man3/ldns_buffer_remaining_at.3.xz /usr/share/man/man3/ldns_buffer_reserve.3.xz /usr/share/man/man3/ldns_buffer_rewind.3.xz /usr/share/man/man3/ldns_buffer_set_capacity.3.xz /usr/share/man/man3/ldns_buffer_set_limit.3.xz /usr/share/man/man3/ldns_buffer_set_position.3.xz /usr/share/man/man3/ldns_buffer_skip.3.xz /usr/share/man/man3/ldns_buffer_status.3.xz /usr/share/man/man3/ldns_buffer_status_ok.3.xz /usr/share/man/man3/ldns_buffer_write.3.xz /usr/share/man/man3/ldns_buffer_write_at.3.xz /usr/share/man/man3/ldns_buffer_write_string.3.xz /usr/share/man/man3/ldns_buffer_write_string_at.3.xz /usr/share/man/man3/ldns_buffer_write_u16.3.xz /usr/share/man/man3/ldns_buffer_write_u16_at.3.xz /usr/share/man/man3/ldns_buffer_write_u8.3.xz /usr/share/man/man3/ldns_buffer_write_u8_at.3.xz /usr/share/man/man3/ldns_calc_keytag.3.xz /usr/share/man/man3/ldns_create_nsec.3.xz /usr/share/man/man3/ldns_dname.3.xz /usr/share/man/man3/ldns_dname2canonical.3.xz /usr/share/man/man3/ldns_dname_cat.3.xz /usr/share/man/man3/ldns_dname_cat_clone.3.xz /usr/share/man/man3/ldns_dname_compare.3.xz /usr/share/man/man3/ldns_dname_interval.3.xz /usr/share/man/man3/ldns_dname_is_subdomain.3.xz /usr/share/man/man3/ldns_dname_label.3.xz /usr/share/man/man3/ldns_dname_label_count.3.xz /usr/share/man/man3/ldns_dname_left_chop.3.xz /usr/share/man/man3/ldns_dname_new.3.xz /usr/share/man/man3/ldns_dname_new_frm_data.3.xz /usr/share/man/man3/ldns_dname_new_frm_str.3.xz /usr/share/man/man3/ldns_dname_str_absolute.3.xz /usr/share/man/man3/ldns_dnssec_build_data_chain.3.xz /usr/share/man/man3/ldns_dnssec_data_chain.3.xz /usr/share/man/man3/ldns_dnssec_data_chain_deep_free.3.xz /usr/share/man/man3/ldns_dnssec_data_chain_free.3.xz /usr/share/man/man3/ldns_dnssec_data_chain_new.3.xz /usr/share/man/man3/ldns_dnssec_data_chain_print.3.xz /usr/share/man/man3/ldns_dnssec_data_chain_struct.3.xz /usr/share/man/man3/ldns_dnssec_derive_trust_tree.3.xz /usr/share/man/man3/ldns_dnssec_derive_trust_tree_dnskey_rrset.3.xz /usr/share/man/man3/ldns_dnssec_derive_trust_tree_ds_rrset.3.xz /usr/share/man/man3/ldns_dnssec_derive_trust_tree_no_sig.3.xz /usr/share/man/man3/ldns_dnssec_derive_trust_tree_normal_rrset.3.xz /usr/share/man/man3/ldns_dnssec_name.3.xz /usr/share/man/man3/ldns_dnssec_name_add_rr.3.xz /usr/share/man/man3/ldns_dnssec_name_cmp.3.xz /usr/share/man/man3/ldns_dnssec_name_find_rrset.3.xz /usr/share/man/man3/ldns_dnssec_name_free.3.xz 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/usr/share/man/man3/ldns_rr_set_class.3.xz /usr/share/man/man3/ldns_rr_set_owner.3.xz /usr/share/man/man3/ldns_rr_set_pop_rr.3.xz /usr/share/man/man3/ldns_rr_set_push_rr.3.xz /usr/share/man/man3/ldns_rr_set_rd_count.3.xz /usr/share/man/man3/ldns_rr_set_rdf.3.xz /usr/share/man/man3/ldns_rr_set_ttl.3.xz /usr/share/man/man3/ldns_rr_set_type.3.xz /usr/share/man/man3/ldns_rr_ttl.3.xz /usr/share/man/man3/ldns_rr_type.3.xz /usr/share/man/man3/ldns_rr_uncompressed_size.3.xz /usr/share/man/man3/ldns_rrsig2buffer_wire.3.xz /usr/share/man/man3/ldns_send.3.xz /usr/share/man/man3/ldns_sign_public.3.xz /usr/share/man/man3/ldns_sign_public_dsa.3.xz /usr/share/man/man3/ldns_sign_public_rsamd5.3.xz /usr/share/man/man3/ldns_sign_public_rsasha1.3.xz /usr/share/man/man3/ldns_status.3.xz /usr/share/man/man3/ldns_str2period.3.xz /usr/share/man/man3/ldns_str_remove_comment.3.xz /usr/share/man/man3/ldns_tcp_connect.3.xz /usr/share/man/man3/ldns_tcp_read_wire.3.xz /usr/share/man/man3/ldns_tcp_send_query.3.xz /usr/share/man/man3/ldns_update_adcount.3.xz /usr/share/man/man3/ldns_update_pkt_new.3.xz /usr/share/man/man3/ldns_update_pkt_tsig_add.3.xz /usr/share/man/man3/ldns_update_prcount.3.xz /usr/share/man/man3/ldns_update_set_adcount.3.xz /usr/share/man/man3/ldns_update_set_prcount.3.xz /usr/share/man/man3/ldns_update_set_upcount.3.xz /usr/share/man/man3/ldns_update_set_zocount.3.xz /usr/share/man/man3/ldns_update_upcount.3.xz /usr/share/man/man3/ldns_update_zocount.3.xz /usr/share/man/man3/ldns_verify.3.xz /usr/share/man/man3/ldns_verify_notime.3.xz /usr/share/man/man3/ldns_verify_rrsig.3.xz /usr/share/man/man3/ldns_verify_rrsig_dsa.3.xz /usr/share/man/man3/ldns_verify_rrsig_keylist.3.xz /usr/share/man/man3/ldns_verify_rrsig_keylist_notime.3.xz /usr/share/man/man3/ldns_verify_rrsig_rsamd5.3.xz /usr/share/man/man3/ldns_verify_rrsig_rsasha1.3.xz /usr/share/man/man3/ldns_wire2dname.3.xz /usr/share/man/man3/ldns_wire2pkt.3.xz /usr/share/man/man3/ldns_wire2rdf.3.xz /usr/share/man/man3/ldns_wire2rr.3.xz /usr/share/man/man3/ldns_zone.3.xz /usr/share/man/man3/ldns_zone_deep_free.3.xz /usr/share/man/man3/ldns_zone_glue_rr_list.3.xz /usr/share/man/man3/ldns_zone_new.3.xz /usr/share/man/man3/ldns_zone_new_frm_fp.3.xz /usr/share/man/man3/ldns_zone_new_frm_fp_l.3.xz /usr/share/man/man3/ldns_zone_print.3.xz /usr/share/man/man3/ldns_zone_push_rr.3.xz /usr/share/man/man3/ldns_zone_push_rr_list.3.xz /usr/share/man/man3/ldns_zone_rr_count.3.xz /usr/share/man/man3/ldns_zone_rrs.3.xz /usr/share/man/man3/ldns_zone_set_rrs.3.xz /usr/share/man/man3/ldns_zone_set_soa.3.xz /usr/share/man/man3/ldns_zone_soa.3.xz /usr/share/man/man3/ldns_zone_sort.3.xz Am I missing something? Or there is a way to force FS to dont use libs in lib directory? LD From mitch.capper at gmail.com Mon Mar 26 20:51:50 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Mon, 26 Mar 2012 09:51:50 -0700 Subject: [Freeswitch-users] Why does FS is not ussin my libs? In-Reply-To: References: Message-ID: FreeSWITCH was specifically designed to NOT use system libraries. There are a lot of reasons for this (and its been covered many times on the ML you can search the archives for previous threads) but the major one is just stability. When we compile against known good libraries that are tested to the exact version with FS it gives us the most stable results and makes debugging a lot easier. You could probably move your own libs into place if you really wanted to try and force it but its highly suggested to leave the FS libs alone especially if you want support. ~Mitch From fdelawarde at wirelessmundi.com Mon Mar 26 21:02:17 2012 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Mon, 26 Mar 2012 19:02:17 +0200 Subject: [Freeswitch-users] mod_callcenter field 'system' In-Reply-To: References: Message-ID: <1332781337.15458.1834.camel@luna.madrid.commsmundi.com> Hey Andrew, It's unused right now, but will be in the future, when multi-box callcenter is implemented! Fran?ois. On Mon, 2012-03-26 at 14:22 +0100, Andrew Cassidy wrote: > Hi guys, > > > Just been looking at my agents table for mod_callcenter. There is a > field called 'system' which has been set to 'single_box'. Anyone know > what this field is for? > > > -- > Andrew Cassidy BSc (Hons) MBCS > Managing Director; Cassidy Web Services Ltd > T: 03300 100 960 F: 03300 100 961 > E: andrew at cassidywebservices.co.uk > W: www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From luis.daniel.lucio at gmail.com Mon Mar 26 21:08:17 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Mon, 26 Mar 2012 13:08:17 -0400 Subject: [Freeswitch-users] Cli command In-Reply-To: <4F696D4D.000010.15156@FLIGHTPC> References: <4F696D4D.000010.15156@FLIGHTPC> Message-ID: gracias Le 21 mars 2012 01:55, Fernando - NextBilling IP Solutions < fernandojdk at gmail.com> a ?crit : > http://wiki.freeswitch.org/wiki/Mod_commands > > ^^ > > ? Atenciosamente, > ** > *Importante:* > Esta mensagem, incluindo todo seu conte?do, cont?m informa??es > confidenciais, legalmente protegidas e destinadas a indiv?duo e prop?sito > espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter > sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o > mais breve poss?vel.****** > As informa??es contidas nesta mensagem e em seu conte?do s?o de > responsabilidade de seu autor, n?o representando necessariamente id?ias, > opini?es, pensamentos ou qualquer forma de posicionamento por parte da > NextBilling IP Solutions. > P* "Antes de imprimir pense em seu compromisso com o Meio Ambiente."* > *-------Original Message-------* > > *From:* Michael Collins > *Date:* 21/03/2012 02:53:02 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Cli command > > wiki.freeswitch.org > left side under "navigation" click "API commands" > > -MC > > On Tue, Mar 20, 2012 at 7:06 PM, Luis Daniel Lucio Quiroz < > luis.daniel.lucio at gmail.com> wrote: > > Just wondering where i can read about cli commands > > LD > > > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/4c89ce94/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 1596 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/4c89ce94/attachment-0001.gif -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 20873 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/4c89ce94/attachment-0001.jpe From luis.daniel.lucio at gmail.com Mon Mar 26 21:10:28 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Mon, 26 Mar 2012 13:10:28 -0400 Subject: [Freeswitch-users] Why does FS is not ussin my libs? In-Reply-To: References: Message-ID: Problem on this is that I0m packaking for a distro, so i can not have same libs twice, it will simple wont work Le 26 mars 2012 12:51, Mitch Capper a ?crit : > hen we compile against known From anthony.minessale at gmail.com Mon Mar 26 21:17:56 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 26 Mar 2012 12:17:56 -0500 Subject: [Freeswitch-users] Why does FS is not ussin my libs? In-Reply-To: References: Message-ID: FS never installs any 3rd party libs. If it uses such a dependency, it statically links it into the .so or the core and uses it privately. On Mon, Mar 26, 2012 at 12:10 PM, Luis Daniel Lucio Quiroz wrote: > Problem on this is that I0m packaking for a distro, so i can not have > same libs twice, it will simple wont work > > > Le 26 mars 2012 12:51, Mitch Capper a ?crit : >> hen we compile against known > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From lazyvirus at gmx.com Mon Mar 26 21:19:23 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 26 Mar 2012 19:19:23 +0200 Subject: [Freeswitch-users] Why does FS is not ussin my libs? In-Reply-To: References: Message-ID: <20120326191923.1f27c99b@anubis.defcon1> On Mon, 26 Mar 2012 13:10:28 -0400 Luis Daniel Lucio Quiroz wrote: > Problem on this is that I0m packaking for a distro, so i can not have > same libs twice, it will simple wont work You shouldn't worry, FS must use its own path setup to find its libs; and that was never a problem on a Debian sid (NO conflicts with almost any libs either FS & Debian installed:) -- From msc at freeswitch.org Mon Mar 26 21:22:05 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Mar 2012 10:22:05 -0700 Subject: [Freeswitch-users] Cli command In-Reply-To: References: <4F696D4D.000010.15156@FLIGHTPC> Message-ID: 2012/3/26 Luis Daniel Lucio Quiroz > gracias > De nada. Also, don't forget that we have a number of Spanish-speaking FreeSWITCHers. Check out #freeswitch-es on irc.freenode.net. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/8a6b9f50/attachment.html From andrew at cassidywebservices.co.uk Mon Mar 26 21:26:28 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 26 Mar 2012 18:26:28 +0100 Subject: [Freeswitch-users] mod_callcenter field 'system' In-Reply-To: <1332781337.15458.1834.camel@luna.madrid.commsmundi.com> References: <1332781337.15458.1834.camel@luna.madrid.commsmundi.com> Message-ID: Cool, sound like an interesting development. I'll keep an eye out for it! On 26 March 2012 18:02, Fran?ois Delawarde wrote: > Hey Andrew, > > It's unused right now, but will be in the future, when multi-box > callcenter is implemented! > > Fran?ois. > > On Mon, 2012-03-26 at 14:22 +0100, Andrew Cassidy wrote: > > Hi guys, > > > > > > Just been looking at my agents table for mod_callcenter. There is a > > field called 'system' which has been set to 'single_box'. Anyone know > > what this field is for? > > > > > > -- > > Andrew Cassidy BSc (Hons) MBCS > > Managing Director; Cassidy Web Services Ltd > > T: 03300 100 960 F: 03300 100 961 > > E: andrew at cassidywebservices.co.uk > > W: www.cassidywebservices.co.uk > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/ed6740e5/attachment.html From msc at freeswitch.org Mon Mar 26 21:39:18 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Mar 2012 10:39:18 -0700 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: References: Message-ID: On Mon, Mar 26, 2012 at 3:01 AM, Daniel Knaggs < Daniel.Knaggs at realitysolutions.co.uk> wrote: > OK, call recording has been setup ? waiting for it to happen now.**** > > ** ** > > ** ** > > Interestingly, before I issue the ?record_session? application (and of > course the ?RECORD_*? variables) I had to execute ?ring_ready? then > ?pre_answer? otherwise the caller gets silence (changing the order of those > two commands results in silence). > I find it odd that just doing a pre_answer wouldn't be sufficient. A pre_answer will send a 183 w/SDP whereas ring_ready simply sends a 180. In any case, I'm glad you got your recordings. I also find it curious that only the "letter" DTMFs are being detected. Let us know if you actually hear those tones in the audio stream. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/e5ed2bc5/attachment.html From msc at freeswitch.org Mon Mar 26 21:40:51 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Mar 2012 10:40:51 -0700 Subject: [Freeswitch-users] Binary download with curl In-Reply-To: References: Message-ID: Perhaps this? http://files-sync.freeswitch.org/windows/installer/x86/ On Mon, Mar 26, 2012 at 1:04 AM, Olli Aro wrote: > ** ** > > Hi all,**** > > ** ** > > Been trying to compile Freeswitch on Windows in order to get in the XML > curl module, however it is not looking easy (76 more fatal errors to figure > out...)**** > > ** ** > > Before I start that task I thought it is always worth asking :)**** > > ** ** > > Would anyone happen to have a download link to already compiled Windows > 32-bit binary for close to recent version of Freeswitch with the XML curl > module?**** > > ** ** > > Regards,**** > > ** ** > > Olli**** > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/08cfabe3/attachment-0001.html From msc at freeswitch.org Mon Mar 26 22:10:57 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Mar 2012 11:10:57 -0700 Subject: [Freeswitch-users] Bridging after sending dtmf In-Reply-To: <74D0BA2985A4B04E8651FA01C8A58F39051F5A6D@vs-win-ex01.corp.parusi.com> References: <74D0BA2985A4B04E8651FA01C8A58F39051F5A6D@vs-win-ex01.corp.parusi.com> Message-ID: Did you post debug logs of the call? -MC On Fri, Mar 16, 2012 at 3:57 PM, Alex Zarubin wrote: > Hello, > > Here are two dialplan examples of the simple thing I'm failing to do - > bridge inbound (external profile) and outbound (internal profile) sip > calls after sending some dtmf sequence to the outbound leg. In both > cases (and several more I've tried) calls are bridged right after > outbound leg picks up and dtmf are sent after the bridge. I would > greatly appreciate your help/hints. > > Thank you. > > Alex > > ---------------------------------------------------------------------- > Example 1. > > > > > > > data="bridge_pre_execute_bleg_app=send_dtmf"/> > data="bridge_pre_execute_bleg_data=w#w${dtmfseq}"/> > data="sofia/internal/${destination}@${destip}"/> > > > > > ---------------------------------------------------------------------- > > Example 2. > > > > > > > > > > > > > > Script lua_async.lua : > > dtmfseq = argv[1]; > phone = argv[2]; > destip = argv[3]; > leg1uuid = argv[4]; > cid = argv[5]; > > ands = ") and ("; > logline = "Lua called with (" .. dtmfseq .. ands .. phone .. ands .. > destip .. ands .. leg1uuid .. ands .. cid .. ")\n"; > freeswitch.console_log("info", logline); > > api = freeswitch.API(); > > sofia_destination = "[origination_caller_id_number=" .. cid .. > "]sofia/internal/" .. phone .. "@" .. destip; > new_session = freeswitch.Session(sofia_destination); > dispo = "None"; > > while (new_session:ready() and dispo ~= "ANSWER") do > dispo = new_session:getVariable("endpoint_disposition") > > freeswitch.consoleLog("INFO", "disposition is '" .. dispo .. "'\n") > os.execute("sleep 1") > end -- while > > leg2uuid = new_session:get_uuid(); > logline = "Lua leg2uuid (" .. leg2uuid .. ")\n"; > freeswitch.console_log("info", logline); > > new_session:setAutoHangup(false); > > exestr_cmd = "uuid_send_dtmf " .. leg2uuid .. " w#w" .. dtmfseq; > logline = "Lua calling api:executeString(" .. exestr_cmd .. ")\n"; > freeswitch.console_log("info", logline); > api:executeString(exestr_cmd); > freeswitch.console_log("info", "Lua after api:executeString(...)\n"); > > exestr_cmd = "uuid_bridge " .. leg1uuid .. " " .. leg2uuid; > logline = "Lua calling api:executeString(" .. exestr_cmd .. ")\n"; > freeswitch.console_log("info", logline); > api:executeString(exestr_cmd); > freeswitch.console_log("info", "Lua after api:executeString(...)\n"); > > > This message and any attachments to it are intended only for the > addressee(s) identified above and may contain CONFIDENTIAL information. It > is not intended for transmission to, or receipt by, any unauthorized > persons. If you are not an intended recipient or an agent responsible for > delivering it to an intended recipient, you have received this e-mail in > error and any dissemination, distribution, or copying of this message or > any attachment to it is strictly prohibited. If you have received this > email in error, please (i) do not read it, (ii) reply to the sender that > you received the message in error, and (iii) erase or destroy the message > from your system. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/9e38d710/attachment.html From Mailings at kh-dev.de Mon Mar 26 22:14:40 2012 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Mon, 26 Mar 2012 18:14:40 +0000 Subject: [Freeswitch-users] Language Files In-Reply-To: References: Message-ID: <8220BA48EF3C9A438124F20C9C76571520EE3E0F@srv01.khdev.corp> Hi, here: http://freeswitch.xpirio.com/ Regards, Klaus Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Volker Wend Gesendet: Freitag, 23. M?rz 2012 22:33 An: FreeSWITCH Users Help Betreff: [Freeswitch-users] Language Files Hi, where can I find german language soundfiles fort the voicemail module ? Br Volker -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/0084dcb7/attachment.html From msc at freeswitch.org Mon Mar 26 22:15:19 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Mar 2012 11:15:19 -0700 Subject: [Freeswitch-users] acl.conf.xml don't understand In-Reply-To: <20120317011748.4e777bfd@anubis.defcon1> References: <20120317011748.4e777bfd@anubis.defcon1> Message-ID: Apologies for the late response... did you get this figured out? Most likely it's because your calls are hitting the public context instead of default. Before recommending a course of action it would probably be good to know what you had in mind for putting that ACL entry in there. -MC On Fri, Mar 16, 2012 at 5:17 PM, Bzzz wrote: > Hi list, > > There's something I don't understand: if I add my LAN cidr > (allow) in the second block of acl.conf.xml (domains), I can't call > anymore. > > > > > > > > >From what I understood of the wiki, this should work because the > block denies everyone but my line allows for LAN cidr. > > I guess I misunderstood something. > > JY > -- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/d9f804bd/attachment.html From gcd at i.ph Mon Mar 26 22:25:52 2012 From: gcd at i.ph (Nandy Dagondon) Date: Tue, 27 Mar 2012 02:25:52 +0800 Subject: [Freeswitch-users] FLEX Client Authentication Message-ID: hello, how can i authenticate Flex client? i have tried the following in the user directory: 1. plain text password => (authentication failed) 2. a1-hash w/ using ssl md5 generated "username:domain:password" value => (empty password not allowed) 3. md5 generated plain text password => (authentication failed) peeking at rtmp.c, i can see it looks for the "password" parameter. so, how to generate the authentication values? i appreciate for your help. tks Anthony for the Flex client. it's cool! -nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/13476a3c/attachment-0001.html From trever.adams at gmail.com Mon Mar 26 22:27:39 2012 From: trever.adams at gmail.com (Trever L. Adams) Date: Mon, 26 Mar 2012 12:27:39 -0600 Subject: [Freeswitch-users] Trouble with transfer and voicemail password Message-ID: <4F70B51B.3080100@gmail.com> Hello everyone, I hope someone can help me. I am using my home as a setup to test FreeSWITCH. I am using FXO for my incoming and FXS to phones around the home. I am trying to do blind transfers when *0# is dialed on the internal phones for calls coming from outside the home. I currently have: The transfer immediately starts ringing the entire house (1000). This is true even when I am trying to get it to transfer to 1003. Also, http://jira.freeswitch.org/browse/OPENZAP-173 seems to be back in that when the incoming phone hangs up, it seems to continue to ring. Maybe I have this setup all wrong. I do not want anyone outside (the call originator) to be able to initiate the transfer. I want the transferring party to be cut out of the call. I want the outside and the *0#INTERNAL_EXT# to be connected. As for voicemail, I all FXS, no SIP for now. How do I set the password? How would I setup a common voicemail box that every thing goes to except one extension? That should have a separate box and password. Thank you for any help. Trever -- I love dogs, but I hate Chihuahuas. A Chihuahua isn't a dog. It's a rat with a thyroid problem. -- Unknown -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/dc3dce78/attachment.bin From lazyvirus at gmx.com Mon Mar 26 22:31:29 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 26 Mar 2012 20:31:29 +0200 Subject: [Freeswitch-users] acl.conf.xml don't understand In-Reply-To: References: <20120317011748.4e777bfd@anubis.defcon1> Message-ID: <20120326203129.0000f78d@anubis.defcon1> On Mon, 26 Mar 2012 11:15:19 -0700 Michael Collins wrote: > Apologies for the late response... did you get this figured out? Most > likely it's because your calls are hitting the public context instead of > default. Before recommending a course of action it would probably be good > to know what you had in mind for putting that ACL entry in there. Well, I was playing with FS to try to understand how it work more deeply. And no, this issue isn't fixed (but that may be connected to fusionPBX, I didn't tried w/ a default configuration). You're right about the public context, I just re-made the test and found this in the logs: mod_dialplan_xml.c:485 Processing jy test <01>->02 in context public For now, this isn't too much a problem as I'm learning how FS work and it is not blocking. BTW, is there an arborescence of FS conf files somewhere? (it is sometimes hard to know who's doing what). Jean-Yves -- From nasida at live.ru Mon Mar 26 22:44:44 2012 From: nasida at live.ru (Yuriy Nasida) Date: Mon, 26 Mar 2012 22:44:44 +0400 Subject: [Freeswitch-users] VoiceChanger for FS In-Reply-To: References: , Message-ID: Guys, thanks for the excellent feedback! Date: Mon, 26 Mar 2012 07:42:21 -0700 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] VoiceChanger for FS 2012/3/26 Yuriy Nasida Hello guys, If anybody know any similar implementation of VoiceChanger (or something like this) for FS ? Thanks. Like Mitch said, you definitely want to check out mod_ladspa. As the name implies it uses LADPSA which means you'll need to be running FS on Linux. I believe mod_ladspa comes with a config for something like "autotune" or "autotalent" or something like that. In any case, it's a great place to start. -MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/6da9c1be/attachment.html From msc at freeswitch.org Mon Mar 26 22:46:02 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Mar 2012 11:46:02 -0700 Subject: [Freeswitch-users] Language Files In-Reply-To: <8220BA48EF3C9A438124F20C9C76571520EE3E0F@srv01.khdev.corp> References: <8220BA48EF3C9A438124F20C9C76571520EE3E0F@srv01.khdev.corp> Message-ID: Would someone mind creating a wiki page for sound and language files and put this in there? We kinda need a place to put language-related sound files and this is a good piece of information to have. Thanks, MC On Mon, Mar 26, 2012 at 11:14 AM, Klaus Hochlehnert wrote: > Hi,**** > > ** ** > > here: http://freeswitch.xpirio.com/**** > > ** ** > > Regards,**** > > Klaus**** > > ** ** > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Volker > Wend > *Gesendet:* Freitag, 23. M?rz 2012 22:33 > *An:* FreeSWITCH Users Help > *Betreff:* [Freeswitch-users] Language Files**** > > ** ** > > Hi, **** > > ** ** > > where can I find german language soundfiles fort the voicemail module ?*** > * > > ** ** > > Br**** > > Volker**** > > ** ** > > ** ** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/65474caf/attachment.html From msc at freeswitch.org Mon Mar 26 22:48:38 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Mar 2012 11:48:38 -0700 Subject: [Freeswitch-users] acl.conf.xml don't understand In-Reply-To: <20120326203129.0000f78d@anubis.defcon1> References: <20120317011748.4e777bfd@anubis.defcon1> <20120326203129.0000f78d@anubis.defcon1> Message-ID: Okay, no worries. FYI, if you want to associate an IP address with a specific user so that all calls from a specific IP address are considered as being "from that user" then do the cidr='x.x.x.x/32' trick in the user directory entry. -MC On Mon, Mar 26, 2012 at 11:31 AM, Bzzz wrote: > On Mon, 26 Mar 2012 11:15:19 -0700 > Michael Collins wrote: > > > Apologies for the late response... did you get this figured out? Most > > likely it's because your calls are hitting the public context instead of > > default. Before recommending a course of action it would probably be good > > to know what you had in mind for putting that ACL entry in there. > > Well, I was playing with FS to try to understand how it work more > deeply. And no, this issue isn't fixed (but that may be connected > to fusionPBX, I didn't tried w/ a default configuration). > > You're right about the public context, I just re-made the test and > found this in the logs: > mod_dialplan_xml.c:485 Processing jy test <01>->02 in context public > > For now, this isn't too much a problem as I'm learning how FS work > and it is not blocking. > > BTW, is there an arborescence of FS conf files somewhere? (it is > sometimes hard to know who's doing what). > > Jean-Yves > -- > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/cb4552fe/attachment.html From lazyvirus at gmx.com Mon Mar 26 22:56:52 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 26 Mar 2012 20:56:52 +0200 Subject: [Freeswitch-users] acl.conf.xml don't understand In-Reply-To: References: <20120317011748.4e777bfd@anubis.defcon1> <20120326203129.0000f78d@anubis.defcon1> Message-ID: <20120326205652.57765ca4@anubis.defcon1> On Mon, 26 Mar 2012 11:48:38 -0700 Michael Collins wrote: > Okay, no worries. FYI, if you want to associate an IP address with a > specific user so that all calls from a specific IP address are considered > as being "from that user" then do the cidr='x.x.x.x/32' trick in the user > directory entry. Thanks, that's good to know 'cos I prefer permanent setups in a network (opposed to ie DHCP). JY -- It's not whether you win or lose, it's how you place the blame. From anita.hall at simmortel.com Mon Mar 26 23:24:27 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Tue, 27 Mar 2012 00:54:27 +0530 Subject: [Freeswitch-users] Fax on Sangoma In-Reply-To: References: Message-ID: Hi The dialplan just sends the control to a python script over ESL which gives rxfax(filename.tiff) I have tried giving the rxfax() command in the dialplan itself with similar results. Many calls drop prematuresly and a good many fail to receive DCS after permitted retries. regards, Anita On Sat, Mar 24, 2012 at 9:24 PM, Anton Kvashenkin wrote: > Can you show dialplan snippet. > 24.03.2012 19:24 ???????????? "Anita Hall" > ???????: > >> Hi >> >> My client had been using an old torrenta cards which game problems with >> pretty much everything :) After evangelizing the cause of Sangoma for more >> than 3 months (yes, they are dumb), they are finally making the transition >> so I am pretty excited. The choice of card is A108D which does a whole >> bunch of DSP in the hardware. >> >> Now, the first hurdle that I have to make Sangoma jump across is getting >> incoming Fax over E1 right. We are using mod_spandsp, of course. So, here I >> will be needing a whole lot of help from the veterans of spandsp and Faxing >> :) I desperately need Sangoma to pass the Fax test or they will give me >> torrenta cards all over again! >> >> The primary cause of failures are - (49) The call dropped prematurely and >> (48) Disconnected after permitted retries. >> >> For example, in this case, can I conclude that the other end did not >> provide a Fax tone or is it something else? >> >> 2aeb5f7c-75b5-11e1-8f36-b3286880c45b EXECUTE FreeTDM/4:2/47615728 >> rxfax(/srv/fax/in/2aeb5f7c-75b5-11e1-8f36-b3286880c45b.tiff) >> 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] >> mod_spandsp_fax.c:1357 Raw read codec activation Success L16 20000 >> 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] >> switch_core_codec.c:216 FreeTDM/4:2/47615728 Push codec L16:70 >> 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] >> mod_spandsp_fax.c:1373 Raw write codec activation Success L16 >> 2012-03-24 18:57:57.844876 [DEBUG] ftmod_wanpipe.c:965 [s4c2][4:2] First >> packet read stats: Rx queue len: 1, Rx queue size: 10 >> 2012-03-24 18:57:57.904879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >> signal status is Carrier up (-2) in state 17 >> 2012-03-24 18:57:57.924878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >> signal status is Carrier down (-1) in state 17 >> 2012-03-24 18:57:58.104877 [DEBUG] ftmod_wanpipe.c:901 [s4c2][4:2] First >> packet write stats: Tx queue len: 1, Tx queue size: 5, Tx idle: 30 >> 2012-03-24 18:57:58.444908 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >> signal status is Carrier up (-2) in state 17 >> 2012-03-24 18:57:58.644907 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >> signal status is Carrier down (-1) in state 17 >> 2012-03-24 18:57:58.664878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >> signal status is Carrier up (-2) in state 1 >> 2012-03-24 18:57:58.764879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >> signal status is Carrier up (-2) in state 17 >> 2012-03-24 18:57:58.764879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >> signal status is Carrier down (-1) in state 17 >> 2012-03-24 18:57:58.784878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >> signal status is Carrier up (-2) in state 17 >> 2012-03-24 18:57:58.804879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >> signal status is Carrier down (-1) in state 17 >> 2012-03-24 18:57:59.244886 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >> signal status is Carrier up (-2) in state 17 >> 2012-03-24 18:57:59.464887 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >> signal status is Carrier down (-1) in state 17 >> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 T4 >> expired in phase T30_PHASE_B_RX, state 17 >> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Retry >> number 1 >> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 >> Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX >> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set rx >> type 0 >> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set tx >> type 4 >> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 >> Sending ident 'Sangoma Fax Ident' >> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Tx: >> CSI without final frame tag >> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Tx: >> ff 03 40 74 6e 65 64 49 20 78 61 46 20 61 6d 6f 67 6e 61 53 20 20 20 >> 2012-03-24 18:58:00.524877 [DEBUG] ftmod_libpri.c:1065 -- Hangup REQ on >> channel 4:1 >> 2012-03-24 18:58:00.524877 [DEBUG] ftmod_libpri.c:1078 [s4c1][4:1] >> Changed state from UP to TERMINATING >> 2012-03-24 18:58:00.524877 [DEBUG] ftdm_state.c:511 [s4c1][4:1] Executing >> state processor for TERMINATING >> >> >> I will need some more hand-holding with logs later :) >> >> >> regards, >> Anita >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/945edadd/attachment.html From anita.hall at simmortel.com Mon Mar 26 23:28:32 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Tue, 27 Mar 2012 00:58:32 +0530 Subject: [Freeswitch-users] no DTMF detection over VoIP In-Reply-To: References: Message-ID: If you are using a PRI line, try taking a q931 trace. It will be able to show exactly what is happening below the hood. regards, Anita On Mon, Mar 26, 2012 at 11:48 AM, Neil Patel wrote: > I guess you do this by using start_dtmf? > I tried adding it to my dialplan and also in my lua IVR script: > session:execute("start_dtmf"); > > But no luck with either. > > > On Sun, Mar 25, 2012 at 8:40 AM, curriegrad2004 wrote: > >> If the call is coming from a TDM circuit, you'll have to do in-band >> DTMF detection in that situation. >> >> On Sat, Mar 24, 2012 at 6:25 PM, Michael Collins >> wrote: >> > Neil, >> > >> > Howdy! Long time no chat. :) >> > >> > Regarding SIP/VoIP: A SIP call will not come in over a PRI circuit. SIP >> is >> > VoIP and PRI is TDM - two totally different methods for carrying >> telephone >> > calls. My guess is that you might have a bit of confusion about where >> and >> > how calls are coming in. Let's start with the basics: Do you have a >> SIP/VoIP >> > provider? If so, did they supply you with phone number(s) for dialing >> in to >> > your system? Have you gone through the configuration process to make >> sure >> > your FS box is properly communicating with the provider? >> > >> > If you need some help you can contact me off list. Of course, this means >> > you'll have to take me to that Mexican food place the next time I'm in >> > Frisco! :) >> > >> > -MC >> > >> > >> > On Sat, Mar 24, 2012 at 11:18 AM, Neil Patel >> wrote: >> >> >> >> I have not modified the sip_profiles directory from standard. There is >> no >> >> variable set for dtmf-type in either internal.xml or external.xml. >> >> >> >> I did a trace using: >> >> sofia global siptrace on >> >> >> >> There wer no SIP packets logged. If the SIP call is going to a PRI >> profile >> >> (freetdm), will there still be SIP traffic? >> >> >> >> Thanks, >> >> Neil >> >> >> >> >> >> On Sat, Mar 24, 2012 at 7:22 PM, Anita Hall >> >> wrote: >> >>> >> >>> What is your config for DTMF in sip profile ? >> >>> >> >>> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#dtmf-type >> >>> >> >>> Use tshark to generate a SIP trace and paste it on >> >>> http://pastebin.freeswitch.org/ and give the link here. >> >>> >> >>> regards, >> >>> Anita >> >>> >> >>> >> >>> >> >>> On Sat, Mar 24, 2012 at 2:43 PM, Neil Patel >> >>> wrote: >> >>>> >> >>>> Hi All, >> >>>> >> >>>> I have a basic IVR application in Lua connected to a PRI line. It >> >>>> currently is not responding to DTMF input given from any VoIP call >> (e.g. >> >>>> Skype). However, it accepts input from local mobile or landline >> calls just >> >>>> fine. >> >>>> >> >>>> I recently pulled latest from git and built; before that, DTMF >> detection >> >>>> from both were working. Is there something I need to configure to >> allow DTMF >> >>>> for VoIP? >> >>>> >> >>>> Thanks, >> >>>> Neil >> >>>> >> >>>> >> >>>> >> _________________________________________________________________________ >> >>>> Professional FreeSWITCH Consulting Services: >> >>>> consulting at freeswitch.org >> >>>> http://www.freeswitchsolutions.com >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> Official FreeSWITCH Sites >> >>>> http://www.freeswitch.org >> >>>> http://wiki.freeswitch.org >> >>>> http://www.cluecon.com >> >>>> >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>> http://www.freeswitch.org >> >>>> >> >>> >> >>> >> >>> >> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/4a72118e/attachment-0001.html From anita.hall at simmortel.com Tue Mar 27 01:14:46 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Tue, 27 Mar 2012 02:44:46 +0530 Subject: [Freeswitch-users] Fax on Sangoma In-Reply-To: References: Message-ID: Here is a log when the call drops in Phase D after having received 4 pages! I am seeing this for the first time that call is dropping in Phase D. I think I could fix this, if someone could throw a pointer or two my way. Should I look into mod_spandsp_fax.c or can this be done elsewhere? f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:52.024887 [DEBUG] mod_spandsp_fax.c:428 ==== Page Received =========================================================== f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:52.024887 [DEBUG] mod_spandsp_fax.c:429 Page no = 4 f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:52.024887 [DEBUG] mod_spandsp_fax.c:430 Image size = 1728 x 999 pixels f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:52.024887 [DEBUG] mod_spandsp_fax.c:431 Image resolution = 8031/m x 3850/m f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:52.024887 [DEBUG] mod_spandsp_fax.c:432 Compression = T.6 (3) f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:52.024887 [DEBUG] mod_spandsp_fax.c:433 Compressed image size = 22935 bytes f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:52.024887 [DEBUG] mod_spandsp_fax.c:434 Bad rows = 0 f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:52.024887 [DEBUG] mod_spandsp_fax.c:435 Longest bad row run = 0 f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:52.024887 [DEBUG] mod_spandsp_fax.c:436 ============================================================================== 2012-03-26 22:02:52.024887 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Changing from state 13 to 14 2012-03-26 22:02:52.024887 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Tx: MCF with final frame tag 2012-03-26 22:02:52.024887 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Tx: ff 13 8c 2012-03-26 22:02:52.084891 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 14 2012-03-26 22:02:52.084891 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Changing from phase T30_PHASE_D_RX to T30_PHASE_D_TX 2012-03-26 22:02:52.084891 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set rx type 0 2012-03-26 22:02:52.084891 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set tx type 4 2012-03-26 22:02:53.144909 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 14 2012-03-26 22:02:53.224929 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 14 2012-03-26 22:02:53.224929 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_D_RX 2012-03-26 22:02:53.224929 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set rx type 4 2012-03-26 22:02:53.224929 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set tx type 0 2012-03-26 22:02:53.224929 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Start T4 2012-03-26 22:02:54.384927 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 14 2012-03-26 22:02:54.584907 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Framing OK (-6) in state 14 2012-03-26 22:02:54.584907 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Start T4A 2012-03-26 22:02:55.604879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Bad HDLC CRC received 2012-03-26 22:02:55.604879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Bad CRC and timer is 7 2012-03-26 22:02:55.664879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 14 2012-03-26 22:02:55.664879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Start T4B 2012-03-26 22:02:55.684877 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 14 2012-03-26 22:02:55.684877 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Stop T4B (1440 remaining) 2012-03-26 22:02:55.704878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 14 2012-03-26 22:02:55.704878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Start T4B 2012-03-26 22:02:55.844910 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 14 2012-03-26 22:02:55.844910 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Stop T4B (480 remaining) 2012-03-26 22:02:55.844910 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 14 2012-03-26 22:02:55.844910 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Start T4B 2012-03-26 22:02:56.024878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 T4B expired in phase T30_PHASE_D_RX, state 14. The line is now quiet. 2012-03-26 22:02:56.024878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 T4 expired in phase T30_PHASE_D_RX, state 14 2012-03-26 22:02:56.024878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Retry number 2 2012-03-26 22:02:56.024878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Repeat command called with nothing to repeat - phase T30_PHASE_D_RX, state 14 2012-03-26 22:02:56.264924 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 14 2012-03-26 22:02:56.284900 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 14 2012-03-26 22:02:56.804883 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 14 2012-03-26 22:02:56.824909 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 14 2012-03-26 22:02:56.864878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 14 2012-03-26 22:02:56.864878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 14 2012-03-26 22:02:56.964906 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 14 2012-03-26 22:02:56.984901 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 14 2012-03-26 22:02:57.084900 [DEBUG] ftmod_libpri.c:1065 -- Hangup REQ on channel 4:1 2012-03-26 22:02:57.084900 [DEBUG] ftmod_libpri.c:1078 [s4c1][4:1] Changed state from UP to TERMINATING 2012-03-26 22:02:57.084900 [DEBUG] ftdm_state.c:511 [s4c1][4:1] Executing state processor for TERMINATING 2012-03-26 22:02:57.084900 [DEBUG] ftmod_libpri.c:679 -- 4:1 STATE [TERMINATING] 2012-03-26 22:02:57.084900 [DEBUG] ftmod_libpri.c:687 [s4c1][4:1] Completed state change from UP to TERMINATING in 0ms 2012-03-26 22:02:57.084900 [DEBUG] ftdm_io.c:5586 [s4c1][4:1] Scheduling safety hangup timer 2012-03-26 22:02:57.084900 [DEBUG] mod_freetdm.c:2416 got clear channel sig [STOP] f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.084900 [DEBUG] switch_channel.c:2848 (FreeTDM/4:1/47615101) Callstate Change ACTIVE -> HANGUP f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.084900 [NOTICE] mod_freetdm.c:2441 Hangup FreeTDM/4:1/47615101 [CS_EXECUTE] [NORMAL_CLEARING] f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.084900 [DEBUG] switch_channel.c:2871 Send signal FreeTDM/4:1/47615101 [KILL] f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.084900 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/4:1/47615101 [BREAK] f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:491 ============================================================================== f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:504 Fax processing not successful - result (49) The call dropped prematurely. f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:509 Remote station id: 05322405354 f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:510 Local station id: Sangoma Fax Ident f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:511 Pages transferred: 4 f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:513 Total fax pages: 4 f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:514 Image resolution: 8031x3850 f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:515 Transfer Rate: 4800 f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:517 ECM status on f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:518 remote country: f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:519 remote vendor: f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:520 remote model: f1a680ea-7760-11e1-b0ea-b3286880c45b 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:522 ============================================================================== 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Changing from state 14 to 32 2012-03-26 22:02:57.104898 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Changing from phase T30_PHASE_D_RX to T30_PHASE_CALL_FINISHED regards, Anita On Tue, Mar 27, 2012 at 12:54 AM, Anita Hall wrote: > Hi > > The dialplan just sends the control to a python script over ESL which > gives rxfax(filename.tiff) > > I have tried giving the rxfax() command in the dialplan itself with > similar results. Many calls drop prematuresly and a good many fail to > receive DCS after permitted retries. > > > regards, > Anita > > > > > On Sat, Mar 24, 2012 at 9:24 PM, Anton Kvashenkin > wrote: > >> Can you show dialplan snippet. >> 24.03.2012 19:24 ???????????? "Anita Hall" >> ???????: >> >>> Hi >>> >>> My client had been using an old torrenta cards which game problems with >>> pretty much everything :) After evangelizing the cause of Sangoma for more >>> than 3 months (yes, they are dumb), they are finally making the transition >>> so I am pretty excited. The choice of card is A108D which does a whole >>> bunch of DSP in the hardware. >>> >>> Now, the first hurdle that I have to make Sangoma jump across is getting >>> incoming Fax over E1 right. We are using mod_spandsp, of course. So, here I >>> will be needing a whole lot of help from the veterans of spandsp and Faxing >>> :) I desperately need Sangoma to pass the Fax test or they will give me >>> torrenta cards all over again! >>> >>> The primary cause of failures are - (49) The call dropped prematurely >>> and (48) Disconnected after permitted retries. >>> >>> For example, in this case, can I conclude that the other end did not >>> provide a Fax tone or is it something else? >>> >>> 2aeb5f7c-75b5-11e1-8f36-b3286880c45b EXECUTE FreeTDM/4:2/47615728 >>> rxfax(/srv/fax/in/2aeb5f7c-75b5-11e1-8f36-b3286880c45b.tiff) >>> 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] >>> mod_spandsp_fax.c:1357 Raw read codec activation Success L16 20000 >>> 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] >>> switch_core_codec.c:216 FreeTDM/4:2/47615728 Push codec L16:70 >>> 2aeb5f7c-75b5-11e1-8f36-b3286880c45b 2012-03-24 18:57:57.824906 [DEBUG] >>> mod_spandsp_fax.c:1373 Raw write codec activation Success L16 >>> 2012-03-24 18:57:57.844876 [DEBUG] ftmod_wanpipe.c:965 [s4c2][4:2] First >>> packet read stats: Rx queue len: 1, Rx queue size: 10 >>> 2012-03-24 18:57:57.904879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >>> signal status is Carrier up (-2) in state 17 >>> 2012-03-24 18:57:57.924878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >>> signal status is Carrier down (-1) in state 17 >>> 2012-03-24 18:57:58.104877 [DEBUG] ftmod_wanpipe.c:901 [s4c2][4:2] First >>> packet write stats: Tx queue len: 1, Tx queue size: 5, Tx idle: 30 >>> 2012-03-24 18:57:58.444908 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >>> signal status is Carrier up (-2) in state 17 >>> 2012-03-24 18:57:58.644907 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >>> signal status is Carrier down (-1) in state 17 >>> 2012-03-24 18:57:58.664878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >>> signal status is Carrier up (-2) in state 1 >>> 2012-03-24 18:57:58.764879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >>> signal status is Carrier up (-2) in state 17 >>> 2012-03-24 18:57:58.764879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >>> signal status is Carrier down (-1) in state 17 >>> 2012-03-24 18:57:58.784878 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >>> signal status is Carrier up (-2) in state 17 >>> 2012-03-24 18:57:58.804879 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >>> signal status is Carrier down (-1) in state 17 >>> 2012-03-24 18:57:59.244886 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >>> signal status is Carrier up (-2) in state 17 >>> 2012-03-24 18:57:59.464887 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC >>> signal status is Carrier down (-1) in state 17 >>> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 T4 >>> expired in phase T30_PHASE_B_RX, state 17 >>> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Retry >>> number 1 >>> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 >>> Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX >>> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set rx >>> type 0 >>> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set tx >>> type 4 >>> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 >>> Sending ident 'Sangoma Fax Ident' >>> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Tx: >>> CSI without final frame tag >>> 2012-03-24 18:57:59.564903 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Tx: >>> ff 03 40 74 6e 65 64 49 20 78 61 46 20 61 6d 6f 67 6e 61 53 20 20 20 >>> 2012-03-24 18:58:00.524877 [DEBUG] ftmod_libpri.c:1065 -- Hangup REQ on >>> channel 4:1 >>> 2012-03-24 18:58:00.524877 [DEBUG] ftmod_libpri.c:1078 [s4c1][4:1] >>> Changed state from UP to TERMINATING >>> 2012-03-24 18:58:00.524877 [DEBUG] ftdm_state.c:511 [s4c1][4:1] >>> Executing state processor for TERMINATING >>> >>> >>> I will need some more hand-holding with logs later :) >>> >>> >>> regards, >>> Anita >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/05a644fe/attachment-0001.html From msc at freeswitch.org Tue Mar 27 04:10:02 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Mar 2012 17:10:02 -0700 Subject: [Freeswitch-users] Trouble with transfer and voicemail password In-Reply-To: <4F70B51B.3080100@gmail.com> References: <4F70B51B.3080100@gmail.com> Message-ID: go ahead and get a debug log of a call coming in, being answered, and the recipient dialing *0#xxx# to send the call elsewhere. Drop it into pastebin.freeswitch.org and use "FreeSWITCH Log" as the syntax highlighting. -MC On Mon, Mar 26, 2012 at 11:27 AM, Trever L. Adams wrote: > Hello everyone, > > I hope someone can help me. I am using my home as a setup to test > FreeSWITCH. > > I am using FXO for my incoming and FXS to phones around the home. I am > trying to do blind transfers when *0# is dialed on the internal phones > for calls coming from outside the home. I currently have: > > > expression="^SETUP_TRANSFER$"> > > data="do_transfer,*0#,exec:execute_extension,DO_TRANSFER XML > Incoming-FXO"/> > data="do_transfer"/> > > > > > > > data="bridge_pre_execute_bleg_app=execute_extension"/> > data="bridge_pre_execute_bleg_data=SETUP_TRANSFER XML Incoming-FXO"/> > > > > > expression="^DO_TRANSFER$"> > > > > > > > The transfer immediately starts ringing the entire house (1000). This > is true even when I am trying to get it to transfer to 1003. > > Also, http://jira.freeswitch.org/browse/OPENZAP-173 seems to be back in > that when the incoming phone hangs up, it seems to continue to ring. > Maybe I have this setup all wrong. > > I do not want anyone outside (the call originator) to be able to > initiate the transfer. I want the transferring party to be cut out of > the call. I want the outside and the *0#INTERNAL_EXT# to be connected. > > As for voicemail, I all FXS, no SIP for now. How do I set the password? > How would I setup a common voicemail box that every thing goes to except > one extension? That should have a separate box and password. > > Thank you for any help. > Trever > -- > I love dogs, but I hate Chihuahuas. A Chihuahua isn't a dog. It's a rat > with a thyroid problem. -- Unknown > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/207a8f8b/attachment.html From luis.daniel.lucio at gmail.com Tue Mar 27 08:23:15 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Mon, 26 Mar 2012 22:23:15 -0600 Subject: [Freeswitch-users] Cli command In-Reply-To: References: <4F696D4D.000010.15156@FLIGHTPC> Message-ID: :) Le 26 mars 2012 11:22, Michael Collins a ?crit : > > > 2012/3/26 Luis Daniel Lucio Quiroz >> >> gracias > > De nada. > > Also, don't forget that we have a number of Spanish-speaking FreeSWITCHers. > Check out #freeswitch-es on irc.freenode.net. > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jnankin at gmail.com Tue Mar 27 05:25:14 2012 From: jnankin at gmail.com (Josh) Date: Tue, 27 Mar 2012 01:25:14 +0000 (UTC) Subject: [Freeswitch-users] Compiling Latest GIT References: <4F481410.1080203@privatedemail.net> Message-ID: Brian Foster writes: > > > Gennady > > -j n isn't supported by the makefile, yet. I'd refrain from using it even if you're just throwing one thread at the cpu. > > -BDF > > > On Tue, Mar 20, 2012 at 7:50 PM, Gennady wrote: > Dome Charoenyost ...> writes: > > > > apt-get install gawk > > and then start first step > I've installed in Ubuntu 64-bit 11.04 without issues before and am now having > the exact same problem as described on a new Ubuntu 64-bit 11.10 server. > I've tried to do a ./bootstrap.sh and ./configure and make -j1 as was suggested > and gawk is installed and set up as alternative. But still same error. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting- YF8E+gPBBv73h3GqohbjpQ at public.gmane.orghttp://www.freeswitchsolutions.com > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > FreeSWITCH-users mailing listFreeSWITCH-users lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch- users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- usershttp://www.freeswitch.org > > > > > -- Brian D. FosterEndigo Computer LLCEmail: bdfoster- 15yuSvdC0LkqDJ6do+/SaQ at public.gmane.orgPhone: 317-800-7876 > Indianapolis, Indiana, USAThis message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at ... > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at ... > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > I'm having this problem as well on 11.10 and have installed gawk and rebootstrapped and reconfigured. Any ideas about this one? From jnankin at gmail.com Tue Mar 27 05:51:56 2012 From: jnankin at gmail.com (Joshua Nankin) Date: Mon, 26 Mar 2012 20:51:56 -0500 Subject: [Freeswitch-users] Trouble building Freeswitch on Ubuntu 11.10 Message-ID: I've followed all the steps at http://wiki.freeswitch.org/wiki/Quick_Start. I've seen similar issues in previous threads, and I've installed gawk and run make with -j1, but I'm still not able to do a clean build: quiet_libtool: compile: gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -fPIC -DPIC -o .libs/mod_xml_rpc.o quiet_libtool: compile: gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -o mod_xml_rpc.o >/dev/null 2>&1 Creating mod_xml_rpc.la... make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Full log at http://joshnankin.com/freeswitch.log -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120326/d2402601/attachment.html From gabe at gundy.org Tue Mar 27 10:13:29 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Tue, 27 Mar 2012 00:13:29 -0600 Subject: [Freeswitch-users] Trouble building Freeswitch on Ubuntu 11.10 In-Reply-To: References: Message-ID: On Mon, Mar 26, 2012 at 7:51 PM, Joshua Nankin wrote: > I've followed all the steps at?http://wiki.freeswitch.org/wiki/Quick_Start. > ?I've seen similar issues in previous threads, and I've installed gawk and > run make with -j1, but I'm still not able to do a clean build: Try this: https://parseltone.org/browser/trunk/misc/build_freeswitch.sh Gabe From peter.olsson at visionutveckling.se Tue Mar 27 10:18:29 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 27 Mar 2012 06:18:29 +0000 Subject: [Freeswitch-users] Trouble building Freeswitch on Ubuntu 11.10 Message-ID: <1FFF97C269757C458224B7C895F35F1507F851@cantor.std.visionutv.se> Please report to Jira, and make sure to attach the full build log - this doesn't show the actual error. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Joshua Nankin Skickat: den 27 mars 2012 03:52 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] Trouble building Freeswitch on Ubuntu 11.10 I've followed all the steps at http://wiki.freeswitch.org/wiki/Quick_Start. I've seen similar issues in previous threads, and I've installed gawk and run make with -j1, but I'm still not able to do a clean build: quiet_libtool: compile: gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -fPIC -DPIC -o .libs/mod_xml_rpc.o quiet_libtool: compile: gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -o mod_xml_rpc.o >/dev/null 2>&1 Creating mod_xml_rpc.la... make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Full log at http://joshnankin.com/freeswitch.log !DSPAM:4f7158e332761051920756! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/8f248e37/attachment-0001.html From peter.olsson at visionutveckling.se Tue Mar 27 12:47:22 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 27 Mar 2012 08:47:22 +0000 Subject: [Freeswitch-users] Binary download with curl Message-ID: <1FFF97C269757C458224B7C895F35F1507FF48@cantor.std.visionutv.se> If you follow the instructions on the wiki the build should work out of the box. The most common problem is that you've enabled autocrlf in git, which causes the build to break. If you still have problems after following the instructions, please report to Jira, since this is something that should be working. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Olli Aro Skickat: den 26 mars 2012 10:05 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] Binary download with curl Hi all, Been trying to compile Freeswitch on Windows in order to get in the XML curl module, however it is not looking easy (76 more fatal errors to figure out...) Before I start that task I thought it is always worth asking :) Would anyone happen to have a download link to already compiled Windows 32-bit binary for close to recent version of Freeswitch with the XML curl module? Regards, Olli !DSPAM:4f707e2932761654319646! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/7f21171e/attachment.html From fuji246 at gmail.com Tue Mar 27 13:44:45 2012 From: fuji246 at gmail.com (Fu Jiantao) Date: Tue, 27 Mar 2012 17:44:45 +0800 Subject: [Freeswitch-users] Binary download with curl In-Reply-To: <1FFF97C269757C458224B7C895F35F1507FF48@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1507FF48@cantor.std.visionutv.se> Message-ID: Hi , I've got some experience to share with you guys, if you're using WIN7, please turn off the "UAC", or else, the download vb script will failed to run. Hope this can help! 2012/3/27 Peter Olsson > If you follow the instructions on the wiki the build should work out of > the box. The most common problem is that you?ve enabled autocrlf in git, > which causes the build to break.**** > > ** ** > > If you still have problems after following the instructions, please report > to Jira, since this is something that should be working.**** > > ** ** > > /Peter**** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Olli Aro > *Skickat:* den 26 mars 2012 10:05 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* [Freeswitch-users] Binary download with curl**** > > ** ** > > ** ** > > Hi all,**** > > ** ** > > Been trying to compile Freeswitch on Windows in order to get in the XML > curl module, however it is not looking easy (76 more fatal errors to figure > out...)**** > > ** ** > > Before I start that task I thought it is always worth asking :)**** > > ** ** > > Would anyone happen to have a download link to already compiled Windows > 32-bit binary for close to recent version of Freeswitch with the XML curl > module?**** > > ** ** > > Regards,**** > > ** ** > > Olli**** > > ** ** > > ** ** > > !DSPAM:4f707e2932761654319646! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/4b1a58ad/attachment.html From anton.jugatsu at gmail.com Tue Mar 27 14:07:57 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Tue, 27 Mar 2012 14:07:57 +0400 Subject: [Freeswitch-users] Problem with outbound call thru ITSP (MERA MVTS3G v.4.4.0-20) Message-ID: Hello guys. I have setup the gateway for outbound calls with credentials provided by my ITSP. http://pastebin.freeswitch.org/18749 Trying to make test call using, for example, originate command originate sofia/gateway/mtt.ru/XXXX &txfax(/tmp/fax.tif) result: ITSP proxy sends 500 Gateway is Invalid http://pastebin.freeswitch.org/18750 But, trying to make call using asterisk with the same credentials suprisenly works. Here dump of success call http://pastebin.freeswitch.org/18751 I also attach logs that provided mys ITSP. Don't mind that there is ip address from privite LAN, the first try was from behind NAT. The second, that i provided above (http://pastebin.freeswitch.org/18750), was from vps with staticly assigned address. For the first glance I thought that the root of this problem was NAT but IMO it's not. Also, my external.xml http://pastebin.freeswitch.org/18752 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/45e50d4a/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: 123.pcap Type: application/octet-stream Size: 44565 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/45e50d4a/attachment-0001.obj From peter.olsson at visionutveckling.se Tue Mar 27 14:08:12 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 27 Mar 2012 10:08:12 +0000 Subject: [Freeswitch-users] Binary download with curl Message-ID: <1FFF97C269757C458224B7C895F35F15080018@cantor.std.visionutv.se> Thanks for the input. But I think this is only true if you put the sources under a protected directory, like "C:\Program Files"? If the sources are put somewhere else it should work even with UAC enabled (at least it does for me :)). /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Fu Jiantao Skickat: den 27 mars 2012 11:45 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Binary download with curl Hi , I've got some experience to share with you guys, if you're using WIN7, please turn off the "UAC", or else, the download vb script will failed to run. Hope this can help! 2012/3/27 Peter Olsson > If you follow the instructions on the wiki the build should work out of the box. The most common problem is that you've enabled autocrlf in git, which causes the build to break. If you still have problems after following the instructions, please report to Jira, since this is something that should be working. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Olli Aro Skickat: den 26 mars 2012 10:05 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] Binary download with curl Hi all, Been trying to compile Freeswitch on Windows in order to get in the XML curl module, however it is not looking easy (76 more fatal errors to figure out...) Before I start that task I thought it is always worth asking :) Would anyone happen to have a download link to already compiled Windows 32-bit binary for close to recent version of Freeswitch with the XML curl module? Regards, Olli _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f718ac732761986176360! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/fc4a044c/attachment.html From trever.adams at gmail.com Tue Mar 27 14:11:02 2012 From: trever.adams at gmail.com (Trever L. Adams) Date: Tue, 27 Mar 2012 04:11:02 -0600 Subject: [Freeswitch-users] Trouble with transfer and voicemail password Message-ID: <4F719236.9030201@gmail.com> > go ahead and get a debug log of a call coming in, being answered, and the > recipient dialing *0#xxx# to send the call elsewhere. Drop it into > pastebin.freeswitch.org and use "FreeSWITCH Log" as the syntax highlighting. > > -MC http://pastebin.freeswitch.org/18753 I am not sure how much the log will help. It doesn't show me pushing anything after the transfer code (*01 as I was experimenting and forgot to change it back to *0# as I had it set before). If I dial any thing 1000, 1000#, 1001, 1001# it all fails and plays the invalid_extension.wav. I have changed the code a little bit. It now reads: It seems that I had a problem with play and get digits not working when I was working on my call screener (I stopped work do to bug openzap-137). I seemed to have gotten around it by rewriting the entire thing in lua. I may be remembering incorrectly as this was months ago. Thank you for any help, Trever -- "Selfishness is really self-destruction in slow motion." -? Elder Neal A. Maxwell - Ensign, May 1999, 23 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/89b9fcc5/attachment.bin From farooqhussain786 at gmail.com Tue Mar 27 14:19:59 2012 From: farooqhussain786 at gmail.com (Farooq Hussain) Date: Tue, 27 Mar 2012 15:19:59 +0500 Subject: [Freeswitch-users] How to configure Bandwidth.com Message-ID: Hello Everyone, Please let me know how would we configure bandwith.com SIP turnk. Please help us. -- Thanks Farooq Hussain -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/d03199c6/attachment.html From jaybinks at gmail.com Tue Mar 27 14:26:21 2012 From: jaybinks at gmail.com (jay binks) Date: Tue, 27 Mar 2012 20:26:21 +1000 Subject: [Freeswitch-users] Problem with outbound call thru ITSP (MERA MVTS3G v.4.4.0-20) In-Reply-To: References: Message-ID: your not getting very far at all.. try enabling more codecs... there is something in your invite ( or SDP ) that they dont like. but their logs tell you nothing more than the pcap you provided. I can assure you MERA DOES work... I send millions of calls to MERA servers every day from my freeswitch boxes. so this is something specific to your setup. Id also suggest the ITSP should be able to give you more info and be a little more helpful. Jay On 27 March 2012 20:07, Anton Kvashenkin wrote: > Hello guys. I have setup the gateway for outbound calls with credentials > provided by my ITSP. > > http://pastebin.freeswitch.org/18749 > > Trying to make test call using, for example, originate command > > originate sofia/gateway/mtt.ru/XXXX &txfax(/tmp/fax.tif) > > result: > > ITSP proxy sends 500 Gateway is Invalid > http://pastebin.freeswitch.org/18750 > > But, trying to make call using asterisk with the same credentials > suprisenly works. Here dump of success call > http://pastebin.freeswitch.org/18751 > > I also attach logs that provided mys ITSP. Don't mind that there is ip > address from privite LAN, the first try was from behind NAT. The second, > that i provided above (http://pastebin.freeswitch.org/18750), was from > vps with staticly assigned address. For the first glance I thought that the > root of this problem was NAT but IMO it's not. > > Also, my external.xml > > http://pastebin.freeswitch.org/18752 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/b0f4a8c6/attachment-0001.html From anton.jugatsu at gmail.com Tue Mar 27 14:27:13 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Tue, 27 Mar 2012 14:27:13 +0400 Subject: [Freeswitch-users] How to configure Bandwidth.com In-Reply-To: References: Message-ID: Use wiki http://wiki.freeswitch.com/wiki/Provider_Configuration:_Bandwidth.com 27 ????? 2012 ?. 14:19 ???????????? Farooq Hussain < farooqhussain786 at gmail.com> ???????: > Hello Everyone, > > Please let me know how would we configure bandwith.com SIP turnk. Please > help us. > > -- > Thanks > > Farooq Hussain > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/03a7406b/attachment.html From farooqhussain786 at gmail.com Tue Mar 27 14:57:00 2012 From: farooqhussain786 at gmail.com (Farooq Hussain) Date: Tue, 27 Mar 2012 15:57:00 +0500 Subject: [Freeswitch-users] How to configure Bandwidth.com In-Reply-To: References: Message-ID: Yeah I know about that link. The information I got from bandwidth.com is only a IP and they say please open you port 5060. They use IP authentication. More help required other then this link. Also on which path I have to configure inbound and outbound rule. please help me 2012/3/27 Anton Kvashenkin > Use wiki > http://wiki.freeswitch.com/wiki/Provider_Configuration:_Bandwidth.com > > 27 ????? 2012 ?. 14:19 ???????????? Farooq Hussain < > farooqhussain786 at gmail.com> ???????: > >> Hello Everyone, >> >> Please let me know how would we configure bandwith.com SIP turnk. Please >> help us. >> >> -- >> Thanks >> >> Farooq Hussain >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thanks Farooq Hussain -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/d3156c46/attachment.html From bdfoster at endigotech.com Tue Mar 27 14:58:43 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 27 Mar 2012 06:58:43 -0400 Subject: [Freeswitch-users] Trouble with transfer and voicemail password In-Reply-To: <4F719236.9030201@gmail.com> References: <4F719236.9030201@gmail.com> Message-ID: If it's anything like the Linksys phones, you might have to change the dialplan on the phone itself in order to dial certain digits and make sure all if it is sent to the switch instead of holding some of the dialed digits back. Double check this to make sure you aren't being sent on a wild goose chase. -BDF On Mar 27, 2012 6:11 AM, "Trever L. Adams" wrote: > > go ahead and get a debug log of a call coming in, being answered, and the > > recipient dialing *0#xxx# to send the call elsewhere. Drop it into > > pastebin.freeswitch.org and use "FreeSWITCH Log" as the syntax > highlighting. > > > > -MC > http://pastebin.freeswitch.org/18753 > > I am not sure how much the log will help. It doesn't show me pushing > anything after the transfer code (*01 as I was experimenting and forgot > to change it back to *0# as I had it set before). > > If I dial any thing 1000, 1000#, 1001, 1001# it all fails and plays the > invalid_extension.wav. I have changed the code a little bit. It now reads: > > > expression="^SETUP_TRANSFER$"> > > data="do_transfer,*01,exec:execute_extension,DO_TRANSFER XML > Incoming-FXO"/> > data="do_transfer"/> > > > > > > data="bridge_pre_execute_bleg_app=execute_extension"/> > data="bridge_pre_execute_bleg_data=SETUP_TRANSFER XML Incoming-FXO"/> > > > > > expression="^DO_TRANSFER$"> > > > > > > > > It seems that I had a problem with play and get digits not working when > I was working on my call screener (I stopped work do to bug > openzap-137). I seemed to have gotten around it by rewriting the entire > thing in lua. I may be remembering incorrectly as this was months ago. > > Thank you for any help, > Trever > -- > "Selfishness is really self-destruction in slow motion." -? Elder Neal > A. Maxwell - Ensign, May 1999, 23 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/e658aa7d/attachment.html From trever.adams at gmail.com Tue Mar 27 15:04:16 2012 From: trever.adams at gmail.com (Trever L. Adams) Date: Tue, 27 Mar 2012 05:04:16 -0600 Subject: [Freeswitch-users] [SPAM] Re: Trouble with transfer and voicemail password In-Reply-To: References: <4F719236.9030201@gmail.com> Message-ID: <4F719EB0.6040803@gmail.com> On 03/27/2012 04:58 AM, Brian Foster wrote: > > If it's anything like the Linksys phones, you might have to change the > dialplan on the phone itself in order to dial certain digits and make > sure all if it is sent to the switch instead of holding some of the > dialed digits back. Double check this to make sure you aren't being > sent on a wild goose chase. > > -BDF > BDF, Thank you, but this are actual POTS phones on FXS ports. Thank you for the suggestion. Trever -- "The world is full of people who have never, since childhood, met an open doorway with an open mind." -- E.B. White -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/be361d68/attachment-0001.bin From bdfoster at endigotech.com Tue Mar 27 15:04:37 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 27 Mar 2012 07:04:37 -0400 Subject: [Freeswitch-users] How to configure Bandwidth.com In-Reply-To: References: Message-ID: You have a few options. 1. You can usually tell the ITSP to send calls to 5080 (just add: 5080 behind your IP) 2. If that doesn't work, there are ways of having everything on one IP/port combo. There are security implications with this if you don't know what you are doing. 3. Safe bet: grab another IP and use it to talk to the ITSP. -BDF On Mar 27, 2012 6:58 AM, "Farooq Hussain" wrote: > Yeah I know about that link. The information I got from bandwidth.com is > only a IP and they say please open you port 5060. They use > IP authentication. > > More help required other then this link. Also on which path I have to > configure inbound and outbound rule. please help me > > 2012/3/27 Anton Kvashenkin > >> Use wiki >> http://wiki.freeswitch.com/wiki/Provider_Configuration:_Bandwidth.com >> >> 27 ????? 2012 ?. 14:19 ???????????? Farooq Hussain < >> farooqhussain786 at gmail.com> ???????: >> >>> Hello Everyone, >>> >>> Please let me know how would we configure bandwith.com SIP turnk. >>> Please help us. >>> >>> -- >>> Thanks >>> >>> Farooq Hussain >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Thanks > > Farooq Hussain > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/f3d3b92f/attachment.html From B.Tietz at pinguin.ag Tue Mar 27 15:07:08 2012 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Tue, 27 Mar 2012 13:07:08 +0200 Subject: [Freeswitch-users] How to configure Bandwidth.com In-Reply-To: References: Message-ID: <07BF4904977CC645B485E970424193AD0FF1324850@localhost> If you have to use IP-Auth make a user for bandwith.com with the IP as cidr as metioned here: http://wiki.freeswitch.org/wiki/Acl#Users for outgoing calls just make a bridge to sofia/external/$1@ regards Benjamin Yeah I know about that link. The information I got from bandwidth.com is only a IP and they say please open you port 5060. They use IP authentication. More help required other then this link. Also on which path I have to configure inbound and outbound rule. please help me 2012/3/27 Anton Kvashenkin > Use wiki http://wiki.freeswitch.com/wiki/Provider_Configuration:_Bandwidth.com 27 ????? 2012 ?. 14:19 ???????????? Farooq Hussain > ???????: Hello Everyone, Please let me know how would we configure bandwith.com SIP turnk. Please help us. -- Thanks Farooq Hussain _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Thanks Farooq Hussain -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/39e117cf/attachment.html From anton.jugatsu at gmail.com Tue Mar 27 15:17:18 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Tue, 27 Mar 2012 15:17:18 +0400 Subject: [Freeswitch-users] Problem with outbound call thru ITSP (MERA MVTS3G v.4.4.0-20) In-Reply-To: References: Message-ID: Thanks for the reply. Here sofia status profile external http://pastebin.freeswitch.org/18754 Hm... You are right... WTF with codecs @ SDP. originate {absolute_codec_string='PCMU,PCMA,G729'}sofia/gateway/ mtt.ru/89093848124 &playback() and INVITE i have U 2012/03/27 11:12:45.244216 62.76.180.83:5080 -> 80.75.130.134:5060 INVITE sip:89093848124 at sip.mtt.ru SIP/2.0. Via: SIP/2.0/UDP 62.76.180.83:5080;rport;branch=z9hG4bKy0HX29D6rc86K. Max-Forwards: 70. From: "" ;tag=7USXQ5jg6caFB. To: . Call-ID: 9ea5c38c-f2a0-122f-fd98-00163e000f5d. CSeq: 26087318 INVITE. Contact: . User-Agent: IP-PBX. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 201. X-FS-Support: update_display,send_info. Remote-Party-ID: ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1332822661 1332822662 IN IP4 62.76.180.83. s=FreeSWITCH. c=IN IP4 62.76.180.83. t=0 0. m=audio 24104 RTP/AVP 0 101 13. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. The same... I'm little bit frustrating :) 27 ????? 2012 ?. 14:26 ???????????? jay binks ???????: > your not getting very far at all.. > > try enabling more codecs... there is something in your invite ( or SDP ) > that they dont like. > but their logs tell you nothing more than the pcap you provided. > > I can assure you MERA DOES work... I send millions of calls to MERA > servers every day from my freeswitch boxes. > so this is something specific to your setup. > > Id also suggest the ITSP should be able to give you more info and be a > little more helpful. > > Jay > > On 27 March 2012 20:07, Anton Kvashenkin wrote: > >> Hello guys. I have setup the gateway for outbound calls with credentials >> provided by my ITSP. >> >> http://pastebin.freeswitch.org/18749 >> >> Trying to make test call using, for example, originate command >> >> originate sofia/gateway/mtt.ru/XXXX &txfax(/tmp/fax.tif) >> >> result: >> >> ITSP proxy sends 500 Gateway is Invalid >> http://pastebin.freeswitch.org/18750 >> >> But, trying to make call using asterisk with the same credentials >> suprisenly works. Here dump of success call >> http://pastebin.freeswitch.org/18751 >> >> I also attach logs that provided mys ITSP. Don't mind that there is ip >> address from privite LAN, the first try was from behind NAT. The second, >> that i provided above (http://pastebin.freeswitch.org/18750), was from >> vps with staticly assigned address. For the first glance I thought that the >> root of this problem was NAT but IMO it's not. >> >> Also, my external.xml >> >> http://pastebin.freeswitch.org/18752 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/26652986/attachment-0001.html From b2m at a-cti.com Tue Mar 27 15:17:47 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Tue, 27 Mar 2012 16:47:47 +0530 Subject: [Freeswitch-users] How to configure Bandwidth.com In-Reply-To: <07BF4904977CC645B485E970424193AD0FF1324850@localhost> References: <07BF4904977CC645B485E970424193AD0FF1324850@localhost> Message-ID: I had the same issue, and it worked when I move the xml to /usr/local/freeswitch/conf/sip_profiles/internal/bandwidth.conf.xml make sure your vars.xml as like this(guess its default) bridge to : sofia/gateway/bandwidth.com/+1xxxxxxxxx Thanks, Bala 2012/3/27 > If you have to use IP-Auth make a user for bandwith.com with the IP as > cidr as metioned here: http://wiki.freeswitch.org/wiki/Acl#Users**** > > ** ** > > for outgoing calls just make a bridge to sofia/external/$1@ > **** > > ** ** > > regards**** > > Benjamin**** > > ** ** > > ** ** > > Yeah I know about that link. The information I got from bandwidth.com is > only a IP and they say please open you port 5060. They use > IP authentication. **** > > ** ** > > More help required other then this link. Also on which path I have to > configure inbound and outbound rule. please help me**** > > 2012/3/27 Anton Kvashenkin **** > > Use wiki > http://wiki.freeswitch.com/wiki/Provider_Configuration:_Bandwidth.com **** > > 27 ????? 2012 ?. 14:19 ???????????? Farooq Hussain < > farooqhussain786 at gmail.com> ???????:**** > > Hello Everyone,**** > > ** ** > > Please let me know how would we configure bandwith.com SIP turnk. Please > help us. > **** > > ** ** > > -- > Thanks > > Farooq Hussain**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Thanks > > Farooq Hussain**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/ef9e8d1f/attachment.html From bdfoster at endigotech.com Tue Mar 27 15:26:42 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 27 Mar 2012 07:26:42 -0400 Subject: [Freeswitch-users] How to configure Bandwidth.com In-Reply-To: <07BF4904977CC645B485E970424193AD0FF1324850@localhost> References: <07BF4904977CC645B485E970424193AD0FF1324850@localhost> Message-ID: You shouldn't need the cidr but it's good if you have a closed system. You wouldn't change the way you route calls, you're going to be doing your matches in the public context against the destination number. 2012/3/27 > If you have to use IP-Auth make a user for bandwith.com with the IP as > cidr as metioned here: http://wiki.freeswitch.org/wiki/Acl#Users**** > > ** ** > > for outgoing calls just make a bridge to sofia/external/$1@ > **** > > ** ** > > regards**** > > Benjamin**** > > ** ** > > ** ** > > Yeah I know about that link. The information I got from bandwidth.com is > only a IP and they say please open you port 5060. They use > IP authentication. **** > > ** ** > > More help required other then this link. Also on which path I have to > configure inbound and outbound rule. please help me**** > > 2012/3/27 Anton Kvashenkin **** > > Use wiki > http://wiki.freeswitch.com/wiki/Provider_Configuration:_Bandwidth.com **** > > 27 ????? 2012 ?. 14:19 ???????????? Farooq Hussain < > farooqhussain786 at gmail.com> ???????:**** > > Hello Everyone,**** > > ** ** > > Please let me know how would we configure bandwith.com SIP turnk. Please > help us. > **** > > ** ** > > -- > Thanks > > Farooq Hussain**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Thanks > > Farooq Hussain**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/c1e4af82/attachment-0001.html From bdfoster at endigotech.com Tue Mar 27 15:28:25 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 27 Mar 2012 07:28:25 -0400 Subject: [Freeswitch-users] How to configure Bandwidth.com In-Reply-To: References: <07BF4904977CC645B485E970424193AD0FF1324850@localhost> Message-ID: Just remember there are some security implications registering to a carrier with your internal profile. Do your research. -BDF 2012/3/27 Balamurugan Mahendran > I had the same issue, and it worked when I move the xml > to /usr/local/freeswitch/conf/sip_profiles/internal/bandwidth.conf.xml > > > > > > > > > > > > > > > > > make sure your vars.xml as like this(guess its default) > > > > > > > > > > > > > > > > > bridge to : > > sofia/gateway/bandwidth.com/+1xxxxxxxxx > > > Thanks, > Bala > > > > 2012/3/27 > > If you have to use IP-Auth make a user for bandwith.com with the IP as >> cidr as metioned here: http://wiki.freeswitch.org/wiki/Acl#Users**** >> >> ** ** >> >> for outgoing calls just make a bridge to sofia/external/$1@ >> **** >> >> ** ** >> >> regards**** >> >> Benjamin**** >> >> ** ** >> >> ** ** >> >> Yeah I know about that link. The information I got from bandwidth.com is >> only a IP and they say please open you port 5060. They use >> IP authentication. **** >> >> ** ** >> >> More help required other then this link. Also on which path I have to >> configure inbound and outbound rule. please help me**** >> >> 2012/3/27 Anton Kvashenkin **** >> >> Use wiki >> http://wiki.freeswitch.com/wiki/Provider_Configuration:_Bandwidth.com *** >> * >> >> 27 ????? 2012 ?. 14:19 ???????????? Farooq Hussain < >> farooqhussain786 at gmail.com> ???????:**** >> >> Hello Everyone,**** >> >> ** ** >> >> Please let me know how would we configure bandwith.com SIP turnk. Please >> help us. >> **** >> >> ** ** >> >> -- >> Thanks >> >> Farooq Hussain**** >> >> ** ** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> >> >> **** >> >> ** ** >> >> -- >> Thanks >> >> Farooq Hussain**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/9dfbf6a8/attachment.html From anton.jugatsu at gmail.com Tue Mar 27 15:32:39 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Tue, 27 Mar 2012 15:32:39 +0400 Subject: [Freeswitch-users] How to configure Bandwidth.com In-Reply-To: References: <07BF4904977CC645B485E970424193AD0FF1324850@localhost> Message-ID: Brian, for example? 27 ????? 2012 ?. 15:28 ???????????? Brian Foster ???????: > Just remember there are some security implications registering to a > carrier with your internal profile. Do your research. > > -BDF > > > 2012/3/27 Balamurugan Mahendran > >> I had the same issue, and it worked when I move the xml >> to /usr/local/freeswitch/conf/sip_profiles/internal/bandwidth.conf.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> make sure your vars.xml as like this(guess its default) >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> bridge to : >> >> sofia/gateway/bandwidth.com/+1xxxxxxxxx >> >> >> Thanks, >> Bala >> >> >> >> 2012/3/27 >> >> If you have to use IP-Auth make a user for bandwith.com with the IP as >>> cidr as metioned here: http://wiki.freeswitch.org/wiki/Acl#Users**** >>> >>> ** ** >>> >>> for outgoing calls just make a bridge to sofia/external/$1@ >>> **** >>> >>> ** ** >>> >>> regards**** >>> >>> Benjamin**** >>> >>> ** ** >>> >>> ** ** >>> >>> Yeah I know about that link. The information I got from bandwidth.comis only a IP and they say please open you port 5060. They use >>> IP authentication. **** >>> >>> ** ** >>> >>> More help required other then this link. Also on which path I have to >>> configure inbound and outbound rule. please help me**** >>> >>> 2012/3/27 Anton Kvashenkin **** >>> >>> Use wiki >>> http://wiki.freeswitch.com/wiki/Provider_Configuration:_Bandwidth.com ** >>> ** >>> >>> 27 ????? 2012 ?. 14:19 ???????????? Farooq Hussain < >>> farooqhussain786 at gmail.com> ???????:**** >>> >>> Hello Everyone,**** >>> >>> ** ** >>> >>> Please let me know how would we configure bandwith.com SIP turnk. >>> Please help us. >>> **** >>> >>> ** ** >>> >>> -- >>> Thanks >>> >>> Farooq Hussain**** >>> >>> ** ** >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org**** >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org**** >>> >>> >>> >>> **** >>> >>> ** ** >>> >>> -- >>> Thanks >>> >>> Farooq Hussain**** >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/3c7501ec/attachment-0001.html From jaybinks at gmail.com Tue Mar 27 15:33:29 2012 From: jaybinks at gmail.com (jay binks) Date: Tue, 27 Mar 2012 21:33:29 +1000 Subject: [Freeswitch-users] Problem with outbound call thru ITSP (MERA MVTS3G v.4.4.0-20) In-Reply-To: References: Message-ID: they may also be rejecting based on your caller id.. seen that before From: ;tag=B36y48m258Uga and Remote-Party-ID: ;party=calling;screen=yes;privacy=off. dont look like its legit caller id ... Jay On 27 March 2012 21:17, Anton Kvashenkin wrote: > Thanks for the reply. Here > > sofia status profile external http://pastebin.freeswitch.org/18754 > > Hm... You are right... WTF with codecs @ SDP. > > originate {absolute_codec_string='PCMU,PCMA,G729'}sofia/gateway/ > mtt.ru/89093848124 &playback() > > and INVITE i have > > U 2012/03/27 11:12:45.244216 62.76.180.83:5080 -> 80.75.130.134:5060 > INVITE sip:89093848124 at sip.mtt.ru SIP/2.0. > Via: SIP/2.0/UDP 62.76.180.83:5080;rport;branch=z9hG4bKy0HX29D6rc86K. > Max-Forwards: 70. > From: "" ;tag=7USXQ5jg6caFB. > To: . > Call-ID: 9ea5c38c-f2a0-122f-fd98-00163e000f5d. > CSeq: 26087318 INVITE. > Contact: . > User-Agent: IP-PBX. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, hold, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 201. > X-FS-Support: update_display,send_info. > Remote-Party-ID: >;party=calling;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1332822661 1332822662 IN IP4 62.76.180.83. > s=FreeSWITCH. > c=IN IP4 62.76.180.83. > t=0 0. > m=audio 24104 RTP/AVP 0 101 13. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > > > The same... I'm little bit frustrating :) > > 27 ????? 2012 ?. 14:26 ???????????? jay binks ???????: > > your not getting very far at all.. >> >> try enabling more codecs... there is something in your invite ( or SDP ) >> that they dont like. >> but their logs tell you nothing more than the pcap you provided. >> >> I can assure you MERA DOES work... I send millions of calls to MERA >> servers every day from my freeswitch boxes. >> so this is something specific to your setup. >> >> Id also suggest the ITSP should be able to give you more info and be a >> little more helpful. >> >> Jay >> >> On 27 March 2012 20:07, Anton Kvashenkin wrote: >> >>> Hello guys. I have setup the gateway for outbound calls with credentials >>> provided by my ITSP. >>> >>> http://pastebin.freeswitch.org/18749 >>> >>> Trying to make test call using, for example, originate command >>> >>> originate sofia/gateway/mtt.ru/XXXX &txfax(/tmp/fax.tif) >>> >>> result: >>> >>> ITSP proxy sends 500 Gateway is Invalid >>> http://pastebin.freeswitch.org/18750 >>> >>> But, trying to make call using asterisk with the same credentials >>> suprisenly works. Here dump of success call >>> http://pastebin.freeswitch.org/18751 >>> >>> I also attach logs that provided mys ITSP. Don't mind that there is ip >>> address from privite LAN, the first try was from behind NAT. The second, >>> that i provided above (http://pastebin.freeswitch.org/18750), was from >>> vps with staticly assigned address. For the first glance I thought that the >>> root of this problem was NAT but IMO it's not. >>> >>> Also, my external.xml >>> >>> http://pastebin.freeswitch.org/18752 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Sincerely >> >> Jay >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/bd24fbbe/attachment.html From avi at avimarcus.net Tue Mar 27 15:54:18 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 27 Mar 2012 13:54:18 +0200 Subject: [Freeswitch-users] How to configure Bandwidth.com In-Reply-To: References: <07BF4904977CC645B485E970424193AD0FF1324850@localhost> Message-ID: .. unless you set their context to "public". Then you're good. -Avi 2012/3/27 Brian Foster > Just remember there are some security implications registering to a > carrier with your internal profile. Do your research. > > -BDF > > > 2012/3/27 Balamurugan Mahendran > >> I had the same issue, and it worked when I move the xml >> to /usr/local/freeswitch/conf/sip_profiles/internal/bandwidth.conf.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> make sure your vars.xml as like this(guess its default) >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> bridge to : >> >> sofia/gateway/bandwidth.com/+1xxxxxxxxx >> >> >> Thanks, >> Bala >> >> >> >> 2012/3/27 >> >> If you have to use IP-Auth make a user for bandwith.com with the IP as >>> cidr as metioned here: http://wiki.freeswitch.org/wiki/Acl#Users**** >>> >>> ** ** >>> >>> for outgoing calls just make a bridge to sofia/external/$1@ >>> **** >>> >>> ** ** >>> >>> regards**** >>> >>> Benjamin**** >>> >>> ** ** >>> >>> ** ** >>> >>> Yeah I know about that link. The information I got from bandwidth.comis only a IP and they say please open you port 5060. They use >>> IP authentication. **** >>> >>> ** ** >>> >>> More help required other then this link. Also on which path I have to >>> configure inbound and outbound rule. please help me**** >>> >>> 2012/3/27 Anton Kvashenkin **** >>> >>> Use wiki >>> http://wiki.freeswitch.com/wiki/Provider_Configuration:_Bandwidth.com ** >>> ** >>> >>> 27 ????? 2012 ?. 14:19 ???????????? Farooq Hussain < >>> farooqhussain786 at gmail.com> ???????:**** >>> >>> Hello Everyone,**** >>> >>> ** ** >>> >>> Please let me know how would we configure bandwith.com SIP turnk. >>> Please help us. >>> **** >>> >>> ** ** >>> >>> -- >>> Thanks >>> >>> Farooq Hussain**** >>> >>> ** ** >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org**** >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org**** >>> >>> >>> >>> **** >>> >>> ** ** >>> >>> -- >>> Thanks >>> >>> Farooq Hussain**** >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/2300bac4/attachment-0001.html From anton.jugatsu at gmail.com Tue Mar 27 16:05:42 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Tue, 27 Mar 2012 16:05:42 +0400 Subject: [Freeswitch-users] Problem with outbound call thru ITSP (MERA MVTS3G v.4.4.0-20) In-Reply-To: References: Message-ID: Big thanks, Jay. Now it works as aspected ;) originate {sip_cid_type=none}sofia/gateway/mtt.ru/XXXX&txfax(/home/kvashenkin/txfax-sample.tiff) U 2012/03/27 11:47:03.635333 62.76.180.83:5080 -> 80.75.130.134:5060 INVITE sip:89093848124 at sip.mtt.ru SIP/2.0. Via: SIP/2.0/UDP 62.76.180.83:5080;rport;branch=z9hG4bK79pvrF70rHvHN. Max-Forwards: 70. From: "" ;tag=NB3BQ3tK7germ. To: . Call-ID: 6997f9d4-f2a5-122f-fd98-00163e000f5d. CSeq: 26088347 INVITE. Contact: . User-Agent: IP-PBX. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 205. X-FS-Support: update_display,send_info. . v=0. o=FreeSWITCH 1332829045 1332829046 IN IP4 62.76.180.83. s=FreeSWITCH. c=IN IP4 62.76.180.83. t=0 0. m=audio 19778 RTP/AVP 0 8 3 101 13. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. I disable Remote-party-id by providing {sip_cid_type=none} variable. The last thing is added _this_ config param (what param) to mtt.xlm gateway. 27 ????? 2012 ?. 15:33 ???????????? jay binks ???????: > they may also be rejecting based on your caller id.. > seen that before > > From: ;tag=B36y48m258Uga > and > Remote-Party-ID: >;party=calling;screen=yes;privacy=off. > > dont look like its legit caller id ... > > > Jay > > > > On 27 March 2012 21:17, Anton Kvashenkin wrote: > >> Thanks for the reply. Here >> >> sofia status profile external http://pastebin.freeswitch.org/18754 >> >> Hm... You are right... WTF with codecs @ SDP. >> >> originate {absolute_codec_string='PCMU,PCMA,G729'}sofia/gateway/ >> mtt.ru/89093848124 &playback() >> >> and INVITE i have >> >> U 2012/03/27 11:12:45.244216 62.76.180.83:5080 -> 80.75.130.134:5060 >> INVITE sip:89093848124 at sip.mtt.ru SIP/2.0. >> Via: SIP/2.0/UDP 62.76.180.83:5080;rport;branch=z9hG4bKy0HX29D6rc86K. >> Max-Forwards: 70. >> From: "" ;tag=7USXQ5jg6caFB. >> To: . >> Call-ID: 9ea5c38c-f2a0-122f-fd98-00163e000f5d. >> CSeq: 26087318 INVITE. >> Contact: . >> User-Agent: IP-PBX. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY. >> Supported: timer, precondition, path, replaces. >> Allow-Events: talk, hold, refer. >> Content-Type: application/sdp. >> Content-Disposition: session. >> Content-Length: 201. >> X-FS-Support: update_display,send_info. >> Remote-Party-ID: > >;party=calling;screen=yes;privacy=off. >> . >> v=0. >> o=FreeSWITCH 1332822661 1332822662 IN IP4 62.76.180.83. >> s=FreeSWITCH. >> c=IN IP4 62.76.180.83. >> t=0 0. >> m=audio 24104 RTP/AVP 0 101 13. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=ptime:20. >> >> >> The same... I'm little bit frustrating :) >> >> 27 ????? 2012 ?. 14:26 ???????????? jay binks ???????: >> >> your not getting very far at all.. >>> >>> try enabling more codecs... there is something in your invite ( or SDP ) >>> that they dont like. >>> but their logs tell you nothing more than the pcap you provided. >>> >>> I can assure you MERA DOES work... I send millions of calls to MERA >>> servers every day from my freeswitch boxes. >>> so this is something specific to your setup. >>> >>> Id also suggest the ITSP should be able to give you more info and be a >>> little more helpful. >>> >>> Jay >>> >>> On 27 March 2012 20:07, Anton Kvashenkin wrote: >>> >>>> Hello guys. I have setup the gateway for outbound calls >>>> with credentials provided by my ITSP. >>>> >>>> http://pastebin.freeswitch.org/18749 >>>> >>>> Trying to make test call using, for example, originate command >>>> >>>> originate sofia/gateway/mtt.ru/XXXX &txfax(/tmp/fax.tif) >>>> >>>> result: >>>> >>>> ITSP proxy sends 500 Gateway is Invalid >>>> http://pastebin.freeswitch.org/18750 >>>> >>>> But, trying to make call using asterisk with the same credentials >>>> suprisenly works. Here dump of success call >>>> http://pastebin.freeswitch.org/18751 >>>> >>>> I also attach logs that provided mys ITSP. Don't mind that there is ip >>>> address from privite LAN, the first try was from behind NAT. The second, >>>> that i provided above (http://pastebin.freeswitch.org/18750), was from >>>> vps with staticly assigned address. For the first glance I thought that the >>>> root of this problem was NAT but IMO it's not. >>>> >>>> Also, my external.xml >>>> >>>> http://pastebin.freeswitch.org/18752 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Sincerely >>> >>> Jay >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/bf8110d0/attachment.html From tpe at actimizer.com Tue Mar 27 16:06:30 2012 From: tpe at actimizer.com (Tor Petterson) Date: Tue, 27 Mar 2012 14:06:30 +0200 Subject: [Freeswitch-users] Large delay/latency when bridging SIP calls In-Reply-To: References: Message-ID: 2012/3/26 Anthony Minessale : > please try HEAD and please report bugs on JIRA not here going forward. It works now thanks! Sorry about not using JIRA will do in the future -- Tor Petterson tpe at actimizer.com Tobaksvejen 25, 2. tv. - 2860 S?borg Telephone: +45 39 55 05 32 www.actimizer.com From vipkilla at gmail.com Tue Mar 27 16:07:46 2012 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 27 Mar 2012 08:07:46 -0400 Subject: [Freeswitch-users] setting caller ID on a call to Asterisk Message-ID: When I send a call from FreeSWITCH to Asterisk, I cannot get Asterisk to properly display the caller ID. I have my FS set to register to Asterisk as a Gateway like sip:00019924 at testdomain.com. Asterisk always seems to use the '00019924' in FROM: 00019924 at testdomain.com as the callerID. I tried setting the 'sip-force-user' directory variable but that did not work. From peter.olsson at visionutveckling.se Tue Mar 27 16:13:26 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 27 Mar 2012 12:13:26 +0000 Subject: [Freeswitch-users] setting caller ID on a call to Asterisk Message-ID: <1FFF97C269757C458224B7C895F35F150804C9@cantor.std.visionutv.se> Try to set caller-id-in-from to true on the gateway config. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Vik Killa Skickat: den 27 mars 2012 14:08 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] setting caller ID on a call to Asterisk When I send a call from FreeSWITCH to Asterisk, I cannot get Asterisk to properly display the caller ID. I have my FS set to register to Asterisk as a Gateway like sip:00019924 at testdomain.com. Asterisk always seems to use the '00019924' in FROM: 00019924 at testdomain.com as the callerID. I tried setting the 'sip-force-user' directory variable but that did not work. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f71abfb32761977921347! From vipkilla at gmail.com Tue Mar 27 16:52:52 2012 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 27 Mar 2012 08:52:52 -0400 Subject: [Freeswitch-users] setting caller ID on a call to Asterisk In-Reply-To: <1FFF97C269757C458224B7C895F35F150804C9@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F150804C9@cantor.std.visionutv.se> Message-ID: That did it! Thanks! On Tue, Mar 27, 2012 at 8:13 AM, Peter Olsson wrote: > Try to set caller-id-in-from to true on the gateway config. > > /Peter > From miha at softnet.si Tue Mar 27 16:58:35 2012 From: miha at softnet.si (Miha) Date: Tue, 27 Mar 2012 14:58:35 +0200 Subject: [Freeswitch-users] P-Asserted-Identity Message-ID: <4F71B97B.1080601@softnet.si> Hi, I was looking in wireshark why some of the other providers are rejecting some of my calls. Some of our client are connected to Freeswitch via cabel-modem with voip support. I found that the problem appears when in sip header sip.P-Asserted-Identity is send and that FS send it to the SBC. The other softswith whitch is not FS and is working properly (this part), also get sip.P-Asserted-Identity in sip header but does not send it to sbc, it change it to P-Charge-info. Is it possbile to change that behaviour on FS so that other providers will not rejecting our calls? Thanks! Miha From fdelawarde at wirelessmundi.com Tue Mar 27 17:09:14 2012 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 27 Mar 2012 15:09:14 +0200 Subject: [Freeswitch-users] P-Asserted-Identity In-Reply-To: <4F71B97B.1080601@softnet.si> References: <4F71B97B.1080601@softnet.si> Message-ID: <1332853754.23470.3.camel@luna.madrid.commsmundi.com> http://wiki.freeswitch.org/wiki/Variable_sip_cid_type On Tue, 2012-03-27 at 14:58 +0200, Miha wrote: > Hi, > > I was looking in wireshark why some of the other providers are rejecting > some of my calls. Some of our client are connected to Freeswitch via > cabel-modem with voip support. > I found that the problem appears when in sip header > sip.P-Asserted-Identity is send and that FS send it to the SBC. > > The other softswith whitch is not FS and is working properly (this > part), also get sip.P-Asserted-Identity in sip header but does not send > it to sbc, it change it to P-Charge-info. > > Is it possbile to change that behaviour on FS so that other providers > will not rejecting our calls? > > Thanks! > > Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anita.hall at simmortel.com Tue Mar 27 17:17:40 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Tue, 27 Mar 2012 18:47:40 +0530 Subject: [Freeswitch-users] The Best Way to Send Fax on Sangoma? Message-ID: Hi What should be the optimal settings for HWEC related parameters in wanpipe1.conf to be able to receive Fax over T.30 on a Sangoma A108DE Card ? In particular, should I enable echo ? If yes, to what parameter ? What other values like RXGAIN, TXGAIN should I tune ? TDMV_HW_DTMF = YES # YES: receive dtmf events from hardware TDMV_HW_FAX_DETECT = YES # YES: receive fax 1100hz events from hardware HWEC_OPERATION_MODE = OCT_NORMAL # OCT_NORMAL: echo cancelation enabled with nlp (default) # OCT_SPEECH: improves software tone detection by disabling NLP (echo possible) # OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions. HWEC_DTMF_REMOVAL = NO # NO: default YES: remove dtmf out of incoming media (must have hwdtmf enabled) HWEC_NOISE_REDUCTION = NO # NO: default YES: reduces noise on the line - could break fax HWEC_ACUSTIC_ECHO = NO # NO: default YES: enables acustic echo cancelation HWEC_NLP_DISABLE = NO # NO: default YES: guarantees software tone detection (possible echo) HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_TX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal HWEC_RX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/3843180e/attachment.html From krice at freeswitch.org Tue Mar 27 17:21:31 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 27 Mar 2012 08:21:31 -0500 Subject: [Freeswitch-users] Trouble building Freeswitch on Ubuntu 11.10 In-Reply-To: Message-ID: Hey Joshua did you open a Jira on this? I think I see the problem but we really need a Jira on this K On 3/26/12 8:51 PM, "Joshua Nankin" wrote: > I've followed all the steps at?http://wiki.freeswitch.org/wiki/Quick_Start. > ?I've seen similar issues in previous threads, and I've installed gawk and run > make with -j1, but I'm still not able to do a clean build: > > quiet_libtool: compile: ?gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse > -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c > -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src > -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ > -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src > -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 > -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c ?-fPIC -DPIC -o > .libs/mod_xml_rpc.o > quiet_libtool: compile: ?gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse > -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c > -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src > -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ > -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src > -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 > -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -o mod_xml_rpc.o > >/dev/null 2>&1 > Creating mod_xml_rpc.la... > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > > Full log at http://joshnankin.com/freeswitch.log > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/ca2aa67c/attachment.html From bwibowo at gmail.com Tue Mar 27 17:27:08 2012 From: bwibowo at gmail.com (budi wibowo) Date: Tue, 27 Mar 2012 20:27:08 +0700 Subject: [Freeswitch-users] double invite Message-ID: dear all i use FreeSWITCH Version 1.0.head (git-9d3401e 2012-03-19 20-06-36 -0500) and connect to some gateway over sip. during test i found - 2 Invite from FS then receive 1 Trying from other gw - FS send cancel and other gw return 481 - FS send ACK , and got return BYE is it common happen? or any config in FS i can use to fine tune the setting? attached is detail pcap file for consideration TIA budi wibowo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/576fe909/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: sip_trace.pcap Type: application/octet-stream Size: 41719 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/576fe909/attachment-0001.obj From miha at softnet.si Tue Mar 27 17:31:45 2012 From: miha at softnet.si (Miha) Date: Tue, 27 Mar 2012 15:31:45 +0200 Subject: [Freeswitch-users] P-Asserted-Identity In-Reply-To: <1332853754.23470.3.camel@luna.madrid.commsmundi.com> References: <4F71B97B.1080601@softnet.si> <1332853754.23470.3.camel@luna.madrid.commsmundi.com> Message-ID: <4F71C141.7050005@softnet.si> On 3/27/2012 3:09 PM, Fran?ois Delawarde wrote: > http://wiki.freeswitch.org/wiki/Variable_sip_cid_type > > > > On Tue, 2012-03-27 at 14:58 +0200, Miha wrote: >> Hi, >> >> I was looking in wireshark why some of the other providers are rejecting >> some of my calls. Some of our client are connected to Freeswitch via >> cabel-modem with voip support. >> I found that the problem appears when in sip header >> sip.P-Asserted-Identity is send and that FS send it to the SBC. >> >> The other softswith whitch is not FS and is working properly (this >> part), also get sip.P-Asserted-Identity in sip header but does not send >> it to sbc, it change it to P-Charge-info. >> >> Is it possbile to change that behaviour on FS so that other providers >> will not rejecting our calls? >> >> Thanks! >> >> Miha >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > @Fran?ois thank you for your quick respon. I was looking on wiki and on google but did not find anything. I guess I was looking at wrong place. Reagrds, Miha From wstephen80 at gmail.com Tue Mar 27 17:40:59 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Tue, 27 Mar 2012 15:40:59 +0200 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1507ED07@cantor.std.visionutv.se> Message-ID: With "1ms-timer" the behaviour is changed but the "timer_test" remain strange: 2012-03-27 15:36:49.609946 [CONSOLE] mod_commands.c:559 Timer Test: 1 sleep 20 1 2012-03-27 15:36:49.630810 [CONSOLE] mod_commands.c:559 Timer Test: 2 sleep 20 21225 2012-03-27 15:36:49.671927 [CONSOLE] mod_commands.c:559 Timer Test: 3 sleep 20 40716 2012-03-27 15:36:49.671927 [CONSOLE] mod_commands.c:559 Timer Test: 4 sleep 20 29 2012-03-27 15:36:49.690968 [CONSOLE] mod_commands.c:559 Timer Test: 5 sleep 20 19451 2012-03-27 15:36:49.714993 [CONSOLE] mod_commands.c:559 Timer Test: 6 sleep 20 23380 2012-03-27 15:36:49.714993 [CONSOLE] mod_commands.c:559 Timer Test: 7 sleep 20 35 2012-03-27 15:36:49.752814 [CONSOLE] mod_commands.c:559 Timer Test: 8 sleep 20 38474 2012-03-27 15:36:49.752814 [CONSOLE] mod_commands.c:559 Timer Test: 9 sleep 20 41 2012-03-27 15:36:49.776956 [CONSOLE] mod_commands.c:559 Timer Test: 10 sleep 20 23408 2012-03-27 15:36:49.798632 [CONSOLE] mod_commands.c:559 Timer Test: 11 sleep 20 22079 2012-03-27 15:36:49.838054 [CONSOLE] mod_commands.c:559 Timer Test: 12 sleep 20 39021 2012-03-27 15:36:49.838054 [CONSOLE] mod_commands.c:559 Timer Test: 13 sleep 20 35 2012-03-27 15:36:49.857783 [CONSOLE] mod_commands.c:559 Timer Test: 14 sleep 20 19946 2012-03-27 15:36:49.887887 [CONSOLE] mod_commands.c:559 Timer Test: 15 sleep 20 29935 2012-03-27 15:36:49.906809 [CONSOLE] mod_commands.c:559 Timer Test: 16 sleep 20 18774 2012-03-27 15:36:49.946751 [CONSOLE] mod_commands.c:559 Timer Test: 17 sleep 20 40290 2012-03-27 15:36:49.946751 [CONSOLE] mod_commands.c:559 Timer Test: 18 sleep 20 32 2012-03-27 15:36:49.966757 [CONSOLE] mod_commands.c:559 Timer Test: 19 sleep 20 19813 2012-03-27 15:36:50.005781 [CONSOLE] mod_commands.c:559 Timer Test: 20 sleep 20 39099 2012-03-27 15:36:50.005781 [CONSOLE] mod_commands.c:559 Timer Test: 21 sleep 20 24 2012-03-27 15:36:50.023894 [CONSOLE] mod_commands.c:559 Timer Test: 22 sleep 20 18116 2012-03-27 15:36:50.046022 [CONSOLE] mod_commands.c:559 Timer Test: 23 sleep 20 22046 2012-03-27 15:36:50.074716 [CONSOLE] mod_commands.c:559 Timer Test: 24 sleep 20 29054 2012-03-27 15:36:50.074716 [CONSOLE] mod_commands.c:559 Timer Test: 25 sleep 20 25 2012-03-27 15:36:50.111794 [CONSOLE] mod_commands.c:559 Timer Test: 26 sleep 20 37413 2012-03-27 15:36:50.131778 [CONSOLE] mod_commands.c:559 Timer Test: 27 sleep 20 19684 2012-03-27 15:36:50.155846 [CONSOLE] mod_commands.c:559 Timer Test: 28 sleep 20 23480 2012-03-27 15:36:50.155846 [CONSOLE] mod_commands.c:559 Timer Test: 29 sleep 20 37 2012-03-27 15:36:50.179738 [CONSOLE] mod_commands.c:559 Timer Test: 30 sleep 20 23811 2012-03-27 15:36:50.196725 [CONSOLE] mod_commands.c:559 Timer Test: 31 sleep 20 18828 2012-03-27 15:36:50.220604 [CONSOLE] mod_commands.c:559 Timer Test: 32 sleep 20 22205 2012-03-27 15:36:50.250696 [CONSOLE] mod_commands.c:559 Timer Test: 33 sleep 20 30868 2012-03-27 15:36:50.275841 [CONSOLE] mod_commands.c:559 Timer Test: 34 sleep 20 24303 2012-03-27 15:36:50.275841 [CONSOLE] mod_commands.c:559 Timer Test: 35 sleep 20 57 2012-03-27 15:36:50.299676 [CONSOLE] mod_commands.c:559 Timer Test: 36 sleep 20 24607 2012-03-27 15:36:50.319195 [CONSOLE] mod_commands.c:559 Timer Test: 37 sleep 20 18660 2012-03-27 15:36:50.336701 [CONSOLE] mod_commands.c:559 Timer Test: 38 sleep 20 18321 2012-03-27 15:36:50.377677 [CONSOLE] mod_commands.c:559 Timer Test: 39 sleep 20 40551 2012-03-27 15:36:50.377677 [CONSOLE] mod_commands.c:559 Timer Test: 40 sleep 20 33 2012-03-27 15:36:50.398692 [CONSOLE] mod_commands.c:559 Timer Test: 41 sleep 20 20552 2012-03-27 15:36:50.421692 [CONSOLE] mod_commands.c:559 Timer Test: 42 sleep 20 23208 2012-03-27 15:36:50.459684 [CONSOLE] mod_commands.c:559 Timer Test: 43 sleep 20 38668 2012-03-27 15:36:50.459684 [CONSOLE] mod_commands.c:559 Timer Test: 44 sleep 20 34 2012-03-27 15:36:50.478817 [CONSOLE] mod_commands.c:559 Timer Test: 45 sleep 20 18432 2012-03-27 15:36:50.519691 [CONSOLE] mod_commands.c:559 Timer Test: 46 sleep 20 41447 2012-03-27 15:36:50.519691 [CONSOLE] mod_commands.c:559 Timer Test: 47 sleep 20 29 2012-03-27 15:36:50.542590 [CONSOLE] mod_commands.c:559 Timer Test: 48 sleep 20 22699 2012-03-27 15:36:50.552674 [CONSOLE] mod_commands.c:559 Timer Test: 49 sleep 20 10034 2012-03-27 15:36:50.590676 [CONSOLE] mod_commands.c:559 Timer Test: 50 sleep 20 38060 2012-03-27 15:36:50.608681 [CONSOLE] mod_commands.c:559 Timer Test: 51 sleep 20 17448 2012-03-27 15:36:50.625755 [CONSOLE] mod_commands.c:559 Timer Test: 52 sleep 20 17767 2012-03-27 15:36:50.643693 [CONSOLE] mod_commands.c:559 Timer Test: 53 sleep 20 17263 2012-03-27 15:36:50.663677 [CONSOLE] mod_commands.c:559 Timer Test: 54 sleep 20 21429 2012-03-27 15:36:50.682676 [CONSOLE] mod_commands.c:559 Timer Test: 55 sleep 20 18072 2012-03-27 15:36:50.723740 [CONSOLE] mod_commands.c:559 Timer Test: 56 sleep 20 40933 2012-03-27 15:36:50.723740 [CONSOLE] mod_commands.c:559 Timer Test: 57 sleep 20 29 2012-03-27 15:36:50.758800 [CONSOLE] mod_commands.c:559 Timer Test: 58 sleep 20 35950 2012-03-27 15:36:50.758800 [CONSOLE] mod_commands.c:559 Timer Test: 59 sleep 20 30 2012-03-27 15:36:50.779674 [CONSOLE] mod_commands.c:559 Timer Test: 60 sleep 20 19772 The CPU load is low (20%) so what parameter can affect this? The CentOS version? The kernel versione? The FS version? The number of cpu cores? Stephen On Mon, Mar 26, 2012 at 4:40 PM, Stephen Wilde wrote: > Ok, I'll try with this parameter, I think I have to do a Freeswitch > restart to make this change... > > Stephen > > > On Mon, Mar 26, 2012 at 4:31 PM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > >> You could try to enable the ?old? 1 ms-timer. It will be a little less >> efficient (a little more CPU load), and I?m not sure it helps, but it?s >> worth a try.**** >> >> ** ** >> >> In switch.conf.xml (under autoload_configs), add this: > name="1ms-timer" value="true"/> - within the settings-tags.**** >> >> ** ** >> >> /Peter**** >> >> ** ** >> >> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Stephen Wilde >> *Skickat:* den 26 mars 2012 16:20 >> >> *Till:* FreeSWITCH Users Help >> *?mne:* Re: [Freeswitch-users] Strange "timer_test" result**** >> >> ** ** >> >> Thank you Peter for your reply!**** >> >> I have already tried the upgrade to latest git, now I'm on FreeSWITCH >> Version 1.1.beta1 (git-c31a799 2012-03-24 14-11-49 -0700) and the result is >> the same.**** >> >> I have tried with few load (during night) and in this case the timer_test >> is perfect.**** >> >> Any suggestion?**** >> >> ** ** >> >> ** ** >> >> On Mon, Mar 26, 2012 at 4:08 PM, Peter Olsson < >> peter.olsson at visionutveckling.se> wrote:**** >> >> Yes, most likely.**** >> >> **** >> >> It?s very strange though, that a machine like this gives so poor timing >> results.**** >> >> **** >> >> The first thing I would do is to upgrade to latest GIT HEAD. A patch was >> commited about two weeks ago, that does the calculation for timer_test more >> properly (so time for logging is not calculated).**** >> >> **** >> >> Have you tried to do the same test with no load at all?**** >> >> **** >> >> /Peter**** >> >> **** >> >> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Stephen Wilde >> *Skickat:* den 26 mars 2012 15:52 >> *Till:* FreeSWITCH Users Help >> *?mne:* Re: [Freeswitch-users] Strange "timer_test" result**** >> >> **** >> >> Can this issue affect the voice quality?**** >> >> **** >> >> On Fri, Mar 23, 2012 at 7:46 PM, Stephen Wilde >> wrote:**** >> >> Real hardware, a dedicate server with 2 Xeon X5670 (a total 24 core each >> one with 12Mb cache at 2.93GHz) that is running at 20% - 25% of load. OS is >> CentOS 5.7 64bit and FS is (git-0626c89 2012-02-29 14-45-39 -0600)**** >> >> **** >> >> On Fri, Mar 23, 2012 at 6:33 PM, Brian Foster >> wrote:**** >> >> Is this on virtualized or real hardware?**** >> >> **** >> >> -BDF**** >> >> On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde >> wrote:**** >> >> I have run a "timer_test" in a dedicated FS server and I see strange >> result: it's normal?**** >> >> **** >> >> Stephen**** >> >> **** >> >> **** >> >> freeswitch at internal> timer_test 20 40**** >> >> Avg: 19.866ms Total Time: 795.880ms**** >> >> **** >> >> 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: >> samplecount after init: 1**** >> >> 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: >> samplecount after first step: 2**** >> >> 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 >> sleep 20 19568**** >> >> 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 >> sleep 20 38231**** >> >> 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 >> sleep 20 18847**** >> >> 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 >> sleep 20 13982**** >> >> 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 >> sleep 20 34793**** >> >> 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 >> sleep 20 2**** >> >> 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 >> sleep 20 23166**** >> >> 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 >> sleep 20 16957**** >> >> 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 >> sleep 20 17643**** >> >> 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 >> sleep 20 18786**** >> >> 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 >> sleep 20 25100**** >> >> 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 >> sleep 20 18552**** >> >> 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 >> sleep 20 18815**** >> >> 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 >> sleep 20 19464**** >> >> 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 >> sleep 20 22012**** >> >> 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 >> sleep 20 13980**** >> >> 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 >> sleep 20 19065**** >> >> 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 >> sleep 20 39585**** >> >> 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 >> sleep 20 2**** >> >> 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 >> sleep 20 26255**** >> >> 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 >> sleep 20 17872**** >> >> 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 >> sleep 20 32191**** >> >> 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 >> sleep 20 22634**** >> >> 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 >> sleep 20 15483**** >> >> 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 >> sleep 20 22813**** >> >> 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 >> sleep 20 17099**** >> >> 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 >> sleep 20 1**** >> >> 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 >> sleep 20 29108**** >> >> 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 >> sleep 20 11492**** >> >> 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 >> sleep 20 20855**** >> >> 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 >> sleep 20 32579**** >> >> 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 >> sleep 20 18173**** >> >> 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 >> sleep 20 22666**** >> >> 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 >> sleep 20 23792**** >> >> 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 >> sleep 20 26158**** >> >> 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 >> sleep 20 13080**** >> >> 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 >> sleep 20 24609**** >> >> 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 >> sleep 20 1**** >> >> 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 >> sleep 20 19413**** >> >> 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 >> sleep 20 19820**** >> >> freeswitch at internal> status**** >> >> UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, >> 696 microseconds**** >> >> FreeSWITCH is ready**** >> >> 955611 session(s) since startup**** >> >> 2192 session(s) 0/50**** >> >> 6000 session(s) max**** >> >> min idle cpu 0.00/74.00**** >> >> **** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> >> >> **** >> >> **** >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version.**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> **** >> >> **** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> ** ** >> >> !DSPAM:4f7079c632761636018988! **** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/06cd0ee1/attachment-0001.html From farooqhussain786 at gmail.com Tue Mar 27 17:47:35 2012 From: farooqhussain786 at gmail.com (Farooq Hussain) Date: Tue, 27 Mar 2012 18:47:35 +0500 Subject: [Freeswitch-users] How to configure Bandwidth.com In-Reply-To: References: <07BF4904977CC645B485E970424193AD0FF1324850@localhost> Message-ID: Balamurgan, I don't have user name my client only provide me IP address. How would i create user on bandwidth.com. I don't find any link on bandwidth.com site. Please help and I very much in trouble. Anyone can provide skype, gmail, for any other instant contact way would be very help full. Thanks farooq 2012/3/27 Balamurugan Mahendran > I had the same issue, and it worked when I move the xml > to /usr/local/freeswitch/conf/sip_profiles/internal/bandwidth.conf.xml > > > > > > > > > > > > > > > > > make sure your vars.xml as like this(guess its default) > > > > > > > > > > > > > > > > > bridge to : > > sofia/gateway/bandwidth.com/+1xxxxxxxxx > > > Thanks, > Bala > > > > 2012/3/27 > > If you have to use IP-Auth make a user for bandwith.com with the IP as >> cidr as metioned here: http://wiki.freeswitch.org/wiki/Acl#Users**** >> >> ** ** >> >> for outgoing calls just make a bridge to sofia/external/$1@ >> **** >> >> ** ** >> >> regards**** >> >> Benjamin**** >> >> ** ** >> >> ** ** >> >> Yeah I know about that link. The information I got from bandwidth.com is >> only a IP and they say please open you port 5060. They use >> IP authentication. **** >> >> ** ** >> >> More help required other then this link. Also on which path I have to >> configure inbound and outbound rule. please help me**** >> >> 2012/3/27 Anton Kvashenkin **** >> >> Use wiki >> http://wiki.freeswitch.com/wiki/Provider_Configuration:_Bandwidth.com *** >> * >> >> 27 ????? 2012 ?. 14:19 ???????????? Farooq Hussain < >> farooqhussain786 at gmail.com> ???????:**** >> >> Hello Everyone,**** >> >> ** ** >> >> Please let me know how would we configure bandwith.com SIP turnk. Please >> help us. >> **** >> >> ** ** >> >> -- >> Thanks >> >> Farooq Hussain**** >> >> ** ** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> >> >> **** >> >> ** ** >> >> -- >> Thanks >> >> Farooq Hussain**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thanks Farooq Hussain -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/58c9683c/attachment.html From peter.olsson at visionutveckling.se Tue Mar 27 17:48:50 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 27 Mar 2012 13:48:50 +0000 Subject: [Freeswitch-users] Strange "timer_test" result Message-ID: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Is this the original CentOS kernel? Also, did you set any strange kernel parameters during startup? I think CentOS kernel out-of-the-box usually handles this just fine. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen Wilde Skickat: den 27 mars 2012 15:41 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Strange "timer_test" result With "1ms-timer" the behaviour is changed but the "timer_test" remain strange: 2012-03-27 15:36:49.609946 [CONSOLE] mod_commands.c:559 Timer Test: 1 sleep 20 1 2012-03-27 15:36:49.630810 [CONSOLE] mod_commands.c:559 Timer Test: 2 sleep 20 21225 2012-03-27 15:36:49.671927 [CONSOLE] mod_commands.c:559 Timer Test: 3 sleep 20 40716 2012-03-27 15:36:49.671927 [CONSOLE] mod_commands.c:559 Timer Test: 4 sleep 20 29 2012-03-27 15:36:49.690968 [CONSOLE] mod_commands.c:559 Timer Test: 5 sleep 20 19451 2012-03-27 15:36:49.714993 [CONSOLE] mod_commands.c:559 Timer Test: 6 sleep 20 23380 2012-03-27 15:36:49.714993 [CONSOLE] mod_commands.c:559 Timer Test: 7 sleep 20 35 2012-03-27 15:36:49.752814 [CONSOLE] mod_commands.c:559 Timer Test: 8 sleep 20 38474 2012-03-27 15:36:49.752814 [CONSOLE] mod_commands.c:559 Timer Test: 9 sleep 20 41 2012-03-27 15:36:49.776956 [CONSOLE] mod_commands.c:559 Timer Test: 10 sleep 20 23408 2012-03-27 15:36:49.798632 [CONSOLE] mod_commands.c:559 Timer Test: 11 sleep 20 22079 2012-03-27 15:36:49.838054 [CONSOLE] mod_commands.c:559 Timer Test: 12 sleep 20 39021 2012-03-27 15:36:49.838054 [CONSOLE] mod_commands.c:559 Timer Test: 13 sleep 20 35 2012-03-27 15:36:49.857783 [CONSOLE] mod_commands.c:559 Timer Test: 14 sleep 20 19946 2012-03-27 15:36:49.887887 [CONSOLE] mod_commands.c:559 Timer Test: 15 sleep 20 29935 2012-03-27 15:36:49.906809 [CONSOLE] mod_commands.c:559 Timer Test: 16 sleep 20 18774 2012-03-27 15:36:49.946751 [CONSOLE] mod_commands.c:559 Timer Test: 17 sleep 20 40290 2012-03-27 15:36:49.946751 [CONSOLE] mod_commands.c:559 Timer Test: 18 sleep 20 32 2012-03-27 15:36:49.966757 [CONSOLE] mod_commands.c:559 Timer Test: 19 sleep 20 19813 2012-03-27 15:36:50.005781 [CONSOLE] mod_commands.c:559 Timer Test: 20 sleep 20 39099 2012-03-27 15:36:50.005781 [CONSOLE] mod_commands.c:559 Timer Test: 21 sleep 20 24 2012-03-27 15:36:50.023894 [CONSOLE] mod_commands.c:559 Timer Test: 22 sleep 20 18116 2012-03-27 15:36:50.046022 [CONSOLE] mod_commands.c:559 Timer Test: 23 sleep 20 22046 2012-03-27 15:36:50.074716 [CONSOLE] mod_commands.c:559 Timer Test: 24 sleep 20 29054 2012-03-27 15:36:50.074716 [CONSOLE] mod_commands.c:559 Timer Test: 25 sleep 20 25 2012-03-27 15:36:50.111794 [CONSOLE] mod_commands.c:559 Timer Test: 26 sleep 20 37413 2012-03-27 15:36:50.131778 [CONSOLE] mod_commands.c:559 Timer Test: 27 sleep 20 19684 2012-03-27 15:36:50.155846 [CONSOLE] mod_commands.c:559 Timer Test: 28 sleep 20 23480 2012-03-27 15:36:50.155846 [CONSOLE] mod_commands.c:559 Timer Test: 29 sleep 20 37 2012-03-27 15:36:50.179738 [CONSOLE] mod_commands.c:559 Timer Test: 30 sleep 20 23811 2012-03-27 15:36:50.196725 [CONSOLE] mod_commands.c:559 Timer Test: 31 sleep 20 18828 2012-03-27 15:36:50.220604 [CONSOLE] mod_commands.c:559 Timer Test: 32 sleep 20 22205 2012-03-27 15:36:50.250696 [CONSOLE] mod_commands.c:559 Timer Test: 33 sleep 20 30868 2012-03-27 15:36:50.275841 [CONSOLE] mod_commands.c:559 Timer Test: 34 sleep 20 24303 2012-03-27 15:36:50.275841 [CONSOLE] mod_commands.c:559 Timer Test: 35 sleep 20 57 2012-03-27 15:36:50.299676 [CONSOLE] mod_commands.c:559 Timer Test: 36 sleep 20 24607 2012-03-27 15:36:50.319195 [CONSOLE] mod_commands.c:559 Timer Test: 37 sleep 20 18660 2012-03-27 15:36:50.336701 [CONSOLE] mod_commands.c:559 Timer Test: 38 sleep 20 18321 2012-03-27 15:36:50.377677 [CONSOLE] mod_commands.c:559 Timer Test: 39 sleep 20 40551 2012-03-27 15:36:50.377677 [CONSOLE] mod_commands.c:559 Timer Test: 40 sleep 20 33 2012-03-27 15:36:50.398692 [CONSOLE] mod_commands.c:559 Timer Test: 41 sleep 20 20552 2012-03-27 15:36:50.421692 [CONSOLE] mod_commands.c:559 Timer Test: 42 sleep 20 23208 2012-03-27 15:36:50.459684 [CONSOLE] mod_commands.c:559 Timer Test: 43 sleep 20 38668 2012-03-27 15:36:50.459684 [CONSOLE] mod_commands.c:559 Timer Test: 44 sleep 20 34 2012-03-27 15:36:50.478817 [CONSOLE] mod_commands.c:559 Timer Test: 45 sleep 20 18432 2012-03-27 15:36:50.519691 [CONSOLE] mod_commands.c:559 Timer Test: 46 sleep 20 41447 2012-03-27 15:36:50.519691 [CONSOLE] mod_commands.c:559 Timer Test: 47 sleep 20 29 2012-03-27 15:36:50.542590 [CONSOLE] mod_commands.c:559 Timer Test: 48 sleep 20 22699 2012-03-27 15:36:50.552674 [CONSOLE] mod_commands.c:559 Timer Test: 49 sleep 20 10034 2012-03-27 15:36:50.590676 [CONSOLE] mod_commands.c:559 Timer Test: 50 sleep 20 38060 2012-03-27 15:36:50.608681 [CONSOLE] mod_commands.c:559 Timer Test: 51 sleep 20 17448 2012-03-27 15:36:50.625755 [CONSOLE] mod_commands.c:559 Timer Test: 52 sleep 20 17767 2012-03-27 15:36:50.643693 [CONSOLE] mod_commands.c:559 Timer Test: 53 sleep 20 17263 2012-03-27 15:36:50.663677 [CONSOLE] mod_commands.c:559 Timer Test: 54 sleep 20 21429 2012-03-27 15:36:50.682676 [CONSOLE] mod_commands.c:559 Timer Test: 55 sleep 20 18072 2012-03-27 15:36:50.723740 [CONSOLE] mod_commands.c:559 Timer Test: 56 sleep 20 40933 2012-03-27 15:36:50.723740 [CONSOLE] mod_commands.c:559 Timer Test: 57 sleep 20 29 2012-03-27 15:36:50.758800 [CONSOLE] mod_commands.c:559 Timer Test: 58 sleep 20 35950 2012-03-27 15:36:50.758800 [CONSOLE] mod_commands.c:559 Timer Test: 59 sleep 20 30 2012-03-27 15:36:50.779674 [CONSOLE] mod_commands.c:559 Timer Test: 60 sleep 20 19772 The CPU load is low (20%) so what parameter can affect this? The CentOS version? The kernel versione? The FS version? The number of cpu cores? Stephen On Mon, Mar 26, 2012 at 4:40 PM, Stephen Wilde > wrote: Ok, I'll try with this parameter, I think I have to do a Freeswitch restart to make this change... Stephen On Mon, Mar 26, 2012 at 4:31 PM, Peter Olsson > wrote: You could try to enable the "old" 1 ms-timer. It will be a little less efficient (a little more CPU load), and I'm not sure it helps, but it's worth a try. In switch.conf.xml (under autoload_configs), add this: - within the settings-tags. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen Wilde Skickat: den 26 mars 2012 16:20 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Strange "timer_test" result Thank you Peter for your reply! I have already tried the upgrade to latest git, now I'm on FreeSWITCH Version 1.1.beta1 (git-c31a799 2012-03-24 14-11-49 -0700) and the result is the same. I have tried with few load (during night) and in this case the timer_test is perfect. Any suggestion? On Mon, Mar 26, 2012 at 4:08 PM, Peter Olsson > wrote: Yes, most likely. It's very strange though, that a machine like this gives so poor timing results. The first thing I would do is to upgrade to latest GIT HEAD. A patch was commited about two weeks ago, that does the calculation for timer_test more properly (so time for logging is not calculated). Have you tried to do the same test with no load at all? /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen Wilde Skickat: den 26 mars 2012 15:52 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Strange "timer_test" result Can this issue affect the voice quality? On Fri, Mar 23, 2012 at 7:46 PM, Stephen Wilde > wrote: Real hardware, a dedicate server with 2 Xeon X5670 (a total 24 core each one with 12Mb cache at 2.93GHz) that is running at 20% - 25% of load. OS is CentOS 5.7 64bit and FS is (git-0626c89 2012-02-29 14-45-39 -0600) On Fri, Mar 23, 2012 at 6:33 PM, Brian Foster > wrote: Is this on virtualized or real hardware? -BDF On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde > wrote: I have run a "timer_test" in a dedicated FS server and I see strange result: it's normal? Stephen freeswitch at internal> timer_test 20 40 Avg: 19.866ms Total Time: 795.880ms 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: samplecount after init: 1 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: samplecount after first step: 2 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 sleep 20 19568 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 sleep 20 38231 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 sleep 20 18847 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 sleep 20 13982 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 sleep 20 34793 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 sleep 20 2 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 sleep 20 23166 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 sleep 20 16957 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 sleep 20 17643 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 sleep 20 18786 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 sleep 20 25100 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 sleep 20 18552 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 sleep 20 18815 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 sleep 20 19464 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 sleep 20 22012 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 sleep 20 13980 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 sleep 20 19065 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 sleep 20 39585 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 sleep 20 2 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 sleep 20 26255 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 sleep 20 17872 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 sleep 20 32191 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 sleep 20 22634 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 sleep 20 15483 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 sleep 20 22813 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 sleep 20 17099 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 sleep 20 1 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 sleep 20 29108 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 sleep 20 11492 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 sleep 20 20855 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 sleep 20 32579 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 sleep 20 18173 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 sleep 20 22666 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 sleep 20 23792 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 sleep 20 26158 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 sleep 20 13080 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 sleep 20 24609 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 sleep 20 1 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 sleep 20 19413 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 sleep 20 19820 freeswitch at internal> status UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, 696 microseconds FreeSWITCH is ready 955611 session(s) since startup 2192 session(s) 0/50 6000 session(s) max min idle cpu 0.00/74.00 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f71c23b32762536418862! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/88ad960e/attachment-0001.html From Daniel.Knaggs at realitysolutions.co.uk Tue Mar 27 13:21:55 2012 From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs) Date: Tue, 27 Mar 2012 09:21:55 +0000 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: References: Message-ID: OK, it does NOT appear in the recordings. Here is a spectrum analysis of the call when it happened: - Frequency (Hz) Level (dB) 15.625000 -59.445583 31.250000 -69.136353 46.875000 -71.569084 62.500000 -72.445877 78.125000 -67.142319 93.750000 -63.964447 109.375000 -61.157677 125.000000 -58.849174 140.625000 -62.083492 156.250000 -51.071819 171.875000 -48.957565 187.500000 -48.156109 203.125000 -48.908588 218.750000 -47.891460 234.375000 -46.257698 250.000000 -43.011139 265.625000 -48.706509 281.250000 -53.241238 296.875000 -52.513802 312.500000 -50.387054 328.125000 -53.275459 343.750000 -56.096489 359.375000 -54.788857 375.000000 -52.283993 390.625000 -50.975601 406.250000 -49.737530 421.875000 -49.870571 437.500000 -50.445431 453.125000 -50.647591 468.750000 -49.450035 484.375000 -48.767914 500.000000 -48.281902 515.625000 -49.419975 531.250000 -52.635048 546.875000 -52.856857 562.500000 -51.675240 578.125000 -50.667599 593.750000 -51.584862 609.375000 -48.992435 625.000000 -48.492191 640.625000 -49.701382 656.250000 -53.005512 671.875000 -54.269699 687.500000 -54.937397 703.125000 -53.922794 718.750000 -51.949009 734.375000 -52.763660 750.000000 -53.731155 765.625000 -54.977795 781.250000 -54.029835 796.875000 -54.375313 812.500000 -54.419693 828.125000 -54.796967 843.750000 -55.052872 859.375000 -53.625359 875.000000 -54.649513 890.625000 -56.736855 906.250000 -57.202946 921.875000 -56.587177 937.500000 -57.263260 953.125000 -56.744713 968.750000 -57.710808 984.375000 -58.118694 1000.000000 -60.595867 1015.625000 -60.736752 1031.250000 -57.966949 1046.875000 -58.101311 1062.500000 -59.093056 1078.125000 -59.889881 1093.750000 -59.972092 1109.375000 -59.728626 1125.000000 -58.060120 1140.625000 -56.506371 1156.250000 -55.109329 1171.875000 -54.278183 1187.500000 -56.115913 1203.125000 -57.535069 1218.750000 -58.304485 1234.375000 -58.614723 1250.000000 -58.873520 1265.625000 -61.268688 1281.250000 -62.702538 1296.875000 -63.828163 1312.500000 -63.712082 1328.125000 -65.020050 1343.750000 -64.481941 1359.375000 -62.652424 1375.000000 -61.640785 1390.625000 -61.719952 1406.250000 -61.451962 1421.875000 -58.490433 1437.500000 -57.193993 1453.125000 -56.973965 1468.750000 -57.940182 1484.375000 -59.935654 1500.000000 -61.829227 1515.625000 -63.259266 1531.250000 -62.169945 1546.875000 -61.958508 1562.500000 -62.018082 1578.125000 -60.562317 1593.750000 -59.146374 1609.375000 -58.868763 1625.000000 -60.028801 1640.625000 -59.241119 1656.250000 -60.101322 1671.875000 -60.429554 1687.500000 -60.538834 1703.125000 -61.073635 1718.750000 -60.375114 1734.375000 -60.152672 1750.000000 -59.529144 1765.625000 -59.783649 1781.250000 -61.850414 1796.875000 -63.506111 1812.500000 -63.869320 1828.125000 -62.442127 1843.750000 -62.519482 1859.375000 -63.814472 1875.000000 -63.300858 1890.625000 -61.998482 1906.250000 -62.199738 1921.875000 -61.745621 1937.500000 -61.611210 1953.125000 -61.174740 1968.750000 -60.504318 1984.375000 -60.277737 2000.000000 -61.696060 2015.625000 -61.693054 2031.250000 -62.755184 2046.875000 -63.983723 2062.500000 -63.584675 2078.125000 -62.943420 2093.750000 -63.238541 2109.375000 -62.929970 2125.000000 -62.243599 2140.625000 -60.967789 2156.250000 -60.915421 2171.875000 -61.997654 2187.500000 -63.702518 2203.125000 -64.316109 2218.750000 -63.577995 2234.375000 -63.250530 2250.000000 -63.107010 2265.625000 -62.953743 2281.250000 -62.933647 2296.875000 -61.956242 2312.500000 -62.948036 2328.125000 -64.375603 2343.750000 -64.296860 2359.375000 -63.552139 2375.000000 -62.556362 2390.625000 -62.979954 2406.250000 -64.505875 2421.875000 -65.626236 2437.500000 -65.732819 2453.125000 -66.130798 2468.750000 -65.920502 2484.375000 -64.183731 2500.000000 -63.387459 2515.625000 -63.110027 2531.250000 -64.085106 2546.875000 -64.269905 2562.500000 -64.181808 2578.125000 -64.600060 2593.750000 -63.998692 2609.375000 -63.854473 2625.000000 -65.015961 2640.625000 -65.751480 2656.250000 -66.293800 2671.875000 -66.494102 2687.500000 -66.300240 2703.125000 -66.383118 2718.750000 -66.466385 2734.375000 -65.733604 2750.000000 -65.110283 2765.625000 -65.537567 2781.250000 -66.125465 2796.875000 -65.979088 2812.500000 -64.833984 2828.125000 -63.773678 2843.750000 -64.419113 2859.375000 -64.800369 2875.000000 -64.710480 2890.625000 -64.088387 2906.250000 -64.790306 2921.875000 -65.160469 2937.500000 -65.285408 2953.125000 -66.030342 2968.750000 -65.027481 2984.375000 -64.623558 3000.000000 -65.082748 3015.625000 -63.680820 3031.250000 -62.836716 3046.875000 -62.210663 3062.500000 -61.578278 3078.125000 -62.397720 3093.750000 -63.185940 3109.375000 -62.439983 3125.000000 -62.382778 3140.625000 -63.123928 3156.250000 -64.276588 3171.875000 -65.444725 3187.500000 -65.891289 3203.125000 -65.480240 3218.750000 -64.761063 3234.375000 -65.140015 3250.000000 -66.010643 3265.625000 -66.964401 3281.250000 -67.296051 3296.875000 -66.430000 3312.500000 -66.564758 3328.125000 -67.878830 3343.750000 -67.748436 3359.375000 -68.965981 3375.000000 -70.426888 3390.625000 -71.400375 3406.250000 -72.067627 3421.875000 -71.944176 3437.500000 -72.285637 3453.125000 -71.983047 3468.750000 -72.565109 3484.375000 -72.350845 3500.000000 -72.335533 3515.625000 -72.608849 3531.250000 -72.417786 3546.875000 -73.100441 3562.500000 -73.461548 3578.125000 -73.558250 3593.750000 -73.218422 3609.375000 -73.994888 3625.000000 -74.379204 3640.625000 -74.896202 3656.250000 -74.944405 3671.875000 -74.958710 3687.500000 -75.029655 3703.125000 -74.314133 3718.750000 -74.855209 3734.375000 -76.021591 3750.000000 -76.441444 3765.625000 -77.122787 3781.250000 -77.733589 3796.875000 -79.138847 3812.500000 -80.206360 3828.125000 -80.391960 3843.750000 -81.214249 3859.375000 -81.434105 3875.000000 -82.234749 3890.625000 -82.529884 3906.250000 -82.929169 3921.875000 -83.735237 3937.500000 -84.770660 3953.125000 -84.203438 3968.750000 -84.455025 3984.375000 -85.116302 2012-03-27 09:50:40.412101 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=320/320/2000 seq=38825 lw=28160 2012-03-27 09:50:40.433101 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=480/480/2000 seq=38826 lw=28320 2012-03-27 09:50:40.454100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=640/640/2000 seq=38827 lw=28480 2012-03-27 09:50:40.475100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=800/800/2000 seq=38828 lw=28640 2012-03-27 09:50:40.496100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=960/960/2000 seq=38829 lw=28800 2012-03-27 09:50:40.517100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1120/1120/2000 seq=38830 lw=28960 2012-03-27 09:50:40.538100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1280/1280/2000 seq=38831 lw=29120 2012-03-27 09:50:40.559099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1440/1440/2000 seq=38832 lw=29280 2012-03-27 09:50:40.580099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1600/1600/2000 seq=38833 lw=29440 2012-03-27 09:50:40.601099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1760/1760/2000 seq=38834 lw=29600 2012-03-27 09:50:40.622098 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1920/1920/2000 seq=38835 lw=29760 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38836 lw=29760 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38837 lw=29760 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38838 lw=29760 Any ideas? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 26 March 2012 18:39 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls On Mon, Mar 26, 2012 at 3:01 AM, Daniel Knaggs > wrote: OK, call recording has been setup ? waiting for it to happen now. Interestingly, before I issue the ?record_session? application (and of course the ?RECORD_*? variables) I had to execute ?ring_ready? then ?pre_answer? otherwise the caller gets silence (changing the order of those two commands results in silence). I find it odd that just doing a pre_answer wouldn't be sufficient. A pre_answer will send a 183 w/SDP whereas ring_ready simply sends a 180. In any case, I'm glad you got your recordings. I also find it curious that only the "letter" DTMFs are being detected. Let us know if you actually hear those tones in the audio stream. -MC [cid:imageda5d6c.PNG at 183a2c3f.4b84f9b4] Daniel Knaggs Software Developer Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE Tel: 01482 828000 / Fax: 01482 373100 Daniel.Knaggs at realitysolutions.co.uk www.realitysolutions.co.uk ________________________________ Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire [cid:image6d7ef5.PNG at b1008b83.44ae8694] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/b692e030/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: imageda5d6c.PNG Type: image/png Size: 22463 bytes Desc: imageda5d6c.PNG Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/b692e030/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: image6d7ef5.PNG Type: image/png Size: 174391 bytes Desc: image6d7ef5.PNG Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/b692e030/attachment-0003.png From jnankin at gmail.com Tue Mar 27 16:19:57 2012 From: jnankin at gmail.com (Joshua Nankin) Date: Tue, 27 Mar 2012 07:19:57 -0500 Subject: [Freeswitch-users] Trouble building Freeswitch on Ubuntu 11.10 In-Reply-To: References: Message-ID: Hi Gabe. That script didn't work for me. Got the following: In file included from nua_subnotref.c:50:0: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:2:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:6:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:7:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:11:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:12:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:16:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:17:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:21:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:22:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:26:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:27:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:31:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:32:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:36:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:37:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:41:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:42:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:46:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:47:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:51:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:52:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:56:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:57:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:61:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:62:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:66:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:67:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:71:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:72:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:76:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:77:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:81:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:82:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:86:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:87:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:91:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:92:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:96:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:97:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:101:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:102:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:106:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:107:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:111:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:112:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:116:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:117:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:121:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:122:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:126:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:127:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:131:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:132:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:252:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:252:47: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:318:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:318:53: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:323:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:323:53: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:362:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:362:54: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:367:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:367:54: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:393:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:397:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_refer_sub_make': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:399:11: error: 'sip_refer_sub_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:399:11: note: each undeclared identifier is reported only once for each function it appears in ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:399:28: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:425:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:429:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_refer_sub_format': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:438:11: error: 'sip_refer_sub_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:438:28: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:431:17: error: variable 'h' set but not used [-Werror=unused-but-set-variable] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:506:36: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:509:37: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:598:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:598:49: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:664:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:664:55: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:669:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:669:55: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:708:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:708:56: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:713:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:713:56: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:739:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:743:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_alert_info_make': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:745:11: error: 'sip_alert_info_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:745:29: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:771:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:775:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_alert_info_format': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:784:11: error: 'sip_alert_info_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:784:29: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:777:17: error: variable 'h' set but not used [-Werror=unused-but-set-variable] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:852:38: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:855:39: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:944:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:944:45: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1010:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1010:51: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1015:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1015:51: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1054:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1054:52: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1059:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1059:52: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1085:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1089:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_reply_to_make': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1091:11: error: 'sip_reply_to_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1091:27: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1117:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1121:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_reply_to_format': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1130:11: error: 'sip_reply_to_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1130:27: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1123:17: error: variable 'h' set but not used [-Werror=unused-but-set-variable] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1198:34: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1201:35: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1636:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1636:67: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1702:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1702:73: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1707:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1707:73: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1746:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1746:74: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1751:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1751:74: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1777:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1781:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_p_asserted_identity_make': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1783:11: error: 'sip_p_asserted_identity_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1783:38: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1809:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1813:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_p_asserted_identity_format': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1822:11: error: 'sip_p_asserted_identity_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1822:38: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1815:17: error: variable 'h' set but not used [-Werror=unused-but-set-variable] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1890:56: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1893:57: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:2945:2: error: #endif without #if nua_subnotref.c: In function 'nua_refer_client_response': nua_subnotref.c:915:7: error: 'sip_refer_sub_t' undeclared (first use in this function) nua_subnotref.c:915:23: error: expected ';' before 'const' nua_subnotref.c:917:11: error: 'rs' undeclared (first use in this function) cc1: all warnings being treated as errors make[10]: *** [nua_subnotref.lo] Error 1 make[10]: Leaving directory `/home/ubuntu/freeswitch_git/libs/sofia-sip/libsofia-sip-ua/nua' make[9]: *** [all] Error 2 make[9]: Leaving directory `/home/ubuntu/freeswitch_git/libs/sofia-sip/libsofia-sip-ua/nua' make[9]: Entering directory `/home/ubuntu/freeswitch_git/libs/sofia-sip/libsofia-sip-ua' make[9]: *** No rule to make target `nua/libnua.la', needed by ` libsofia-sip-ua.la'. Stop. make[9]: Leaving directory `/home/ubuntu/freeswitch_git/libs/sofia-sip/libsofia-sip-ua' make[8]: *** [all-recursive] Error 1 make[8]: Leaving directory `/home/ubuntu/freeswitch_git/libs/sofia-sip/libsofia-sip-ua' Making all in packages make[8]: Entering directory `/home/ubuntu/freeswitch_git/libs/sofia-sip' make[8]: Leaving directory `/home/ubuntu/freeswitch_git/libs/sofia-sip' make[7]: *** [all-recursive] Error 1 make[7]: Leaving directory `/home/ubuntu/freeswitch_git/libs/sofia-sip' make[6]: *** [all] Error 2 make[6]: Leaving directory `/home/ubuntu/freeswitch_git/libs/sofia-sip' make[5]: *** [/home/ubuntu/freeswitch_git/libs/sofia-sip/libsofia-sip-ua/ libsofia-sip-ua.la] Error 2 make[5]: Leaving directory `/home/ubuntu/freeswitch_git/src/mod/endpoints/mod_sofia' make[4]: *** [endpoints/mod_sofia-all] Error 1 make[4]: Leaving directory `/home/ubuntu/freeswitch_git/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/home/ubuntu/freeswitch_git/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/home/ubuntu/freeswitch_git' make[1]: *** [all] Error 2 make[1]: Leaving directory `/home/ubuntu/freeswitch_git' make: *** [build-stamp] Error 2 dpkg-buildpackage: error: debian/rules build gave error exit status 2 debuild: fatal error at line 1348: dpkg-buildpackage -rfakeroot -D -us -uc -i -b failed mv: cannot stat `freeswitch[_-]*.deb': No such file or directory On Tue, Mar 27, 2012 at 1:13 AM, Gabriel Gunderson wrote: > On Mon, Mar 26, 2012 at 7:51 PM, Joshua Nankin wrote: > > I've followed all the steps at > http://wiki.freeswitch.org/wiki/Quick_Start. > > I've seen similar issues in previous threads, and I've installed gawk > and > > run make with -j1, but I'm still not able to do a clean build: > > Try this: > https://parseltone.org/browser/trunk/misc/build_freeswitch.sh > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/07289b8f/attachment-0001.html From jnankin at gmail.com Tue Mar 27 16:30:10 2012 From: jnankin at gmail.com (Joshua Nankin) Date: Tue, 27 Mar 2012 07:30:10 -0500 Subject: [Freeswitch-users] Trouble building Freeswitch on Ubuntu 11.10 In-Reply-To: <1FFF97C269757C458224B7C895F35F1507F851@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1507F851@cantor.std.visionutv.se> Message-ID: Opened a ticket in Jira: http://jira.freeswitch.org/browse/FS-4047 On Tue, Mar 27, 2012 at 1:18 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Please report to Jira, and make sure to attach the full build log ? this > doesn?t show the actual error.**** > > ** ** > > /Peter**** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Joshua Nankin > *Skickat:* den 27 mars 2012 03:52 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* [Freeswitch-users] Trouble building Freeswitch on Ubuntu 11.10**** > > ** ** > > I've followed all the steps at http://wiki.freeswitch.org/wiki/Quick_Start. > I've seen similar issues in previous threads, and I've installed gawk and > run make with -j1, but I'm still not able to do a clean build:**** > > ** ** > > quiet_libtool: compile: gcc -w > -I../../../../libs/xmlrpc-c/lib/expat/xmlparse > -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c > -I../../../../libs/xmlrpc-c/include > -I../../../../libs/xmlrpc-c/lib/abyss/src > -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ > -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -fPIC -DPIC > -o .libs/mod_xml_rpc.o**** > > quiet_libtool: compile: gcc -w > -I../../../../libs/xmlrpc-c/lib/expat/xmlparse > -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c > -I../../../../libs/xmlrpc-c/include > -I../../../../libs/xmlrpc-c/lib/abyss/src > -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ > -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -o > mod_xml_rpc.o >/dev/null 2>&1**** > > Creating mod_xml_rpc.la...**** > > make[1]: *** [all-recursive] Error 1**** > > make: *** [all] Error 2**** > > ** ** > > ** ** > > Full log at http://joshnankin.com/freeswitch.log**** > > !DSPAM:4f7158e332761051920756! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/7ee810a7/attachment.html From wstephen80 at gmail.com Tue Mar 27 18:16:06 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Tue, 27 Mar 2012 16:16:06 +0200 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Message-ID: Yes, it's the original kernel. I start linux without any additional parameter. I have this problem in 4 of my server and no issue with only 1 server: Problem with timer_test on: Server.1, dual Xeon X5660 (12 core in HT = 24), CentOS 5.7, kernel 2.6.18-274.3.1.el5, FS commit 2012-03-26, cpu load of 35% Server.2, dual Xeon X5520 (8 core in HT = 16), CentOS 5.5, kernel 2.6.18-194.17.4.el5, FS commit 2012-03-26, cpu load of 50% Server.3, dual Xeon E5606 (8 core), CentOS 5.7, kernel 2.6.18-274.17.1.el5, FS commit 2012-02-24, cpu load of 35% Server.4, dual Xeon X5670 (12 core in HT = 24), CentOS 5.7, kernel 2.6.18-274.12.1.el5, FS commit 2012-02-24, cpu load of 25% No problem on this server: Server.5, dual Xeon X5520 (8 core in HT = 16), CentOS 5.5, kernel 2.6.18-194.3.1.el5, FS commit 2011-10-24, cpu load of 50% Stephen On Tue, Mar 27, 2012 at 3:48 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Is this the original CentOS kernel? Also, did you set any strange kernel > parameters during startup? I think CentOS kernel out-of-the-box usually > handles this just fine.**** > > ** ** > > /Peter**** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Stephen Wilde > *Skickat:* den 27 mars 2012 15:41 > > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] Strange "timer_test" result**** > > ** ** > > With "1ms-timer" the behaviour is changed but the "timer_test" remain > strange:**** > > ** ** > > 2012-03-27 15:36:49.609946 [CONSOLE] mod_commands.c:559 Timer Test: 1 > sleep 20 1**** > > 2012-03-27 15:36:49.630810 [CONSOLE] mod_commands.c:559 Timer Test: 2 > sleep 20 21225**** > > 2012-03-27 15:36:49.671927 [CONSOLE] mod_commands.c:559 Timer Test: 3 > sleep 20 40716**** > > 2012-03-27 15:36:49.671927 [CONSOLE] mod_commands.c:559 Timer Test: 4 > sleep 20 29**** > > 2012-03-27 15:36:49.690968 [CONSOLE] mod_commands.c:559 Timer Test: 5 > sleep 20 19451**** > > 2012-03-27 15:36:49.714993 [CONSOLE] mod_commands.c:559 Timer Test: 6 > sleep 20 23380**** > > 2012-03-27 15:36:49.714993 [CONSOLE] mod_commands.c:559 Timer Test: 7 > sleep 20 35**** > > 2012-03-27 15:36:49.752814 [CONSOLE] mod_commands.c:559 Timer Test: 8 > sleep 20 38474**** > > 2012-03-27 15:36:49.752814 [CONSOLE] mod_commands.c:559 Timer Test: 9 > sleep 20 41**** > > 2012-03-27 15:36:49.776956 [CONSOLE] mod_commands.c:559 Timer Test: 10 > sleep 20 23408**** > > 2012-03-27 15:36:49.798632 [CONSOLE] mod_commands.c:559 Timer Test: 11 > sleep 20 22079**** > > 2012-03-27 15:36:49.838054 [CONSOLE] mod_commands.c:559 Timer Test: 12 > sleep 20 39021**** > > 2012-03-27 15:36:49.838054 [CONSOLE] mod_commands.c:559 Timer Test: 13 > sleep 20 35**** > > 2012-03-27 15:36:49.857783 [CONSOLE] mod_commands.c:559 Timer Test: 14 > sleep 20 19946**** > > 2012-03-27 15:36:49.887887 [CONSOLE] mod_commands.c:559 Timer Test: 15 > sleep 20 29935**** > > 2012-03-27 15:36:49.906809 [CONSOLE] mod_commands.c:559 Timer Test: 16 > sleep 20 18774**** > > 2012-03-27 15:36:49.946751 [CONSOLE] mod_commands.c:559 Timer Test: 17 > sleep 20 40290**** > > 2012-03-27 15:36:49.946751 [CONSOLE] mod_commands.c:559 Timer Test: 18 > sleep 20 32**** > > 2012-03-27 15:36:49.966757 [CONSOLE] mod_commands.c:559 Timer Test: 19 > sleep 20 19813**** > > 2012-03-27 15:36:50.005781 [CONSOLE] mod_commands.c:559 Timer Test: 20 > sleep 20 39099**** > > 2012-03-27 15:36:50.005781 [CONSOLE] mod_commands.c:559 Timer Test: 21 > sleep 20 24**** > > 2012-03-27 15:36:50.023894 [CONSOLE] mod_commands.c:559 Timer Test: 22 > sleep 20 18116**** > > 2012-03-27 15:36:50.046022 [CONSOLE] mod_commands.c:559 Timer Test: 23 > sleep 20 22046**** > > 2012-03-27 15:36:50.074716 [CONSOLE] mod_commands.c:559 Timer Test: 24 > sleep 20 29054**** > > 2012-03-27 15:36:50.074716 [CONSOLE] mod_commands.c:559 Timer Test: 25 > sleep 20 25**** > > 2012-03-27 15:36:50.111794 [CONSOLE] mod_commands.c:559 Timer Test: 26 > sleep 20 37413**** > > 2012-03-27 15:36:50.131778 [CONSOLE] mod_commands.c:559 Timer Test: 27 > sleep 20 19684**** > > 2012-03-27 15:36:50.155846 [CONSOLE] mod_commands.c:559 Timer Test: 28 > sleep 20 23480**** > > 2012-03-27 15:36:50.155846 [CONSOLE] mod_commands.c:559 Timer Test: 29 > sleep 20 37**** > > 2012-03-27 15:36:50.179738 [CONSOLE] mod_commands.c:559 Timer Test: 30 > sleep 20 23811**** > > 2012-03-27 15:36:50.196725 [CONSOLE] mod_commands.c:559 Timer Test: 31 > sleep 20 18828**** > > 2012-03-27 15:36:50.220604 [CONSOLE] mod_commands.c:559 Timer Test: 32 > sleep 20 22205**** > > 2012-03-27 15:36:50.250696 [CONSOLE] mod_commands.c:559 Timer Test: 33 > sleep 20 30868**** > > 2012-03-27 15:36:50.275841 [CONSOLE] mod_commands.c:559 Timer Test: 34 > sleep 20 24303**** > > 2012-03-27 15:36:50.275841 [CONSOLE] mod_commands.c:559 Timer Test: 35 > sleep 20 57**** > > 2012-03-27 15:36:50.299676 [CONSOLE] mod_commands.c:559 Timer Test: 36 > sleep 20 24607**** > > 2012-03-27 15:36:50.319195 [CONSOLE] mod_commands.c:559 Timer Test: 37 > sleep 20 18660**** > > 2012-03-27 15:36:50.336701 [CONSOLE] mod_commands.c:559 Timer Test: 38 > sleep 20 18321**** > > 2012-03-27 15:36:50.377677 [CONSOLE] mod_commands.c:559 Timer Test: 39 > sleep 20 40551**** > > 2012-03-27 15:36:50.377677 [CONSOLE] mod_commands.c:559 Timer Test: 40 > sleep 20 33**** > > 2012-03-27 15:36:50.398692 [CONSOLE] mod_commands.c:559 Timer Test: 41 > sleep 20 20552**** > > 2012-03-27 15:36:50.421692 [CONSOLE] mod_commands.c:559 Timer Test: 42 > sleep 20 23208**** > > 2012-03-27 15:36:50.459684 [CONSOLE] mod_commands.c:559 Timer Test: 43 > sleep 20 38668**** > > 2012-03-27 15:36:50.459684 [CONSOLE] mod_commands.c:559 Timer Test: 44 > sleep 20 34**** > > 2012-03-27 15:36:50.478817 [CONSOLE] mod_commands.c:559 Timer Test: 45 > sleep 20 18432**** > > 2012-03-27 15:36:50.519691 [CONSOLE] mod_commands.c:559 Timer Test: 46 > sleep 20 41447**** > > 2012-03-27 15:36:50.519691 [CONSOLE] mod_commands.c:559 Timer Test: 47 > sleep 20 29**** > > 2012-03-27 15:36:50.542590 [CONSOLE] mod_commands.c:559 Timer Test: 48 > sleep 20 22699**** > > 2012-03-27 15:36:50.552674 [CONSOLE] mod_commands.c:559 Timer Test: 49 > sleep 20 10034**** > > 2012-03-27 15:36:50.590676 [CONSOLE] mod_commands.c:559 Timer Test: 50 > sleep 20 38060**** > > 2012-03-27 15:36:50.608681 [CONSOLE] mod_commands.c:559 Timer Test: 51 > sleep 20 17448**** > > 2012-03-27 15:36:50.625755 [CONSOLE] mod_commands.c:559 Timer Test: 52 > sleep 20 17767**** > > 2012-03-27 15:36:50.643693 [CONSOLE] mod_commands.c:559 Timer Test: 53 > sleep 20 17263**** > > 2012-03-27 15:36:50.663677 [CONSOLE] mod_commands.c:559 Timer Test: 54 > sleep 20 21429**** > > 2012-03-27 15:36:50.682676 [CONSOLE] mod_commands.c:559 Timer Test: 55 > sleep 20 18072**** > > 2012-03-27 15:36:50.723740 [CONSOLE] mod_commands.c:559 Timer Test: 56 > sleep 20 40933**** > > 2012-03-27 15:36:50.723740 [CONSOLE] mod_commands.c:559 Timer Test: 57 > sleep 20 29**** > > 2012-03-27 15:36:50.758800 [CONSOLE] mod_commands.c:559 Timer Test: 58 > sleep 20 35950**** > > 2012-03-27 15:36:50.758800 [CONSOLE] mod_commands.c:559 Timer Test: 59 > sleep 20 30**** > > 2012-03-27 15:36:50.779674 [CONSOLE] mod_commands.c:559 Timer Test: 60 > sleep 20 19772**** > > ** ** > > The CPU load is low (20%) so what parameter can affect this? The CentOS > version? The kernel versione? The FS version? The number of cpu cores?**** > > **** > > Stephen**** > > ** ** > > On Mon, Mar 26, 2012 at 4:40 PM, Stephen Wilde > wrote:**** > > Ok, I'll try with this parameter, I think I have to do a Freeswitch > restart to make this change...**** > > ** ** > > Stephen**** > > ** ** > > On Mon, Mar 26, 2012 at 4:31 PM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote:**** > > You could try to enable the ?old? 1 ms-timer. It will be a little less > efficient (a little more CPU load), and I?m not sure it helps, but it?s > worth a try.**** > > **** > > In switch.conf.xml (under autoload_configs), add this: name="1ms-timer" value="true"/> - within the settings-tags.**** > > **** > > /Peter**** > > **** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Stephen Wilde > *Skickat:* den 26 mars 2012 16:20**** > > > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] Strange "timer_test" result**** > > **** > > Thank you Peter for your reply!**** > > I have already tried the upgrade to latest git, now I'm on FreeSWITCH > Version 1.1.beta1 (git-c31a799 2012-03-24 14-11-49 -0700) and the result is > the same.**** > > I have tried with few load (during night) and in this case the timer_test > is perfect.**** > > Any suggestion?**** > > **** > > **** > > On Mon, Mar 26, 2012 at 4:08 PM, Peter Olsson < > peter.olsson at visionutveckling.se> wrote:**** > > Yes, most likely.**** > > **** > > It?s very strange though, that a machine like this gives so poor timing > results.**** > > **** > > The first thing I would do is to upgrade to latest GIT HEAD. A patch was > commited about two weeks ago, that does the calculation for timer_test more > properly (so time for logging is not calculated).**** > > **** > > Have you tried to do the same test with no load at all?**** > > **** > > /Peter**** > > **** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Stephen Wilde > *Skickat:* den 26 mars 2012 15:52 > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] Strange "timer_test" result**** > > **** > > Can this issue affect the voice quality?**** > > **** > > On Fri, Mar 23, 2012 at 7:46 PM, Stephen Wilde > wrote:**** > > Real hardware, a dedicate server with 2 Xeon X5670 (a total 24 core each > one with 12Mb cache at 2.93GHz) that is running at 20% - 25% of load. OS is > CentOS 5.7 64bit and FS is (git-0626c89 2012-02-29 14-45-39 -0600)**** > > **** > > On Fri, Mar 23, 2012 at 6:33 PM, Brian Foster > wrote:**** > > Is this on virtualized or real hardware?**** > > **** > > -BDF**** > > On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde > wrote:**** > > I have run a "timer_test" in a dedicated FS server and I see strange > result: it's normal?**** > > **** > > Stephen**** > > **** > > **** > > freeswitch at internal> timer_test 20 40**** > > Avg: 19.866ms Total Time: 795.880ms**** > > **** > > 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: > samplecount after init: 1**** > > 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: > samplecount after first step: 2**** > > 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 > sleep 20 19568**** > > 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 > sleep 20 38231**** > > 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 > sleep 20 18847**** > > 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 > sleep 20 13982**** > > 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 > sleep 20 34793**** > > 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 > sleep 20 2**** > > 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 > sleep 20 23166**** > > 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 > sleep 20 16957**** > > 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 > sleep 20 17643**** > > 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 > sleep 20 18786**** > > 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 > sleep 20 25100**** > > 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 > sleep 20 18552**** > > 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 > sleep 20 18815**** > > 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 > sleep 20 19464**** > > 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 > sleep 20 22012**** > > 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 > sleep 20 13980**** > > 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 > sleep 20 19065**** > > 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 > sleep 20 39585**** > > 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 > sleep 20 2**** > > 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 > sleep 20 26255**** > > 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 > sleep 20 17872**** > > 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 > sleep 20 32191**** > > 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 > sleep 20 22634**** > > 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 > sleep 20 15483**** > > 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 > sleep 20 22813**** > > 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 > sleep 20 17099**** > > 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 > sleep 20 1**** > > 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 > sleep 20 29108**** > > 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 > sleep 20 11492**** > > 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 > sleep 20 20855**** > > 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 > sleep 20 32579**** > > 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 > sleep 20 18173**** > > 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 > sleep 20 22666**** > > 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 > sleep 20 23792**** > > 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 > sleep 20 26158**** > > 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 > sleep 20 13080**** > > 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 > sleep 20 24609**** > > 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 > sleep 20 1**** > > 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 > sleep 20 19413**** > > 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 > sleep 20 19820**** > > freeswitch at internal> status**** > > UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, > 696 microseconds**** > > FreeSWITCH is ready**** > > 955611 session(s) since startup**** > > 2192 session(s) 0/50**** > > 6000 session(s) max**** > > min idle cpu 0.00/74.00**** > > **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > **** > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version.**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > **** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > ** ** > > !DSPAM:4f71c23b32762536418862! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/53d85379/attachment-0001.html From avi at avimarcus.net Tue Mar 27 18:22:00 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 27 Mar 2012 16:22:00 +0200 Subject: [Freeswitch-users] How to configure Bandwidth.com In-Reply-To: References: <07BF4904977CC645B485E970424193AD0FF1324850@localhost> Message-ID: Farooq, it seems to me you don't have a basic understanding of SIP authentication, registration, sending calls, etc. This mailing list is to help you with issues using FreeSWITCH, not to teach you from scratch. Please learn about basic SIP on the FreeSWITCH wiki, from the FreeSWITCH book, etc. If you wish to have someone set up your system for you, contact me or consulting at freeswitch.org -Avi Marcus 2012/3/27 Farooq Hussain > Balamurgan, > > I don't have user name my client only provide me IP address. How would i > create user on bandwidth.com. I don't find any link on bandwidth.com site.. > > Please help and I very much in trouble. Anyone can provide skype, gmail, > for any other instant contact way would be very help full. > > Thanks > farooq > > > 2012/3/27 Balamurugan Mahendran > >> I had the same issue, and it worked when I move the xml >> to /usr/local/freeswitch/conf/sip_profiles/internal/bandwidth.conf.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> make sure your vars.xml as like this(guess its default) >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> bridge to : >> >> sofia/gateway/bandwidth.com/+1xxxxxxxxx >> >> >> Thanks, >> Bala >> >> >> >> 2012/3/27 >> >> If you have to use IP-Auth make a user for bandwith.com with the IP as >>> cidr as metioned here: http://wiki.freeswitch.org/wiki/Acl#Users**** >>> >>> ** ** >>> >>> for outgoing calls just make a bridge to sofia/external/$1@ >>> **** >>> >>> ** ** >>> >>> regards**** >>> >>> Benjamin**** >>> >>> ** ** >>> >>> ** ** >>> >>> Yeah I know about that link. The information I got from bandwidth.comis only a IP and they say please open you port 5060. They use >>> IP authentication. **** >>> >>> ** ** >>> >>> More help required other then this link. Also on which path I have to >>> configure inbound and outbound rule. please help me**** >>> >>> 2012/3/27 Anton Kvashenkin **** >>> >>> Use wiki >>> http://wiki.freeswitch.com/wiki/Provider_Configuration:_Bandwidth.com ** >>> ** >>> >>> 27 ????? 2012 ?. 14:19 ???????????? Farooq Hussain < >>> farooqhussain786 at gmail.com> ???????:**** >>> >>> Hello Everyone,**** >>> >>> ** ** >>> >>> Please let me know how would we configure bandwith.com SIP turnk. >>> Please help us. >>> **** >>> >>> ** ** >>> >>> -- >>> Thanks >>> >>> Farooq Hussain**** >>> >>> ** ** >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org**** >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org**** >>> >>> >>> >>> **** >>> >>> ** ** >>> >>> -- >>> Thanks >>> >>> Farooq Hussain**** >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Thanks > > Farooq Hussain > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/84c4a08d/attachment.html From anthony.minessale at gmail.com Tue Mar 27 19:21:16 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 27 Mar 2012 10:21:16 -0500 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Message-ID: compare the quality of the motherboards. try enable-clock-nanosleep=false in switch.conf.xml also try -heavy_timer (last resort) you may want to explore a newer kernel that supports timerfd (centos 6 for instance) On Tue, Mar 27, 2012 at 9:16 AM, Stephen Wilde wrote: > Yes, it's the original kernel. I start linux without any additional > parameter. > > I have this problem in 4 of my server and no issue with only 1 server: > > Problem with timer_test on: > > Server.1, dual Xeon X5660 (12 core in HT = 24), CentOS 5.7, > kernel?2.6.18-274.3.1.el5, FS commit 2012-03-26, cpu load of 35% > Server.2, dual Xeon X5520 (8 core in HT = 16), CentOS 5.5, > kernel?2.6.18-194.17.4.el5, FS commit??2012-03-26, cpu load of 50% > Server.3,?dual Xeon E5606?(8 core), CentOS 5.7, kernel 2.6.18-274.17.1.el5, > FS commit ?2012-02-24, cpu load of 35% > Server.4, dual?Xeon? X5670?(12 core in HT = 24), CentOS 5.7, kernel > 2.6.18-274.12.1.el5, FS commit ?2012-02-24, cpu load of 25% > > No problem on this server: > > Server.5, dual?Xeon??X5520?(8 core in HT = 16), CentOS 5.5, kernel > ?2.6.18-194.3.1.el5, FS commit ?2011-10-24, cpu load of 50% > > Stephen > > > > On Tue, Mar 27, 2012 at 3:48 PM, Peter Olsson > wrote: >> >> Is this the original CentOS kernel? Also, did you set any strange kernel >> parameters during startup? I think CentOS kernel out-of-the-box usually >> handles this just fine. >> >> >> >> /Peter >> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen Wilde >> Skickat: den 27 mars 2012 15:41 >> >> >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] Strange "timer_test" result >> >> >> >> With "1ms-timer" the behaviour is changed but the "timer_test" remain >> strange: >> >> >> >> 2012-03-27 15:36:49.609946 [CONSOLE] mod_commands.c:559 Timer Test: 1 >> sleep 20 1 >> >> 2012-03-27 15:36:49.630810 [CONSOLE] mod_commands.c:559 Timer Test: 2 >> sleep 20 21225 >> >> 2012-03-27 15:36:49.671927 [CONSOLE] mod_commands.c:559 Timer Test: 3 >> sleep 20 40716 >> >> 2012-03-27 15:36:49.671927 [CONSOLE] mod_commands.c:559 Timer Test: 4 >> sleep 20 29 >> >> 2012-03-27 15:36:49.690968 [CONSOLE] mod_commands.c:559 Timer Test: 5 >> sleep 20 19451 >> >> 2012-03-27 15:36:49.714993 [CONSOLE] mod_commands.c:559 Timer Test: 6 >> sleep 20 23380 >> >> 2012-03-27 15:36:49.714993 [CONSOLE] mod_commands.c:559 Timer Test: 7 >> sleep 20 35 >> >> 2012-03-27 15:36:49.752814 [CONSOLE] mod_commands.c:559 Timer Test: 8 >> sleep 20 38474 >> >> 2012-03-27 15:36:49.752814 [CONSOLE] mod_commands.c:559 Timer Test: 9 >> sleep 20 41 >> >> 2012-03-27 15:36:49.776956 [CONSOLE] mod_commands.c:559 Timer Test: 10 >> sleep 20 23408 >> >> 2012-03-27 15:36:49.798632 [CONSOLE] mod_commands.c:559 Timer Test: 11 >> sleep 20 22079 >> >> 2012-03-27 15:36:49.838054 [CONSOLE] mod_commands.c:559 Timer Test: 12 >> sleep 20 39021 >> >> 2012-03-27 15:36:49.838054 [CONSOLE] mod_commands.c:559 Timer Test: 13 >> sleep 20 35 >> >> 2012-03-27 15:36:49.857783 [CONSOLE] mod_commands.c:559 Timer Test: 14 >> sleep 20 19946 >> >> 2012-03-27 15:36:49.887887 [CONSOLE] mod_commands.c:559 Timer Test: 15 >> sleep 20 29935 >> >> 2012-03-27 15:36:49.906809 [CONSOLE] mod_commands.c:559 Timer Test: 16 >> sleep 20 18774 >> >> 2012-03-27 15:36:49.946751 [CONSOLE] mod_commands.c:559 Timer Test: 17 >> sleep 20 40290 >> >> 2012-03-27 15:36:49.946751 [CONSOLE] mod_commands.c:559 Timer Test: 18 >> sleep 20 32 >> >> 2012-03-27 15:36:49.966757 [CONSOLE] mod_commands.c:559 Timer Test: 19 >> sleep 20 19813 >> >> 2012-03-27 15:36:50.005781 [CONSOLE] mod_commands.c:559 Timer Test: 20 >> sleep 20 39099 >> >> 2012-03-27 15:36:50.005781 [CONSOLE] mod_commands.c:559 Timer Test: 21 >> sleep 20 24 >> >> 2012-03-27 15:36:50.023894 [CONSOLE] mod_commands.c:559 Timer Test: 22 >> sleep 20 18116 >> >> 2012-03-27 15:36:50.046022 [CONSOLE] mod_commands.c:559 Timer Test: 23 >> sleep 20 22046 >> >> 2012-03-27 15:36:50.074716 [CONSOLE] mod_commands.c:559 Timer Test: 24 >> sleep 20 29054 >> >> 2012-03-27 15:36:50.074716 [CONSOLE] mod_commands.c:559 Timer Test: 25 >> sleep 20 25 >> >> 2012-03-27 15:36:50.111794 [CONSOLE] mod_commands.c:559 Timer Test: 26 >> sleep 20 37413 >> >> 2012-03-27 15:36:50.131778 [CONSOLE] mod_commands.c:559 Timer Test: 27 >> sleep 20 19684 >> >> 2012-03-27 15:36:50.155846 [CONSOLE] mod_commands.c:559 Timer Test: 28 >> sleep 20 23480 >> >> 2012-03-27 15:36:50.155846 [CONSOLE] mod_commands.c:559 Timer Test: 29 >> sleep 20 37 >> >> 2012-03-27 15:36:50.179738 [CONSOLE] mod_commands.c:559 Timer Test: 30 >> sleep 20 23811 >> >> 2012-03-27 15:36:50.196725 [CONSOLE] mod_commands.c:559 Timer Test: 31 >> sleep 20 18828 >> >> 2012-03-27 15:36:50.220604 [CONSOLE] mod_commands.c:559 Timer Test: 32 >> sleep 20 22205 >> >> 2012-03-27 15:36:50.250696 [CONSOLE] mod_commands.c:559 Timer Test: 33 >> sleep 20 30868 >> >> 2012-03-27 15:36:50.275841 [CONSOLE] mod_commands.c:559 Timer Test: 34 >> sleep 20 24303 >> >> 2012-03-27 15:36:50.275841 [CONSOLE] mod_commands.c:559 Timer Test: 35 >> sleep 20 57 >> >> 2012-03-27 15:36:50.299676 [CONSOLE] mod_commands.c:559 Timer Test: 36 >> sleep 20 24607 >> >> 2012-03-27 15:36:50.319195 [CONSOLE] mod_commands.c:559 Timer Test: 37 >> sleep 20 18660 >> >> 2012-03-27 15:36:50.336701 [CONSOLE] mod_commands.c:559 Timer Test: 38 >> sleep 20 18321 >> >> 2012-03-27 15:36:50.377677 [CONSOLE] mod_commands.c:559 Timer Test: 39 >> sleep 20 40551 >> >> 2012-03-27 15:36:50.377677 [CONSOLE] mod_commands.c:559 Timer Test: 40 >> sleep 20 33 >> >> 2012-03-27 15:36:50.398692 [CONSOLE] mod_commands.c:559 Timer Test: 41 >> sleep 20 20552 >> >> 2012-03-27 15:36:50.421692 [CONSOLE] mod_commands.c:559 Timer Test: 42 >> sleep 20 23208 >> >> 2012-03-27 15:36:50.459684 [CONSOLE] mod_commands.c:559 Timer Test: 43 >> sleep 20 38668 >> >> 2012-03-27 15:36:50.459684 [CONSOLE] mod_commands.c:559 Timer Test: 44 >> sleep 20 34 >> >> 2012-03-27 15:36:50.478817 [CONSOLE] mod_commands.c:559 Timer Test: 45 >> sleep 20 18432 >> >> 2012-03-27 15:36:50.519691 [CONSOLE] mod_commands.c:559 Timer Test: 46 >> sleep 20 41447 >> >> 2012-03-27 15:36:50.519691 [CONSOLE] mod_commands.c:559 Timer Test: 47 >> sleep 20 29 >> >> 2012-03-27 15:36:50.542590 [CONSOLE] mod_commands.c:559 Timer Test: 48 >> sleep 20 22699 >> >> 2012-03-27 15:36:50.552674 [CONSOLE] mod_commands.c:559 Timer Test: 49 >> sleep 20 10034 >> >> 2012-03-27 15:36:50.590676 [CONSOLE] mod_commands.c:559 Timer Test: 50 >> sleep 20 38060 >> >> 2012-03-27 15:36:50.608681 [CONSOLE] mod_commands.c:559 Timer Test: 51 >> sleep 20 17448 >> >> 2012-03-27 15:36:50.625755 [CONSOLE] mod_commands.c:559 Timer Test: 52 >> sleep 20 17767 >> >> 2012-03-27 15:36:50.643693 [CONSOLE] mod_commands.c:559 Timer Test: 53 >> sleep 20 17263 >> >> 2012-03-27 15:36:50.663677 [CONSOLE] mod_commands.c:559 Timer Test: 54 >> sleep 20 21429 >> >> 2012-03-27 15:36:50.682676 [CONSOLE] mod_commands.c:559 Timer Test: 55 >> sleep 20 18072 >> >> 2012-03-27 15:36:50.723740 [CONSOLE] mod_commands.c:559 Timer Test: 56 >> sleep 20 40933 >> >> 2012-03-27 15:36:50.723740 [CONSOLE] mod_commands.c:559 Timer Test: 57 >> sleep 20 29 >> >> 2012-03-27 15:36:50.758800 [CONSOLE] mod_commands.c:559 Timer Test: 58 >> sleep 20 35950 >> >> 2012-03-27 15:36:50.758800 [CONSOLE] mod_commands.c:559 Timer Test: 59 >> sleep 20 30 >> >> 2012-03-27 15:36:50.779674 [CONSOLE] mod_commands.c:559 Timer Test: 60 >> sleep 20 19772 >> >> >> >> The CPU load is low (20%) so what parameter can affect this? The CentOS >> version? The kernel versione? The FS version? The number of cpu cores? >> >> >> >> Stephen >> >> >> >> On Mon, Mar 26, 2012 at 4:40 PM, Stephen Wilde >> wrote: >> >> Ok, I'll try with this parameter, I think I have to do a Freeswitch >> restart to make this change... >> >> >> >> Stephen >> >> >> >> On Mon, Mar 26, 2012 at 4:31 PM, Peter Olsson >> wrote: >> >> You could try to enable the ?old? 1 ms-timer. It will be a little less >> efficient (a little more CPU load), and I?m not sure it helps, but it?s >> worth a try. >> >> >> >> In switch.conf.xml (under autoload_configs), add this: > name="1ms-timer" value="true"/> - within the settings-tags. >> >> >> >> /Peter >> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen Wilde >> Skickat: den 26 mars 2012 16:20 >> >> >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] Strange "timer_test" result >> >> >> >> Thank you Peter for your reply! >> >> I have already tried the upgrade to latest git, now I'm on?FreeSWITCH >> Version 1.1.beta1 (git-c31a799 2012-03-24 14-11-49 -0700) and the result is >> the same. >> >> I have tried with few load (during night) and in this case the timer_test >> is perfect. >> >> Any suggestion? >> >> >> >> >> >> On Mon, Mar 26, 2012 at 4:08 PM, Peter Olsson >> wrote: >> >> Yes, most likely. >> >> >> >> It?s very strange though, that a machine like this gives so poor timing >> results. >> >> >> >> The first thing I would do is to upgrade to latest GIT HEAD. A patch was >> commited about two weeks ago, that does the calculation for timer_test more >> properly (so time for logging is not calculated). >> >> >> >> Have you tried to do the same test with no load at all? >> >> >> >> /Peter >> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen Wilde >> Skickat: den 26 mars 2012 15:52 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] Strange "timer_test" result >> >> >> >> Can this issue affect the voice quality? >> >> >> >> On Fri, Mar 23, 2012 at 7:46 PM, Stephen Wilde >> wrote: >> >> Real hardware, a dedicate server with 2 Xeon X5670 (a total 24 core each >> one with 12Mb cache at 2.93GHz) that is running at 20% - 25% of load. OS is >> CentOS 5.7 64bit and FS is?(git-0626c89 2012-02-29 14-45-39 -0600) >> >> >> >> On Fri, Mar 23, 2012 at 6:33 PM, Brian Foster >> wrote: >> >> Is this on virtualized or real hardware? >> >> >> >> -BDF >> >> On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde >> wrote: >> >> I have run a "timer_test" in a dedicated FS server and I see strange >> result: it's normal? >> >> >> >> Stephen >> >> >> >> >> >> freeswitch at internal> timer_test 20 40 >> >> Avg: 19.866ms Total Time: 795.880ms >> >> >> >> 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: >> samplecount after init: 1 >> >> 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: >> samplecount after first step: 2 >> >> 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 >> sleep 20 19568 >> >> 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 >> sleep 20 38231 >> >> 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 >> sleep 20 18847 >> >> 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 >> sleep 20 13982 >> >> 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 >> sleep 20 34793 >> >> 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 >> sleep 20 2 >> >> 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 >> sleep 20 23166 >> >> 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 >> sleep 20 16957 >> >> 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 >> sleep 20 17643 >> >> 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 >> sleep 20 18786 >> >> 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 >> sleep 20 25100 >> >> 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 >> sleep 20 18552 >> >> 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 >> sleep 20 18815 >> >> 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 >> sleep 20 19464 >> >> 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 >> sleep 20 22012 >> >> 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 >> sleep 20 13980 >> >> 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 >> sleep 20 19065 >> >> 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 >> sleep 20 39585 >> >> 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 >> sleep 20 2 >> >> 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 >> sleep 20 26255 >> >> 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 >> sleep 20 17872 >> >> 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 >> sleep 20 32191 >> >> 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 >> sleep 20 22634 >> >> 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 >> sleep 20 15483 >> >> 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 >> sleep 20 22813 >> >> 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 >> sleep 20 17099 >> >> 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 >> sleep 20 1 >> >> 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 >> sleep 20 29108 >> >> 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 >> sleep 20 11492 >> >> 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 >> sleep 20 20855 >> >> 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 >> sleep 20 32579 >> >> 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 >> sleep 20 18173 >> >> 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 >> sleep 20 22666 >> >> 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 >> sleep 20 23792 >> >> 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 >> sleep 20 26158 >> >> 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 >> sleep 20 13080 >> >> 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 >> sleep 20 24609 >> >> 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 >> sleep 20 1 >> >> 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 >> sleep 20 19413 >> >> 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 >> sleep 20 19820 >> >> freeswitch at internal> status >> >> UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, >> 696 microseconds >> >> FreeSWITCH is ready >> >> 955611 session(s) since startup >> >> 2192 session(s) 0/50 >> >> 6000 session(s) max >> >> min idle cpu 0.00/74.00 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> !DSPAM:4f71c23b32762536418862! >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From wstephen80 at gmail.com Tue Mar 27 19:37:52 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Tue, 27 Mar 2012 17:37:52 +0200 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Message-ID: Thank you for your reply. The motherboard cannot be the issue because 2 of these servers (server.2 and server.5) are exactly the same hardware (HP DL380 G6). I'll try with enable-clock-nanosleep=false and I'll try also with -heavy_timer cmd line parameter. More difficult to test timerfd (these server are in production and it's not simple to do this upgrade). In the working server is running a very old FS version (2011-10-24), I'll try also to run this version in a not working server. Stephen On Tue, Mar 27, 2012 at 5:21 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > compare the quality of the motherboards. > > try enable-clock-nanosleep=false in switch.conf.xml > > also try -heavy_timer (last resort) > > you may want to explore a newer kernel that supports timerfd (centos 6 > for instance) > > > On Tue, Mar 27, 2012 at 9:16 AM, Stephen Wilde > wrote: > > Yes, it's the original kernel. I start linux without any additional > > parameter. > > > > I have this problem in 4 of my server and no issue with only 1 server: > > > > Problem with timer_test on: > > > > Server.1, dual Xeon X5660 (12 core in HT = 24), CentOS 5.7, > > kernel 2.6.18-274.3.1.el5, FS commit 2012-03-26, cpu load of 35% > > Server.2, dual Xeon X5520 (8 core in HT = 16), CentOS 5.5, > > kernel 2.6.18-194.17.4.el5, FS commit 2012-03-26, cpu load of 50% > > Server.3, dual Xeon E5606 (8 core), CentOS 5.7, kernel > 2.6.18-274.17.1.el5, > > FS commit 2012-02-24, cpu load of 35% > > Server.4, dual Xeon X5670 (12 core in HT = 24), CentOS 5.7, kernel > > 2.6.18-274.12.1.el5, FS commit 2012-02-24, cpu load of 25% > > > > No problem on this server: > > > > Server.5, dual Xeon X5520 (8 core in HT = 16), CentOS 5.5, kernel > > 2.6.18-194.3.1.el5, FS commit 2011-10-24, cpu load of 50% > > > > Stephen > > > > > > > > On Tue, Mar 27, 2012 at 3:48 PM, Peter Olsson > > wrote: > >> > >> Is this the original CentOS kernel? Also, did you set any strange kernel > >> parameters during startup? I think CentOS kernel out-of-the-box usually > >> handles this just fine. > >> > >> > >> > >> /Peter > >> > >> > >> > >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen > Wilde > >> Skickat: den 27 mars 2012 15:41 > >> > >> > >> Till: FreeSWITCH Users Help > >> ?mne: Re: [Freeswitch-users] Strange "timer_test" result > >> > >> > >> > >> With "1ms-timer" the behaviour is changed but the "timer_test" remain > >> strange: > >> > >> > >> > >> 2012-03-27 15:36:49.609946 [CONSOLE] mod_commands.c:559 Timer Test: 1 > >> sleep 20 1 > >> > >> 2012-03-27 15:36:49.630810 [CONSOLE] mod_commands.c:559 Timer Test: 2 > >> sleep 20 21225 > >> > >> 2012-03-27 15:36:49.671927 [CONSOLE] mod_commands.c:559 Timer Test: 3 > >> sleep 20 40716 > >> > >> 2012-03-27 15:36:49.671927 [CONSOLE] mod_commands.c:559 Timer Test: 4 > >> sleep 20 29 > >> > >> 2012-03-27 15:36:49.690968 [CONSOLE] mod_commands.c:559 Timer Test: 5 > >> sleep 20 19451 > >> > >> 2012-03-27 15:36:49.714993 [CONSOLE] mod_commands.c:559 Timer Test: 6 > >> sleep 20 23380 > >> > >> 2012-03-27 15:36:49.714993 [CONSOLE] mod_commands.c:559 Timer Test: 7 > >> sleep 20 35 > >> > >> 2012-03-27 15:36:49.752814 [CONSOLE] mod_commands.c:559 Timer Test: 8 > >> sleep 20 38474 > >> > >> 2012-03-27 15:36:49.752814 [CONSOLE] mod_commands.c:559 Timer Test: 9 > >> sleep 20 41 > >> > >> 2012-03-27 15:36:49.776956 [CONSOLE] mod_commands.c:559 Timer Test: 10 > >> sleep 20 23408 > >> > >> 2012-03-27 15:36:49.798632 [CONSOLE] mod_commands.c:559 Timer Test: 11 > >> sleep 20 22079 > >> > >> 2012-03-27 15:36:49.838054 [CONSOLE] mod_commands.c:559 Timer Test: 12 > >> sleep 20 39021 > >> > >> 2012-03-27 15:36:49.838054 [CONSOLE] mod_commands.c:559 Timer Test: 13 > >> sleep 20 35 > >> > >> 2012-03-27 15:36:49.857783 [CONSOLE] mod_commands.c:559 Timer Test: 14 > >> sleep 20 19946 > >> > >> 2012-03-27 15:36:49.887887 [CONSOLE] mod_commands.c:559 Timer Test: 15 > >> sleep 20 29935 > >> > >> 2012-03-27 15:36:49.906809 [CONSOLE] mod_commands.c:559 Timer Test: 16 > >> sleep 20 18774 > >> > >> 2012-03-27 15:36:49.946751 [CONSOLE] mod_commands.c:559 Timer Test: 17 > >> sleep 20 40290 > >> > >> 2012-03-27 15:36:49.946751 [CONSOLE] mod_commands.c:559 Timer Test: 18 > >> sleep 20 32 > >> > >> 2012-03-27 15:36:49.966757 [CONSOLE] mod_commands.c:559 Timer Test: 19 > >> sleep 20 19813 > >> > >> 2012-03-27 15:36:50.005781 [CONSOLE] mod_commands.c:559 Timer Test: 20 > >> sleep 20 39099 > >> > >> 2012-03-27 15:36:50.005781 [CONSOLE] mod_commands.c:559 Timer Test: 21 > >> sleep 20 24 > >> > >> 2012-03-27 15:36:50.023894 [CONSOLE] mod_commands.c:559 Timer Test: 22 > >> sleep 20 18116 > >> > >> 2012-03-27 15:36:50.046022 [CONSOLE] mod_commands.c:559 Timer Test: 23 > >> sleep 20 22046 > >> > >> 2012-03-27 15:36:50.074716 [CONSOLE] mod_commands.c:559 Timer Test: 24 > >> sleep 20 29054 > >> > >> 2012-03-27 15:36:50.074716 [CONSOLE] mod_commands.c:559 Timer Test: 25 > >> sleep 20 25 > >> > >> 2012-03-27 15:36:50.111794 [CONSOLE] mod_commands.c:559 Timer Test: 26 > >> sleep 20 37413 > >> > >> 2012-03-27 15:36:50.131778 [CONSOLE] mod_commands.c:559 Timer Test: 27 > >> sleep 20 19684 > >> > >> 2012-03-27 15:36:50.155846 [CONSOLE] mod_commands.c:559 Timer Test: 28 > >> sleep 20 23480 > >> > >> 2012-03-27 15:36:50.155846 [CONSOLE] mod_commands.c:559 Timer Test: 29 > >> sleep 20 37 > >> > >> 2012-03-27 15:36:50.179738 [CONSOLE] mod_commands.c:559 Timer Test: 30 > >> sleep 20 23811 > >> > >> 2012-03-27 15:36:50.196725 [CONSOLE] mod_commands.c:559 Timer Test: 31 > >> sleep 20 18828 > >> > >> 2012-03-27 15:36:50.220604 [CONSOLE] mod_commands.c:559 Timer Test: 32 > >> sleep 20 22205 > >> > >> 2012-03-27 15:36:50.250696 [CONSOLE] mod_commands.c:559 Timer Test: 33 > >> sleep 20 30868 > >> > >> 2012-03-27 15:36:50.275841 [CONSOLE] mod_commands.c:559 Timer Test: 34 > >> sleep 20 24303 > >> > >> 2012-03-27 15:36:50.275841 [CONSOLE] mod_commands.c:559 Timer Test: 35 > >> sleep 20 57 > >> > >> 2012-03-27 15:36:50.299676 [CONSOLE] mod_commands.c:559 Timer Test: 36 > >> sleep 20 24607 > >> > >> 2012-03-27 15:36:50.319195 [CONSOLE] mod_commands.c:559 Timer Test: 37 > >> sleep 20 18660 > >> > >> 2012-03-27 15:36:50.336701 [CONSOLE] mod_commands.c:559 Timer Test: 38 > >> sleep 20 18321 > >> > >> 2012-03-27 15:36:50.377677 [CONSOLE] mod_commands.c:559 Timer Test: 39 > >> sleep 20 40551 > >> > >> 2012-03-27 15:36:50.377677 [CONSOLE] mod_commands.c:559 Timer Test: 40 > >> sleep 20 33 > >> > >> 2012-03-27 15:36:50.398692 [CONSOLE] mod_commands.c:559 Timer Test: 41 > >> sleep 20 20552 > >> > >> 2012-03-27 15:36:50.421692 [CONSOLE] mod_commands.c:559 Timer Test: 42 > >> sleep 20 23208 > >> > >> 2012-03-27 15:36:50.459684 [CONSOLE] mod_commands.c:559 Timer Test: 43 > >> sleep 20 38668 > >> > >> 2012-03-27 15:36:50.459684 [CONSOLE] mod_commands.c:559 Timer Test: 44 > >> sleep 20 34 > >> > >> 2012-03-27 15:36:50.478817 [CONSOLE] mod_commands.c:559 Timer Test: 45 > >> sleep 20 18432 > >> > >> 2012-03-27 15:36:50.519691 [CONSOLE] mod_commands.c:559 Timer Test: 46 > >> sleep 20 41447 > >> > >> 2012-03-27 15:36:50.519691 [CONSOLE] mod_commands.c:559 Timer Test: 47 > >> sleep 20 29 > >> > >> 2012-03-27 15:36:50.542590 [CONSOLE] mod_commands.c:559 Timer Test: 48 > >> sleep 20 22699 > >> > >> 2012-03-27 15:36:50.552674 [CONSOLE] mod_commands.c:559 Timer Test: 49 > >> sleep 20 10034 > >> > >> 2012-03-27 15:36:50.590676 [CONSOLE] mod_commands.c:559 Timer Test: 50 > >> sleep 20 38060 > >> > >> 2012-03-27 15:36:50.608681 [CONSOLE] mod_commands.c:559 Timer Test: 51 > >> sleep 20 17448 > >> > >> 2012-03-27 15:36:50.625755 [CONSOLE] mod_commands.c:559 Timer Test: 52 > >> sleep 20 17767 > >> > >> 2012-03-27 15:36:50.643693 [CONSOLE] mod_commands.c:559 Timer Test: 53 > >> sleep 20 17263 > >> > >> 2012-03-27 15:36:50.663677 [CONSOLE] mod_commands.c:559 Timer Test: 54 > >> sleep 20 21429 > >> > >> 2012-03-27 15:36:50.682676 [CONSOLE] mod_commands.c:559 Timer Test: 55 > >> sleep 20 18072 > >> > >> 2012-03-27 15:36:50.723740 [CONSOLE] mod_commands.c:559 Timer Test: 56 > >> sleep 20 40933 > >> > >> 2012-03-27 15:36:50.723740 [CONSOLE] mod_commands.c:559 Timer Test: 57 > >> sleep 20 29 > >> > >> 2012-03-27 15:36:50.758800 [CONSOLE] mod_commands.c:559 Timer Test: 58 > >> sleep 20 35950 > >> > >> 2012-03-27 15:36:50.758800 [CONSOLE] mod_commands.c:559 Timer Test: 59 > >> sleep 20 30 > >> > >> 2012-03-27 15:36:50.779674 [CONSOLE] mod_commands.c:559 Timer Test: 60 > >> sleep 20 19772 > >> > >> > >> > >> The CPU load is low (20%) so what parameter can affect this? The CentOS > >> version? The kernel versione? The FS version? The number of cpu cores? > >> > >> > >> > >> Stephen > >> > >> > >> > >> On Mon, Mar 26, 2012 at 4:40 PM, Stephen Wilde > >> wrote: > >> > >> Ok, I'll try with this parameter, I think I have to do a Freeswitch > >> restart to make this change... > >> > >> > >> > >> Stephen > >> > >> > >> > >> On Mon, Mar 26, 2012 at 4:31 PM, Peter Olsson > >> wrote: > >> > >> You could try to enable the ?old? 1 ms-timer. It will be a little less > >> efficient (a little more CPU load), and I?m not sure it helps, but it?s > >> worth a try. > >> > >> > >> > >> In switch.conf.xml (under autoload_configs), add this: >> name="1ms-timer" value="true"/> - within the settings-tags. > >> > >> > >> > >> /Peter > >> > >> > >> > >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen > Wilde > >> Skickat: den 26 mars 2012 16:20 > >> > >> > >> Till: FreeSWITCH Users Help > >> ?mne: Re: [Freeswitch-users] Strange "timer_test" result > >> > >> > >> > >> Thank you Peter for your reply! > >> > >> I have already tried the upgrade to latest git, now I'm on FreeSWITCH > >> Version 1.1.beta1 (git-c31a799 2012-03-24 14-11-49 -0700) and the > result is > >> the same. > >> > >> I have tried with few load (during night) and in this case the > timer_test > >> is perfect. > >> > >> Any suggestion? > >> > >> > >> > >> > >> > >> On Mon, Mar 26, 2012 at 4:08 PM, Peter Olsson > >> wrote: > >> > >> Yes, most likely. > >> > >> > >> > >> It?s very strange though, that a machine like this gives so poor timing > >> results. > >> > >> > >> > >> The first thing I would do is to upgrade to latest GIT HEAD. A patch was > >> commited about two weeks ago, that does the calculation for timer_test > more > >> properly (so time for logging is not calculated). > >> > >> > >> > >> Have you tried to do the same test with no load at all? > >> > >> > >> > >> /Peter > >> > >> > >> > >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen > Wilde > >> Skickat: den 26 mars 2012 15:52 > >> Till: FreeSWITCH Users Help > >> ?mne: Re: [Freeswitch-users] Strange "timer_test" result > >> > >> > >> > >> Can this issue affect the voice quality? > >> > >> > >> > >> On Fri, Mar 23, 2012 at 7:46 PM, Stephen Wilde > >> wrote: > >> > >> Real hardware, a dedicate server with 2 Xeon X5670 (a total 24 core each > >> one with 12Mb cache at 2.93GHz) that is running at 20% - 25% of load. > OS is > >> CentOS 5.7 64bit and FS is (git-0626c89 2012-02-29 14-45-39 -0600) > >> > >> > >> > >> On Fri, Mar 23, 2012 at 6:33 PM, Brian Foster > >> wrote: > >> > >> Is this on virtualized or real hardware? > >> > >> > >> > >> -BDF > >> > >> On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde > >> wrote: > >> > >> I have run a "timer_test" in a dedicated FS server and I see strange > >> result: it's normal? > >> > >> > >> > >> Stephen > >> > >> > >> > >> > >> > >> freeswitch at internal> timer_test 20 40 > >> > >> Avg: 19.866ms Total Time: 795.880ms > >> > >> > >> > >> 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: > >> samplecount after init: 1 > >> > >> 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: > >> samplecount after first step: 2 > >> > >> 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 > >> sleep 20 19568 > >> > >> 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 > >> sleep 20 38231 > >> > >> 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 > >> sleep 20 18847 > >> > >> 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 > >> sleep 20 13982 > >> > >> 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 > >> sleep 20 34793 > >> > >> 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 > >> sleep 20 2 > >> > >> 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 > >> sleep 20 23166 > >> > >> 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 > >> sleep 20 16957 > >> > >> 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 > >> sleep 20 17643 > >> > >> 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 > >> sleep 20 18786 > >> > >> 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 > >> sleep 20 25100 > >> > >> 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 > >> sleep 20 18552 > >> > >> 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 > >> sleep 20 18815 > >> > >> 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 > >> sleep 20 19464 > >> > >> 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 > >> sleep 20 22012 > >> > >> 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 > >> sleep 20 13980 > >> > >> 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 > >> sleep 20 19065 > >> > >> 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 > >> sleep 20 39585 > >> > >> 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 > >> sleep 20 2 > >> > >> 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 > >> sleep 20 26255 > >> > >> 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 > >> sleep 20 17872 > >> > >> 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 > >> sleep 20 32191 > >> > >> 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 > >> sleep 20 22634 > >> > >> 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 > >> sleep 20 15483 > >> > >> 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 > >> sleep 20 22813 > >> > >> 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 > >> sleep 20 17099 > >> > >> 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 > >> sleep 20 1 > >> > >> 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 > >> sleep 20 29108 > >> > >> 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 > >> sleep 20 11492 > >> > >> 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 > >> sleep 20 20855 > >> > >> 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 > >> sleep 20 32579 > >> > >> 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 > >> sleep 20 18173 > >> > >> 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 > >> sleep 20 22666 > >> > >> 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 > >> sleep 20 23792 > >> > >> 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 > >> sleep 20 26158 > >> > >> 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 > >> sleep 20 13080 > >> > >> 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 > >> sleep 20 24609 > >> > >> 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 > >> sleep 20 1 > >> > >> 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 > >> sleep 20 19413 > >> > >> 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 > >> sleep 20 19820 > >> > >> freeswitch at internal> status > >> > >> UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, > >> 696 microseconds > >> > >> FreeSWITCH is ready > >> > >> 955611 session(s) since startup > >> > >> 2192 session(s) 0/50 > >> > >> 6000 session(s) max > >> > >> min idle cpu 0.00/74.00 > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> > >> -- > >> Brian D. Foster > >> Endigo Computer LLC > >> Email: bdfoster at endigotech.com > >> Phone: 317-800-7876 > >> Indianapolis, Indiana, USA > >> > >> This message contains confidential information and is intended for those > >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. > If > >> you are not the intended recipient you are notified that disclosing, > >> copying, distributing or taking any action in reliance on the contents > of > >> this information is strictly prohibited. E-mail transmission cannot be > >> guaranteed to be secure or error-free as information could be > intercepted, > >> corrupted, lost, destroyed, arrive late or incomplete, or contain > viruses. > >> The sender therefore does not accept liability for any errors or > omissions > >> in the contents of this message, which arise as a result of e-mail > >> transmission. If verification is required please request a hard-copy > >> version. > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> > >> !DSPAM:4f71c23b32762536418862! > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/28cc6cec/attachment-0001.html From anthony.minessale at gmail.com Tue Mar 27 19:48:28 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 27 Mar 2012 10:48:28 -0500 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Message-ID: I don't think the timing code has changes since that october version really. The only other possibility is maybe something else on the box is using up the cpu. When I run the test on my box circa 2008 with centos 5.7 i get a completely clean score. >From my experience, the only boxes that act bad in this case are one of the following: 32 bit linux on 64 bit hw bad motherborard (may be fixed with the disable nanosleep) overloaded box freeswitch at DeathSTAR> timer_test 2012-03-27 10:41:33.788056 [CONSOLE] mod_commands.c:559 Timer Test: 1 sleep 20 19942 2012-03-27 10:41:33.808060 [CONSOLE] mod_commands.c:559 Timer Test: 2 sleep 20 20004 2012-03-27 10:41:33.828064 [CONSOLE] mod_commands.c:559 Timer Test: 3 sleep 20 20004 2012-03-27 10:41:33.848068 [CONSOLE] mod_commands.c:559 Timer Test: 4 sleep 20 20004 2012-03-27 10:41:33.868072 [CONSOLE] mod_commands.c:559 Timer Test: 5 sleep 20 20002 2012-03-27 10:41:33.888076 [CONSOLE] mod_commands.c:559 Timer Test: 6 sleep 20 20009 2012-03-27 10:41:33.908079 [CONSOLE] mod_commands.c:559 Timer Test: 7 sleep 20 20011 2012-03-27 10:41:33.928088 [CONSOLE] mod_commands.c:559 Timer Test: 8 sleep 20 19998 2012-03-27 10:41:33.948088 [CONSOLE] mod_commands.c:559 Timer Test: 9 sleep 20 20000 2012-03-27 10:41:33.968091 [CONSOLE] mod_commands.c:559 Timer Test: 10 sleep 20 20001 2012-03-27 10:41:33.988098 [CONSOLE] mod_commands.c:559 Timer Test: 11 sleep 20 20003 2012-03-27 10:41:34.008098 [CONSOLE] mod_commands.c:559 Timer Test: 12 sleep 20 20001 2012-03-27 10:41:34.028102 [CONSOLE] mod_commands.c:559 Timer Test: 13 sleep 20 20003 2012-03-27 10:41:34.048106 [CONSOLE] mod_commands.c:559 Timer Test: 14 sleep 20 20004 2012-03-27 10:41:34.068110 [CONSOLE] mod_commands.c:559 Timer Test: 15 sleep 20 20005 2012-03-27 10:41:34.088114 [CONSOLE] mod_commands.c:559 Timer Test: 16 sleep 20 20004 2012-03-27 10:41:34.108118 [CONSOLE] mod_commands.c:559 Timer Test: 17 sleep 20 20004 2012-03-27 10:41:34.128122 [CONSOLE] mod_commands.c:559 Timer Test: 18 sleep 20 20003 2012-03-27 10:41:34.148126 [CONSOLE] mod_commands.c:559 Timer Test: 19 sleep 20 20004 2012-03-27 10:41:34.168129 [CONSOLE] mod_commands.c:559 Timer Test: 20 sleep 20 20003 2012-03-27 10:41:34.188133 [CONSOLE] mod_commands.c:559 Timer Test: 21 sleep 20 20005 2012-03-27 10:41:34.208137 [CONSOLE] mod_commands.c:559 Timer Test: 22 sleep 20 20003 2012-03-27 10:41:34.228141 [CONSOLE] mod_commands.c:559 Timer Test: 23 sleep 20 20004 2012-03-27 10:41:34.248145 [CONSOLE] mod_commands.c:559 Timer Test: 24 sleep 20 20004 2012-03-27 10:41:34.268148 [CONSOLE] mod_commands.c:559 Timer Test: 25 sleep 20 20004 2012-03-27 10:41:34.288152 [CONSOLE] mod_commands.c:559 Timer Test: 26 sleep 20 20003 2012-03-27 10:41:34.308156 [CONSOLE] mod_commands.c:559 Timer Test: 27 sleep 20 20005 2012-03-27 10:41:34.328160 [CONSOLE] mod_commands.c:559 Timer Test: 28 sleep 20 20001 2012-03-27 10:41:34.348164 [CONSOLE] mod_commands.c:559 Timer Test: 29 sleep 20 20003 2012-03-27 10:41:34.368167 [CONSOLE] mod_commands.c:559 Timer Test: 30 sleep 20 20004 2012-03-27 10:41:34.388171 [CONSOLE] mod_commands.c:559 Timer Test: 31 sleep 20 20004 2012-03-27 10:41:34.408175 [CONSOLE] mod_commands.c:559 Timer Test: 32 sleep 20 20003 2012-03-27 10:41:34.428179 [CONSOLE] mod_commands.c:559 Timer Test: 33 sleep 20 20004 2012-03-27 10:41:34.448183 [CONSOLE] mod_commands.c:559 Timer Test: 34 sleep 20 20004 2012-03-27 10:41:34.468186 [CONSOLE] mod_commands.c:559 Timer Test: 35 sleep 20 20004 2012-03-27 10:41:34.488190 [CONSOLE] mod_commands.c:559 Timer Test: 36 sleep 20 20004 2012-03-27 10:41:34.508194 [CONSOLE] mod_commands.c:559 Timer Test: 37 sleep 20 20004 2012-03-27 10:41:34.528198 [CONSOLE] mod_commands.c:559 Timer Test: 38 sleep 20 20004 2012-03-27 10:41:34.548202 [CONSOLE] mod_commands.c:559 Timer Test: 39 sleep 20 20003 2012-03-27 10:41:34.568205 [CONSOLE] mod_commands.c:559 Timer Test: 40 sleep 20 20004 2012-03-27 10:41:34.588209 [CONSOLE] mod_commands.c:559 Timer Test: 41 sleep 20 20005 2012-03-27 10:41:34.608213 [CONSOLE] mod_commands.c:559 Timer Test: 42 sleep 20 20003 2012-03-27 10:41:34.628217 [CONSOLE] mod_commands.c:559 Timer Test: 43 sleep 20 20004 2012-03-27 10:41:34.648221 [CONSOLE] mod_commands.c:559 Timer Test: 44 sleep 20 20004 2012-03-27 10:41:34.668225 [CONSOLE] mod_commands.c:559 Timer Test: 45 sleep 20 20004 2012-03-27 10:41:34.688228 [CONSOLE] mod_commands.c:559 Timer Test: 46 sleep 20 20003 2012-03-27 10:41:34.708339 [CONSOLE] mod_commands.c:559 Timer Test: 47 sleep 20 20107 2012-03-27 10:41:34.728236 [CONSOLE] mod_commands.c:559 Timer Test: 48 sleep 20 19902 2012-03-27 10:41:34.748240 [CONSOLE] mod_commands.c:559 Timer Test: 49 sleep 20 20024 2012-03-27 10:41:34.768244 [CONSOLE] mod_commands.c:559 Timer Test: 50 sleep 20 20043 Avg: 20.003ms Total Time: 1000.182ms On Tue, Mar 27, 2012 at 10:37 AM, Stephen Wilde wrote: > Thank you for your reply. > The motherboard cannot be the issue because 2 of these servers (server.2 and > server.5) are exactly the same hardware (HP DL380 G6). > I'll try with?enable-clock-nanosleep=false?and I'll try also > with?-heavy_timer cmd line parameter. > More difficult to test timerfd (these server are in production and it's not > simple to do this upgrade). > In the working server is running a very old FS version (2011-10-24), I'll > try also to run this version in a not working server. > > Stephen > > > On Tue, Mar 27, 2012 at 5:21 PM, Anthony Minessale > wrote: >> >> compare the quality of the motherboards. >> >> try enable-clock-nanosleep=false in switch.conf.xml >> >> also try -heavy_timer (last resort) >> >> you may want to explore a newer kernel that supports timerfd (centos 6 >> for instance) >> >> >> On Tue, Mar 27, 2012 at 9:16 AM, Stephen Wilde >> wrote: >> > Yes, it's the original kernel. I start linux without any additional >> > parameter. >> > >> > I have this problem in 4 of my server and no issue with only 1 server: >> > >> > Problem with timer_test on: >> > >> > Server.1, dual Xeon X5660 (12 core in HT = 24), CentOS 5.7, >> > kernel?2.6.18-274.3.1.el5, FS commit 2012-03-26, cpu load of 35% >> > Server.2, dual Xeon X5520 (8 core in HT = 16), CentOS 5.5, >> > kernel?2.6.18-194.17.4.el5, FS commit??2012-03-26, cpu load of 50% >> > Server.3,?dual Xeon E5606?(8 core), CentOS 5.7, kernel >> > 2.6.18-274.17.1.el5, >> > FS commit ?2012-02-24, cpu load of 35% >> > Server.4, dual?Xeon? X5670?(12 core in HT = 24), CentOS 5.7, kernel >> > 2.6.18-274.12.1.el5, FS commit ?2012-02-24, cpu load of 25% >> > >> > No problem on this server: >> > >> > Server.5, dual?Xeon??X5520?(8 core in HT = 16), CentOS 5.5, kernel >> > ?2.6.18-194.3.1.el5, FS commit ?2011-10-24, cpu load of 50% >> > >> > Stephen >> > >> > >> > >> > On Tue, Mar 27, 2012 at 3:48 PM, Peter Olsson >> > wrote: >> >> >> >> Is this the original CentOS kernel? Also, did you set any strange >> >> kernel >> >> parameters during startup? I think CentOS kernel out-of-the-box usually >> >> handles this just fine. >> >> >> >> >> >> >> >> /Peter >> >> >> >> >> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen >> >> Wilde >> >> Skickat: den 27 mars 2012 15:41 >> >> >> >> >> >> Till: FreeSWITCH Users Help >> >> ?mne: Re: [Freeswitch-users] Strange "timer_test" result >> >> >> >> >> >> >> >> With "1ms-timer" the behaviour is changed but the "timer_test" remain >> >> strange: >> >> >> >> >> >> >> >> 2012-03-27 15:36:49.609946 [CONSOLE] mod_commands.c:559 Timer Test: 1 >> >> sleep 20 1 >> >> >> >> 2012-03-27 15:36:49.630810 [CONSOLE] mod_commands.c:559 Timer Test: 2 >> >> sleep 20 21225 >> >> >> >> 2012-03-27 15:36:49.671927 [CONSOLE] mod_commands.c:559 Timer Test: 3 >> >> sleep 20 40716 >> >> >> >> 2012-03-27 15:36:49.671927 [CONSOLE] mod_commands.c:559 Timer Test: 4 >> >> sleep 20 29 >> >> >> >> 2012-03-27 15:36:49.690968 [CONSOLE] mod_commands.c:559 Timer Test: 5 >> >> sleep 20 19451 >> >> >> >> 2012-03-27 15:36:49.714993 [CONSOLE] mod_commands.c:559 Timer Test: 6 >> >> sleep 20 23380 >> >> >> >> 2012-03-27 15:36:49.714993 [CONSOLE] mod_commands.c:559 Timer Test: 7 >> >> sleep 20 35 >> >> >> >> 2012-03-27 15:36:49.752814 [CONSOLE] mod_commands.c:559 Timer Test: 8 >> >> sleep 20 38474 >> >> >> >> 2012-03-27 15:36:49.752814 [CONSOLE] mod_commands.c:559 Timer Test: 9 >> >> sleep 20 41 >> >> >> >> 2012-03-27 15:36:49.776956 [CONSOLE] mod_commands.c:559 Timer Test: 10 >> >> sleep 20 23408 >> >> >> >> 2012-03-27 15:36:49.798632 [CONSOLE] mod_commands.c:559 Timer Test: 11 >> >> sleep 20 22079 >> >> >> >> 2012-03-27 15:36:49.838054 [CONSOLE] mod_commands.c:559 Timer Test: 12 >> >> sleep 20 39021 >> >> >> >> 2012-03-27 15:36:49.838054 [CONSOLE] mod_commands.c:559 Timer Test: 13 >> >> sleep 20 35 >> >> >> >> 2012-03-27 15:36:49.857783 [CONSOLE] mod_commands.c:559 Timer Test: 14 >> >> sleep 20 19946 >> >> >> >> 2012-03-27 15:36:49.887887 [CONSOLE] mod_commands.c:559 Timer Test: 15 >> >> sleep 20 29935 >> >> >> >> 2012-03-27 15:36:49.906809 [CONSOLE] mod_commands.c:559 Timer Test: 16 >> >> sleep 20 18774 >> >> >> >> 2012-03-27 15:36:49.946751 [CONSOLE] mod_commands.c:559 Timer Test: 17 >> >> sleep 20 40290 >> >> >> >> 2012-03-27 15:36:49.946751 [CONSOLE] mod_commands.c:559 Timer Test: 18 >> >> sleep 20 32 >> >> >> >> 2012-03-27 15:36:49.966757 [CONSOLE] mod_commands.c:559 Timer Test: 19 >> >> sleep 20 19813 >> >> >> >> 2012-03-27 15:36:50.005781 [CONSOLE] mod_commands.c:559 Timer Test: 20 >> >> sleep 20 39099 >> >> >> >> 2012-03-27 15:36:50.005781 [CONSOLE] mod_commands.c:559 Timer Test: 21 >> >> sleep 20 24 >> >> >> >> 2012-03-27 15:36:50.023894 [CONSOLE] mod_commands.c:559 Timer Test: 22 >> >> sleep 20 18116 >> >> >> >> 2012-03-27 15:36:50.046022 [CONSOLE] mod_commands.c:559 Timer Test: 23 >> >> sleep 20 22046 >> >> >> >> 2012-03-27 15:36:50.074716 [CONSOLE] mod_commands.c:559 Timer Test: 24 >> >> sleep 20 29054 >> >> >> >> 2012-03-27 15:36:50.074716 [CONSOLE] mod_commands.c:559 Timer Test: 25 >> >> sleep 20 25 >> >> >> >> 2012-03-27 15:36:50.111794 [CONSOLE] mod_commands.c:559 Timer Test: 26 >> >> sleep 20 37413 >> >> >> >> 2012-03-27 15:36:50.131778 [CONSOLE] mod_commands.c:559 Timer Test: 27 >> >> sleep 20 19684 >> >> >> >> 2012-03-27 15:36:50.155846 [CONSOLE] mod_commands.c:559 Timer Test: 28 >> >> sleep 20 23480 >> >> >> >> 2012-03-27 15:36:50.155846 [CONSOLE] mod_commands.c:559 Timer Test: 29 >> >> sleep 20 37 >> >> >> >> 2012-03-27 15:36:50.179738 [CONSOLE] mod_commands.c:559 Timer Test: 30 >> >> sleep 20 23811 >> >> >> >> 2012-03-27 15:36:50.196725 [CONSOLE] mod_commands.c:559 Timer Test: 31 >> >> sleep 20 18828 >> >> >> >> 2012-03-27 15:36:50.220604 [CONSOLE] mod_commands.c:559 Timer Test: 32 >> >> sleep 20 22205 >> >> >> >> 2012-03-27 15:36:50.250696 [CONSOLE] mod_commands.c:559 Timer Test: 33 >> >> sleep 20 30868 >> >> >> >> 2012-03-27 15:36:50.275841 [CONSOLE] mod_commands.c:559 Timer Test: 34 >> >> sleep 20 24303 >> >> >> >> 2012-03-27 15:36:50.275841 [CONSOLE] mod_commands.c:559 Timer Test: 35 >> >> sleep 20 57 >> >> >> >> 2012-03-27 15:36:50.299676 [CONSOLE] mod_commands.c:559 Timer Test: 36 >> >> sleep 20 24607 >> >> >> >> 2012-03-27 15:36:50.319195 [CONSOLE] mod_commands.c:559 Timer Test: 37 >> >> sleep 20 18660 >> >> >> >> 2012-03-27 15:36:50.336701 [CONSOLE] mod_commands.c:559 Timer Test: 38 >> >> sleep 20 18321 >> >> >> >> 2012-03-27 15:36:50.377677 [CONSOLE] mod_commands.c:559 Timer Test: 39 >> >> sleep 20 40551 >> >> >> >> 2012-03-27 15:36:50.377677 [CONSOLE] mod_commands.c:559 Timer Test: 40 >> >> sleep 20 33 >> >> >> >> 2012-03-27 15:36:50.398692 [CONSOLE] mod_commands.c:559 Timer Test: 41 >> >> sleep 20 20552 >> >> >> >> 2012-03-27 15:36:50.421692 [CONSOLE] mod_commands.c:559 Timer Test: 42 >> >> sleep 20 23208 >> >> >> >> 2012-03-27 15:36:50.459684 [CONSOLE] mod_commands.c:559 Timer Test: 43 >> >> sleep 20 38668 >> >> >> >> 2012-03-27 15:36:50.459684 [CONSOLE] mod_commands.c:559 Timer Test: 44 >> >> sleep 20 34 >> >> >> >> 2012-03-27 15:36:50.478817 [CONSOLE] mod_commands.c:559 Timer Test: 45 >> >> sleep 20 18432 >> >> >> >> 2012-03-27 15:36:50.519691 [CONSOLE] mod_commands.c:559 Timer Test: 46 >> >> sleep 20 41447 >> >> >> >> 2012-03-27 15:36:50.519691 [CONSOLE] mod_commands.c:559 Timer Test: 47 >> >> sleep 20 29 >> >> >> >> 2012-03-27 15:36:50.542590 [CONSOLE] mod_commands.c:559 Timer Test: 48 >> >> sleep 20 22699 >> >> >> >> 2012-03-27 15:36:50.552674 [CONSOLE] mod_commands.c:559 Timer Test: 49 >> >> sleep 20 10034 >> >> >> >> 2012-03-27 15:36:50.590676 [CONSOLE] mod_commands.c:559 Timer Test: 50 >> >> sleep 20 38060 >> >> >> >> 2012-03-27 15:36:50.608681 [CONSOLE] mod_commands.c:559 Timer Test: 51 >> >> sleep 20 17448 >> >> >> >> 2012-03-27 15:36:50.625755 [CONSOLE] mod_commands.c:559 Timer Test: 52 >> >> sleep 20 17767 >> >> >> >> 2012-03-27 15:36:50.643693 [CONSOLE] mod_commands.c:559 Timer Test: 53 >> >> sleep 20 17263 >> >> >> >> 2012-03-27 15:36:50.663677 [CONSOLE] mod_commands.c:559 Timer Test: 54 >> >> sleep 20 21429 >> >> >> >> 2012-03-27 15:36:50.682676 [CONSOLE] mod_commands.c:559 Timer Test: 55 >> >> sleep 20 18072 >> >> >> >> 2012-03-27 15:36:50.723740 [CONSOLE] mod_commands.c:559 Timer Test: 56 >> >> sleep 20 40933 >> >> >> >> 2012-03-27 15:36:50.723740 [CONSOLE] mod_commands.c:559 Timer Test: 57 >> >> sleep 20 29 >> >> >> >> 2012-03-27 15:36:50.758800 [CONSOLE] mod_commands.c:559 Timer Test: 58 >> >> sleep 20 35950 >> >> >> >> 2012-03-27 15:36:50.758800 [CONSOLE] mod_commands.c:559 Timer Test: 59 >> >> sleep 20 30 >> >> >> >> 2012-03-27 15:36:50.779674 [CONSOLE] mod_commands.c:559 Timer Test: 60 >> >> sleep 20 19772 >> >> >> >> >> >> >> >> The CPU load is low (20%) so what parameter can affect this? The CentOS >> >> version? The kernel versione? The FS version? The number of cpu cores? >> >> >> >> >> >> >> >> Stephen >> >> >> >> >> >> >> >> On Mon, Mar 26, 2012 at 4:40 PM, Stephen Wilde >> >> wrote: >> >> >> >> Ok, I'll try with this parameter, I think I have to do a Freeswitch >> >> restart to make this change... >> >> >> >> >> >> >> >> Stephen >> >> >> >> >> >> >> >> On Mon, Mar 26, 2012 at 4:31 PM, Peter Olsson >> >> wrote: >> >> >> >> You could try to enable the ?old? 1 ms-timer. It will be a little less >> >> efficient (a little more CPU load), and I?m not sure it helps, but it?s >> >> worth a try. >> >> >> >> >> >> >> >> In switch.conf.xml (under autoload_configs), add this: > >> name="1ms-timer" value="true"/> - within the settings-tags. >> >> >> >> >> >> >> >> /Peter >> >> >> >> >> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen >> >> Wilde >> >> Skickat: den 26 mars 2012 16:20 >> >> >> >> >> >> Till: FreeSWITCH Users Help >> >> ?mne: Re: [Freeswitch-users] Strange "timer_test" result >> >> >> >> >> >> >> >> Thank you Peter for your reply! >> >> >> >> I have already tried the upgrade to latest git, now I'm on?FreeSWITCH >> >> Version 1.1.beta1 (git-c31a799 2012-03-24 14-11-49 -0700) and the >> >> result is >> >> the same. >> >> >> >> I have tried with few load (during night) and in this case the >> >> timer_test >> >> is perfect. >> >> >> >> Any suggestion? >> >> >> >> >> >> >> >> >> >> >> >> On Mon, Mar 26, 2012 at 4:08 PM, Peter Olsson >> >> wrote: >> >> >> >> Yes, most likely. >> >> >> >> >> >> >> >> It?s very strange though, that a machine like this gives so poor timing >> >> results. >> >> >> >> >> >> >> >> The first thing I would do is to upgrade to latest GIT HEAD. A patch >> >> was >> >> commited about two weeks ago, that does the calculation for timer_test >> >> more >> >> properly (so time for logging is not calculated). >> >> >> >> >> >> >> >> Have you tried to do the same test with no load at all? >> >> >> >> >> >> >> >> /Peter >> >> >> >> >> >> >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen >> >> Wilde >> >> Skickat: den 26 mars 2012 15:52 >> >> Till: FreeSWITCH Users Help >> >> ?mne: Re: [Freeswitch-users] Strange "timer_test" result >> >> >> >> >> >> >> >> Can this issue affect the voice quality? >> >> >> >> >> >> >> >> On Fri, Mar 23, 2012 at 7:46 PM, Stephen Wilde >> >> wrote: >> >> >> >> Real hardware, a dedicate server with 2 Xeon X5670 (a total 24 core >> >> each >> >> one with 12Mb cache at 2.93GHz) that is running at 20% - 25% of load. >> >> OS is >> >> CentOS 5.7 64bit and FS is?(git-0626c89 2012-02-29 14-45-39 -0600) >> >> >> >> >> >> >> >> On Fri, Mar 23, 2012 at 6:33 PM, Brian Foster >> >> wrote: >> >> >> >> Is this on virtualized or real hardware? >> >> >> >> >> >> >> >> -BDF >> >> >> >> On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde >> >> wrote: >> >> >> >> I have run a "timer_test" in a dedicated FS server and I see strange >> >> result: it's normal? >> >> >> >> >> >> >> >> Stephen >> >> >> >> >> >> >> >> >> >> >> >> freeswitch at internal> timer_test 20 40 >> >> >> >> Avg: 19.866ms Total Time: 795.880ms >> >> >> >> >> >> >> >> 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: >> >> samplecount after init: 1 >> >> >> >> 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: >> >> samplecount after first step: 2 >> >> >> >> 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 >> >> sleep 20 19568 >> >> >> >> 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 >> >> sleep 20 38231 >> >> >> >> 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 >> >> sleep 20 18847 >> >> >> >> 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 >> >> sleep 20 13982 >> >> >> >> 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 >> >> sleep 20 34793 >> >> >> >> 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 >> >> sleep 20 2 >> >> >> >> 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 >> >> sleep 20 23166 >> >> >> >> 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 >> >> sleep 20 16957 >> >> >> >> 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 >> >> sleep 20 17643 >> >> >> >> 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 >> >> sleep 20 18786 >> >> >> >> 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 >> >> sleep 20 25100 >> >> >> >> 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 >> >> sleep 20 18552 >> >> >> >> 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 >> >> sleep 20 18815 >> >> >> >> 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 >> >> sleep 20 19464 >> >> >> >> 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 >> >> sleep 20 22012 >> >> >> >> 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 >> >> sleep 20 13980 >> >> >> >> 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 >> >> sleep 20 19065 >> >> >> >> 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 >> >> sleep 20 39585 >> >> >> >> 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 >> >> sleep 20 2 >> >> >> >> 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 >> >> sleep 20 26255 >> >> >> >> 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 >> >> sleep 20 17872 >> >> >> >> 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 >> >> sleep 20 32191 >> >> >> >> 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 >> >> sleep 20 22634 >> >> >> >> 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 >> >> sleep 20 15483 >> >> >> >> 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 >> >> sleep 20 22813 >> >> >> >> 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 >> >> sleep 20 17099 >> >> >> >> 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 >> >> sleep 20 1 >> >> >> >> 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 >> >> sleep 20 29108 >> >> >> >> 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 >> >> sleep 20 11492 >> >> >> >> 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 >> >> sleep 20 20855 >> >> >> >> 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 >> >> sleep 20 32579 >> >> >> >> 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 >> >> sleep 20 18173 >> >> >> >> 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 >> >> sleep 20 22666 >> >> >> >> 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 >> >> sleep 20 23792 >> >> >> >> 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 >> >> sleep 20 26158 >> >> >> >> 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 >> >> sleep 20 13080 >> >> >> >> 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 >> >> sleep 20 24609 >> >> >> >> 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 >> >> sleep 20 1 >> >> >> >> 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 >> >> sleep 20 19413 >> >> >> >> 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 >> >> sleep 20 19820 >> >> >> >> freeswitch at internal> status >> >> >> >> UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, >> >> 696 microseconds >> >> >> >> FreeSWITCH is ready >> >> >> >> 955611 session(s) since startup >> >> >> >> 2192 session(s) 0/50 >> >> >> >> 6000 session(s) max >> >> >> >> min idle cpu 0.00/74.00 >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> Brian D. Foster >> >> Endigo Computer LLC >> >> Email: bdfoster at endigotech.com >> >> Phone: 317-800-7876 >> >> Indianapolis, Indiana, USA >> >> >> >> This message contains confidential information and is intended for >> >> those >> >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. >> >> If >> >> you are not the intended recipient you are notified that disclosing, >> >> copying, distributing or taking any action in reliance on the contents >> >> of >> >> this information is strictly prohibited. E-mail transmission cannot be >> >> guaranteed to be secure or error-free as information could be >> >> intercepted, >> >> corrupted, lost, destroyed, arrive late or incomplete, or contain >> >> viruses. >> >> The sender therefore does not accept liability for any errors or >> >> omissions >> >> in the contents of this message, which arise as a result of e-mail >> >> transmission. If verification is required please request a hard-copy >> >> version. >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> >> >> !DSPAM:4f71c23b32762536418862! >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From wstephen80 at gmail.com Tue Mar 27 20:07:51 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Tue, 27 Mar 2012 18:07:51 +0200 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Message-ID: Ok, I'll try disabling the nanosleep. In server.2 the current load of the box is 35% and after FreeSwitch (at 750% of cpu), in top there is Sangoma Transcoding server (4% of cpu). The result of "timer_test 20 50" is: http://pastebin.freeswitch.org/18756 Here the result of "timer_test 10 50" : http://pastebin.freeswitch.org/18757 It's normal that only the timer_test on 20 is bad where 10ms is good and also is good 40,60,120?: Stephen On Tue, Mar 27, 2012 at 5:48 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I don't think the timing code has changes since that october version > really. > The only other possibility is maybe something else on the box is using > up the cpu. > > > When I run the test on my box circa 2008 with centos 5.7 i get a > completely clean score. > >From my experience, the only boxes that act bad in this case are one > of the following: > > 32 bit linux on 64 bit hw > bad motherborard (may be fixed with the disable nanosleep) > overloaded box > > freeswitch at DeathSTAR> timer_test > 2012-03-27 10:41:33.788056 [CONSOLE] mod_commands.c:559 Timer Test: 1 > sleep 20 19942 > 2012-03-27 10:41:33.808060 [CONSOLE] mod_commands.c:559 Timer Test: 2 > sleep 20 20004 > 2012-03-27 10:41:33.828064 [CONSOLE] mod_commands.c:559 Timer Test: 3 > sleep 20 20004 > 2012-03-27 10:41:33.848068 [CONSOLE] mod_commands.c:559 Timer Test: 4 > sleep 20 20004 > 2012-03-27 10:41:33.868072 [CONSOLE] mod_commands.c:559 Timer Test: 5 > sleep 20 20002 > 2012-03-27 10:41:33.888076 [CONSOLE] mod_commands.c:559 Timer Test: 6 > sleep 20 20009 > 2012-03-27 10:41:33.908079 [CONSOLE] mod_commands.c:559 Timer Test: 7 > sleep 20 20011 > 2012-03-27 10:41:33.928088 [CONSOLE] mod_commands.c:559 Timer Test: 8 > sleep 20 19998 > 2012-03-27 10:41:33.948088 [CONSOLE] mod_commands.c:559 Timer Test: 9 > sleep 20 20000 > 2012-03-27 10:41:33.968091 [CONSOLE] mod_commands.c:559 Timer Test: 10 > sleep 20 20001 > 2012-03-27 10:41:33.988098 [CONSOLE] mod_commands.c:559 Timer Test: 11 > sleep 20 20003 > 2012-03-27 10:41:34.008098 [CONSOLE] mod_commands.c:559 Timer Test: 12 > sleep 20 20001 > 2012-03-27 10:41:34.028102 [CONSOLE] mod_commands.c:559 Timer Test: 13 > sleep 20 20003 > 2012-03-27 10:41:34.048106 [CONSOLE] mod_commands.c:559 Timer Test: 14 > sleep 20 20004 > 2012-03-27 10:41:34.068110 [CONSOLE] mod_commands.c:559 Timer Test: 15 > sleep 20 20005 > 2012-03-27 10:41:34.088114 [CONSOLE] mod_commands.c:559 Timer Test: 16 > sleep 20 20004 > 2012-03-27 10:41:34.108118 [CONSOLE] mod_commands.c:559 Timer Test: 17 > sleep 20 20004 > 2012-03-27 10:41:34.128122 [CONSOLE] mod_commands.c:559 Timer Test: 18 > sleep 20 20003 > 2012-03-27 10:41:34.148126 [CONSOLE] mod_commands.c:559 Timer Test: 19 > sleep 20 20004 > 2012-03-27 10:41:34.168129 [CONSOLE] mod_commands.c:559 Timer Test: 20 > sleep 20 20003 > 2012-03-27 10:41:34.188133 [CONSOLE] mod_commands.c:559 Timer Test: 21 > sleep 20 20005 > 2012-03-27 10:41:34.208137 [CONSOLE] mod_commands.c:559 Timer Test: 22 > sleep 20 20003 > 2012-03-27 10:41:34.228141 [CONSOLE] mod_commands.c:559 Timer Test: 23 > sleep 20 20004 > 2012-03-27 10:41:34.248145 [CONSOLE] mod_commands.c:559 Timer Test: 24 > sleep 20 20004 > 2012-03-27 10:41:34.268148 [CONSOLE] mod_commands.c:559 Timer Test: 25 > sleep 20 20004 > 2012-03-27 10:41:34.288152 [CONSOLE] mod_commands.c:559 Timer Test: 26 > sleep 20 20003 > 2012-03-27 10:41:34.308156 [CONSOLE] mod_commands.c:559 Timer Test: 27 > sleep 20 20005 > 2012-03-27 10:41:34.328160 [CONSOLE] mod_commands.c:559 Timer Test: 28 > sleep 20 20001 > 2012-03-27 10:41:34.348164 [CONSOLE] mod_commands.c:559 Timer Test: 29 > sleep 20 20003 > 2012-03-27 10:41:34.368167 [CONSOLE] mod_commands.c:559 Timer Test: 30 > sleep 20 20004 > 2012-03-27 10:41:34.388171 [CONSOLE] mod_commands.c:559 Timer Test: 31 > sleep 20 20004 > 2012-03-27 10:41:34.408175 [CONSOLE] mod_commands.c:559 Timer Test: 32 > sleep 20 20003 > 2012-03-27 10:41:34.428179 [CONSOLE] mod_commands.c:559 Timer Test: 33 > sleep 20 20004 > 2012-03-27 10:41:34.448183 [CONSOLE] mod_commands.c:559 Timer Test: 34 > sleep 20 20004 > 2012-03-27 10:41:34.468186 [CONSOLE] mod_commands.c:559 Timer Test: 35 > sleep 20 20004 > 2012-03-27 10:41:34.488190 [CONSOLE] mod_commands.c:559 Timer Test: 36 > sleep 20 20004 > 2012-03-27 10:41:34.508194 [CONSOLE] mod_commands.c:559 Timer Test: 37 > sleep 20 20004 > 2012-03-27 10:41:34.528198 [CONSOLE] mod_commands.c:559 Timer Test: 38 > sleep 20 20004 > 2012-03-27 10:41:34.548202 [CONSOLE] mod_commands.c:559 Timer Test: 39 > sleep 20 20003 > 2012-03-27 10:41:34.568205 [CONSOLE] mod_commands.c:559 Timer Test: 40 > sleep 20 20004 > 2012-03-27 10:41:34.588209 [CONSOLE] mod_commands.c:559 Timer Test: 41 > sleep 20 20005 > 2012-03-27 10:41:34.608213 [CONSOLE] mod_commands.c:559 Timer Test: 42 > sleep 20 20003 > 2012-03-27 10:41:34.628217 [CONSOLE] mod_commands.c:559 Timer Test: 43 > sleep 20 20004 > 2012-03-27 10:41:34.648221 [CONSOLE] mod_commands.c:559 Timer Test: 44 > sleep 20 20004 > 2012-03-27 10:41:34.668225 [CONSOLE] mod_commands.c:559 Timer Test: 45 > sleep 20 20004 > 2012-03-27 10:41:34.688228 [CONSOLE] mod_commands.c:559 Timer Test: 46 > sleep 20 20003 > 2012-03-27 10:41:34.708339 [CONSOLE] mod_commands.c:559 Timer Test: 47 > sleep 20 20107 > 2012-03-27 10:41:34.728236 [CONSOLE] mod_commands.c:559 Timer Test: 48 > sleep 20 19902 > 2012-03-27 10:41:34.748240 [CONSOLE] mod_commands.c:559 Timer Test: 49 > sleep 20 20024 > 2012-03-27 10:41:34.768244 [CONSOLE] mod_commands.c:559 Timer Test: 50 > sleep 20 20043 > > Avg: 20.003ms Total Time: 1000.182ms > > > > On Tue, Mar 27, 2012 at 10:37 AM, Stephen Wilde > wrote: > > Thank you for your reply. > > The motherboard cannot be the issue because 2 of these servers (server.2 > and > > server.5) are exactly the same hardware (HP DL380 G6). > > I'll try with enable-clock-nanosleep=false and I'll try also > > with -heavy_timer cmd line parameter. > > More difficult to test timerfd (these server are in production and it's > not > > simple to do this upgrade). > > In the working server is running a very old FS version (2011-10-24), I'll > > try also to run this version in a not working server. > > > > Stephen > > > > > > On Tue, Mar 27, 2012 at 5:21 PM, Anthony Minessale > > wrote: > >> > >> compare the quality of the motherboards. > >> > >> try enable-clock-nanosleep=false in switch.conf.xml > >> > >> also try -heavy_timer (last resort) > >> > >> you may want to explore a newer kernel that supports timerfd (centos 6 > >> for instance) > >> > >> > >> On Tue, Mar 27, 2012 at 9:16 AM, Stephen Wilde > >> wrote: > >> > Yes, it's the original kernel. I start linux without any additional > >> > parameter. > >> > > >> > I have this problem in 4 of my server and no issue with only 1 server: > >> > > >> > Problem with timer_test on: > >> > > >> > Server.1, dual Xeon X5660 (12 core in HT = 24), CentOS 5.7, > >> > kernel 2.6.18-274.3.1.el5, FS commit 2012-03-26, cpu load of 35% > >> > Server.2, dual Xeon X5520 (8 core in HT = 16), CentOS 5.5, > >> > kernel 2.6.18-194.17.4.el5, FS commit 2012-03-26, cpu load of 50% > >> > Server.3, dual Xeon E5606 (8 core), CentOS 5.7, kernel > >> > 2.6.18-274.17.1.el5, > >> > FS commit 2012-02-24, cpu load of 35% > >> > Server.4, dual Xeon X5670 (12 core in HT = 24), CentOS 5.7, kernel > >> > 2.6.18-274.12.1.el5, FS commit 2012-02-24, cpu load of 25% > >> > > >> > No problem on this server: > >> > > >> > Server.5, dual Xeon X5520 (8 core in HT = 16), CentOS 5.5, kernel > >> > 2.6.18-194.3.1.el5, FS commit 2011-10-24, cpu load of 50% > >> > > >> > Stephen > >> > > >> > > >> > > >> > On Tue, Mar 27, 2012 at 3:48 PM, Peter Olsson > >> > wrote: > >> >> > >> >> Is this the original CentOS kernel? Also, did you set any strange > >> >> kernel > >> >> parameters during startup? I think CentOS kernel out-of-the-box > usually > >> >> handles this just fine. > >> >> > >> >> > >> >> > >> >> /Peter > >> >> > >> >> > >> >> > >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org > >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen > >> >> Wilde > >> >> Skickat: den 27 mars 2012 15:41 > >> >> > >> >> > >> >> Till: FreeSWITCH Users Help > >> >> ?mne: Re: [Freeswitch-users] Strange "timer_test" result > >> >> > >> >> > >> >> > >> >> With "1ms-timer" the behaviour is changed but the "timer_test" remain > >> >> strange: > >> >> > >> >> > >> >> > >> >> 2012-03-27 15:36:49.609946 [CONSOLE] mod_commands.c:559 Timer Test: 1 > >> >> sleep 20 1 > >> >> > >> >> 2012-03-27 15:36:49.630810 [CONSOLE] mod_commands.c:559 Timer Test: 2 > >> >> sleep 20 21225 > >> >> > >> >> 2012-03-27 15:36:49.671927 [CONSOLE] mod_commands.c:559 Timer Test: 3 > >> >> sleep 20 40716 > >> >> > >> >> 2012-03-27 15:36:49.671927 [CONSOLE] mod_commands.c:559 Timer Test: 4 > >> >> sleep 20 29 > >> >> > >> >> 2012-03-27 15:36:49.690968 [CONSOLE] mod_commands.c:559 Timer Test: 5 > >> >> sleep 20 19451 > >> >> > >> >> 2012-03-27 15:36:49.714993 [CONSOLE] mod_commands.c:559 Timer Test: 6 > >> >> sleep 20 23380 > >> >> > >> >> 2012-03-27 15:36:49.714993 [CONSOLE] mod_commands.c:559 Timer Test: 7 > >> >> sleep 20 35 > >> >> > >> >> 2012-03-27 15:36:49.752814 [CONSOLE] mod_commands.c:559 Timer Test: 8 > >> >> sleep 20 38474 > >> >> > >> >> 2012-03-27 15:36:49.752814 [CONSOLE] mod_commands.c:559 Timer Test: 9 > >> >> sleep 20 41 > >> >> > >> >> 2012-03-27 15:36:49.776956 [CONSOLE] mod_commands.c:559 Timer Test: > 10 > >> >> sleep 20 23408 > >> >> > >> >> 2012-03-27 15:36:49.798632 [CONSOLE] mod_commands.c:559 Timer Test: > 11 > >> >> sleep 20 22079 > >> >> > >> >> 2012-03-27 15:36:49.838054 [CONSOLE] mod_commands.c:559 Timer Test: > 12 > >> >> sleep 20 39021 > >> >> > >> >> 2012-03-27 15:36:49.838054 [CONSOLE] mod_commands.c:559 Timer Test: > 13 > >> >> sleep 20 35 > >> >> > >> >> 2012-03-27 15:36:49.857783 [CONSOLE] mod_commands.c:559 Timer Test: > 14 > >> >> sleep 20 19946 > >> >> > >> >> 2012-03-27 15:36:49.887887 [CONSOLE] mod_commands.c:559 Timer Test: > 15 > >> >> sleep 20 29935 > >> >> > >> >> 2012-03-27 15:36:49.906809 [CONSOLE] mod_commands.c:559 Timer Test: > 16 > >> >> sleep 20 18774 > >> >> > >> >> 2012-03-27 15:36:49.946751 [CONSOLE] mod_commands.c:559 Timer Test: > 17 > >> >> sleep 20 40290 > >> >> > >> >> 2012-03-27 15:36:49.946751 [CONSOLE] mod_commands.c:559 Timer Test: > 18 > >> >> sleep 20 32 > >> >> > >> >> 2012-03-27 15:36:49.966757 [CONSOLE] mod_commands.c:559 Timer Test: > 19 > >> >> sleep 20 19813 > >> >> > >> >> 2012-03-27 15:36:50.005781 [CONSOLE] mod_commands.c:559 Timer Test: > 20 > >> >> sleep 20 39099 > >> >> > >> >> 2012-03-27 15:36:50.005781 [CONSOLE] mod_commands.c:559 Timer Test: > 21 > >> >> sleep 20 24 > >> >> > >> >> 2012-03-27 15:36:50.023894 [CONSOLE] mod_commands.c:559 Timer Test: > 22 > >> >> sleep 20 18116 > >> >> > >> >> 2012-03-27 15:36:50.046022 [CONSOLE] mod_commands.c:559 Timer Test: > 23 > >> >> sleep 20 22046 > >> >> > >> >> 2012-03-27 15:36:50.074716 [CONSOLE] mod_commands.c:559 Timer Test: > 24 > >> >> sleep 20 29054 > >> >> > >> >> 2012-03-27 15:36:50.074716 [CONSOLE] mod_commands.c:559 Timer Test: > 25 > >> >> sleep 20 25 > >> >> > >> >> 2012-03-27 15:36:50.111794 [CONSOLE] mod_commands.c:559 Timer Test: > 26 > >> >> sleep 20 37413 > >> >> > >> >> 2012-03-27 15:36:50.131778 [CONSOLE] mod_commands.c:559 Timer Test: > 27 > >> >> sleep 20 19684 > >> >> > >> >> 2012-03-27 15:36:50.155846 [CONSOLE] mod_commands.c:559 Timer Test: > 28 > >> >> sleep 20 23480 > >> >> > >> >> 2012-03-27 15:36:50.155846 [CONSOLE] mod_commands.c:559 Timer Test: > 29 > >> >> sleep 20 37 > >> >> > >> >> 2012-03-27 15:36:50.179738 [CONSOLE] mod_commands.c:559 Timer Test: > 30 > >> >> sleep 20 23811 > >> >> > >> >> 2012-03-27 15:36:50.196725 [CONSOLE] mod_commands.c:559 Timer Test: > 31 > >> >> sleep 20 18828 > >> >> > >> >> 2012-03-27 15:36:50.220604 [CONSOLE] mod_commands.c:559 Timer Test: > 32 > >> >> sleep 20 22205 > >> >> > >> >> 2012-03-27 15:36:50.250696 [CONSOLE] mod_commands.c:559 Timer Test: > 33 > >> >> sleep 20 30868 > >> >> > >> >> 2012-03-27 15:36:50.275841 [CONSOLE] mod_commands.c:559 Timer Test: > 34 > >> >> sleep 20 24303 > >> >> > >> >> 2012-03-27 15:36:50.275841 [CONSOLE] mod_commands.c:559 Timer Test: > 35 > >> >> sleep 20 57 > >> >> > >> >> 2012-03-27 15:36:50.299676 [CONSOLE] mod_commands.c:559 Timer Test: > 36 > >> >> sleep 20 24607 > >> >> > >> >> 2012-03-27 15:36:50.319195 [CONSOLE] mod_commands.c:559 Timer Test: > 37 > >> >> sleep 20 18660 > >> >> > >> >> 2012-03-27 15:36:50.336701 [CONSOLE] mod_commands.c:559 Timer Test: > 38 > >> >> sleep 20 18321 > >> >> > >> >> 2012-03-27 15:36:50.377677 [CONSOLE] mod_commands.c:559 Timer Test: > 39 > >> >> sleep 20 40551 > >> >> > >> >> 2012-03-27 15:36:50.377677 [CONSOLE] mod_commands.c:559 Timer Test: > 40 > >> >> sleep 20 33 > >> >> > >> >> 2012-03-27 15:36:50.398692 [CONSOLE] mod_commands.c:559 Timer Test: > 41 > >> >> sleep 20 20552 > >> >> > >> >> 2012-03-27 15:36:50.421692 [CONSOLE] mod_commands.c:559 Timer Test: > 42 > >> >> sleep 20 23208 > >> >> > >> >> 2012-03-27 15:36:50.459684 [CONSOLE] mod_commands.c:559 Timer Test: > 43 > >> >> sleep 20 38668 > >> >> > >> >> 2012-03-27 15:36:50.459684 [CONSOLE] mod_commands.c:559 Timer Test: > 44 > >> >> sleep 20 34 > >> >> > >> >> 2012-03-27 15:36:50.478817 [CONSOLE] mod_commands.c:559 Timer Test: > 45 > >> >> sleep 20 18432 > >> >> > >> >> 2012-03-27 15:36:50.519691 [CONSOLE] mod_commands.c:559 Timer Test: > 46 > >> >> sleep 20 41447 > >> >> > >> >> 2012-03-27 15:36:50.519691 [CONSOLE] mod_commands.c:559 Timer Test: > 47 > >> >> sleep 20 29 > >> >> > >> >> 2012-03-27 15:36:50.542590 [CONSOLE] mod_commands.c:559 Timer Test: > 48 > >> >> sleep 20 22699 > >> >> > >> >> 2012-03-27 15:36:50.552674 [CONSOLE] mod_commands.c:559 Timer Test: > 49 > >> >> sleep 20 10034 > >> >> > >> >> 2012-03-27 15:36:50.590676 [CONSOLE] mod_commands.c:559 Timer Test: > 50 > >> >> sleep 20 38060 > >> >> > >> >> 2012-03-27 15:36:50.608681 [CONSOLE] mod_commands.c:559 Timer Test: > 51 > >> >> sleep 20 17448 > >> >> > >> >> 2012-03-27 15:36:50.625755 [CONSOLE] mod_commands.c:559 Timer Test: > 52 > >> >> sleep 20 17767 > >> >> > >> >> 2012-03-27 15:36:50.643693 [CONSOLE] mod_commands.c:559 Timer Test: > 53 > >> >> sleep 20 17263 > >> >> > >> >> 2012-03-27 15:36:50.663677 [CONSOLE] mod_commands.c:559 Timer Test: > 54 > >> >> sleep 20 21429 > >> >> > >> >> 2012-03-27 15:36:50.682676 [CONSOLE] mod_commands.c:559 Timer Test: > 55 > >> >> sleep 20 18072 > >> >> > >> >> 2012-03-27 15:36:50.723740 [CONSOLE] mod_commands.c:559 Timer Test: > 56 > >> >> sleep 20 40933 > >> >> > >> >> 2012-03-27 15:36:50.723740 [CONSOLE] mod_commands.c:559 Timer Test: > 57 > >> >> sleep 20 29 > >> >> > >> >> 2012-03-27 15:36:50.758800 [CONSOLE] mod_commands.c:559 Timer Test: > 58 > >> >> sleep 20 35950 > >> >> > >> >> 2012-03-27 15:36:50.758800 [CONSOLE] mod_commands.c:559 Timer Test: > 59 > >> >> sleep 20 30 > >> >> > >> >> 2012-03-27 15:36:50.779674 [CONSOLE] mod_commands.c:559 Timer Test: > 60 > >> >> sleep 20 19772 > >> >> > >> >> > >> >> > >> >> The CPU load is low (20%) so what parameter can affect this? The > CentOS > >> >> version? The kernel versione? The FS version? The number of cpu > cores? > >> >> > >> >> > >> >> > >> >> Stephen > >> >> > >> >> > >> >> > >> >> On Mon, Mar 26, 2012 at 4:40 PM, Stephen Wilde > > >> >> wrote: > >> >> > >> >> Ok, I'll try with this parameter, I think I have to do a Freeswitch > >> >> restart to make this change... > >> >> > >> >> > >> >> > >> >> Stephen > >> >> > >> >> > >> >> > >> >> On Mon, Mar 26, 2012 at 4:31 PM, Peter Olsson > >> >> wrote: > >> >> > >> >> You could try to enable the ?old? 1 ms-timer. It will be a little > less > >> >> efficient (a little more CPU load), and I?m not sure it helps, but > it?s > >> >> worth a try. > >> >> > >> >> > >> >> > >> >> In switch.conf.xml (under autoload_configs), add this: >> >> name="1ms-timer" value="true"/> - within the settings-tags. > >> >> > >> >> > >> >> > >> >> /Peter > >> >> > >> >> > >> >> > >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org > >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen > >> >> Wilde > >> >> Skickat: den 26 mars 2012 16:20 > >> >> > >> >> > >> >> Till: FreeSWITCH Users Help > >> >> ?mne: Re: [Freeswitch-users] Strange "timer_test" result > >> >> > >> >> > >> >> > >> >> Thank you Peter for your reply! > >> >> > >> >> I have already tried the upgrade to latest git, now I'm on FreeSWITCH > >> >> Version 1.1.beta1 (git-c31a799 2012-03-24 14-11-49 -0700) and the > >> >> result is > >> >> the same. > >> >> > >> >> I have tried with few load (during night) and in this case the > >> >> timer_test > >> >> is perfect. > >> >> > >> >> Any suggestion? > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> On Mon, Mar 26, 2012 at 4:08 PM, Peter Olsson > >> >> wrote: > >> >> > >> >> Yes, most likely. > >> >> > >> >> > >> >> > >> >> It?s very strange though, that a machine like this gives so poor > timing > >> >> results. > >> >> > >> >> > >> >> > >> >> The first thing I would do is to upgrade to latest GIT HEAD. A patch > >> >> was > >> >> commited about two weeks ago, that does the calculation for > timer_test > >> >> more > >> >> properly (so time for logging is not calculated). > >> >> > >> >> > >> >> > >> >> Have you tried to do the same test with no load at all? > >> >> > >> >> > >> >> > >> >> /Peter > >> >> > >> >> > >> >> > >> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org > >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Stephen > >> >> Wilde > >> >> Skickat: den 26 mars 2012 15:52 > >> >> Till: FreeSWITCH Users Help > >> >> ?mne: Re: [Freeswitch-users] Strange "timer_test" result > >> >> > >> >> > >> >> > >> >> Can this issue affect the voice quality? > >> >> > >> >> > >> >> > >> >> On Fri, Mar 23, 2012 at 7:46 PM, Stephen Wilde > > >> >> wrote: > >> >> > >> >> Real hardware, a dedicate server with 2 Xeon X5670 (a total 24 core > >> >> each > >> >> one with 12Mb cache at 2.93GHz) that is running at 20% - 25% of load. > >> >> OS is > >> >> CentOS 5.7 64bit and FS is (git-0626c89 2012-02-29 14-45-39 -0600) > >> >> > >> >> > >> >> > >> >> On Fri, Mar 23, 2012 at 6:33 PM, Brian Foster < > bdfoster at endigotech.com> > >> >> wrote: > >> >> > >> >> Is this on virtualized or real hardware? > >> >> > >> >> > >> >> > >> >> -BDF > >> >> > >> >> On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde > > >> >> wrote: > >> >> > >> >> I have run a "timer_test" in a dedicated FS server and I see strange > >> >> result: it's normal? > >> >> > >> >> > >> >> > >> >> Stephen > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> freeswitch at internal> timer_test 20 40 > >> >> > >> >> Avg: 19.866ms Total Time: 795.880ms > >> >> > >> >> > >> >> > >> >> 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: > >> >> samplecount after init: 1 > >> >> > >> >> 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: > >> >> samplecount after first step: 2 > >> >> > >> >> 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 > >> >> sleep 20 19568 > >> >> > >> >> 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 > >> >> sleep 20 38231 > >> >> > >> >> 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 > >> >> sleep 20 18847 > >> >> > >> >> 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 > >> >> sleep 20 13982 > >> >> > >> >> 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 > >> >> sleep 20 34793 > >> >> > >> >> 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 > >> >> sleep 20 2 > >> >> > >> >> 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 > >> >> sleep 20 23166 > >> >> > >> >> 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 > >> >> sleep 20 16957 > >> >> > >> >> 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 > >> >> sleep 20 17643 > >> >> > >> >> 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: > 10 > >> >> sleep 20 18786 > >> >> > >> >> 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: > 11 > >> >> sleep 20 25100 > >> >> > >> >> 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: > 12 > >> >> sleep 20 18552 > >> >> > >> >> 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: > 13 > >> >> sleep 20 18815 > >> >> > >> >> 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: > 14 > >> >> sleep 20 19464 > >> >> > >> >> 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: > 15 > >> >> sleep 20 22012 > >> >> > >> >> 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: > 16 > >> >> sleep 20 13980 > >> >> > >> >> 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: > 17 > >> >> sleep 20 19065 > >> >> > >> >> 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: > 18 > >> >> sleep 20 39585 > >> >> > >> >> 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: > 19 > >> >> sleep 20 2 > >> >> > >> >> 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: > 20 > >> >> sleep 20 26255 > >> >> > >> >> 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: > 21 > >> >> sleep 20 17872 > >> >> > >> >> 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: > 22 > >> >> sleep 20 32191 > >> >> > >> >> 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: > 23 > >> >> sleep 20 22634 > >> >> > >> >> 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: > 24 > >> >> sleep 20 15483 > >> >> > >> >> 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: > 25 > >> >> sleep 20 22813 > >> >> > >> >> 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: > 26 > >> >> sleep 20 17099 > >> >> > >> >> 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: > 27 > >> >> sleep 20 1 > >> >> > >> >> 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: > 28 > >> >> sleep 20 29108 > >> >> > >> >> 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: > 29 > >> >> sleep 20 11492 > >> >> > >> >> 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: > 30 > >> >> sleep 20 20855 > >> >> > >> >> 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: > 31 > >> >> sleep 20 32579 > >> >> > >> >> 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: > 32 > >> >> sleep 20 18173 > >> >> > >> >> 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: > 33 > >> >> sleep 20 22666 > >> >> > >> >> 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: > 34 > >> >> sleep 20 23792 > >> >> > >> >> 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: > 35 > >> >> sleep 20 26158 > >> >> > >> >> 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: > 36 > >> >> sleep 20 13080 > >> >> > >> >> 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: > 37 > >> >> sleep 20 24609 > >> >> > >> >> 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: > 38 > >> >> sleep 20 1 > >> >> > >> >> 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: > 39 > >> >> sleep 20 19413 > >> >> > >> >> 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: > 40 > >> >> sleep 20 19820 > >> >> > >> >> freeswitch at internal> status > >> >> > >> >> UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 > milliseconds, > >> >> 696 microseconds > >> >> > >> >> FreeSWITCH is ready > >> >> > >> >> 955611 session(s) since startup > >> >> > >> >> 2192 session(s) 0/50 > >> >> > >> >> 6000 session(s) max > >> >> > >> >> min idle cpu 0.00/74.00 > >> >> > >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ > >> >> Professional FreeSWITCH Consulting Services: > >> >> consulting at freeswitch.org > >> >> http://www.freeswitchsolutions.com > >> >> > >> >> > >> >> > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org > >> >> http://wiki.freeswitch.org > >> >> http://www.cluecon.com > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> -- > >> >> Brian D. Foster > >> >> Endigo Computer LLC > >> >> Email: bdfoster at endigotech.com > >> >> Phone: 317-800-7876 > >> >> Indianapolis, Indiana, USA > >> >> > >> >> This message contains confidential information and is intended for > >> >> those > >> >> listed in the "To:", "CC:", and/or "BCC:" fields of the message > header. > >> >> If > >> >> you are not the intended recipient you are notified that disclosing, > >> >> copying, distributing or taking any action in reliance on the > contents > >> >> of > >> >> this information is strictly prohibited. E-mail transmission cannot > be > >> >> guaranteed to be secure or error-free as information could be > >> >> intercepted, > >> >> corrupted, lost, destroyed, arrive late or incomplete, or contain > >> >> viruses. > >> >> The sender therefore does not accept liability for any errors or > >> >> omissions > >> >> in the contents of this message, which arise as a result of e-mail > >> >> transmission. If verification is required please request a hard-copy > >> >> version. > >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ > >> >> Professional FreeSWITCH Consulting Services: > >> >> consulting at freeswitch.org > >> >> http://www.freeswitchsolutions.com > >> >> > >> >> > >> >> > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org > >> >> http://wiki.freeswitch.org > >> >> http://www.cluecon.com > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ > >> >> Professional FreeSWITCH Consulting Services: > >> >> consulting at freeswitch.org > >> >> http://www.freeswitchsolutions.com > >> >> > >> >> > >> >> > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org > >> >> http://wiki.freeswitch.org > >> >> http://www.cluecon.com > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ > >> >> Professional FreeSWITCH Consulting Services: > >> >> consulting at freeswitch.org > >> >> http://www.freeswitchsolutions.com > >> >> > >> >> > >> >> > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org > >> >> http://wiki.freeswitch.org > >> >> http://www.cluecon.com > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> !DSPAM:4f71c23b32762536418862! > >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ > >> >> Professional FreeSWITCH Consulting Services: > >> >> consulting at freeswitch.org > >> >> http://www.freeswitchsolutions.com > >> >> > >> >> > >> >> > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org > >> >> http://wiki.freeswitch.org > >> >> http://www.cluecon.com > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> > > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888 at conference.freeswitch.org > >> googletalk:conf+888 at conference.freeswitch.org > >> pstn:+19193869900 > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/628c19f2/attachment-0001.html From msc at freeswitch.org Tue Mar 27 20:09:45 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 27 Mar 2012 09:09:45 -0700 Subject: [Freeswitch-users] Trouble with transfer and voicemail password In-Reply-To: <4F719236.9030201@gmail.com> References: <4F719236.9030201@gmail.com> Message-ID: Trever, I think there is a better way to do this. Look at this example: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action#Catching_the_Digits_Dialed I think if you use the last digits dialed you should be able to avoid the play_and_get_digits altogether. You'll need to strip off the leading and trailing digits but that shouldn't be too difficult. -MC On Tue, Mar 27, 2012 at 3:11 AM, Trever L. Adams wrote: > > go ahead and get a debug log of a call coming in, being answered, and the > > recipient dialing *0#xxx# to send the call elsewhere. Drop it into > > pastebin.freeswitch.org and use "FreeSWITCH Log" as the syntax > highlighting. > > > > -MC > http://pastebin.freeswitch.org/18753 > > I am not sure how much the log will help. It doesn't show me pushing > anything after the transfer code (*01 as I was experimenting and forgot > to change it back to *0# as I had it set before). > > If I dial any thing 1000, 1000#, 1001, 1001# it all fails and plays the > invalid_extension.wav. I have changed the code a little bit. It now reads: > > > expression="^SETUP_TRANSFER$"> > > data="do_transfer,*01,exec:execute_extension,DO_TRANSFER XML > Incoming-FXO"/> > data="do_transfer"/> > > > > > > data="bridge_pre_execute_bleg_app=execute_extension"/> > data="bridge_pre_execute_bleg_data=SETUP_TRANSFER XML Incoming-FXO"/> > > > > > expression="^DO_TRANSFER$"> > > > > > > > > It seems that I had a problem with play and get digits not working when > I was working on my call screener (I stopped work do to bug > openzap-137). I seemed to have gotten around it by rewriting the entire > thing in lua. I may be remembering incorrectly as this was months ago. > > Thank you for any help, > Trever > -- > "Selfishness is really self-destruction in slow motion." -? Elder Neal > A. Maxwell - Ensign, May 1999, 23 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/7d167f52/attachment.html From cmrienzo at gmail.com Tue Mar 27 20:28:32 2012 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 27 Mar 2012 12:28:32 -0400 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Message-ID: You can also try mod_posix_timer if you have it built. My CentOS 5.4 server performs better under load using it with heavy-timer vs using the default timer. "timer_test 20 50 posix" On Tue, Mar 27, 2012 at 12:07 PM, Stephen Wilde wrote: > Ok, I'll try disabling the nanosleep. > > In server.2 the current load of the box is 35% and after FreeSwitch (at > 750% of cpu), in top there is Sangoma Transcoding server (4% of cpu). > > > The result of "timer_test 20 50" is: > > http://pastebin.freeswitch.org/18756 > > > Here the result of "timer_test 10 50" : > > http://pastebin.freeswitch.org/18757 > > It's normal that only the timer_test on 20 is bad where 10ms is good and > also is good 40,60,120?: > > Stephen > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/6db66183/attachment.html From wstephen80 at gmail.com Tue Mar 27 20:34:38 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Tue, 27 Mar 2012 18:34:38 +0200 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Message-ID: I have tried to use mod_posix_timer but with this timer I had some FS crashes (segmentation fault) with core dump so I'm moved back to soft timers. Stephen On Tue, Mar 27, 2012 at 6:28 PM, Christopher Rienzo wrote: > You can also try mod_posix_timer if you have it built. My CentOS 5.4 > server performs better under load using it with heavy-timer vs using the > default timer. > > "timer_test 20 50 posix" > > > > > On Tue, Mar 27, 2012 at 12:07 PM, Stephen Wilde wrote: > >> Ok, I'll try disabling the nanosleep. >> >> In server.2 the current load of the box is 35% and after FreeSwitch (at >> 750% of cpu), in top there is Sangoma Transcoding server (4% of cpu). >> >> >> The result of "timer_test 20 50" is: >> >> http://pastebin.freeswitch.org/18756 >> >> >> Here the result of "timer_test 10 50" : >> >> http://pastebin.freeswitch.org/18757 >> >> It's normal that only the timer_test on 20 is bad where 10ms is good and >> also is good 40,60,120?: >> >> Stephen >> >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/b156a4c9/attachment.html From cmrienzo at gmail.com Tue Mar 27 21:14:19 2012 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 27 Mar 2012 13:14:19 -0400 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Message-ID: Please report crashes in jira, otherwise they'll never get fixed. On Tue, Mar 27, 2012 at 12:34 PM, Stephen Wilde wrote: > I have tried to use mod_posix_timer but with this timer I had some FS > crashes (segmentation fault) with core dump so I'm moved back to soft > timers. > > Stephen > > On Tue, Mar 27, 2012 at 6:28 PM, Christopher Rienzo wrote: > >> You can also try mod_posix_timer if you have it built. My CentOS 5.4 >> server performs better under load using it with heavy-timer vs using the >> default timer. >> >> "timer_test 20 50 posix" >> >> >> >> >> On Tue, Mar 27, 2012 at 12:07 PM, Stephen Wilde wrote: >> >>> Ok, I'll try disabling the nanosleep. >>> >>> In server.2 the current load of the box is 35% and after FreeSwitch (at >>> 750% of cpu), in top there is Sangoma Transcoding server (4% of cpu). >>> >>> >>> The result of "timer_test 20 50" is: >>> >>> http://pastebin.freeswitch.org/18756 >>> >>> >>> Here the result of "timer_test 10 50" : >>> >>> http://pastebin.freeswitch.org/18757 >>> >>> It's normal that only the timer_test on 20 is bad where 10ms is good and >>> also is good 40,60,120?: >>> >>> Stephen >>> >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/06cd7afd/attachment-0001.html From lloyd.aloysius at gmail.com Tue Mar 27 21:29:35 2012 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Tue, 27 Mar 2012 13:29:35 -0400 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Message-ID: Tony, Your recommendation for using the new kernel, is stable for a production environment? Thanks Lloyd On Tue, Mar 27, 2012 at 1:14 PM, Christopher Rienzo wrote: > Please report crashes in jira, otherwise they'll never get fixed. > > > > On Tue, Mar 27, 2012 at 12:34 PM, Stephen Wilde wrote: > >> I have tried to use mod_posix_timer but with this timer I had some FS >> crashes (segmentation fault) with core dump so I'm moved back to soft >> timers. >> >> Stephen >> >> On Tue, Mar 27, 2012 at 6:28 PM, Christopher Rienzo wrote: >> >>> You can also try mod_posix_timer if you have it built. My CentOS 5.4 >>> server performs better under load using it with heavy-timer vs using the >>> default timer. >>> >>> "timer_test 20 50 posix" >>> >>> >>> >>> >>> On Tue, Mar 27, 2012 at 12:07 PM, Stephen Wilde wrote: >>> >>>> Ok, I'll try disabling the nanosleep. >>>> >>>> In server.2 the current load of the box is 35% and after FreeSwitch (at >>>> 750% of cpu), in top there is Sangoma Transcoding server (4% of cpu). >>>> >>>> >>>> The result of "timer_test 20 50" is: >>>> >>>> http://pastebin.freeswitch.org/18756 >>>> >>>> >>>> Here the result of "timer_test 10 50" : >>>> >>>> http://pastebin.freeswitch.org/18757 >>>> >>>> It's normal that only the timer_test on 20 is bad where 10ms is good >>>> and also is good 40,60,120?: >>>> >>>> Stephen >>>> >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/e2551bc0/attachment.html From msc at freeswitch.org Tue Mar 27 21:34:49 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 27 Mar 2012 10:34:49 -0700 Subject: [Freeswitch-users] Congrats to Gabe Gunderson - New Baby Girl Message-ID: Hey all, Just a hat tip to Gabe Gunderson (IRC: gundy) whose wife delivered a baby girl yesterday! Now that that business is done he's back to the office to work on the ClueCon website. :P Congrats Gabe! We'll expect to see some pics when we're all having a beer at ClueCon. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/470cb926/attachment.html From anthony.minessale at gmail.com Tue Mar 27 22:09:59 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 27 Mar 2012 13:09:59 -0500 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Message-ID: It should be, the timerfd has been around since kernel >= 2.6.25 and libc >= 2.8 On Tue, Mar 27, 2012 at 12:29 PM, Lloyd Aloysius wrote: > Tony, > > Your recommendation for using the new kernel, is stable for a production > environment? > > > Thanks > Lloyd > > > On Tue, Mar 27, 2012 at 1:14 PM, Christopher Rienzo > wrote: >> >> Please report crashes in jira, otherwise they'll never get fixed. >> >> >> >> On Tue, Mar 27, 2012 at 12:34 PM, Stephen Wilde >> wrote: >>> >>> I have tried to use mod_posix_timer but with this timer I had some FS >>> crashes (segmentation fault) with core dump so I'm moved back to soft >>> timers. >>> >>> Stephen >>> >>> On Tue, Mar 27, 2012 at 6:28 PM, Christopher Rienzo >>> wrote: >>>> >>>> You can also try mod_posix_timer if you have it built.? My CentOS 5.4 >>>> server performs better under load using it with heavy-timer vs using the >>>> default timer. >>>> >>>> "timer_test 20 50 posix" >>>> >>>> >>>> >>>> >>>> On Tue, Mar 27, 2012 at 12:07 PM, Stephen Wilde >>>> wrote: >>>>> >>>>> Ok, I'll try disabling the nanosleep. >>>>> >>>>> In server.2 the current load of the box is 35% and after FreeSwitch (at >>>>> 750% of cpu), in top there is Sangoma Transcoding server (4% of cpu). >>>>> >>>>> >>>>> The result of "timer_test 20 50" is: >>>>> >>>>> http://pastebin.freeswitch.org/18756 >>>>> >>>>> >>>>> Here the result of "timer_test 10 50" : >>>>> >>>>> http://pastebin.freeswitch.org/18757 >>>>> >>>>> It's normal that only the timer_test on 20 is bad where 10ms is good >>>>> and also is good 40,60,120?: >>>>> >>>>> Stephen >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From wstephen80 at gmail.com Tue Mar 27 22:24:17 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Tue, 27 Mar 2012 20:24:17 +0200 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Message-ID: The core dump was pretty unusable (stack corrupted) so I had some difficult to report to jira a useful info. On Tue, Mar 27, 2012 at 7:14 PM, Christopher Rienzo wrote: > Please report crashes in jira, otherwise they'll never get fixed. > > > > On Tue, Mar 27, 2012 at 12:34 PM, Stephen Wilde wrote: > >> I have tried to use mod_posix_timer but with this timer I had some FS >> crashes (segmentation fault) with core dump so I'm moved back to soft >> timers. >> >> Stephen >> >> On Tue, Mar 27, 2012 at 6:28 PM, Christopher Rienzo wrote: >> >>> You can also try mod_posix_timer if you have it built. My CentOS 5.4 >>> server performs better under load using it with heavy-timer vs using the >>> default timer. >>> >>> "timer_test 20 50 posix" >>> >>> >>> >>> >>> On Tue, Mar 27, 2012 at 12:07 PM, Stephen Wilde wrote: >>> >>>> Ok, I'll try disabling the nanosleep. >>>> >>>> In server.2 the current load of the box is 35% and after FreeSwitch (at >>>> 750% of cpu), in top there is Sangoma Transcoding server (4% of cpu). >>>> >>>> >>>> The result of "timer_test 20 50" is: >>>> >>>> http://pastebin.freeswitch.org/18756 >>>> >>>> >>>> Here the result of "timer_test 10 50" : >>>> >>>> http://pastebin.freeswitch.org/18757 >>>> >>>> It's normal that only the timer_test on 20 is bad where 10ms is good >>>> and also is good 40,60,120?: >>>> >>>> Stephen >>>> >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/a70e0888/attachment-0001.html From anthony.minessale at gmail.com Tue Mar 27 23:32:41 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 27 Mar 2012 14:32:41 -0500 Subject: [Freeswitch-users] double invite In-Reply-To: References: Message-ID: I don't seen any traffic from FS in this trace I see user agent MERA sending the invites. maybe you have something in the way. On Tue, Mar 27, 2012 at 8:27 AM, budi wibowo wrote: > dear all > i use?FreeSWITCH Version 1.0.head (git-9d3401e 2012-03-19 20-06-36 -0500) > and connect to some gateway over sip. > during test i found > - 2 Invite from FS then receive 1 Trying from other gw > - FS send cancel and other gw return 481 > - FS send ACK , and got return BYE > is it common happen? or any config in FS i can use to fine tune the setting? > attached is detail pcap file for consideration > > TIA > > budi wibowo > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From lloyd.aloysius at gmail.com Wed Mar 28 00:31:56 2012 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Tue, 27 Mar 2012 16:31:56 -0400 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Message-ID: Tony, Thank you. On Tue, Mar 27, 2012 at 2:24 PM, Stephen Wilde wrote: > The core dump was pretty unusable (stack corrupted) so I had some > difficult to report to jira a useful info. > > > On Tue, Mar 27, 2012 at 7:14 PM, Christopher Rienzo wrote: > >> Please report crashes in jira, otherwise they'll never get fixed. >> >> >> >> On Tue, Mar 27, 2012 at 12:34 PM, Stephen Wilde wrote: >> >>> I have tried to use mod_posix_timer but with this timer I had some FS >>> crashes (segmentation fault) with core dump so I'm moved back to soft >>> timers. >>> >>> Stephen >>> >>> On Tue, Mar 27, 2012 at 6:28 PM, Christopher Rienzo wrote: >>> >>>> You can also try mod_posix_timer if you have it built. My CentOS 5.4 >>>> server performs better under load using it with heavy-timer vs using the >>>> default timer. >>>> >>>> "timer_test 20 50 posix" >>>> >>>> >>>> >>>> >>>> On Tue, Mar 27, 2012 at 12:07 PM, Stephen Wilde wrote: >>>> >>>>> Ok, I'll try disabling the nanosleep. >>>>> >>>>> In server.2 the current load of the box is 35% and after FreeSwitch >>>>> (at 750% of cpu), in top there is Sangoma Transcoding server (4% of cpu). >>>>> >>>>> >>>>> The result of "timer_test 20 50" is: >>>>> >>>>> http://pastebin.freeswitch.org/18756 >>>>> >>>>> >>>>> Here the result of "timer_test 10 50" : >>>>> >>>>> http://pastebin.freeswitch.org/18757 >>>>> >>>>> It's normal that only the timer_test on 20 is bad where 10ms is good >>>>> and also is good 40,60,120?: >>>>> >>>>> Stephen >>>>> >>>>> >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/e7e5e2a0/attachment.html From cmrienzo at gmail.com Wed Mar 28 00:52:20 2012 From: cmrienzo at gmail.com (cmrienzo at gmail.com) Date: Tue, 27 Mar 2012 16:52:20 -0400 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Message-ID: <0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com> No big deal. I fixed a few issues a while back, so maybe your problem is resolved. Check the revision history to see if the changes happened after your crash. On Mar 27, 2012, at 14:24, Stephen Wilde wrote: > The core dump was pretty unusable (stack corrupted) so I had some difficult to report to jira a useful info. > > On Tue, Mar 27, 2012 at 7:14 PM, Christopher Rienzo wrote: > Please report crashes in jira, otherwise they'll never get fixed. > > > > On Tue, Mar 27, 2012 at 12:34 PM, Stephen Wilde wrote: > I have tried to use mod_posix_timer but with this timer I had some FS crashes (segmentation fault) with core dump so I'm moved back to soft timers. > > Stephen > > On Tue, Mar 27, 2012 at 6:28 PM, Christopher Rienzo wrote: > You can also try mod_posix_timer if you have it built. My CentOS 5.4 server performs better under load using it with heavy-timer vs using the default timer. > > "timer_test 20 50 posix" > > > > > On Tue, Mar 27, 2012 at 12:07 PM, Stephen Wilde wrote: > Ok, I'll try disabling the nanosleep. > > In server.2 the current load of the box is 35% and after FreeSwitch (at 750% of cpu), in top there is Sangoma Transcoding server (4% of cpu). > > > The result of "timer_test 20 50" is: > > http://pastebin.freeswitch.org/18756 > > > Here the result of "timer_test 10 50" : > > http://pastebin.freeswitch.org/18757 > > It's normal that only the timer_test on 20 is bad where 10ms is good and also is good 40,60,120?: > > Stephen > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/ce460614/attachment-0001.html From wstephen80 at gmail.com Wed Mar 28 01:00:31 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Tue, 27 Mar 2012 23:00:31 +0200 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: <0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com> References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> <0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com> Message-ID: Ok, thank you, I'll do a try with latest release. It remains my doubt on the result of timer_test with 10ms that is ok where the result with 20ms is not ok: there is an explanation? Stephen On Tue, Mar 27, 2012 at 10:52 PM, wrote: > No big deal. I fixed a few issues a while back, so maybe your problem is > resolved. Check the revision history to see if the changes happened after > your crash. > > > > > On Mar 27, 2012, at 14:24, Stephen Wilde wrote: > > The core dump was pretty unusable (stack corrupted) so I had some > difficult to report to jira a useful info. > > On Tue, Mar 27, 2012 at 7:14 PM, Christopher Rienzo wrote: > >> Please report crashes in jira, otherwise they'll never get fixed. >> >> >> >> On Tue, Mar 27, 2012 at 12:34 PM, Stephen Wilde wrote: >> >>> I have tried to use mod_posix_timer but with this timer I had some FS >>> crashes (segmentation fault) with core dump so I'm moved back to soft >>> timers. >>> >>> Stephen >>> >>> On Tue, Mar 27, 2012 at 6:28 PM, Christopher Rienzo wrote: >>> >>>> You can also try mod_posix_timer if you have it built. My CentOS 5.4 >>>> server performs better under load using it with heavy-timer vs using the >>>> default timer. >>>> >>>> "timer_test 20 50 posix" >>>> >>>> >>>> >>>> >>>> On Tue, Mar 27, 2012 at 12:07 PM, Stephen Wilde wrote: >>>> >>>>> Ok, I'll try disabling the nanosleep. >>>>> >>>>> In server.2 the current load of the box is 35% and after FreeSwitch >>>>> (at 750% of cpu), in top there is Sangoma Transcoding server (4% of cpu). >>>>> >>>>> >>>>> The result of "timer_test 20 50" is: >>>>> >>>>> http://pastebin.freeswitch.org/18756 >>>>> >>>>> >>>>> Here the result of "timer_test 10 50" : >>>>> >>>>> http://pastebin.freeswitch.org/18757 >>>>> >>>>> It's normal that only the timer_test on 20 is bad where 10ms is good >>>>> and also is good 40,60,120?: >>>>> >>>>> Stephen >>>>> >>>>> >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/4cf4fef8/attachment.html From anthony.minessale at gmail.com Wed Mar 28 01:03:09 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 27 Mar 2012 16:03:09 -0500 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> <0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com> Message-ID: I provided you with a list, did you cross them all off? On Tue, Mar 27, 2012 at 4:00 PM, Stephen Wilde wrote: > Ok, thank you, I'll do a try with latest release. > It remains my doubt on the result of timer_test with 10ms that is ok where > the result with 20ms is not ok: there is an explanation? > > Stephen > > On Tue, Mar 27, 2012 at 10:52 PM, wrote: >> >> No big deal. ?I fixed a few issues a while back, so maybe your problem is >> resolved. ?Check the revision history to see if the changes happened after >> your crash. >> >> >> >> >> On Mar 27, 2012, at 14:24, Stephen Wilde wrote: >> >> The core dump was pretty unusable (stack corrupted) so I had some >> difficult to report to jira a useful info. >> >> On Tue, Mar 27, 2012 at 7:14 PM, Christopher Rienzo >> wrote: >>> >>> Please report crashes in jira, otherwise they'll never get fixed. >>> >>> >>> >>> On Tue, Mar 27, 2012 at 12:34 PM, Stephen Wilde >>> wrote: >>>> >>>> I have tried to use mod_posix_timer but with this timer I had some FS >>>> crashes (segmentation fault) with core dump so I'm moved back to soft >>>> timers. >>>> >>>> Stephen >>>> >>>> On Tue, Mar 27, 2012 at 6:28 PM, Christopher Rienzo >>>> wrote: >>>>> >>>>> You can also try mod_posix_timer if you have it built.? My CentOS 5.4 >>>>> server performs better under load using it with heavy-timer vs using the >>>>> default timer. >>>>> >>>>> "timer_test 20 50 posix" >>>>> >>>>> >>>>> >>>>> >>>>> On Tue, Mar 27, 2012 at 12:07 PM, Stephen Wilde >>>>> wrote: >>>>>> >>>>>> Ok, I'll try disabling the nanosleep. >>>>>> >>>>>> In server.2 the current load of the box is 35% and after FreeSwitch >>>>>> (at 750% of cpu), in top there is Sangoma Transcoding server (4% of cpu). >>>>>> >>>>>> >>>>>> The result of "timer_test 20 50" is: >>>>>> >>>>>> http://pastebin.freeswitch.org/18756 >>>>>> >>>>>> >>>>>> Here the result of "timer_test 10 50" : >>>>>> >>>>>> http://pastebin.freeswitch.org/18757 >>>>>> >>>>>> It's normal that only the timer_test on 20 is bad where 10ms is good >>>>>> and also is good 40,60,120?: >>>>>> >>>>>> Stephen >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From wstephen80 at gmail.com Wed Mar 28 01:10:52 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Tue, 27 Mar 2012 23:10:52 +0200 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> <0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com> Message-ID: To apply the changes I have to do a FS restart on the production server that I'll do during night. On Tue, Mar 27, 2012 at 11:03 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I provided you with a list, did you cross them all off? > > > On Tue, Mar 27, 2012 at 4:00 PM, Stephen Wilde > wrote: > > Ok, thank you, I'll do a try with latest release. > > It remains my doubt on the result of timer_test with 10ms that is ok > where > > the result with 20ms is not ok: there is an explanation? > > > > Stephen > > > > On Tue, Mar 27, 2012 at 10:52 PM, wrote: > >> > >> No big deal. I fixed a few issues a while back, so maybe your problem > is > >> resolved. Check the revision history to see if the changes happened > after > >> your crash. > >> > >> > >> > >> > >> On Mar 27, 2012, at 14:24, Stephen Wilde wrote: > >> > >> The core dump was pretty unusable (stack corrupted) so I had some > >> difficult to report to jira a useful info. > >> > >> On Tue, Mar 27, 2012 at 7:14 PM, Christopher Rienzo > > >> wrote: > >>> > >>> Please report crashes in jira, otherwise they'll never get fixed. > >>> > >>> > >>> > >>> On Tue, Mar 27, 2012 at 12:34 PM, Stephen Wilde > >>> wrote: > >>>> > >>>> I have tried to use mod_posix_timer but with this timer I had some FS > >>>> crashes (segmentation fault) with core dump so I'm moved back to soft > >>>> timers. > >>>> > >>>> Stephen > >>>> > >>>> On Tue, Mar 27, 2012 at 6:28 PM, Christopher Rienzo < > cmrienzo at gmail.com> > >>>> wrote: > >>>>> > >>>>> You can also try mod_posix_timer if you have it built. My CentOS 5.4 > >>>>> server performs better under load using it with heavy-timer vs using > the > >>>>> default timer. > >>>>> > >>>>> "timer_test 20 50 posix" > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> On Tue, Mar 27, 2012 at 12:07 PM, Stephen Wilde < > wstephen80 at gmail.com> > >>>>> wrote: > >>>>>> > >>>>>> Ok, I'll try disabling the nanosleep. > >>>>>> > >>>>>> In server.2 the current load of the box is 35% and after FreeSwitch > >>>>>> (at 750% of cpu), in top there is Sangoma Transcoding server (4% of > cpu). > >>>>>> > >>>>>> > >>>>>> The result of "timer_test 20 50" is: > >>>>>> > >>>>>> http://pastebin.freeswitch.org/18756 > >>>>>> > >>>>>> > >>>>>> Here the result of "timer_test 10 50" : > >>>>>> > >>>>>> http://pastebin.freeswitch.org/18757 > >>>>>> > >>>>>> It's normal that only the timer_test on 20 is bad where 10ms is good > >>>>>> and also is good 40,60,120?: > >>>>>> > >>>>>> Stephen > >>>>>> > >>>>>> > >>>>> > >>>>> > >>>>> > _________________________________________________________________________ > >>>>> Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://wiki.freeswitch.org > >>>>> http://www.cluecon.com > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org > >>>>> > >>>> > >>>> > >>>> > >>>> > _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> http://www.cluecon.com > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120327/f0040ae7/attachment-0001.html From shane.harrison at paragon.co.nz Wed Mar 28 02:07:09 2012 From: shane.harrison at paragon.co.nz (Shane Harrison) Date: Wed, 28 Mar 2012 11:07:09 +1300 Subject: [Freeswitch-users] [SPAM] Re: Trouble with transfer and voicemail password In-Reply-To: <4F719EB0.6040803@gmail.com> References: <4F719236.9030201@gmail.com> <4F719EB0.6040803@gmail.com> Message-ID: Hi Trevor, Sounds like the same issue I have. I posted around 3 weeks ago but didn't get a reply. I was doing attended transfer on a POT. Basically the dialplan detected the attended transfer DTMF sequence and entered the correct dialplan extension but any digits pressed after that didn't seem to reach that dialplan extension. I am not sure if it is a FreeTDM issue or a dialplan issue. I could see no evidence in any logs that the digits were being delivered. I am about to get back to debugging the problem soon. My last thought was that I still have a default.xml in my dialplan and maybe it is doing things I am not aware of in some way. Sorry I can't offer any real help as I am fairly new to FS - but if it is the same problem then at least you know you aren't going mad, comfort in numbers etc :-) Cheers Shane On Wed, Mar 28, 2012 at 12:04 AM, Trever L. Adams wrote: > On 03/27/2012 04:58 AM, Brian Foster wrote: > > > > If it's anything like the Linksys phones, you might have to change the > > dialplan on the phone itself in order to dial certain digits and make > > sure all if it is sent to the switch instead of holding some of the > > dialed digits back. Double check this to make sure you aren't being > > sent on a wild goose chase. > > > > -BDF > > > BDF, > > Thank you, but this are actual POTS phones on FXS ports. Thank you for > the suggestion. > > Trever > -- > "The world is full of people who have never, since childhood, met an > open doorway with an open mind." -- E.B. White > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Paragon Electronic Design Ltd L6 Crest House 92 Queens Drive P0 Box 30449 Lower Hutt 5040 +64 4 5703870 Extn 875 +64 21 608919 (mobile) "Solving your problems with the right technology" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120328/46bee476/attachment.html From bwibowo at gmail.com Wed Mar 28 02:16:20 2012 From: bwibowo at gmail.com (budi wibowo) Date: Wed, 28 Mar 2012 05:16:20 +0700 Subject: [Freeswitch-users] double invite In-Reply-To: References: Message-ID: the RTP session is not happening in this case. as i know 1 invite will have 1 Trying as response. if the invite is more than 1, is it common and acceptable? On Wed, Mar 28, 2012 at 2:32 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I don't seen any traffic from FS in this trace I see user agent MERA > sending the invites. > maybe you have something in the way. > > On Tue, Mar 27, 2012 at 8:27 AM, budi wibowo wrote: > > dear all > > i use FreeSWITCH Version 1.0.head (git-9d3401e 2012-03-19 20-06-36 -0500) > > and connect to some gateway over sip. > > during test i found > > - 2 Invite from FS then receive 1 Trying from other gw > > - FS send cancel and other gw return 481 > > - FS send ACK , and got return BYE > > is it common happen? or any config in FS i can use to fine tune the > setting? > > attached is detail pcap file for consideration > > > > TIA > > > > budi wibowo > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120328/b26d1118/attachment.html From gabe at gundy.org Wed Mar 28 12:07:46 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 28 Mar 2012 02:07:46 -0600 Subject: [Freeswitch-users] Congrats to Gabe Gunderson - New Baby Girl In-Reply-To: References: Message-ID: On Tue, Mar 27, 2012 at 11:34 AM, Michael Collins wrote: > Just a hat tip to Gabe Gunderson (IRC: gundy) whose wife delivered a baby > girl yesterday! Now that that business is done he's back to the office to > work on the ClueCon website. :P Imagine my surprise when I jump on the list and spotted this :) Thanks for the congrats! And yes, the ClueCon site is going to get *lots* of attention tomorrow. > Congrats Gabe! We'll expect to see some pics when we're all having a beer at > ClueCon. Pics go here: http://gundy.org/ We'll mingle for sure, but club soda or Diet Coke for me ;) We really are looking forward to the next ClueCon. We had a great time last year and are counting down the days until this year's conference! Best, Gabe From anton.jugatsu at gmail.com Wed Mar 28 12:33:59 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Wed, 28 Mar 2012 12:33:59 +0400 Subject: [Freeswitch-users] Best practice to use t38_gateway for transcoding from T.38 to ALAW Message-ID: Hello, list. My ITSP supports T.38 for faxing pretty well and i _can_ send fax using ATA D-Link 5008s with t38_passthru Now i want to use FreeSWITCH-box as the fax gateway to send all faxes to asterisk box using transcoding from T.38 to just audio. My setup for outbound call: FAX --> ATA --> T.38 --> FS --> T.38 --> ITSP --> customer PSTN FAX This setup, using t38_passthru, works pretty reliable. Now i wanna: FAX --> ATA (fax disabled for purpose to not RE-INVITE) --> ALAW --> FS --> T.38 --> ITSP --> customer PSTN FAX or FAX --> ATA (fax disabled for purpose to not RE-INVITE) --> ALAW --> Asterisk --> ALAW --> FS --> T.38 --> ITSP --> customer PSTN FAX asterisk as a registrar for all endpoints @ office and FS (virtual machine) act as simple *sic* fax gateway. So, I swear, i tried all the configuratings from wiki ( http://i1110.photobucket.com/albums/h447/jugatsu85/280312-1.jpg) but it doesn't work at all. According to your requestd i'll post all pcap/flat dumps (actually, when i resolve a issue with my ITSP sending 403 Forbidden ^_^) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120328/c2dbe124/attachment.html From freeswitch-list at puzzled.xs4all.nl Wed Mar 28 15:59:32 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 28 Mar 2012 13:59:32 +0200 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> Message-ID: <4F72FD24.4020604@puzzled.xs4all.nl> On 03/27/2012 05:37 PM, Stephen Wilde wrote: > Thank you for your reply. > The motherboard cannot be the issue because 2 of these servers (server.2 > and server.5) are exactly the same hardware (HP DL380 G6). In the past I have seen similar hardware where one worked and the other did not. Yet all diagnostics said both boxes were fine. After a lot of digging the hardware in one of the boxes was faulty. Iirc it was PCIe related. So if you don't solve your problem, even with Anthony's recommendations, and you have ruled out any software issue(s) then perhaps it could be a hardware issue. Are all firmware versions similar in both boxes? Last I looked for a DL360 G6 there were firmware fixes for a bunch of nasty problems. Regards, Patrick From wstephen80 at gmail.com Wed Mar 28 17:31:34 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Wed, 28 Mar 2012 15:31:34 +0200 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> <0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com> Message-ID: I have tried: 1. 2. "heavy-timer" cmd line param 3. 1+2 In all cases, when the load is high (30% cpu), the test_timer result shows very irregular timings. Now I have started FS with mod_posix_timer and until now it works fine and also the test_timer result is acceptable: http://pastebin.freeswitch.org/18763 so for me the solution is the mod_posix_timer. Now I'm trying to install a CentOS 6 server to use timerfd. Stephen On Tue, Mar 27, 2012 at 11:03 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I provided you with a list, did you cross them all off? > > > On Tue, Mar 27, 2012 at 4:00 PM, Stephen Wilde > wrote: > > Ok, thank you, I'll do a try with latest release. > > It remains my doubt on the result of timer_test with 10ms that is ok > where > > the result with 20ms is not ok: there is an explanation? > > > > Stephen > > > > On Tue, Mar 27, 2012 at 10:52 PM, wrote: > >> > >> No big deal. I fixed a few issues a while back, so maybe your problem > is > >> resolved. Check the revision history to see if the changes happened > after > >> your crash. > >> > >> > >> > >> > >> On Mar 27, 2012, at 14:24, Stephen Wilde wrote: > >> > >> The core dump was pretty unusable (stack corrupted) so I had some > >> difficult to report to jira a useful info. > >> > >> On Tue, Mar 27, 2012 at 7:14 PM, Christopher Rienzo > > >> wrote: > >>> > >>> Please report crashes in jira, otherwise they'll never get fixed. > >>> > >>> > >>> > >>> On Tue, Mar 27, 2012 at 12:34 PM, Stephen Wilde > >>> wrote: > >>>> > >>>> I have tried to use mod_posix_timer but with this timer I had some FS > >>>> crashes (segmentation fault) with core dump so I'm moved back to soft > >>>> timers. > >>>> > >>>> Stephen > >>>> > >>>> On Tue, Mar 27, 2012 at 6:28 PM, Christopher Rienzo < > cmrienzo at gmail.com> > >>>> wrote: > >>>>> > >>>>> You can also try mod_posix_timer if you have it built. My CentOS 5.4 > >>>>> server performs better under load using it with heavy-timer vs using > the > >>>>> default timer. > >>>>> > >>>>> "timer_test 20 50 posix" > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> On Tue, Mar 27, 2012 at 12:07 PM, Stephen Wilde < > wstephen80 at gmail.com> > >>>>> wrote: > >>>>>> > >>>>>> Ok, I'll try disabling the nanosleep. > >>>>>> > >>>>>> In server.2 the current load of the box is 35% and after FreeSwitch > >>>>>> (at 750% of cpu), in top there is Sangoma Transcoding server (4% of > cpu). > >>>>>> > >>>>>> > >>>>>> The result of "timer_test 20 50" is: > >>>>>> > >>>>>> http://pastebin.freeswitch.org/18756 > >>>>>> > >>>>>> > >>>>>> Here the result of "timer_test 10 50" : > >>>>>> > >>>>>> http://pastebin.freeswitch.org/18757 > >>>>>> > >>>>>> It's normal that only the timer_test on 20 is bad where 10ms is good > >>>>>> and also is good 40,60,120?: > >>>>>> > >>>>>> Stephen > >>>>>> > >>>>>> > >>>>> > >>>>> > >>>>> > _________________________________________________________________________ > >>>>> Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://wiki.freeswitch.org > >>>>> http://www.cluecon.com > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org > >>>>> > >>>> > >>>> > >>>> > >>>> > _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> http://www.cluecon.com > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120328/dfd4169c/attachment-0001.html From wstephen80 at gmail.com Wed Mar 28 17:37:38 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Wed, 28 Mar 2012 15:37:38 +0200 Subject: [Freeswitch-users] Strange "timer_test" result In-Reply-To: <4F72FD24.4020604@puzzled.xs4all.nl> References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se> <4F72FD24.4020604@puzzled.xs4all.nl> Message-ID: Thank you Patrick, I'm sure that it's something hardware related because with same load, different hardware show different behaviour but I don't think it's a fault but something related to the architecture. For example to me it seems that HP server are more regular then Fujitsu or Dell at the same load. On Wed, Mar 28, 2012 at 1:59 PM, Patrick Lists < freeswitch-list at puzzled.xs4all.nl> wrote: > On 03/27/2012 05:37 PM, Stephen Wilde wrote: > > Thank you for your reply. > > The motherboard cannot be the issue because 2 of these servers (server.2 > > and server.5) are exactly the same hardware (HP DL380 G6). > > In the past I have seen similar hardware where one worked and the other > did not. Yet all diagnostics said both boxes were fine. After a lot of > digging the hardware in one of the boxes was faulty. Iirc it was PCIe > related. So if you don't solve your problem, even with Anthony's > recommendations, and you have ruled out any software issue(s) then > perhaps it could be a hardware issue. > > Are all firmware versions similar in both boxes? Last I looked for a > DL360 G6 there were firmware fixes for a bunch of nasty problems. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120328/989ca6e9/attachment.html From lazyvirus at gmx.com Wed Mar 28 17:40:35 2012 From: lazyvirus at gmx.com (Bzzz) Date: Wed, 28 Mar 2012 15:40:35 +0200 Subject: [Freeswitch-users] Congrats to Gabe Gunderson - New Baby Girl In-Reply-To: References: Message-ID: <20120328154035.617e3dff@anubis.defcon1> On Tue, 27 Mar 2012 10:34:49 -0700 Michael Collins wrote: > > Just a hat tip to Gabe Gunderson (IRC: gundy) whose wife delivered a baby > girl yesterday! Hehe, CONNECTION SUCCEED ;) Congrats to the mother & Gabe. JY -- From fs-list at communicatefreely.net Wed Mar 28 18:02:10 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Wed, 28 Mar 2012 10:02:10 -0400 Subject: [Freeswitch-users] One-legged callee_id Message-ID: <4F7319E2.5010107@communicatefreely.net> Hello, I have been making extensive use of the callee_id_name and callee_id_number variables so that our IP phones update the display to show who they are actually talking to on the other end. Generally, it works very well, but it only seems to work if there is a bridge. Is there a way to set the callee_id in a single legged call? ie. I want the name of the company to come up when the caller is in the main IVR, or the name of the conference to come up if they are in the conference room. Is there a way to do this? Thanks! -Tim From krice at freeswitch.org Wed Mar 28 19:05:27 2012 From: krice at freeswitch.org (Ken Rice) Date: Wed, 28 Mar 2012 10:05:27 -0500 Subject: [Freeswitch-users] Open Bugs on Jira and Call for help Message-ID: If you have open bugs on Jira, please make sure they are up to date on any all information.... If you wish to address a specific bug, please visit us on the weekly conference call so we can address them... We?re getting close to time for rolling Release Candidate for 1.2 and we want to address as many of these as possible... Also, I?m looking for some configuration documentation help. If you?ve pulled git in the past few days you?ll notice that in the module src directories there is now a conf directory, we need some help with making sure all the configuration options are documented and what their default settings are. The end goal is to get all the options documented in their default states. What skills do you need? The ability to read a little C, and the ability to use a text editor to edit/create configs... K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120328/47c3c451/attachment.html From fuji246 at gmail.com Wed Mar 28 19:13:02 2012 From: fuji246 at gmail.com (Fu Jiantao) Date: Wed, 28 Mar 2012 23:13:02 +0800 Subject: [Freeswitch-users] Binary download with curl In-Reply-To: <1FFF97C269757C458224B7C895F35F15080018@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F15080018@cantor.std.visionutv.se> Message-ID: I think you should clean, and try rebuild, or else, try to get clean copy, and don't forget to set core.autocrlf to false http://wiki.freeswitch.org/wiki/Installation_for_Windows 2012/3/27 Peter Olsson > Thanks for the input. But I think this is only true if you put the > sources under a protected directory, like ?C:\Program Files?? If the > sources are put somewhere else it should work even with UAC enabled (at > least it does for me :)).**** > > ** ** > > /Peter**** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Fu Jiantao > *Skickat:* den 27 mars 2012 11:45 > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] Binary download with curl**** > > ** ** > > Hi , > > I've got some experience to share with you guys, if you're using WIN7, > please turn off the "UAC", or else, the download vb script will failed to > run. > > Hope this can help! > > > **** > > 2012/3/27 Peter Olsson **** > > If you follow the instructions on the wiki the build should work out of > the box. The most common problem is that you?ve enabled autocrlf in git, > which causes the build to break.**** > > **** > > If you still have problems after following the instructions, please report > to Jira, since this is something that should be working.**** > > **** > > /Peter**** > > **** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Olli Aro > *Skickat:* den 26 mars 2012 10:05 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* [Freeswitch-users] Binary download with curl**** > > **** > > **** > > Hi all,**** > > **** > > Been trying to compile Freeswitch on Windows in order to get in the XML > curl module, however it is not looking easy (76 more fatal errors to figure > out...)**** > > **** > > Before I start that task I thought it is always worth asking :)**** > > **** > > Would anyone happen to have a download link to already compiled Windows > 32-bit binary for close to recent version of Freeswitch with the XML curl > module?**** > > **** > > Regards,**** > > **** > > Olli**** > > **** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > !DSPAM:4f718ac732761986176360! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120328/84e72c80/attachment-0001.html From genachka at gmail.com Wed Mar 28 19:20:04 2012 From: genachka at gmail.com (Gennady) Date: Wed, 28 Mar 2012 15:20:04 +0000 (UTC) Subject: [Freeswitch-users] Compiling Latest GIT References: <4F481410.1080203@privatedemail.net> Message-ID: Brian Foster writes: > > > Gennady > > -j n isn't supported by the makefile, yet. I'd refrain from using it even if you're just throwing one thread at the cpu. > > -BDF > > > On Tue, Mar 20, 2012 at 7:50 PM, Gennady wrote: > Dome Charoenyost ...> writes: > > > > apt-get install gawk > > and then start first step > I've installed in Ubuntu 64-bit 11.04 without issues before and am now having > the exact same problem as described on a new Ubuntu 64-bit 11.10 server. > I've tried to do a ./bootstrap.sh and ./configure and make -j1 as was suggested > and gawk is installed and set up as alternative. But still same error. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting- YF8E+gPBBv73h3GqohbjpQ at public.gmane.orghttp://www.freeswitchsolutions.com > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > FreeSWITCH-users mailing listFreeSWITCH-users lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch- users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- usershttp://www.freeswitch.org > > > > > -- Brian D. FosterEndigo Computer LLCEmail: bdfoster- 15yuSvdC0LkqDJ6do+/SaQ at public.gmane.orgPhone: 317-800-7876 > Indianapolis, Indiana, USAThis message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at ... > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at ... > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > I only tried the -j switch as it was suggested in the thread. But I was getting the error without it. I gave up trying to get it working on 11.10 64bit and installed 11.10 32bit and am up and running without issues. From trever.adams at gmail.com Wed Mar 28 20:24:57 2012 From: trever.adams at gmail.com (Trever L. Adams) Date: Wed, 28 Mar 2012 10:24:57 -0600 Subject: [Freeswitch-users] multiple bridge_pre_execute_bleg_app Message-ID: <4F733B59.1050506@gmail.com> Hello Everyone, I am trying hard to do my various setups in a moduluar way. Unfortunately, I have hit a problem where I am binding internal transfers on incoming calls to *01 and *00 to crap on those violating the do not call list. Both are being done in two different xml files in the Incoming dialplan with bridge_pre_execute_bleg_app. Both work if only one of the two is in the dialplan. Put both in and only the second file processed is picked up. Is it possible to do both in different files and have one somehow readd the original entry? Thank you, Trever From dujinfang at gmail.com Wed Mar 28 20:41:23 2012 From: dujinfang at gmail.com (Seven Du) Date: Thu, 29 Mar 2012 00:41:23 +0800 Subject: [Freeswitch-users] Open Bugs on Jira and Call for help In-Reply-To: References: Message-ID: <19C7DA849AE64EB0AABCD4009CA85F25@gmail.com> Cool. I'll update some of my open bugs on jira. And for documentation help, I suggest also address the indents, especially don't mix spaces and tabs, I guess all like tabs? Also, is there any coordination of which conf files who's working on or should all choose randomly? And show we commit to git or report on jira? On Wednesday, March 28, 2012 at 11:05 PM, Ken Rice wrote: > Open Bugs on Jira and Call for help If you have open bugs on Jira, please make sure they are up to date on any all information.... > > If you wish to address a specific bug, please visit us on the weekly conference call so we can address them... We?re getting close to time for rolling Release Candidate for 1.2 and we want to address as many of these as possible... > > Also, I?m looking for some configuration documentation help. If you?ve pulled git in the past few days you?ll notice that in the module src directories there is now a conf directory, we need some help with making sure all the configuration options are documented and what their default settings are. The end goal is to get all the options documented in their default states. > What skills do you need? The ability to read a little C, and the ability to use a text editor to edit/create configs... > > K > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/ab8a5119/attachment.html From trever.adams at gmail.com Wed Mar 28 20:41:41 2012 From: trever.adams at gmail.com (Trever L. Adams) Date: Wed, 28 Mar 2012 10:41:41 -0600 Subject: [Freeswitch-users] Trouble with transfer and voicemail password In-Reply-To: References: <4F719236.9030201@gmail.com> Message-ID: <4F733F45.2020705@gmail.com> On 03/27/2012 10:09 AM, Michael Collins wrote: > Trever, > > I think there is a better way to do this. Look at this example: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action#Catching_the_Digits_Dialed > > I think if you use the last digits dialed you should be able to avoid > the play_and_get_digits altogether. You'll need to strip off the > leading and trailing digits but that shouldn't be too difficult. > > -MC > Michael, Thank you for the response. The problem with your suggestion is that what I am trying to do is in the documentation, slightly different situation, at http://wiki.freeswitch.org/wiki/Conference_Add_Call_Example Additionally, since this is a POTS setup, your suggestion puts more tones in the callers ear. One final problem is that is that I am not sure how to strip off the *0# from the extension. I know the regex would be ^*0#(\d{4})$. I am just not sure how to match that against last_matching_digits and save it or use $1 elsewhere. Thank you, Trever From curriegrad2004 at gmail.com Wed Mar 28 20:43:12 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 28 Mar 2012 09:43:12 -0700 Subject: [Freeswitch-users] Compiling Latest GIT In-Reply-To: References: <4F481410.1080203@privatedemail.net> Message-ID: The -j flag just works fine. If it doesn't work then fall back to single make thread. On 2012-03-28 8:40 AM, "Gennady" wrote: > Brian Foster writes: > > > > > > > Gennady > > > > -j n isn't supported by the makefile, yet. I'd refrain from using it > even if > you're just throwing one thread at the cpu. > > > > -BDF > > > > > > On Tue, Mar 20, 2012 at 7:50 PM, Gennady Re5JQEeQqe8AvxtiuMwx3w at public.gmane.org> wrote: > > Dome Charoenyost ...> writes: > > > > > > apt-get install gawk > > > and then start first step > > I've installed in Ubuntu 64-bit 11.04 without issues before and am now > having > > the exact same problem as described on a new Ubuntu 64-bit 11.10 server. > > I've tried to do a ./bootstrap.sh and ./configure and make -j1 as was > suggested > > and gawk is installed and set up as alternative. But still same error. > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services:consulting- > YF8E+gPBBv73h3GqohbjpQ at public.gmane.orghttp://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > Official FreeSWITCH > Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp:// > www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users > lists.freeswitch.orghttp:// > lists.freeswitch.org/mailman/listinfo/freeswitch- > users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > usershttp://www.freeswitch.org > > > > > > > > > > -- Brian D. FosterEndigo Computer LLCEmail: bdfoster- > 15yuSvdC0LkqDJ6do+/SaQ at public.gmane.orgPhone: 317-800-7876 > > Indianapolis, Indiana, USAThis message contains confidential information > and > is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of > the > message header. If you are not the intended recipient you are notified that > disclosing, copying, distributing or taking any action in reliance on the > contents of this information is strictly prohibited. E-mail transmission > cannot > be guaranteed to be secure or error-free as information could be > intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The > sender therefore does not accept liability for any errors or omissions in > the > contents of this message, which arise as a result of e-mail transmission. > If > verification is required please request a hard-copy version. > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at ... > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at ... > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > I only tried the -j switch as it was suggested in the thread. But I was > getting > the error without it. > > I gave up trying to get it working on 11.10 64bit and installed 11.10 > 32bit and > am up and running without issues. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120328/02858d8b/attachment-0001.html From krice at freeswitch.org Wed Mar 28 20:59:20 2012 From: krice at freeswitch.org (Ken Rice) Date: Wed, 28 Mar 2012 11:59:20 -0500 Subject: [Freeswitch-users] Open Bugs on Jira and Call for help In-Reply-To: <19C7DA849AE64EB0AABCD4009CA85F25@gmail.com> Message-ID: Actually we can opena tracking bug on Jira or just email me the diffs I can review and commit that way... Chjeck out git-format-patch its a good way to format a patch so you get credit also K On 3/28/12 11:41 AM, "Seven Du" wrote: > > Cool. I'll update some of my open bugs on jira. And for documentation help, I > suggest also address the indents, especially don't mix spaces and tabs, I > guess all like tabs? Also, is there any coordination of which conf files who's > working on or should all choose randomly? And show we commit to git or report > on jira? > > > > On Wednesday, March 28, 2012 at 11:05 PM, Ken Rice wrote: > >> >> Open Bugs on Jira and Call for help If you have open bugs on Jira, please >> make sure they are up to date on any all information.... >> >> If you wish to address a specific bug, please visit us on the weekly >> conference call so we can address them... We?re getting close to time for >> rolling Release Candidate for 1.2 and we want to address as many of these as >> possible... >> >> Also, I?m looking for some configuration documentation help. If you?ve pulled >> git in the past few days you?ll notice that in the module src directories >> there is now a conf directory, we need some help with making sure all the >> configuration options are documented and what their default settings are. >> The end goal is to get all the options documented in their default states. >> What skills do you need? The ability to read a little C, and the ability to >> use a text editor to edit/create configs... >> >> K >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120328/7e6d83f5/attachment.html From vipkilla at gmail.com Wed Mar 28 23:17:43 2012 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 28 Mar 2012 15:17:43 -0400 Subject: [Freeswitch-users] mod_callcenter and moh-sound Message-ID: The wiki says you can use the channel variable ${hold_music} (http://wiki.freeswitch.org/wiki/Mod_callcenter#moh-sound) Yet when you use You will get: mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/${hold_music}.PCMU If you use It will play the hold music set in the global variable. Is there any way for mod_callcenter to take a channel variable as the music on hold? From vipkilla at gmail.com Wed Mar 28 23:32:58 2012 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 28 Mar 2012 15:32:58 -0400 Subject: [Freeswitch-users] mod_callcenter and moh-sound In-Reply-To: References: Message-ID: This was answered in IRC: use the channel variable: cc_moh_override like or or Thanks Moc, this module is badass! On Wed, Mar 28, 2012 at 3:17 PM, Vik Killa wrote: > The wiki says you can use the channel variable ${hold_music} > (http://wiki.freeswitch.org/wiki/Mod_callcenter#moh-sound) > Yet when you use > You will get: > mod_native_file.c:74 Error opening > /usr/local/freeswitch/sounds/en/us/callie/${hold_music}.PCMU > > If you use > It will play the hold music set in the global variable. > Is there any way for mod_callcenter to take a channel variable as the > music on hold? From Rob.Moore at Aeriandi.com Thu Mar 29 02:57:49 2012 From: Rob.Moore at Aeriandi.com (Rob Moore) Date: Wed, 28 Mar 2012 22:57:49 +0000 Subject: [Freeswitch-users] Registration VIA TCP Message-ID: <49C5FCA19A8A114493EBAACA42FE589910503568@1AERDCEXCHMBX1.AER.AERCO.local> Hi Guys, Bit of an odd one here. We've been using UDP as the Transport Protocol for a Good while with a number of different SIP Phones but main Yealink SIP-T22P's. Due to some recently changes around the network needing to make things a little more robust we've had to look at shifting registration to TCP (and eventually to TLS but I have this issue to master first). We've had our SIPPhones working on multiple sites using UDP on a setup similar to this: [sipPhone]----------[SonicwallTMG]--------------(internet)---------------[SonicwallTMG] --------[Freeswitch] When changing to TCP phones can register fine and call to the outside world via freeswitch and our gateway providers however when attempting to make a call from the freeswitch box to the sip phone the call times out with NORMAL_TEMPORARY_FAILURE. This would suggest that the problem is related to NATing but we have exactly the same NAT rules for UDP as TCP in both firewalls and calls from the Freeswitch box with UDP can reach the sipPhone fine. Also doing a local test with TCP/TLS in our office works fine with STUN disabled and no other firewalls / routers in the way. I've tried a combination of various phones and configurations all with STUN enabled but as yet to no avail. I'm going to dig deeper on this tomorrow however I thought I'd send something out here to see if there is something simple that I've missed. The problem stinks of an issue with basic configuration of either Freeswitch or the firewalls we have in place but I can't see anything immediately that could be causing an issue as fundamental as this. If Anyone has any suggestions or has experienced this before I'd appreciate any help or direction on offer! Thanks in Advance Rob. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120328/4a584c5e/attachment.html From sherifomran2000 at yahoo.com Thu Mar 29 05:49:58 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Wed, 28 Mar 2012 18:49:58 -0700 (PDT) Subject: [Freeswitch-users] Forward calls from opensips to freeswitch Message-ID: <1332985798.30098.YahooMailClassic@web110803.mail.gq1.yahoo.com> Hello guys, any body knows how to forward calls from opensips dispatcher to freeswitch? Opensips + freeswitch are running on the same server. 1- Opensips dispatcher set to send calls to 127.0.0.1:5070 2- Freeswitch is set to use internal profile at 5070 (please help to check) and other settings? thanks Sherif Omran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120328/ddba614b/attachment.html From mitch.capper at gmail.com Thu Mar 29 06:02:45 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Wed, 28 Mar 2012 19:02:45 -0700 Subject: [Freeswitch-users] Registration VIA TCP In-Reply-To: <49C5FCA19A8A114493EBAACA42FE589910503568@1AERDCEXCHMBX1.AER.AERCO.local> References: <49C5FCA19A8A114493EBAACA42FE589910503568@1AERDCEXCHMBX1.AER.AERCO.local> Message-ID: Rob this is related to FS-3877 (Reuse TCP connection for both directions) freeswitch will attempt to establish a TCP connection back to the sip phone when sending it that incoming call and that is not working causing the failure would be the best guess. Ensuring the proper contact port may allow it to reuse this connection and not try to create this reverse connect, but otherwise ensuring the FS server can properly connect back to the client will probably solve it. Crank sofia loglevel should show the exact issue. ~Mitch From mitch.capper at gmail.com Thu Mar 29 06:10:28 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Wed, 28 Mar 2012 19:10:28 -0700 Subject: [Freeswitch-users] Forward calls from opensips to freeswitch In-Reply-To: <1332985798.30098.YahooMailClassic@web110803.mail.gq1.yahoo.com> References: <1332985798.30098.YahooMailClassic@web110803.mail.gq1.yahoo.com> Message-ID: Normally the internal profile is the authed profile so make sure to use ip or cred acls to properly have opensips be able to send on the internal profile (or disable that check) but otherwise you should be fine. ~mitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120328/1eaf4f98/attachment.html From bdfoster at endigotech.com Thu Mar 29 08:40:40 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 29 Mar 2012 00:40:40 -0400 Subject: [Freeswitch-users] Forward calls from opensips to freeswitch In-Reply-To: References: <1332985798.30098.YahooMailClassic@web110803.mail.gq1.yahoo.com> Message-ID: ...or just send the calls to the external profile. Also, unless you've got two internal profiles, you aren't likely able to dispatch calls to localhost AND serve users on it's actual IP. I'd just have a duplicate external profile and set it to accept calls on 127.0.0.1:5080 (or whatever port you want to use). -BDF On Wed, Mar 28, 2012 at 10:10 PM, Mitch Capper wrote: > Normally the internal profile is the authed profile so make sure to use ip > or cred acls to properly have opensips be able to send on the internal > profile (or disable that check) but otherwise you should be fine. > > ~mitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/1c1c4e09/attachment.html From joe.jflemmings at gmail.com Thu Mar 29 10:56:02 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Wed, 28 Mar 2012 23:56:02 -0700 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. Message-ID: What is the most scalable way to control FreeSwitch Routing dynamically (apart from XML configs) that has the least load on FreeSwitch and scales the best. This is to help achieve the most Call Setups Per Second (CPS) eg xml_rpc, lua, event socket etc Joe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120328/d9be8c82/attachment.html From markus.lindenberg at gmail.com Thu Mar 29 11:32:20 2012 From: markus.lindenberg at gmail.com (Markus Lindenberg) Date: Thu, 29 Mar 2012 09:32:20 +0200 Subject: [Freeswitch-users] One-legged callee_id In-Reply-To: <4F7319E2.5010107@communicatefreely.net> References: <4F7319E2.5010107@communicatefreely.net> Message-ID: Hi Tim, would love this as well. I guess the answer application should see if effective_callee_id_* is set and trigger a display update on the calling phone. It would also be nice to be able to trigger display updates to the a leg during (ivr) execution. Asterisk's CONNECTEDLINE() allows all of this. It's using Reinvites/Updates w/ PAI-Header and the syntax is cumbersome as everything asterisk is, but at least it works with most phones. (Snom 3xx, Snom M3, Siemens OpenStage, Polycom confirmed as i'm working with these every day) FreeSWITCH display updates seem to rely on workarounds (grep polycom or snom in mod_sofia). I would like to be able to disable the INFO/sipfrag thing for Snom, use PAI instead of RDNIS in the A-Leg and send Reinvites for Display Update etc., but it all seems to be hard coded right now. Regards, Markus On Wed, Mar 28, 2012 at 16:02, Tim St. Pierre wrote: > Hello, > > I have been making extensive use of the callee_id_name and > callee_id_number variables so that our IP phones update the display to > show who they are actually talking to on the other end. > > Generally, it works very well, but it only seems to work if there is a > bridge. ?Is there a way to set the callee_id in a single legged call? > ie. I want the name of the company to come up when the caller is in the > main IVR, or the name of the conference to come up if they are in the > conference room. > > Is there a way to do this? > > Thanks! > > -Tim > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From andrew at cassidywebservices.co.uk Thu Mar 29 12:49:19 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 29 Mar 2012 09:49:19 +0100 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. In-Reply-To: References: Message-ID: I use xml_curl, reads config from a central server so is replicated across you whole cluster, and directory/dialplan are requested every call (but supply filter parameters) and in effect dynamic if you generate these from a web service. On 29 March 2012 07:56, Joe Flemmings wrote: > What is the most scalable way to control FreeSwitch Routing dynamically > (apart from XML configs) that has the least load on FreeSwitch and scales > the best. This is to help achieve the most Call Setups Per Second (CPS) > > eg xml_rpc, lua, event socket etc > > > > Joe > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Andrew Cassidy BSc (Hons) MBCS Managing Director; Cassidy Web Services Ltd T: 03300 100 960 F: 03300 100 961 E: andrew at cassidywebservices.co.uk W: www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/91dc2454/attachment.html From sharad at coraltele.com Thu Mar 29 15:04:10 2012 From: sharad at coraltele.com (Sharad Garg) Date: Thu, 29 Mar 2012 16:34:10 +0530 Subject: [Freeswitch-users] Session-wise Language Setting References: Message-ID: <4CB4E6075A9F4F9BA1C8A57FF2A06735@sharad> Hi All Just wondering whether we can define the language from beginning of the call means when a call is originated or landed to Freeswitch, can we define the language of all the prompts whether the call should be processed in English or Russian or so on.? Regards Sharad ----- Original Message ----- From: To: Sent: Thursday, March 29, 2012 7:33 AM Subject: FreeSWITCH-users Digest, Vol 69, Issue 282 > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > -------------------------------------------------------------------------------- > Today's Topics: > > 1. Re: Open Bugs on Jira and Call for help (Ken Rice) > 2. mod_callcenter and moh-sound (Vik Killa) > 3. Re: mod_callcenter and moh-sound (Vik Killa) > 4. Registration VIA TCP (Rob Moore) > 5. Forward calls from opensips to freeswitch (Sherif Omran) > 6. Re: Registration VIA TCP (Mitch Capper) > -------------------------------------------------------------------------------- > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Rob.Moore at Aeriandi.com Thu Mar 29 16:16:13 2012 From: Rob.Moore at Aeriandi.com (Rob Moore) Date: Thu, 29 Mar 2012 12:16:13 +0000 Subject: [Freeswitch-users] Registration VIA TCP In-Reply-To: References: <49C5FCA19A8A114493EBAACA42FE589910503568@1AERDCEXCHMBX1.AER.AERCO.local> Message-ID: <49C5FCA19A8A114493EBAACA42FE58991050380F@1AERDCEXCHMBX1.AER.AERCO.local> Awesome thanks Mitch, I'll give that a blast. Thanks for the advice, I'll reply with how I get on in a bit -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch Capper Sent: 29 March 2012 03:03 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Registration VIA TCP Rob this is related to FS-3877 (Reuse TCP connection for both directions) freeswitch will attempt to establish a TCP connection back to the sip phone when sending it that incoming call and that is not working causing the failure would be the best guess. Ensuring the proper contact port may allow it to reuse this connection and not try to create this reverse connect, but otherwise ensuring the FS server can properly connect back to the client will probably solve it. Crank sofia loglevel should show the exact issue. ~Mitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From trever.adams at gmail.com Thu Mar 29 16:30:36 2012 From: trever.adams at gmail.com (Trever L. Adams) Date: Thu, 29 Mar 2012 06:30:36 -0600 Subject: [Freeswitch-users] [SOLVED] Re: multiple bridge_pre_execute_bleg_app In-Reply-To: <4F733B59.1050506@gmail.com> References: <4F733B59.1050506@gmail.com> Message-ID: <4F7455EC.8060205@gmail.com> On 03/28/2012 10:24 AM, Trever L. Adams wrote: > Hello Everyone, > > I am trying hard to do my various setups in a moduluar way. > Unfortunately, I have hit a problem where I am binding internal > transfers on incoming calls to *01 and *00 to crap on those violating > the do not call list. > > Both are being done in two different xml files in the Incoming > dialplan with bridge_pre_execute_bleg_app. Both work if only one of > the two is in the dialplan. Put both in and only the second file > processed is picked up. > > Is it possible to do both in different files and have one somehow > readd the original entry? > > Thank you, > Trever This actually turned out to be quite easy. If you are creating binds or whatever in a modular way and need multiple setup runs, here is how I solved it. Please, note, that 3:1 is my eternal POTS line. So, this binds on all incoming calls. Every modular setup should have something like this: This will make sure that it is always setup whether or not you remove any given files. Then, in each file, have an extension like this: Change "SETUP_INCOMING_FXO_BRIDGE_PRE_EXEC" in both examples to be whatever you want. Make sure that the binding extension (the second above) all have continue="true". This will allow them to all match. Thanks for those who helped. Trever -- "They that can give up essential liberty to obtain a little temporary safety deserve neither liberty nor safety." -- Benjamin Franklin, 1759 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/fcd624e5/attachment.bin From trever.adams at gmail.com Thu Mar 29 16:53:03 2012 From: trever.adams at gmail.com (Trever L. Adams) Date: Thu, 29 Mar 2012 06:53:03 -0600 Subject: [Freeswitch-users] process-rxfax.py and announcing that a fax was detected/connection grabbed Message-ID: <4F745B2F.8050805@gmail.com> Hello everyone, I am new to FreeSWITCH (several months, but a few bugs kept me really from digging in until now). I am setting up auto-rx for faxes for a home PBX. It works. However, the person who answers the phone hears a beep, then a click. I am hoping there is a way to cause FreeSwitch to play an announcement from inside the process-rxfax.py or the dialplan calling it so that the person who answers the phone knows a fax was detected and that they can hangup. This should only play to the internal user, and not affect rxfax at all. Is this possible? How would I go about splitting the legs of the connection or what not to do that? Thank you, Trever -- "Arbitrary power is most easily established on the ruins of liberty abused to licentiousness." -- George Washington -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/0be6dcc6/attachment.bin From imran.moinuddin at nexdegree.com Thu Mar 29 17:56:44 2012 From: imran.moinuddin at nexdegree.com (Imran Moinuddin) Date: Thu, 29 Mar 2012 18:56:44 +0500 Subject: [Freeswitch-users] Acoustic Echo Cancellation on Freeswitch side Message-ID: Hi there, I was wondering whether there were any echo cancellation alternatives available that could potentially work on the Freeswitch server side. This is mainly to cater for people that dial into my service using just their laptop microphone + speakers and their client app doesn't support echo cancellation. I realize that this could be an expensive proposition (from a computational perspective) but curious if the option (commercial or otherwise) exists. Thanks! Imran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/2d2d22a5/attachment.html From fs-list at communicatefreely.net Thu Mar 29 18:17:35 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Thu, 29 Mar 2012 10:17:35 -0400 Subject: [Freeswitch-users] Session-wise Language Setting In-Reply-To: <4CB4E6075A9F4F9BA1C8A57FF2A06735@sharad> References: <4CB4E6075A9F4F9BA1C8A57FF2A06735@sharad> Message-ID: <4F746EFF.3050305@communicatefreely.net> Yes! On incoming calls, I set default_language to either en or fr (I'm in Canada), depending on the DID. I can also change that variable in an IVR if the caller wants to change the language. On outgoing calls, I set the same variable in the directory, so that any call that a phone makes will have a language set. This is great in a multi-lingual environment, as each user can have a language preference that will be used for voice mail and other system prompts. We store that in a database and also have the provisioning system look to that same variable, so the screen labels and text on the phone is the same language. -Tim Sharad Garg wrote: > Hi All > > Just wondering whether we can define the language from beginning of the call > means when a call is originated or landed to Freeswitch, can we define the > language of all the prompts whether the call should be processed in English > or Russian or so on.? > > Regards > Sharad > > > ----- Original Message ----- > From: > To: > Sent: Thursday, March 29, 2012 7:33 AM > Subject: FreeSWITCH-users Digest, Vol 69, Issue 282 > > > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> > > > -------------------------------------------------------------------------------- > > > >> Today's Topics: >> >> 1. Re: Open Bugs on Jira and Call for help (Ken Rice) >> 2. mod_callcenter and moh-sound (Vik Killa) >> 3. Re: mod_callcenter and moh-sound (Vik Killa) >> 4. Registration VIA TCP (Rob Moore) >> 5. Forward calls from opensips to freeswitch (Sherif Omran) >> 6. Re: Registration VIA TCP (Mitch Capper) >> >> > > > -------------------------------------------------------------------------------- > > > >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From lazyvirus at gmx.com Thu Mar 29 18:37:29 2012 From: lazyvirus at gmx.com (Bzzz) Date: Thu, 29 Mar 2012 16:37:29 +0200 Subject: [Freeswitch-users] Acoustic Echo Cancellation on Freeswitch side In-Reply-To: References: Message-ID: <20120329163729.09b3d0db@anubis.defcon1> On Thu, 29 Mar 2012 18:56:44 +0500 Imran Moinuddin wrote: > > I was wondering whether there were any echo cancellation alternatives > available that could potentially work on the Freeswitch server side. May be you should search for: voip echo oslec > This is mainly to cater for people that dial into my service using just > their laptop microphone + speakers and their client app doesn't support > echo cancellation. > > I realize that this could be an expensive proposition (from a computational > perspective) but curious if the option (commercial or otherwise) exists. A (good) commercial proposition should be to ask these people to buy a correct headset... -- A snake lurks in the grass. -- Publius Vergilius Maro (Virgil) From anita.hall at simmortel.com Thu Mar 29 18:39:56 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Thu, 29 Mar 2012 20:09:56 +0530 Subject: [Freeswitch-users] Unexpected DCS after Expected DCS !!!! Message-ID: Hi As seen in the log below, I got an unexpected DCS after the expected DCS which led to disconnect and hangup. The set-up is rather unusual. FS <-> PSTN <-> Mobile (call forwarding) <-> PSTN <-> FS What should I blame? Echo, network delay or some other funky ghost? :) And what is the way out of this misery? If I ignore such unexpected DCS, do I risk some other error ? Thanks ! 86d8d618-79aa-11e1-8571-47d63f03be09 EXECUTE OpenZAP/2:1/30713846 rxfax(/srv/fax/in/86d8d618-79aa-11e1-8571-47d63f03be09.tiff) 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:51:52.718371 [DEBUG] mod_spandsp_fax.c:1083 Raw read codec activation Success L16 20000 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:51:52.718371 [DEBUG] switch_core_codec.c:116 OpenZAP/2:1/30713846 Push codec L16:10 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:51:52.718371 [DEBUG] mod_spandsp_fax.c:1099 Raw write codec activation Success L16 2012-03-29 19:51:52.722424 [DEBUG] ozmod_libpri.c:106 < Protocol Discriminator: Q.931 (8) len=5 2012-03-29 19:51:52.722424 [DEBUG] ozmod_libpri.c:106 < Call Ref: len= 2 (reference 136/0x88) (Originator) 2012-03-29 19:51:52.722424 [DEBUG] ozmod_libpri.c:106 < Message type: CONNECT ACKNOWLEDGE (15) 2012-03-29 19:51:55.738290 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send complete in phase T30_PHASE_A_CED, state 1 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Starting answer mode 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from phase T30_PHASE_A_CED to T30_PHASE_B_TX 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx type 0 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx type 4 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Start T2 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from state 1 to 17 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Sending ident 'chennai ident' 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: CSI without final frame tag 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: ff 03 40 74 6e 65 64 69 20 69 61 6e 6e 65 68 63 20 20 20 20 20 20 20 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 DIS: 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= Store and forward Internet fax (T.37): Not set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .0..= Real-time Internet fax (T.38): Not set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 0...= 3G mobile network: Not set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. ....= V.8 capabilities: Not set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. ....= Preferred octets: 256 octets 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= Ready to transmit a fax document (polling): Not set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..1.= Can receive fax: Set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= 2-D coding: Set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1% 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 10..= Recording length: Unlimited 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..0.= Compressed/uncompressed mode: Compressed 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .1..= Error correction mode (ECM): ECM 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .1.. ....= T.6 coding: Set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= "Field not valid" supported: Not set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..0.= Multiple selective polling: Not set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .0..= Polled sub-address: Not set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 0...= T.43 coding: Not set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...0 ....= Plane interleave: Not set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. ....= Reserved for the use of extended voice coding set: Not set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...1= R8x15.4lines/mm: Set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..0.= 300x300pels/25.4mm: Not set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 0...= Inch-based resolution preferred: Not set 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...1 ....= Metric-based resolution preferred: Set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. ....= Selective polling: Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= Sub-addressing: Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..0.= Password: Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .0..= Ready to transmit a data file (polling): Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...0 ....= Binary file transfer (BFT): Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. ....= Document transfer mode (DTM): Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. ....= Electronic data interchange (EDI): Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= Basic transfer mode (BTM): Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .0..= Ready to transfer a character or mixed mode document (polling): Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 0...= Character mode: Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. ....= Mixed mode (Annex E/T.4): Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= Processable mode 26 (Rec. T.505): Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..0.= Digital network capability: Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .0..= Duplex capability: Half only 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 0...= JPEG coding: Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...0 ....= Full colour mode: Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. ....= 12bits/pel component: Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= No subsampling (1:1:1): Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..0.= Custom illuminant: Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .0..= Custom gamut range: Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 1...= North American Letter (215.9mm x 279.4mm): Set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...1 ....= North American Legal (215.9mm x 355.6mm): Set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 0... ....= Extension indicator: Not set 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: DIS with final frame tag 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: ff 13 80 00 ee fa c4 80 95 80 80 80 18 2012-03-29 19:51:57.838516 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17 2012-03-29 19:51:57.918984 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17 2012-03-29 19:51:57.918984 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX 2012-03-29 19:51:57.918984 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx type 4 2012-03-29 19:51:57.918984 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx type 0 2012-03-29 19:51:57.918984 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Start T4 2012-03-29 19:52:01.377951 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17 2012-03-29 19:52:01.378941 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Retry number 1 2012-03-29 19:52:01.378941 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX 2012-03-29 19:52:01.378941 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx type 0 2012-03-29 19:52:01.378941 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx type 4 2012-03-29 19:52:01.378941 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Sending ident 'chennai ident' 2012-03-29 19:52:01.378941 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: CSI without final frame tag 2012-03-29 19:52:01.378941 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: ff 03 40 74 6e 65 64 69 20 69 61 6e 6e 65 68 63 20 20 20 20 20 20 20 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 DIS: 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= Store and forward Internet fax (T.37): Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .0..= Real-time Internet fax (T.38): Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 0...= 3G mobile network: Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. ....= V.8 capabilities: Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. ....= Preferred octets: 256 octets 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= Ready to transmit a fax document (polling): Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..1.= Can receive fax: Set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= 2-D coding: Set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1% 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 10..= Recording length: Unlimited 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..0.= Compressed/uncompressed mode: Compressed 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .1..= Error correction mode (ECM): ECM 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .1.. ....= T.6 coding: Set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= "Field not valid" supported: Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..0.= Multiple selective polling: Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .0..= Polled sub-address: Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 0...= T.43 coding: Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...0 ....= Plane interleave: Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. ....= Reserved for the use of extended voice coding set: Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...1= R8x15.4lines/mm: Set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..0.= 300x300pels/25.4mm: Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 0...= Inch-based resolution preferred: Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...1 ....= Metric-based resolution preferred: Set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. ....= Selective polling: Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= Sub-addressing: Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..0.= Password: Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .0..= Ready to transmit a data file (polling): Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...0 ....= Binary file transfer (BFT): Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. ....= Document transfer mode (DTM): Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. ....= Electronic data interchange (EDI): Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= Basic transfer mode (BTM): Not set 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .0..= Ready to transfer a character or mixed mode document (polling): Not set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 0...= Character mode: Not set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. ....= Mixed mode (Annex E/T.4): Not set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= Processable mode 26 (Rec. T.505): Not set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..0.= Digital network capability: Not set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .0..= Duplex capability: Half only 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 0...= JPEG coding: Not set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...0 ....= Full colour mode: Not set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. ....= 12bits/pel component: Not set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= No subsampling (1:1:1): Not set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..0.= Custom illuminant: Not set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .0..= Custom gamut range: Not set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 1...= North American Letter (215.9mm x 279.4mm): Set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...1 ....= North American Legal (215.9mm x 355.6mm): Set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 0... ....= Extension indicator: Not set 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: DIS with final frame tag 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: ff 13 80 00 ee fa c4 80 95 80 80 80 18 2012-03-29 19:52:03.438160 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17 2012-03-29 19:52:03.518033 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17 2012-03-29 19:52:03.518033 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX 2012-03-29 19:52:03.518033 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx type 4 2012-03-29 19:52:03.518033 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx type 0 2012-03-29 19:52:03.518033 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Start T4 2012-03-29 19:52:03.998988 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17 2012-03-29 19:52:04.078900 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state 17 2012-03-29 19:52:04.278765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Framing OK (-6) in state 17 2012-03-29 19:52:04.278765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Start T4A 2012-03-29 19:52:05.538614 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Stop T4A (13920 remaining) 2012-03-29 19:52:05.538614 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: TSI without final frame tag 2012-03-29 19:52:05.538614 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: ff 03 43 74 6e 65 64 49 20 78 61 46 20 61 6d 6f 67 6e 61 53 20 20 20 2012-03-29 19:52:05.538614 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Remote gave TSI as: "Sangoma Fax Ident" 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Stop none (0 remaining) 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: DCS with final frame tag 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: ff 13 83 00 22 f8 44 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx final frame in state 17 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 DCS: 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ...0= Store and forward Internet fax (T.37): Not set 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .0..= Real-time Internet fax (T.38): Not set 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 0...= 3G mobile network: Not set 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..1.= Receive fax: Set 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..10 00..= Selected data signalling rate: V.17 14400bps 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Not set 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 0... ....= 2-D coding: Not set 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..00= Recording width: 215mm +- 1% 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 10..= Recording length: Unlimited 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .111 ....= Minimum scan line time: 0ms 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... ....= Extension indicator: Set 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... ..0.= Compressed/uncompressed mode: Compressed 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... .1..= Error correction mode (ECM): ECM 2012-03-29 19:52:05.859258 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... 0...= Frame size: 256 octets 2012-03-29 19:52:05.859258 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .1.. ....= T.6 coding: Set 2012-03-29 19:52:05.859258 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 0... ....= Extension indicator: Not set 2012-03-29 19:52:05.859258 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Selected compression T.6 (3) 2012-03-29 19:52:05.859258 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Get document at 14400bps, modem 7 2012-03-29 19:52:05.859258 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from state 17 to 7 2012-03-29 19:52:05.859258 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Start T2 2012-03-29 19:52:05.898219 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7 2012-03-29 19:52:05.898219 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_C_NON_ECM_RX 2012-03-29 19:52:05.898219 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx type 0 2012-03-29 19:52:05.898219 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx type 7 2012-03-29 19:52:05.898219 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx type 0 2012-03-29 19:52:05.958033 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7 2012-03-29 19:52:05.958033 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7 2012-03-29 19:52:06.078320 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Non-ECM signal status is Training in progress (-3) in state 7 2012-03-29 19:52:07.138102 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state 7 2012-03-29 19:52:07.317993 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7 2012-03-29 19:52:07.378781 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state 7 2012-03-29 19:52:08.878420 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7 2012-03-29 19:52:08.878420 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7 2012-03-29 19:52:12.438211 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7 2012-03-29 19:52:12.438211 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7 2012-03-29 19:52:12.578006 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7 2012-03-29 19:52:12.657852 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Framing OK (-6) in state 7 2012-03-29 19:52:12.657852 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Start T2A 2012-03-29 19:52:13.998262 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Stop T2A (13280 remaining) 2012-03-29 19:52:13.998262 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: TSI without final frame tag 2012-03-29 19:52:13.998262 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: ff 03 43 74 6e 65 64 49 20 78 61 46 20 61 6d 6f 67 6e 61 53 20 20 20 2012-03-29 19:52:13.998262 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Remote gave TSI as: "Sangoma Fax Ident" 2012-03-29 19:52:13.998262 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Switching from V.17 + V.21 to V.21 (-21.45dBm0) 2012-03-29 19:52:14.318385 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state 7 2012-03-29 19:52:14.378445 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7 2012-03-29 19:52:14.438480 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7 2012-03-29 19:52:15.878450 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state 7 2012-03-29 19:52:16.258042 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state 7 2012-03-29 19:52:17.358521 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7 2012-03-29 19:52:20.898906 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7 2012-03-29 19:52:21.118003 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Framing OK (-6) in state 7 2012-03-29 19:52:22.458937 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Stop none (0 remaining) 2012-03-29 19:52:22.458937 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: TSI without final frame tag 2012-03-29 19:52:22.458937 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: ff 03 43 74 6e 65 64 49 20 78 61 46 20 61 6d 6f 67 6e 61 53 20 20 20 2012-03-29 19:52:22.458937 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Remote gave TSI as: "Sangoma Fax Ident" 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Stop none (0 remaining) 2012-03-29 19:52:22.458937 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: TSI without final frame tag 2012-03-29 19:52:22.458937 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: ff 03 43 74 6e 65 64 49 20 78 61 46 20 61 6d 6f 67 6e 61 53 20 20 20 2012-03-29 19:52:22.458937 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Remote gave TSI as: "Sangoma Fax Ident" 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Stop none (0 remaining) 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: DCS with final frame tag 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: ff 13 83 00 22 f8 44 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx final frame in state 7 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Unexpected DCS frame in state 7 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from state 7 to 3 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: DCN with final frame tag 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: ff 13 fa 2012-03-29 19:52:22.838011 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier down (-1) in state 3 2012-03-29 19:52:22.839003 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from phase T30_PHASE_C_NON_ECM_RX to T30_PHASE_D_TX 2012-03-29 19:52:22.839003 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx type 0 2012-03-29 19:52:22.839003 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx type 4 2012-03-29 19:52:23.898025 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3 2012-03-29 19:52:23.978342 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3 2012-03-29 19:52:23.978342 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Disconnecting 2012-03-29 19:52:23.978342 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E 2012-03-29 19:52:23.978342 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx type 0 2012-03-29 19:52:23.978342 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx type 1 2012-03-29 19:52:23.978342 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from state 3 to 2 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send complete in phase T30_PHASE_E, state 2 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:321 ============================================================================== 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:333 Fax processing not successful - result (13) Unexpected message received. 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:338 Remote station id: Sangoma Fax Ident 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:339 Local station id: chennai ident 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:340 Pages transferred: 0 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:342 Total fax pages: 0 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:343 Image resolution: 8031x7700 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:344 Transfer Rate: 14400 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:346 ECM status on 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:347 remote country: 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:348 remote vendor: 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:349 remote model: 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:351 ============================================================================== 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from state 2 to 32 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from phase T30_PHASE_E to T30_PHASE_CALL_FINISHED 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx type 9 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX FAX exchange complete 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx type 9 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX FAX exchange complete 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.998043 [DEBUG] switch_core_codec.c:140 OpenZAP/2:1/30713846 Restore previous codec PCMA:8. 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:25.008098 [DEBUG] switch_core_session.c:885 Send signal OpenZAP/2:1/30713846 [BREAK] 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:25.018941 [DEBUG] switch_ivr.c:551 OpenZAP/2:1/30713846 Command Execute hangup() 86d8d618-79aa-11e1-8571-47d63f03be09 EXECUTE OpenZAP/2:1/30713846 hangup() regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/48ecbb17/attachment-0001.html From cmrienzo at gmail.com Thu Mar 29 18:46:34 2012 From: cmrienzo at gmail.com (cmrienzo at gmail.com) Date: Thu, 29 Mar 2012 10:46:34 -0400 Subject: [Freeswitch-users] Acoustic Echo Cancellation on Freeswitch side In-Reply-To: <20120329163729.09b3d0db@anubis.defcon1> References: <20120329163729.09b3d0db@anubis.defcon1> Message-ID: <68602BA4-4F86-4851-BD02-57D0BA6B345E@gmail.com> Oslec is a line echo canceller and its license is incompatible with FreeSWITCH. Acoustic echo cancellation needs to be done by the handset. Speex preprocessor has an echo canceller, but I don't know how good it is. On Mar 29, 2012, at 10:37, Bzzz wrote: > On Thu, 29 Mar 2012 18:56:44 +0500 > Imran Moinuddin wrote: > >> >> I was wondering whether there were any echo cancellation alternatives >> available that could potentially work on the Freeswitch server side. > > May be you should search for: voip echo oslec > >> This is mainly to cater for people that dial into my service using just >> their laptop microphone + speakers and their client app doesn't support >> echo cancellation. >> >> I realize that this could be an expensive proposition (from a computational >> perspective) but curious if the option (commercial or otherwise) exists. > > A (good) commercial proposition should be to ask these people to buy > a correct headset... > > -- > A snake lurks in the grass. > -- Publius Vergilius Maro (Virgil) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fdelawarde at wirelessmundi.com Thu Mar 29 18:48:58 2012 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Thu, 29 Mar 2012 16:48:58 +0200 Subject: [Freeswitch-users] Acoustic Echo Cancellation on Freeswitch side In-Reply-To: <20120329163729.09b3d0db@anubis.defcon1> References: <20120329163729.09b3d0db@anubis.defcon1> Message-ID: <1333032538.23470.164.camel@luna.madrid.commsmundi.com> On Thu, 2012-03-29 at 16:37 +0200, Bzzz wrote: > On Thu, 29 Mar 2012 18:56:44 +0500 > Imran Moinuddin wrote: > > This is mainly to cater for people that dial into my service using just > > their laptop microphone + speakers and their client app doesn't support > > echo cancellation. > > > > I realize that this could be an expensive proposition (from a computational > > perspective) but curious if the option (commercial or otherwise) exists. > > A (good) commercial proposition should be to ask these people to buy > a correct headset... Or use a softphone that has echo cancellation (see counterpath), or maybe even use a webphone with mod_rtmp as flash might also have echo cancel features. But none of those can replace a good headset. From acrow at integrafin.co.uk Thu Mar 29 18:52:14 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Thu, 29 Mar 2012 15:52:14 +0100 Subject: [Freeswitch-users] Dialplan/PHP esl variable problem. Message-ID: <4F74771E.5050808@integrafin.co.uk> Hi all, I have a dialplan that calls a PHP script through CURL: (notice the play_and_get_digits is commented out) and in PHP I connect via ESL via this function and code: function saySomething($uuid) { $sock = new ESLconnection('localhost','8021','ClueCon'); sleep (2); $res = $sock->execute("play_and_get_digits","4 11 1 5000 # ivr/ivr-please_enter_the_phone_number.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+", "$uuid"); //$foo = $uuid . ' target_num'; sleep (2); $res=$sock->sendRecv("api uuid_getvar $uuid target_num"); $ret = $res->getBody(); $sock->disconnect(); //echo $ret; return ($ret); } if ($action == 'test') { $res = saySomething($_REQUEST['uuid']); echo $res; } However if I run the play_and_get_digits from the PHP the subsequent uuid_getvar returns _undef_. If I comment out the play_and_get_digits in PHP and uncomment it in the dialplan, the uuid_getvar *does* return the digits pressed. Is there something I'm doing wrong here? Thanks Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From jmesquita at freeswitch.org Thu Mar 29 19:04:24 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Thu, 29 Mar 2012 12:04:24 -0300 Subject: [Freeswitch-users] Acoustic Echo Cancellation on Freeswitch side In-Reply-To: <1333032538.23470.164.camel@luna.madrid.commsmundi.com> References: <20120329163729.09b3d0db@anubis.defcon1> <1333032538.23470.164.camel@luna.madrid.commsmundi.com> Message-ID: This is a tough one. I played with speex preprocessor while trying to develop FSComm and had no success? I don't know much on the subject and my inability to solve this problem cause be to halt the development of FSComm for a while since I don't see myself wearing a headset all day while coding and FSComm was a project that I was developing for myself. A friend of mine that develops Blink(saghul) told me that they yanked the WebRTC echo cancelled and it works like a charm. Too bad it seems to only work right on Mac due to the lack of realtime audio on other platforms. Again, I don't know much about the subject to really know what he is talking about but I thought it was worth mentioning here. For Imran, acoustic echo cancelers cannot be done anywhere BUT on the device that is capturing the signal. Once it is on the line, it is not echo anymore. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Thursday, March 29, 2012 at 11:48 AM, Fran?ois Delawarde wrote: > On Thu, 2012-03-29 at 16:37 +0200, Bzzz wrote: > > On Thu, 29 Mar 2012 18:56:44 +0500 > > Imran Moinuddin wrote: > > > This is mainly to cater for people that dial into my service using just > > > their laptop microphone + speakers and their client app doesn't support > > > echo cancellation. > > > > > > I realize that this could be an expensive proposition (from a computational > > > perspective) but curious if the option (commercial or otherwise) exists. > > > > > > > > > A (good) commercial proposition should be to ask these people to buy > > a correct headset... > > > > > Or use a softphone that has echo cancellation (see counterpath), or > maybe even use a webphone with mod_rtmp as flash might also have echo > cancel features. > > But none of those can replace a good headset. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/b71d4480/attachment.html From peter.olsson at visionutveckling.se Thu Mar 29 19:08:37 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 29 Mar 2012 15:08:37 +0000 Subject: [Freeswitch-users] Dialplan/PHP esl variable problem. In-Reply-To: <4F74771E.5050808@integrafin.co.uk> References: <4F74771E.5050808@integrafin.co.uk> Message-ID: <1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se> ESL execute is async, so you will need to wait for the correct event to be returned, that tells you it's finished. EXECUTE_COMPLETE I think the event is called.. /Peter 29 mar 2012 kl. 16:57 skrev "Alex Crow" : > Hi all, > > I have a dialplan that calls a PHP script through CURL: > > > > > > > data="http://localhost/provision/index.php?action=test&uuid=${uuid}&exten=${caller_id_number}" > /> > > > > > > > (notice the play_and_get_digits is commented out) > > and in PHP I connect via ESL via this function and code: > > function saySomething($uuid) { > > $sock = new ESLconnection('localhost','8021','ClueCon'); > > sleep (2); > > $res = $sock->execute("play_and_get_digits","4 11 1 5000 # > ivr/ivr-please_enter_the_phone_number.wav > ivr/ivr-that_was_an_invalid_entry.wav target_num \d+", "$uuid"); > > //$foo = $uuid . ' target_num'; > > sleep (2); > > $res=$sock->sendRecv("api uuid_getvar $uuid target_num"); > > > $ret = $res->getBody(); > > $sock->disconnect(); > > //echo $ret; > > return ($ret); > > } > > if ($action == 'test') { > $res = saySomething($_REQUEST['uuid']); > echo $res; > } > > However if I run the play_and_get_digits from the PHP the subsequent > uuid_getvar returns _undef_. > > If I comment out the play_and_get_digits in PHP and uncomment it in the > dialplan, the uuid_getvar *does* return the digits pressed. > > Is there something I'm doing wrong here? > > Thanks > > Alex > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f7475d932769462314482! > From jalsot at gmail.com Thu Mar 29 19:10:36 2012 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Thu, 29 Mar 2012 17:10:36 +0200 Subject: [Freeswitch-users] Acoustic Echo Cancellation on Freeswitch side In-Reply-To: References: <20120329163729.09b3d0db@anubis.defcon1> <1333032538.23470.164.camel@luna.madrid.commsmundi.com> Message-ID: Hello, Anybody having experience with PBXMate? ( http://www.solicall.com/products.html) It seems to have AEC.... Regards, Tamas 2012/3/29 Jo?o Mesquita > This is a tough one. I played with speex preprocessor while trying to > develop FSComm and had no success? I don't know much on the subject and my > inability to solve this problem cause be to halt the development of FSComm > for a while since I don't see myself wearing a headset all day while coding > and FSComm was a project that I was developing for myself. > > A friend of mine that develops Blink(saghul) told me that they yanked the > WebRTC echo cancelled and it works like a charm. Too bad it seems to only > work right on Mac due to the lack of realtime audio on other platforms. > Again, I don't know much about the subject to really know what he is > talking about but I thought it was worth mentioning here. > > For Imran, acoustic echo cancelers cannot be done anywhere BUT on the > device that is capturing the signal. Once it is on the line, it is not echo > anymore. > > Regards, > > -- > Jo?o Mesquita > Sent with Sparrow > > On Thursday, March 29, 2012 at 11:48 AM, Fran?ois Delawarde wrote: > > On Thu, 2012-03-29 at 16:37 +0200, Bzzz wrote: > > On Thu, 29 Mar 2012 18:56:44 +0500 > Imran Moinuddin wrote: > > This is mainly to cater for people that dial into my service using just > their laptop microphone + speakers and their client app doesn't support > echo cancellation. > > I realize that this could be an expensive proposition (from a computational > perspective) but curious if the option (commercial or otherwise) exists. > > > A (good) commercial proposition should be to ask these people to buy > a correct headset... > > > Or use a softphone that has echo cancellation (see counterpath), or > maybe even use a webphone with mod_rtmp as flash might also have echo > cancel features. > > But none of those can replace a good headset. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/9e59e03d/attachment-0001.html From andrew at cassidywebservices.co.uk Thu Mar 29 19:14:27 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 29 Mar 2012 16:14:27 +0100 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after~30seconds In-Reply-To: References: <1363342dbcd.1079533506949741633.-8314378552953017400@zoho.com> <4F6970A2.000019.15156@FLIGHTPC> Message-ID: I have some more information about absolutely's problem. The call is established fine, there is two-way audio, all loooks fine, until about 60 seconds into the call when FreeSWITCH issues what appears to be unsolicited INVITES to which it recieves no response from the carrier and as such, it drops the call. Any suggestions? On 22 March 2012 15:22, Brian Foster wrote: > Yea I see another who's having reportedly the same issue. > > absolutely, how are you coming along? > > -BDF > > > On Thu, Mar 22, 2012 at 5:41 AM, Andrew Cassidy < > andrew at cassidywebservices.co.uk> wrote: > >> Glad you got your problem sorted in the end Brian. >> >> Do I also remember seeing in this thread someone else with a similar >> issue? >> >> >> On 21 March 2012 20:37, Brian Foster wrote: >> >>> I'm going to go ahead and mark this issue as solved. The solution was to >>> do a make current. There was a bug in FreeSWITCH that was fixed before >>> Anthony went on vacation. >>> >>> If anyone was on conference or if your name is Brian West, if there was >>> a specific revision that this was fixed on please chime in for anyone else >>> that may have this issue. >>> >>> Fwiw I actually went ahead and started with a clean install (due to >>> deleting the source folder and also not knowing what all I messed with to >>> try and solve this issue). After I did the install, both the external and >>> internal profile were getting the local ipv4 address. To fix this, I went >>> to conf/sip_profiles/external.xml and changed the ext-sip-ip and ext-rtp-ip >>> from $${local_ip_v4} to stun:stun.freeswitch.org. After restarting the >>> profile, the external profile was getting my external IP and my internal >>> profile was getting the local IP. >>> >>> Thanks to everyone on the conference and on the thread for helping out. >>> Brian, for repayment, I'll get you that pic you've always wanted lol. >>> >>> -BDF >>> On Mar 21, 2012 8:43 AM, "Andrew Cassidy" < >>> andrew at cassidywebservices.co.uk> wrote: >>> >>>> I think we also need to check those variables on the internal profile. >>>> As said previously it looks like it's doing the tun on the internal profile >>>> which is why the ack is being send to the wrong place. >>>> >>>> On 21 March 2012 06:09, Fernando - NextBilling IP Solutions < >>>> fernandojdk at gmail.com> wrote: >>>> >>>>> What you have in your external profile on variables ext-rtp-ip >>>>> end ext-sip-ip? >>>>> ? >>>>> ** >>>>> >>>>> * >>>>> Best Regards, >>>>> Fernando da Silva Santos >>>>> NextBilling IP Solutions LTDA >>>>> Phone: +55 21 2143-9000 >>>>> MSN: fernandojdk at gmail.com >>>>> www.nextbilling.com.br >>>>> Rio de Janeiro, Brazil, BR >>>>> Importante: >>>>> Esta mensagem, incluindo todo seu conte?do, cont?m informa??es >>>>> confidenciais, legalmente protegidas e destinadas a indiv?duo e prop?sito >>>>> espec?ficos. Caso a tenha recebido por engano, lembramos do car?ter >>>>> sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o >>>>> mais breve poss?vel. >>>>> As informa??es contidas nesta mensagem e em seu conte?do s?o de >>>>> responsabilidade de seu autor, n?o representando necessariamente id?ias, >>>>> opini?es, pensamentos ou qualquer forma de posicionamento por parte da >>>>> NextBilling IP Solutions. >>>>> P "Antes de imprimir pense em seu compromisso com o Meio Ambiente." >>>>> * >>>>> *-------Original Message-------* >>>>> >>>>> *From:* dingdong >>>>> *Date:* 21/03/2012 02:57:02 >>>>> *To:* FreeSWITCH Users Help >>>>> *Subject:* Re: [Freeswitch-users] NAT issues - Outbound Call drops >>>>> after~30seconds >>>>> >>>>> id like to know more bout wireshark and sip debugging,please include >>>>> this on the conf call topics >>>>> >>>>> ---- On Tue, 20 Mar 2012 10:15:20 -0700 *Michael Collins < >>>>> msc at freeswitch.org>* wrote ---- >>>>> >>>>> This is why I love Wireshark so much! Look at this purdee graph it >>>>> makes: >>>>> >>>>> >>>>> >>>>> >>>>> See all those 200 OK's that your FS is sending to the Grandstream? >>>>> Guess what your GS is sending in response to those: NADA! If you look at >>>>> the BYE that FS sends to the GS you'll even see the reason: >>>>> >>>>> SIP;cause=408;text=\"ACK Timeout\" >>>>> >>>>> FS never gets an ACK back from the GS. So the question is: why? I'm >>>>> unfamiliar with the GS so I'll have to defer to those with more experience >>>>> than I. However, I think you'll find that tcpdumps and analyzing w/ >>>>> Wireshark is extremely helpful. (Open the pcap, click "Telephony > VoIP >>>>> calls" and then a new dialog opens. In this case it shows two calls - >>>>> meaning two call legs. Click "Select All" then click "Flow" and you'll get >>>>> the cool graph. Click around and see what other stuff does. :) >>>>> >>>>> I'm thinking of doing a FreeSWITCH conference call presentation on the >>>>> subject of collecting pcaps and doing Wireshark analysis. Let me know if >>>>> you guys think that's a good presentation. >>>>> >>>>> -MC >>>>> >>>>> >>>>> On Tue, Mar 20, 2012 at 10:00 AM, Brian Foster < >>>>> bdfoster at endigotech.com> wrote: >>>>> Andrew, >>>>> >>>>> root at homeserver:/usr/local/stund# ./client stunserver.org >>>>> STUN client version 0.97 >>>>> Primary: Independent Mapping, Independent Filter, preserves ports, >>>>> will hairpin >>>>> Return value is 0x000003 >>>>> >>>>> http://da1.endigovoip.com/dump.pcap >>>>> >>>>> Kristian, >>>>> >>>>> http://pastebin.freeswitch.org/18708 >>>>> >>>>> Michael, >>>>> >>>>> I did replace the IP's for security purposes, but now I've realized >>>>> that it's needed and it's not really that big of a deal. I'll end up >>>>> changing the Flowroute creds after this is fixed up. The prior siptrace is >>>>> exactly one call (two legs). I don't think it's a carrier issue, as I've >>>>> tried calling a buddy's server direct sip with the same issues. >>>>> >>>>> -BDF >>>>> >>>>> On Tue, Mar 20, 2012 at 11:34 AM, Michael Collins wrote: >>>>> We have scores of machines behind NAT talking to Flowroute with no >>>>> problems, so there's got to be something potentially non-obvious but easy >>>>> that needs to be set/unset. I noticed in the SIP trace that there are >>>>> several calls. It's hard to know what's what. I think your best bet is a >>>>> pcap analyzed with Wireshark, as was mentioned elsewhere in this thread. I >>>>> also noticed that you redacted IP addrs - you won't be able to do this with >>>>> a pcap. If security is an issue then I'd say get the pcap and let us know >>>>> here on the list, then those who can have a look will email you privately >>>>> and you can send the pcap file to them. >>>>> >>>>> -MC >>>>> >>>>> >>>>> On Mon, Mar 19, 2012 at 12:12 PM, Brian Foster < >>>>> bdfoster at endigotech.com> wrote: >>>>> Alright, so I admit... I'm a little rusty when it comes to NAT, etc. >>>>> I've only set up FS so far on machines with no NAT, so this is sort of a >>>>> new experience for me. >>>>> >>>>> I have a FreeSWITCH server located on the same local network as all of >>>>> my phones here at the house. When I try to make a call to Flowroute, after >>>>> about 30 seconds the call drops. It also does the exact same thing when I >>>>> call a buddy's server directly via SIP. >>>>> >>>>> Here's a siptrace of the call (I didn't think that the actual FS log >>>>> would be much help): >>>>> http://pastebin.freeswitch.org/18697 >>>>> >>>>> ...and here's a paste of 'sofia status': >>>>> http://pastebin.freeswitch.org/18698 >>>>> >>>>> ...and just for good measure, here's a paste of vars.xml: >>>>> http://pastebin.freeswitch.org/18699 >>>>> >>>>> >>>>> -- >>>>> Brian D. Foster >>>>> Endigo Computer LLC >>>>> Email: bdfoster at endigotech.com >>>>> Phone: 317-800-7876 >>>>> Indianapolis, Indiana, USA >>>>> >>>>> This message contains confidential information and is intended for >>>>> those listed in the "To:", "CC:", and/or "BCC:" fields of the message >>>>> header. If you are not the intended recipient you are notified that >>>>> disclosing, copying, distributing or taking any action in reliance on the >>>>> contents of this information is strictly prohibited. E-mail transmission >>>>> cannot be guaranteed to be secure or error-free as information could be >>>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or >>>>> contain viruses. The sender therefore does not accept liability for any >>>>> errors or omissions in the contents of this message, which arise as a >>>>> result of e-mail transmission. If verification is required please request a >>>>> hard-copy version. >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Brian D. Foster >>>>> Endigo Computer LLC >>>>> Email: bdfoster at endigotech.com >>>>> Phone: 317-800-7876 >>>>> Indianapolis, Indiana, USA >>>>> >>>>> This message contains confidential information and is intended for >>>>> those listed in the "To:", "CC:", and/or "BCC:" fields of the message >>>>> header. If you are not the intended recipient you are notified that >>>>> disclosing, copying, distributing or taking any action in reliance on the >>>>> contents of this information is strictly prohibited. E-mail transmission >>>>> cannot be guaranteed to be secure or error-free as information could be >>>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or >>>>> contain viruses. The sender therefore does not accept liability for any >>>>> errors or omissions in the contents of this message, which arise as a >>>>> result of e-mail transmission. If verification is required please request a >>>>> hard-copy version. >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Andrew Cassidy BSc (Hons) MBCS >>>> Managing Director; Cassidy Web Services Ltd >>>> T: 03300 100 960 F: 03300 100 961 >>>> E: andrew at cassidywebservices.co.uk >>>> W: www.cassidywebservices.co.uk >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Andrew Cassidy BSc (Hons) MBCS >> Managing Director; Cassidy Web Services Ltd >> T: 03300 100 960 F: 03300 100 961 >> E: andrew at cassidywebservices.co.uk >> W: www.cassidywebservices.co.uk >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS* Managing Director *T* 03300 100 960 *F* 03300 100 961 *E* andrew at cassidywebservices.co.uk *W* www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/91f2da55/attachment-0001.html From me at nevian.org Thu Mar 29 20:00:47 2012 From: me at nevian.org (Serge S. Yuriev) Date: Thu, 29 Mar 2012 20:00:47 +0400 Subject: [Freeswitch-users] FS-3109 Message-ID: <776081333036847@web23.yandex.ru> Hi, Very old issue is still here. Can anyone take a look? -- wbr, Serge From fs-list at communicatefreely.net Thu Mar 29 20:05:07 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Thu, 29 Mar 2012 12:05:07 -0400 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. In-Reply-To: References: Message-ID: <4F748833.3030604@communicatefreely.net> Yes, I agree there. If you empty out your dialplan and only use curl, then FS hardly has to do anything to route the call, it just waits for a response and then executes the little snippet that comes back. Make your web script really efficient and have it send back a list of dialplan instructions. It's still a good idea to have a bit of XML dialplan locally to handle emergency routing, ie. 911 and 611 if possible. That way, if something happens to your web servers, people can at least call for help, and maybe still call you to complain about it. Then again, maybe you don't want that :-) -Tim Andrew Cassidy wrote: > I use xml_curl, reads config from a central server so is replicated > across you whole cluster, and directory/dialplan are requested every > call (but supply filter parameters) and in effect dynamic if you > generate these from a web service. > > On 29 March 2012 07:56, Joe Flemmings > wrote: > > What is the most scalable way to control FreeSwitch Routing > dynamically (apart from XML configs) that has the least load on > FreeSwitch and scales the best. This is to help achieve the most > Call Setups Per Second (CPS) > > eg xml_rpc, lua, event socket etc > > > > Joe > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Andrew Cassidy BSc (Hons) MBCS > Managing Director; Cassidy Web Services Ltd > T: 03300 100 960 F: 03300 100 961 > E: andrew at cassidywebservices.co.uk > > W: www.cassidywebservices.co.uk > > ------------------------------------------------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From acrow at integrafin.co.uk Thu Mar 29 20:43:55 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Thu, 29 Mar 2012 17:43:55 +0100 Subject: [Freeswitch-users] Dialplan/PHP esl variable problem. In-Reply-To: <1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se> References: <4F74771E.5050808@integrafin.co.uk> <1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se> Message-ID: <4F74914B.7040001@integrafin.co.uk> On 29/03/12 16:08, Peter Olsson wrote: > ESL execute is async, so you will need to wait for the correct event to be returned, that tells you it's finished. EXECUTE_COMPLETE I think the event is called.. > > /Peter > > 29 mar 2012 kl. 16:57 skrev "Alex Crow": > > Many thanks Peter. If you or anyone else can answer as well - I am also using PHP for directory via xml_curl. I see on the Wiki example for auth requests: [sip_contact_user] => 1004 [sip_contact_host] => 192.168.1.100 I am thinking this might be usable to figure out when a phone registers if it is an "outside" phone (ie checking if it is either on a public IP or presenting a private IP outside of company LAN ranges, and as such should have as part of the returned XML to make sure audio gets through for all external clients. Does that sound reasonable? Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From anthony.minessale at gmail.com Thu Mar 29 20:49:15 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 29 Mar 2012 11:49:15 -0500 Subject: [Freeswitch-users] One-legged callee_id In-Reply-To: References: <4F7319E2.5010107@communicatefreely.net> Message-ID: Its not a workaround, its because each phone does it differently and other phones consider each other's way a fatal err and hangup calls. set effective_callee_id_name and effective_callee_id_number variables before you execute ring_ready, pre_answer or answer or before you run an app that may answer the call and you will get what you are looking for. If the call is already up you can use the send_display app And from ESL or cli uuid_display This will be the name|1234 On Thu, Mar 29, 2012 at 2:32 AM, Markus Lindenberg wrote: > Hi Tim, > > would love this as well. > > I guess the answer application should see if effective_callee_id_* is > set and trigger a display update on the calling phone. It would also > be nice to be able to trigger display updates to the a leg during > (ivr) execution. > > Asterisk's CONNECTEDLINE() allows all of this. It's using > Reinvites/Updates w/ PAI-Header and the syntax is cumbersome as > everything asterisk is, but at least it works with most phones. (Snom > 3xx, Snom M3, Siemens OpenStage, Polycom confirmed as i'm working with > these every day) > > FreeSWITCH display updates seem to rely on workarounds (grep polycom > or snom in mod_sofia). I would like to be able to disable the > INFO/sipfrag thing for Snom, use PAI instead of RDNIS in the A-Leg and > send Reinvites for Display Update etc., but it all seems to be hard > coded right now. > > Regards, Markus > > On Wed, Mar 28, 2012 at 16:02, Tim St. Pierre > wrote: >> Hello, >> >> I have been making extensive use of the callee_id_name and >> callee_id_number variables so that our IP phones update the display to >> show who they are actually talking to on the other end. >> >> Generally, it works very well, but it only seems to work if there is a >> bridge. ?Is there a way to set the callee_id in a single legged call? >> ie. I want the name of the company to come up when the caller is in the >> main IVR, or the name of the conference to come up if they are in the >> conference room. >> >> Is there a way to do this? >> >> Thanks! >> >> -Tim >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From lloyd.aloysius at gmail.com Thu Mar 29 21:48:00 2012 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Thu, 29 Mar 2012 13:48:00 -0400 Subject: [Freeswitch-users] xml_curl & user_data , user_exists - return Message-ID: Hi All, I am trying to use the xml_curl & user_data .. retrieve the user information. User registration is fine. . But user_data 21 at mydomain.com attr id return -ERR no reply user_exists id eddie mydomain.com return false Am I missing something here? Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/9e274e1b/attachment.html From bdfoster at endigotech.com Thu Mar 29 22:19:14 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 29 Mar 2012 14:19:14 -0400 Subject: [Freeswitch-users] xml_curl & user_data , user_exists - return In-Reply-To: References: Message-ID: Does that actually work with xml_curl? Might only work on just plain static xml files. -BDF On Thu, Mar 29, 2012 at 1:48 PM, Lloyd Aloysius wrote: > Hi All, > > I am trying to use the xml_curl & user_data .. retrieve the user > information. > > User registration is fine. > > . > > But > > user_data 21 at mydomain.com attr id > > return -ERR no reply > > user_exists id eddie mydomain.com > > return false > > > Am I missing something here? > > Thanks > Lloyd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/fc5e9d0e/attachment.html From d.campbell at ampersand.com Thu Mar 29 21:25:27 2012 From: d.campbell at ampersand.com (Doug Campbell) Date: Thu, 29 Mar 2012 13:25:27 -0400 Subject: [Freeswitch-users] Unexpected behavior with hostname in socket application Message-ID: <4F749B07.6050700@ampersand.com> I am getting unexpected behavior on the hostname used in the socket application. I have a trivial dialplan: This outbound event socket gets connected fine on my CentOS 6.2 machine. Though the freeswitch.log file seems to change "127.0.0.1" to "localhost": EXECUTE sofia/phoneglue/9789735994 socket(localhost:4574 async) However, with the identical software (including fs built from git today), on my RHEL 6.2 machine that has a DNS glitch resolving "localhost" badly (don't ask), it fails. The freeswitch.log file says: EXECUTE sofia/phoneglue/9789735994 socket(localhost:4574 async) 2012-03-29 10:05:08.200547 [ERR] mod_event_socket.c:458 Socket Error! Why is it changing "127.0.0.1" to "localhost"? Shouldn't it be using the raw IP when it is so specified? Without it doing this, my RHEL box is doomed. Much thanks for any help, Doug From bdfoster at endigotech.com Thu Mar 29 23:07:40 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 29 Mar 2012 15:07:40 -0400 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. In-Reply-To: <4F748833.3030604@communicatefreely.net> References: <4F748833.3030604@communicatefreely.net> Message-ID: xml_curl leaves much to be desired. For one, when you're dealing with a lot of traffic, it gets pretty slow. Come up with something that accomplishes these goals: - Fast - Redundant - High Availability - No single point of failure - Simple as can be It's a tall order and requires a ton of skills and some excellent planning. It is not for the faint of heart. I've been dealing with this problem for many months, and I still really don't have a solution. I've got other goals besides these, but I'd start with that list. -BDF On Thu, Mar 29, 2012 at 12:05 PM, Tim St. Pierre < fs-list at communicatefreely.net> wrote: > Yes, I agree there. > > If you empty out your dialplan and only use curl, then FS hardly has to > do anything to route the call, it just waits for a response and then > executes the little snippet that comes back. Make your web script > really efficient and have it send back a list of dialplan instructions. > > It's still a good idea to have a bit of XML dialplan locally to handle > emergency routing, ie. 911 and 611 if possible. That way, if something > happens to your web servers, people can at least call for help, and > maybe still call you to complain about it. Then again, maybe you don't > want that :-) > > -Tim > > Andrew Cassidy wrote: > > I use xml_curl, reads config from a central server so is replicated > > across you whole cluster, and directory/dialplan are requested every > > call (but supply filter parameters) and in effect dynamic if you > > generate these from a web service. > > > > On 29 March 2012 07:56, Joe Flemmings > > wrote: > > > > What is the most scalable way to control FreeSwitch Routing > > dynamically (apart from XML configs) that has the least load on > > FreeSwitch and scales the best. This is to help achieve the most > > Call Setups Per Second (CPS) > > > > eg xml_rpc, lua, event socket etc > > > > > > > > Joe > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Andrew Cassidy BSc (Hons) MBCS > > Managing Director; Cassidy Web Services Ltd > > T: 03300 100 960 F: 03300 100 961 > > E: andrew at cassidywebservices.co.uk > > > > W: www.cassidywebservices.co.uk > > > > ------------------------------------------------------------------------ > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/afc6b5fb/attachment.html From joe.jflemmings at gmail.com Thu Mar 29 23:15:35 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Thu, 29 Mar 2012 12:15:35 -0700 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. In-Reply-To: References: <4F748833.3030604@communicatefreely.net> Message-ID: My question was very specific, speed or system resource utilization. Which has the least load of FreeSwitch? I can always look for different ways to accomplish other tasks like redundancy etc. I have tried both xml_curl and lua and under high load the system level calls are killing me. I was wondering if event socket are a better way to go/ so between lua, xml_curl and event socket. On Thu, Mar 29, 2012 at 12:07 PM, Brian Foster wrote: > xml_curl leaves much to be desired. For one, when you're dealing with a > lot of traffic, it gets pretty slow. > > Come up with something that accomplishes these goals: > > - Fast > - Redundant > - High Availability > - No single point of failure > - Simple as can be > > It's a tall order and requires a ton of skills and some excellent > planning. It is not for the faint of heart. I've been dealing with this > problem for many months, and I still really don't have a solution. I've got > other goals besides these, but I'd start with that list. > > -BDF > > On Thu, Mar 29, 2012 at 12:05 PM, Tim St. Pierre < > fs-list at communicatefreely.net> wrote: > >> Yes, I agree there. >> >> If you empty out your dialplan and only use curl, then FS hardly has to >> do anything to route the call, it just waits for a response and then >> executes the little snippet that comes back. Make your web script >> really efficient and have it send back a list of dialplan instructions. >> >> It's still a good idea to have a bit of XML dialplan locally to handle >> emergency routing, ie. 911 and 611 if possible. That way, if something >> happens to your web servers, people can at least call for help, and >> maybe still call you to complain about it. Then again, maybe you don't >> want that :-) >> >> -Tim >> >> Andrew Cassidy wrote: >> > I use xml_curl, reads config from a central server so is replicated >> > across you whole cluster, and directory/dialplan are requested every >> > call (but supply filter parameters) and in effect dynamic if you >> > generate these from a web service. >> > >> > On 29 March 2012 07:56, Joe Flemmings > > > wrote: >> > >> > What is the most scalable way to control FreeSwitch Routing >> > dynamically (apart from XML configs) that has the least load on >> > FreeSwitch and scales the best. This is to help achieve the most >> > Call Setups Per Second (CPS) >> > >> > eg xml_rpc, lua, event socket etc >> > >> > >> > >> > Joe >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > Andrew Cassidy BSc (Hons) MBCS >> > Managing Director; Cassidy Web Services Ltd >> > T: 03300 100 960 F: 03300 100 961 >> > E: andrew at cassidywebservices.co.uk >> > >> > W: www.cassidywebservices.co.uk >> > >> > ------------------------------------------------------------------------ >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/ff7074d8/attachment-0001.html From Hector.Geraldino at ipsoft.com Thu Mar 29 23:26:49 2012 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Thu, 29 Mar 2012 15:26:49 -0400 Subject: [Freeswitch-users] Unexpected behavior with hostname in socket application In-Reply-To: <4F749B07.6050700@ampersand.com> References: <4F749B07.6050700@ampersand.com> Message-ID: <6A6B4C284AD15042B429EB9D904544AD022D77D397@NY1-EXMB-01.ip-soft.net> The obvious answer would be: fix the DNS settings :) (or use the ip address of the network, not the loopback ip) Anyway, I'm curious about this one. What happens if you try to do a "telnet localhost 4574" and then a "telnet 127.0.0.1 4574" on the same machine? Works in one case and not in the other? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Doug Campbell Sent: Thursday, March 29, 2012 1:25 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Unexpected behavior with hostname in socket application I am getting unexpected behavior on the hostname used in the socket application. I have a trivial dialplan: This outbound event socket gets connected fine on my CentOS 6.2 machine. Though the freeswitch.log file seems to change "127.0.0.1" to "localhost": EXECUTE sofia/phoneglue/9789735994 socket(localhost:4574 async) However, with the identical software (including fs built from git today), on my RHEL 6.2 machine that has a DNS glitch resolving "localhost" badly (don't ask), it fails. The freeswitch.log file says: EXECUTE sofia/phoneglue/9789735994 socket(localhost:4574 async) 2012-03-29 10:05:08.200547 [ERR] mod_event_socket.c:458 Socket Error! Why is it changing "127.0.0.1" to "localhost"? Shouldn't it be using the raw IP when it is so specified? Without it doing this, my RHEL box is doomed. Much thanks for any help, Doug _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Thu Mar 29 23:30:35 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 29 Mar 2012 12:30:35 -0700 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. In-Reply-To: References: <4F748833.3030604@communicatefreely.net> Message-ID: On Thu, Mar 29, 2012 at 12:15 PM, Joe Flemmings wrote: > My question was very specific, speed or system resource utilization. > > Which has the least load of FreeSwitch? I can always look for different > ways to accomplish other tasks like redundancy etc. > > I have tried both xml_curl and lua and under high load the system level > calls are killing me. > I was wondering if event socket are a better way to go/ > > so between lua, xml_curl and event socket.http://www.freeswitch.org > > I'm curious - what system level calls are killing your system when you use xml_curl? (Or Lua, etc.) I can refer you to Ken Rice who does thousands upon thousands of calls a day, many simultaneously. I'm sure he's got some good insights on this. Also, make sure that you're not on crappy hardware or doing something silly like running a 32 bit OS on 64 bit hardware... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/bd52be95/attachment.html From joe.jflemmings at gmail.com Thu Mar 29 23:37:58 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Thu, 29 Mar 2012 12:37:58 -0700 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. In-Reply-To: References: <4F748833.3030604@communicatefreely.net> Message-ID: The problem is not the number of simultaneous calls but cps. I'm doing thousands of simultaneous calls too and that's not an issue. I'm using 64 bit machines and just started tests this week. The most expensive system call right now is select and trying to isolate where its coming from. The issue is not the number of calls but cps. On Thu, Mar 29, 2012 at 12:30 PM, Michael Collins wrote: > > > On Thu, Mar 29, 2012 at 12:15 PM, Joe Flemmings wrote: > >> My question was very specific, speed or system resource utilization. >> >> Which has the least load of FreeSwitch? I can always look for different >> ways to accomplish other tasks like redundancy etc. >> >> I have tried both xml_curl and lua and under high load the system level >> calls are killing me. >> I was wondering if event socket are a better way to go/ >> >> so between lua, xml_curl and event socket.http://www.freeswitch.org >> >> > I'm curious - what system level calls are killing your system when you use > xml_curl? (Or Lua, etc.) I can refer you to Ken Rice who does thousands > upon thousands of calls a day, many simultaneously. I'm sure he's got some > good insights on this. Also, make sure that you're not on crappy hardware > or doing something silly like running a 32 bit OS on 64 bit hardware... > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/5ec3fbea/attachment.html From lloyd.aloysius at sunteltech.ca Thu Mar 29 23:42:46 2012 From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius) Date: Thu, 29 Mar 2012 15:42:46 -0400 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. In-Reply-To: References: <4F748833.3030604@communicatefreely.net> Message-ID: What is the cps you are expecting? * * On Thu, Mar 29, 2012 at 3:37 PM, Joe Flemmings wrote: > The problem is not the number of simultaneous calls but cps. I'm doing > thousands of simultaneous calls too and that's not an issue. I'm using 64 > bit machines and just started tests this week. The most expensive system > call right now is select and trying to isolate where its coming from. The > issue is not the number of calls but cps. > > > On Thu, Mar 29, 2012 at 12:30 PM, Michael Collins wrote: > >> >> >> On Thu, Mar 29, 2012 at 12:15 PM, Joe Flemmings > > wrote: >> >>> My question was very specific, speed or system resource utilization. >>> >>> Which has the least load of FreeSwitch? I can always look for different >>> ways to accomplish other tasks like redundancy etc. >>> >>> I have tried both xml_curl and lua and under high load the system level >>> calls are killing me. >>> I was wondering if event socket are a better way to go/ >>> >>> so between lua, xml_curl and event socket.http://www.freeswitch.org >>> >>> >> I'm curious - what system level calls are killing your system when you >> use xml_curl? (Or Lua, etc.) I can refer you to Ken Rice who does thousands >> upon thousands of calls a day, many simultaneously. I'm sure he's got some >> good insights on this. Also, make sure that you're not on crappy hardware >> or doing something silly like running a 32 bit OS on 64 bit hardware... >> >> -MC >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/0a54946c/attachment.html From joe.jflemmings at gmail.com Thu Mar 29 23:47:50 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Thu, 29 Mar 2012 12:47:50 -0700 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. In-Reply-To: References: <4F748833.3030604@communicatefreely.net> Message-ID: Right now having issue from about 80-90. The kind of traffic we run i high bust and short traffic. so cps is most important for me. I'm also looking into different Linux options to improve this. Joe On Thu, Mar 29, 2012 at 12:42 PM, Lloyd Aloysius < lloyd.aloysius at sunteltech.ca> wrote: > What is the cps you are expecting? > > > * * > > > > On Thu, Mar 29, 2012 at 3:37 PM, Joe Flemmings wrote: > >> The problem is not the number of simultaneous calls but cps. I'm doing >> thousands of simultaneous calls too and that's not an issue. I'm using 64 >> bit machines and just started tests this week. The most expensive system >> call right now is select and trying to isolate where its coming from. The >> issue is not the number of calls but cps. >> >> >> On Thu, Mar 29, 2012 at 12:30 PM, Michael Collins wrote: >> >>> >>> >>> On Thu, Mar 29, 2012 at 12:15 PM, Joe Flemmings < >>> joe.jflemmings at gmail.com> wrote: >>> >>>> My question was very specific, speed or system resource utilization. >>>> >>>> Which has the least load of FreeSwitch? I can always look for different >>>> ways to accomplish other tasks like redundancy etc. >>>> >>>> I have tried both xml_curl and lua and under high load the system level >>>> calls are killing me. >>>> I was wondering if event socket are a better way to go/ >>>> >>>> so between lua, xml_curl and event socket.http://www.freeswitch.org >>>> >>>> >>> I'm curious - what system level calls are killing your system when you >>> use xml_curl? (Or Lua, etc.) I can refer you to Ken Rice who does thousands >>> upon thousands of calls a day, many simultaneously. I'm sure he's got some >>> good insights on this. Also, make sure that you're not on crappy hardware >>> or doing something silly like running a 32 bit OS on 64 bit hardware... >>> >>> -MC >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/d90447e4/attachment-0001.html From anita.hall at simmortel.com Thu Mar 29 23:48:05 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Fri, 30 Mar 2012 01:18:05 +0530 Subject: [Freeswitch-users] Unexpected DCS after Expected DCS !!!! In-Reply-To: References: Message-ID: Update: Strange! I am able to reproduce this consistently again and again! regards, Anita On Thu, Mar 29, 2012 at 8:09 PM, Anita Hall wrote: > Hi > > As seen in the log below, I got an unexpected DCS after the expected DCS > which led to disconnect and hangup. > > The set-up is rather unusual. > FS <-> PSTN <-> Mobile (call forwarding) <-> PSTN <-> FS > > What should I blame? Echo, network delay or some other funky ghost? :) > > And what is the way out of this misery? If I ignore such unexpected DCS, > do I risk some other error ? Thanks ! > > 86d8d618-79aa-11e1-8571-47d63f03be09 EXECUTE OpenZAP/2:1/30713846 > rxfax(/srv/fax/in/86d8d618-79aa-11e1-8571-47d63f03be09.tiff) > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:51:52.718371 [DEBUG] > mod_spandsp_fax.c:1083 Raw read codec activation Success L16 20000 > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:51:52.718371 [DEBUG] > switch_core_codec.c:116 OpenZAP/2:1/30713846 Push codec L16:10 > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:51:52.718371 [DEBUG] > mod_spandsp_fax.c:1099 Raw write codec activation Success L16 > 2012-03-29 19:51:52.722424 [DEBUG] ozmod_libpri.c:106 < Protocol > Discriminator: Q.931 (8) len=5 > 2012-03-29 19:51:52.722424 [DEBUG] ozmod_libpri.c:106 < Call Ref: len= 2 > (reference 136/0x88) (Originator) > 2012-03-29 19:51:52.722424 [DEBUG] ozmod_libpri.c:106 < Message type: > CONNECT ACKNOWLEDGE (15) > 2012-03-29 19:51:55.738290 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 1 > 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send > complete in phase T30_PHASE_A_CED, state 1 > 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Starting answer mode > 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Changing from phase T30_PHASE_A_CED to T30_PHASE_B_TX > 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx > type 0 > 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx > type 4 > 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Start T2 > 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Changing from state 1 to 17 > 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Sending > ident 'chennai ident' > 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: > CSI without final frame tag > 2012-03-29 19:51:55.798543 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: ff > 03 40 74 6e 65 64 69 20 69 61 6e 6e 65 68 63 20 20 20 20 20 20 20 > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send > complete in phase T30_PHASE_B_TX, state 17 > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 DIS: > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= Store and forward Internet fax (T.37): Not set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .0..= Real-time Internet fax (T.38): Not set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 0...= 3G mobile network: Not set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. > ....= V.8 capabilities: Not set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. > ....= Preferred octets: 256 octets > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= Ready to transmit a fax document (polling): Not set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..1.= Can receive fax: Set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..10 > 11..= Supported data signalling rates: V.27 ter, V.29, and V.17 > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .1.. > ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= 2-D coding: Set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1% > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 10..= Recording length: Unlimited > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .111 > ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85 > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..0.= Compressed/uncompressed mode: Compressed > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .1..= Error correction mode (ECM): ECM > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .1.. > ....= T.6 coding: Set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= "Field not valid" supported: Not set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..0.= Multiple selective polling: Not set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .0..= Polled sub-address: Not set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 0...= T.43 coding: Not set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...0 > ....= Plane interleave: Not set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. > ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. > ....= Reserved for the use of extended voice coding set: Not set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...1= R8x15.4lines/mm: Set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..0.= 300x300pels/25.4mm: Not set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 0...= Inch-based resolution preferred: Not set > 2012-03-29 19:51:57.378942 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...1 > ....= Metric-based resolution preferred: Set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. > ....= Minimum scan line time for higher resolutions: T15.4 = T7.7 > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. > ....= Selective polling: Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= Sub-addressing: Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..0.= Password: Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .0..= Ready to transmit a data file (polling): Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...0 > ....= Binary file transfer (BFT): Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. > ....= Document transfer mode (DTM): Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. > ....= Electronic data interchange (EDI): Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= Basic transfer mode (BTM): Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .0..= Ready to transfer a character or mixed mode document (polling): Not > set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 0...= Character mode: Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. > ....= Mixed mode (Annex E/T.4): Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= Processable mode 26 (Rec. T.505): Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..0.= Digital network capability: Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .0..= Duplex capability: Half only > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 0...= JPEG coding: Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...0 > ....= Full colour mode: Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. > ....= 12bits/pel component: Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= No subsampling (1:1:1): Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..0.= Custom illuminant: Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .0..= Custom gamut range: Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 1...= North American Letter (215.9mm x 279.4mm): Set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...1 > ....= North American Legal (215.9mm x 355.6mm): Set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. > ....= Single-progression sequential coding (Rec. T.85) basic: Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. > ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 0... > ....= Extension indicator: Not set > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: > DIS with final frame tag > 2012-03-29 19:51:57.379933 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: ff > 13 80 00 ee fa c4 80 95 80 80 80 18 > 2012-03-29 19:51:57.838516 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send > complete in phase T30_PHASE_B_TX, state 17 > 2012-03-29 19:51:57.918984 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send > complete in phase T30_PHASE_B_TX, state 17 > 2012-03-29 19:51:57.918984 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX > 2012-03-29 19:51:57.918984 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx > type 4 > 2012-03-29 19:51:57.918984 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx > type 0 > 2012-03-29 19:51:57.918984 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Start T4 > 2012-03-29 19:52:01.377951 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 T4 > expired in phase T30_PHASE_B_RX, state 17 > 2012-03-29 19:52:01.378941 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Retry > number 1 > 2012-03-29 19:52:01.378941 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX > 2012-03-29 19:52:01.378941 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx > type 0 > 2012-03-29 19:52:01.378941 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx > type 4 > 2012-03-29 19:52:01.378941 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Sending > ident 'chennai ident' > 2012-03-29 19:52:01.378941 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: > CSI without final frame tag > 2012-03-29 19:52:01.378941 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: ff > 03 40 74 6e 65 64 69 20 69 61 6e 6e 65 68 63 20 20 20 20 20 20 20 > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send > complete in phase T30_PHASE_B_TX, state 17 > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 DIS: > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= Store and forward Internet fax (T.37): Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .0..= Real-time Internet fax (T.38): Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 0...= 3G mobile network: Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. > ....= V.8 capabilities: Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. > ....= Preferred octets: 256 octets > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= Ready to transmit a fax document (polling): Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..1.= Can receive fax: Set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..10 > 11..= Supported data signalling rates: V.27 ter, V.29, and V.17 > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .1.. > ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= 2-D coding: Set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1% > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 10..= Recording length: Unlimited > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .111 > ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85 > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..0.= Compressed/uncompressed mode: Compressed > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .1..= Error correction mode (ECM): ECM > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .1.. > ....= T.6 coding: Set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= "Field not valid" supported: Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..0.= Multiple selective polling: Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .0..= Polled sub-address: Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 0...= T.43 coding: Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...0 > ....= Plane interleave: Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. > ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. > ....= Reserved for the use of extended voice coding set: Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...1= R8x15.4lines/mm: Set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..0.= 300x300pels/25.4mm: Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 0...= Inch-based resolution preferred: Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...1 > ....= Metric-based resolution preferred: Set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. > ....= Minimum scan line time for higher resolutions: T15.4 = T7.7 > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. > ....= Selective polling: Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= Sub-addressing: Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..0.= Password: Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .0..= Ready to transmit a data file (polling): Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...0 > ....= Binary file transfer (BFT): Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. > ....= Document transfer mode (DTM): Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. > ....= Electronic data interchange (EDI): Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= Basic transfer mode (BTM): Not set > 2012-03-29 19:52:02.958770 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .0..= Ready to transfer a character or mixed mode document (polling): Not > set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 0...= Character mode: Not set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. > ....= Mixed mode (Annex E/T.4): Not set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= Processable mode 26 (Rec. T.505): Not set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..0.= Digital network capability: Not set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .0..= Duplex capability: Half only > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 0...= JPEG coding: Not set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...0 > ....= Full colour mode: Not set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. > ....= 12bits/pel component: Not set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= No subsampling (1:1:1): Not set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..0.= Custom illuminant: Not set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .0..= Custom gamut range: Not set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 1...= North American Letter (215.9mm x 279.4mm): Set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ...1 > ....= North American Legal (215.9mm x 355.6mm): Set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..0. > ....= Single-progression sequential coding (Rec. T.85) basic: Not set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. > ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 0... > ....= Extension indicator: Not set > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: > DIS with final frame tag > 2012-03-29 19:52:02.959765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: ff > 13 80 00 ee fa c4 80 95 80 80 80 18 > 2012-03-29 19:52:03.438160 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send > complete in phase T30_PHASE_B_TX, state 17 > 2012-03-29 19:52:03.518033 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send > complete in phase T30_PHASE_B_TX, state 17 > 2012-03-29 19:52:03.518033 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX > 2012-03-29 19:52:03.518033 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx > type 4 > 2012-03-29 19:52:03.518033 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx > type 0 > 2012-03-29 19:52:03.518033 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Start T4 > 2012-03-29 19:52:03.998988 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 17 > 2012-03-29 19:52:04.078900 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Abort (-8) in state 17 > 2012-03-29 19:52:04.278765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Framing OK (-6) in state 17 > 2012-03-29 19:52:04.278765 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Start > T4A > 2012-03-29 19:52:05.538614 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Stop > T4A (13920 remaining) > 2012-03-29 19:52:05.538614 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: > TSI without final frame tag > 2012-03-29 19:52:05.538614 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: ff > 03 43 74 6e 65 64 49 20 78 61 46 20 61 6d 6f 67 6e 61 53 20 20 20 > 2012-03-29 19:52:05.538614 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Remote > gave TSI as: "Sangoma Fax Ident" > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Stop > none (0 remaining) > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: > DCS with final frame tag > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: ff > 13 83 00 22 f8 44 > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx > final frame in state 17 > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 DCS: > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ...0= Store and forward Internet fax (T.37): Not set > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .0..= Real-time Internet fax (T.38): Not set > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 0...= 3G mobile network: Not set > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..1.= Receive fax: Set > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 ..10 > 00..= Selected data signalling rate: V.17 14400bps > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .0.. > ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Not set > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 0... > ....= 2-D coding: Not set > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..00= Recording width: 215mm +- 1% > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 10..= Recording length: Unlimited > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .111 > ....= Minimum scan line time: 0ms > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 1... > ....= Extension indicator: Set > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > ..0.= Compressed/uncompressed mode: Compressed > 2012-03-29 19:52:05.858271 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > .1..= Error correction mode (ECM): ECM > 2012-03-29 19:52:05.859258 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .... > 0...= Frame size: 256 octets > 2012-03-29 19:52:05.859258 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 .1.. > ....= T.6 coding: Set > 2012-03-29 19:52:05.859258 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 0... > ....= Extension indicator: Not set > 2012-03-29 19:52:05.859258 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Selected compression T.6 (3) > 2012-03-29 19:52:05.859258 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Get > document at 14400bps, modem 7 > 2012-03-29 19:52:05.859258 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Changing from state 17 to 7 > 2012-03-29 19:52:05.859258 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Start T2 > 2012-03-29 19:52:05.898219 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 7 > 2012-03-29 19:52:05.898219 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Changing from phase T30_PHASE_B_RX to T30_PHASE_C_NON_ECM_RX > 2012-03-29 19:52:05.898219 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx > type 0 > 2012-03-29 19:52:05.898219 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx > type 7 > 2012-03-29 19:52:05.898219 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx > type 0 > 2012-03-29 19:52:05.958033 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Non-ECM > signal status is Carrier up (-2) in state 7 > 2012-03-29 19:52:05.958033 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 7 > 2012-03-29 19:52:06.078320 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Non-ECM > signal status is Training in progress (-3) in state 7 > 2012-03-29 19:52:07.138102 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Abort (-8) in state 7 > 2012-03-29 19:52:07.317993 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Non-ECM > signal status is Training failed (-5) in state 7 > 2012-03-29 19:52:07.378781 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Abort (-8) in state 7 > 2012-03-29 19:52:08.878420 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Non-ECM > signal status is Carrier down (-1) in state 7 > 2012-03-29 19:52:08.878420 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 7 > 2012-03-29 19:52:12.438211 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Non-ECM > signal status is Carrier up (-2) in state 7 > 2012-03-29 19:52:12.438211 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 7 > 2012-03-29 19:52:12.578006 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Non-ECM > signal status is Training failed (-5) in state 7 > 2012-03-29 19:52:12.657852 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Framing OK (-6) in state 7 > 2012-03-29 19:52:12.657852 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Start > T2A > 2012-03-29 19:52:13.998262 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Stop > T2A (13280 remaining) > 2012-03-29 19:52:13.998262 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: > TSI without final frame tag > 2012-03-29 19:52:13.998262 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: ff > 03 43 74 6e 65 64 49 20 78 61 46 20 61 6d 6f 67 6e 61 53 20 20 20 > 2012-03-29 19:52:13.998262 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Remote > gave TSI as: "Sangoma Fax Ident" > 2012-03-29 19:52:13.998262 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX > Switching from V.17 + V.21 to V.21 (-21.45dBm0) > 2012-03-29 19:52:14.318385 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Abort (-8) in state 7 > 2012-03-29 19:52:14.378445 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 7 > 2012-03-29 19:52:14.438480 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 7 > 2012-03-29 19:52:15.878450 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Abort (-8) in state 7 > 2012-03-29 19:52:16.258042 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Abort (-8) in state 7 > 2012-03-29 19:52:17.358521 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 7 > 2012-03-29 19:52:20.898906 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Carrier up (-2) in state 7 > 2012-03-29 19:52:21.118003 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Framing OK (-6) in state 7 > 2012-03-29 19:52:22.458937 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Stop > none (0 remaining) > 2012-03-29 19:52:22.458937 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: > TSI without final frame tag > 2012-03-29 19:52:22.458937 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: ff > 03 43 74 6e 65 64 49 20 78 61 46 20 61 6d 6f 67 6e 61 53 20 20 20 > 2012-03-29 19:52:22.458937 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Remote > gave TSI as: "Sangoma Fax Ident" > 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Stop > none (0 remaining) > 2012-03-29 19:52:22.458937 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: > TSI without final frame tag > 2012-03-29 19:52:22.458937 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: ff > 03 43 74 6e 65 64 49 20 78 61 46 20 61 6d 6f 67 6e 61 53 20 20 20 > 2012-03-29 19:52:22.458937 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Remote > gave TSI as: "Sangoma Fax Ident" > 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Stop > none (0 remaining) > 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: > DCS with final frame tag > 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx: ff > 13 83 00 22 f8 44 > 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Rx > final frame in state 7 > 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Unexpected DCS frame in state 7 > 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Changing from state 7 to 3 > 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: > DCN with final frame tag > 2012-03-29 19:52:22.758302 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Tx: ff > 13 fa > 2012-03-29 19:52:22.838011 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC > signal status is Carrier down (-1) in state 3 > 2012-03-29 19:52:22.839003 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Changing from phase T30_PHASE_C_NON_ECM_RX to T30_PHASE_D_TX > 2012-03-29 19:52:22.839003 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx > type 0 > 2012-03-29 19:52:22.839003 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx > type 4 > 2012-03-29 19:52:23.898025 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send > complete in phase T30_PHASE_D_TX, state 3 > 2012-03-29 19:52:23.978342 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send > complete in phase T30_PHASE_D_TX, state 3 > 2012-03-29 19:52:23.978342 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Disconnecting > 2012-03-29 19:52:23.978342 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Changing from phase T30_PHASE_D_TX to T30_PHASE_E > 2012-03-29 19:52:23.978342 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx > type 0 > 2012-03-29 19:52:23.978342 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx > type 1 > 2012-03-29 19:52:23.978342 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Changing from state 3 to 2 > 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send > complete in phase T30_PHASE_E, state 2 > 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:321 > ============================================================================== > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] > mod_spandsp_fax.c:333 Fax processing not successful - result (13) > Unexpected message received. > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] > mod_spandsp_fax.c:338 Remote station id: Sangoma Fax Ident > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] > mod_spandsp_fax.c:339 Local station id: chennai ident > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] > mod_spandsp_fax.c:340 Pages transferred: 0 > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] > mod_spandsp_fax.c:342 Total fax pages: 0 > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] > mod_spandsp_fax.c:343 Image resolution: 8031x7700 > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] > mod_spandsp_fax.c:344 Transfer Rate: 14400 > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] > mod_spandsp_fax.c:346 ECM status on > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] > mod_spandsp_fax.c:347 remote country: > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] > mod_spandsp_fax.c:348 remote vendor: > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] > mod_spandsp_fax.c:349 remote model: > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.978861 [DEBUG] > mod_spandsp_fax.c:351 > ============================================================================== > 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Changing from state 2 to 32 > 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 > Changing from phase T30_PHASE_E to T30_PHASE_CALL_FINISHED > 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx > type 9 > 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX FAX > exchange complete > 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx > type 9 > 2012-03-29 19:52:24.978861 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX FAX > exchange complete > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:24.998043 [DEBUG] > switch_core_codec.c:140 OpenZAP/2:1/30713846 Restore previous codec PCMA:8. > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:25.008098 [DEBUG] > switch_core_session.c:885 Send signal OpenZAP/2:1/30713846 [BREAK] > 86d8d618-79aa-11e1-8571-47d63f03be09 2012-03-29 19:52:25.018941 [DEBUG] > switch_ivr.c:551 OpenZAP/2:1/30713846 Command Execute hangup() > 86d8d618-79aa-11e1-8571-47d63f03be09 EXECUTE OpenZAP/2:1/30713846 hangup() > > > > > regards, > Anita > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/4890794d/attachment-0001.html From msc at freeswitch.org Fri Mar 30 00:00:33 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 29 Mar 2012 13:00:33 -0700 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. In-Reply-To: References: <4F748833.3030604@communicatefreely.net> Message-ID: How about putting the database in a ramdrive? -MC On Thu, Mar 29, 2012 at 12:47 PM, Joe Flemmings wrote: > Right now having issue from about 80-90. The kind of traffic we run i high > bust and short traffic. so cps is most important for me. I'm also looking > into different Linux options to improve this. > > Joe > > > On Thu, Mar 29, 2012 at 12:42 PM, Lloyd Aloysius < > lloyd.aloysius at sunteltech.ca> wrote: > >> What is the cps you are expecting? >> >> >> * * >> >> >> >> On Thu, Mar 29, 2012 at 3:37 PM, Joe Flemmings wrote: >> >>> The problem is not the number of simultaneous calls but cps. I'm doing >>> thousands of simultaneous calls too and that's not an issue. I'm using 64 >>> bit machines and just started tests this week. The most expensive system >>> call right now is select and trying to isolate where its coming from. The >>> issue is not the number of calls but cps. >>> >>> >>> On Thu, Mar 29, 2012 at 12:30 PM, Michael Collins wrote: >>> >>>> >>>> >>>> On Thu, Mar 29, 2012 at 12:15 PM, Joe Flemmings < >>>> joe.jflemmings at gmail.com> wrote: >>>> >>>>> My question was very specific, speed or system resource utilization. >>>>> >>>>> Which has the least load of FreeSwitch? I can always look for >>>>> different ways to accomplish other tasks like redundancy etc. >>>>> >>>>> I have tried both xml_curl and lua and under high load the system >>>>> level calls are killing me. >>>>> I was wondering if event socket are a better way to go/ >>>>> >>>>> so between lua, xml_curl and event socket.http://www.freeswitch.org >>>>> >>>>> >>>> I'm curious - what system level calls are killing your system when you >>>> use xml_curl? (Or Lua, etc.) I can refer you to Ken Rice who does thousands >>>> upon thousands of calls a day, many simultaneously. I'm sure he's got some >>>> good insights on this. Also, make sure that you're not on crappy hardware >>>> or doing something silly like running a 32 bit OS on 64 bit hardware... >>>> >>>> -MC >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/cb9bf925/attachment.html From msc at freeswitch.org Fri Mar 30 00:03:09 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 29 Mar 2012 13:03:09 -0700 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after~30seconds In-Reply-To: References: <1363342dbcd.1079533506949741633.-8314378552953017400@zoho.com> <4F6970A2.000019.15156@FLIGHTPC> Message-ID: On Thu, Mar 29, 2012 at 8:14 AM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > I have some more information about absolutely's problem. > > The call is established fine, there is two-way audio, all loooks fine, > until about 60 seconds into the call when FreeSWITCH issues what appears to > be unsolicited INVITES to which it recieves no response from the carrier > and as such, it drops the call. > > Any suggestions? > > I'd get the console log and sip trace and throw it up on pb so we could see that in action. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/7a36ae89/attachment.html From avi at avimarcus.net Fri Mar 30 00:03:50 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 29 Mar 2012 22:03:50 +0200 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. In-Reply-To: References: <4F748833.3030604@communicatefreely.net> Message-ID: How "expensive"/complex is each call routing? Are you doing complex LCR and pricing lookups? Simple routing on regex? A LONG regex list? 1 sql query to pick a route based on the incoming DID...? -Avi On Thu, Mar 29, 2012 at 9:47 PM, Joe Flemmings wrote: > Right now having issue from about 80-90. The kind of traffic we run i high > bust and short traffic. so cps is most important for me. I'm also looking > into different Linux options to improve this. > > Joe > > > On Thu, Mar 29, 2012 at 12:42 PM, Lloyd Aloysius < > lloyd.aloysius at sunteltech.ca> wrote: > >> What is the cps you are expecting? >> >> >> * * >> >> >> >> On Thu, Mar 29, 2012 at 3:37 PM, Joe Flemmings wrote: >> >>> The problem is not the number of simultaneous calls but cps. I'm doing >>> thousands of simultaneous calls too and that's not an issue. I'm using 64 >>> bit machines and just started tests this week. The most expensive system >>> call right now is select and trying to isolate where its coming from. The >>> issue is not the number of calls but cps. >>> >>> >>> On Thu, Mar 29, 2012 at 12:30 PM, Michael Collins wrote: >>> >>>> >>>> >>>> On Thu, Mar 29, 2012 at 12:15 PM, Joe Flemmings < >>>> joe.jflemmings at gmail.com> wrote: >>>> >>>>> My question was very specific, speed or system resource utilization. >>>>> >>>>> Which has the least load of FreeSwitch? I can always look for >>>>> different ways to accomplish other tasks like redundancy etc. >>>>> >>>>> I have tried both xml_curl and lua and under high load the system >>>>> level calls are killing me. >>>>> I was wondering if event socket are a better way to go/ >>>>> >>>>> so between lua, xml_curl and event socket.http://www.freeswitch.org >>>>> >>>>> >>>> I'm curious - what system level calls are killing your system when you >>>> use xml_curl? (Or Lua, etc.) I can refer you to Ken Rice who does thousands >>>> upon thousands of calls a day, many simultaneously. I'm sure he's got some >>>> good insights on this. Also, make sure that you're not on crappy hardware >>>> or doing something silly like running a 32 bit OS on 64 bit hardware... >>>> >>>> -MC >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/76927bfd/attachment-0001.html From kris at kriskinc.com Fri Mar 30 00:08:38 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 29 Mar 2012 16:08:38 -0400 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. In-Reply-To: References: <4F748833.3030604@communicatefreely.net> Message-ID: For CPS, generally speaking, the FreeSWITCH dynamic configuration methods rank in this order of performance (best to worst): - mod_xml_curl - lua - event socket I haven't tested any others but xml_curl should be fine at 80-90 cps. How are you profiling this to make sure your issue isn't elsewhere? On Thu, Mar 29, 2012 at 3:47 PM, Joe Flemmings wrote: > Right now having issue from about 80-90. The kind of traffic we run i high > bust and short traffic. so cps is most important for me. I'm also looking > into different Linux options to improve this. > > Joe > -- Kristian Kielhofner From markus.lindenberg at gmail.com Fri Mar 30 00:09:16 2012 From: markus.lindenberg at gmail.com (Markus Lindenberg) Date: Thu, 29 Mar 2012 22:09:16 +0200 Subject: [Freeswitch-users] One-legged callee_id In-Reply-To: References: <4F7319E2.5010107@communicatefreely.net> Message-ID: Great, thanks, confirmed that works for me. After i downgraded Snom firmware to 8.4.22 (later Firmware is broken in different ways on 370 and 360), send_display works (also for one-legged calls) and connectedline gets updated after bridge/transfer. The only thing that doesn't work for Snom phones is connected lind info in ringing phase. Freeswitch is sending RPID to the caller, Snom expects PAI. Polycom works great. Aastra doesn't display anything at all before 200OK, i'll have to investigate what they expect. Setting sip_cid_type just affects the called party. If i tell the Snom phones that they're connected to a Cisco Callmanager, they will read RPID. But i don't know about possible side effects. On Thu, Mar 29, 2012 at 18:49, Anthony Minessale wrote: > Its not a workaround, its because each phone does it differently and > other phones consider each other's way a fatal err and hangup calls. > > set effective_callee_id_name and effective_callee_id_number variables > before you execute ring_ready, pre_answer or answer or before you run > an app that may answer the call and you will get what you are looking > for. > > If the call is already up you can use the send_display app > > > > And from ESL or cli > > uuid_display This will be the name|1234 > > > > On Thu, Mar 29, 2012 at 2:32 AM, Markus Lindenberg > wrote: >> Hi Tim, >> >> would love this as well. >> >> I guess the answer application should see if effective_callee_id_* is >> set and trigger a display update on the calling phone. It would also >> be nice to be able to trigger display updates to the a leg during >> (ivr) execution. >> >> Asterisk's CONNECTEDLINE() allows all of this. It's using >> Reinvites/Updates w/ PAI-Header and the syntax is cumbersome as >> everything asterisk is, but at least it works with most phones. (Snom >> 3xx, Snom M3, Siemens OpenStage, Polycom confirmed as i'm working with >> these every day) >> >> FreeSWITCH display updates seem to rely on workarounds (grep polycom >> or snom in mod_sofia). I would like to be able to disable the >> INFO/sipfrag thing for Snom, use PAI instead of RDNIS in the A-Leg and >> send Reinvites for Display Update etc., but it all seems to be hard >> coded right now. >> >> Regards, Markus >> >> On Wed, Mar 28, 2012 at 16:02, Tim St. Pierre >> wrote: >>> Hello, >>> >>> I have been making extensive use of the callee_id_name and >>> callee_id_number variables so that our IP phones update the display to >>> show who they are actually talking to on the other end. >>> >>> Generally, it works very well, but it only seems to work if there is a >>> bridge. ?Is there a way to set the callee_id in a single legged call? >>> ie. I want the name of the company to come up when the caller is in the >>> main IVR, or the name of the conference to come up if they are in the >>> conference room. >>> >>> Is there a way to do this? >>> >>> Thanks! >>> >>> -Tim >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From potxoka at gmail.com Fri Mar 30 01:09:54 2012 From: potxoka at gmail.com (Anto) Date: Thu, 29 Mar 2012 23:09:54 +0200 Subject: [Freeswitch-users] Codec Negotiation. Bug? Message-ID: Hello I'm still fighting with the codec negotiation with carriers (as I am unable to understand how in FreeSWITCH) and I have seen something, I do not know if it's a bug. If I use a variable in the specification of output codecs do not see a corresponding message sip, however if specified in the parameter, if negotiated with the carrier. http://pastebin.freeswitch.org/18782 ----------------------------------------------------------------------------------------------- http://pastebin.freeswitch.org/18784 Do not understand why in one case send sip message and not in another (in this cancel the call because it does not handle the negotiation). What can be?. Thanks !!! Best regards Anto From lloyd.aloysius at sunteltech.ca Fri Mar 30 01:37:54 2012 From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius) Date: Thu, 29 Mar 2012 17:37:54 -0400 Subject: [Freeswitch-users] xml_curl & user_data , user_exists - return In-Reply-To: References: Message-ID: It should work with xml_curl. Wiki says *Using this user_data function in combination with mod_xml_curl will generate an additional request each time the user_data function is called. Note that it is already called once in the Local Extension section to determine the callgroup* But is not working in my case. On Thu, Mar 29, 2012 at 2:19 PM, Brian Foster wrote: > Does that actually work with xml_curl? > > Might only work on just plain static xml files. > > -BDF > > > On Thu, Mar 29, 2012 at 1:48 PM, Lloyd Aloysius wrote: > >> Hi All, >> >> I am trying to use the xml_curl & user_data .. retrieve the user >> information. >> >> User registration is fine. >> >> . >> >> But >> >> user_data 21 at mydomain.com attr id >> >> return -ERR no reply >> >> user_exists id eddie mydomain.com >> >> return false >> >> >> Am I missing something here? >> >> Thanks >> Lloyd >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/9f9810a8/attachment.html From joe.jflemmings at gmail.com Fri Mar 30 02:53:34 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Thu, 29 Mar 2012 15:53:34 -0700 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. In-Reply-To: References: <4F748833.3030604@communicatefreely.net> Message-ID: Have already done this. On Thu, Mar 29, 2012 at 1:00 PM, Michael Collins wrote: > How about putting the database in a ramdrive? > -MC > > > On Thu, Mar 29, 2012 at 12:47 PM, Joe Flemmings wrote: > >> Right now having issue from about 80-90. The kind of traffic we run i >> high bust and short traffic. so cps is most important for me. I'm also >> looking into different Linux options to improve this. >> >> Joe >> >> >> On Thu, Mar 29, 2012 at 12:42 PM, Lloyd Aloysius < >> lloyd.aloysius at sunteltech.ca> wrote: >> >>> What is the cps you are expecting? >>> >>> >>> * * >>> >>> >>> >>> On Thu, Mar 29, 2012 at 3:37 PM, Joe Flemmings >> > wrote: >>> >>>> The problem is not the number of simultaneous calls but cps. I'm doing >>>> thousands of simultaneous calls too and that's not an issue. I'm using 64 >>>> bit machines and just started tests this week. The most expensive system >>>> call right now is select and trying to isolate where its coming from. The >>>> issue is not the number of calls but cps. >>>> >>>> >>>> On Thu, Mar 29, 2012 at 12:30 PM, Michael Collins wrote: >>>> >>>>> >>>>> >>>>> On Thu, Mar 29, 2012 at 12:15 PM, Joe Flemmings < >>>>> joe.jflemmings at gmail.com> wrote: >>>>> >>>>>> My question was very specific, speed or system resource utilization. >>>>>> >>>>>> Which has the least load of FreeSwitch? I can always look for >>>>>> different ways to accomplish other tasks like redundancy etc. >>>>>> >>>>>> I have tried both xml_curl and lua and under high load the system >>>>>> level calls are killing me. >>>>>> I was wondering if event socket are a better way to go/ >>>>>> >>>>>> so between lua, xml_curl and event socket.http://www.freeswitch.org >>>>>> >>>>>> >>>>> I'm curious - what system level calls are killing your system when you >>>>> use xml_curl? (Or Lua, etc.) I can refer you to Ken Rice who does thousands >>>>> upon thousands of calls a day, many simultaneously. I'm sure he's got some >>>>> good insights on this. Also, make sure that you're not on crappy hardware >>>>> or doing something silly like running a 32 bit OS on 64 bit hardware... >>>>> >>>>> -MC >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/337e6b59/attachment-0001.html From joe.jflemmings at gmail.com Fri Mar 30 02:55:11 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Thu, 29 Mar 2012 15:55:11 -0700 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. In-Reply-To: References: <4F748833.3030604@communicatefreely.net> Message-ID: For testing was only doing very simple routing. i'll need to do some tests on a call that i was making that could be an issue. So still need to do some tests On Thu, Mar 29, 2012 at 1:03 PM, Avi Marcus wrote: > How "expensive"/complex is each call routing? Are you doing complex LCR > and pricing lookups? Simple routing on regex? A LONG regex list? 1 sql > query to pick a route based on the incoming DID...? > > -Avi > > > On Thu, Mar 29, 2012 at 9:47 PM, Joe Flemmings wrote: > >> Right now having issue from about 80-90. The kind of traffic we run i >> high bust and short traffic. so cps is most important for me. I'm also >> looking into different Linux options to improve this. >> >> Joe >> >> >> On Thu, Mar 29, 2012 at 12:42 PM, Lloyd Aloysius < >> lloyd.aloysius at sunteltech.ca> wrote: >> >>> What is the cps you are expecting? >>> >>> >>> * * >>> >>> >>> >>> On Thu, Mar 29, 2012 at 3:37 PM, Joe Flemmings >> > wrote: >>> >>>> The problem is not the number of simultaneous calls but cps. I'm doing >>>> thousands of simultaneous calls too and that's not an issue. I'm using 64 >>>> bit machines and just started tests this week. The most expensive system >>>> call right now is select and trying to isolate where its coming from. The >>>> issue is not the number of calls but cps. >>>> >>>> >>>> On Thu, Mar 29, 2012 at 12:30 PM, Michael Collins wrote: >>>> >>>>> >>>>> >>>>> On Thu, Mar 29, 2012 at 12:15 PM, Joe Flemmings < >>>>> joe.jflemmings at gmail.com> wrote: >>>>> >>>>>> My question was very specific, speed or system resource utilization. >>>>>> >>>>>> Which has the least load of FreeSwitch? I can always look for >>>>>> different ways to accomplish other tasks like redundancy etc. >>>>>> >>>>>> I have tried both xml_curl and lua and under high load the system >>>>>> level calls are killing me. >>>>>> I was wondering if event socket are a better way to go/ >>>>>> >>>>>> so between lua, xml_curl and event socket.http://www.freeswitch.org >>>>>> >>>>>> >>>>> I'm curious - what system level calls are killing your system when you >>>>> use xml_curl? (Or Lua, etc.) I can refer you to Ken Rice who does thousands >>>>> upon thousands of calls a day, many simultaneously. I'm sure he's got some >>>>> good insights on this. Also, make sure that you're not on crappy hardware >>>>> or doing something silly like running a 32 bit OS on 64 bit hardware... >>>>> >>>>> -MC >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/df67d454/attachment.html From joe.jflemmings at gmail.com Fri Mar 30 02:56:18 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Thu, 29 Mar 2012 15:56:18 -0700 Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch. In-Reply-To: References: <4F748833.3030604@communicatefreely.net> Message-ID: Thank you, this is what i was asking for. On Thu, Mar 29, 2012 at 1:08 PM, Kristian Kielhofner wrote: > For CPS, generally speaking, the FreeSWITCH dynamic configuration > methods rank in this order of performance (best to worst): > > - mod_xml_curl > - lua > - event socket > > I haven't tested any others but xml_curl should be fine at 80-90 cps. > How are you profiling this to make sure your issue isn't elsewhere? > > On Thu, Mar 29, 2012 at 3:47 PM, Joe Flemmings > wrote: > > Right now having issue from about 80-90. The kind of traffic we run i > high > > bust and short traffic. so cps is most important for me. I'm also looking > > into different Linux options to improve this. > > > > Joe > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/8a5e8eaa/attachment.html From lesley.pervis at gmail.com Fri Mar 30 04:02:51 2012 From: lesley.pervis at gmail.com (Lesley Pervis) Date: Thu, 29 Mar 2012 18:02:51 -0600 Subject: [Freeswitch-users] sip_exclude_contact in mad boss scenario does not work Message-ID: I'm using a small variation on the mad boss extension in the default dialplan, much like this, except instead of using groups, I'm using multiple user/100X@${domain} lines. http://wiki.freeswitch.org/wiki/Variable_conference_auto_outcall_prefix The problem is that the sip_exclude contact does not work as I expect. I expect that the conference is NOT bridged to the caller, but it is. I can see in the logs that a conference leg to the calling phone is set up, and the phone answers and puts it on hold. This means there are two conference legs up on the calling phone: the one that initiated the page, and another deaf leg. When the first leg hangs up, the deaf one is on hold, and to end the conference, I have to resume that leg and hang it up. Anyone know why this isn't working for me? FreeSWITCH version: 1.0.head (git-d8d4d20 2012-03-14 19-00-26 +0000) Thanks, Les -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/b87b75e1/attachment-0001.html From msc at freeswitch.org Fri Mar 30 06:03:28 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 29 Mar 2012 19:03:28 -0700 Subject: [Freeswitch-users] sip_exclude_contact in mad boss scenario does not work In-Reply-To: References: Message-ID: Grab a console debug log and throw it on pastebin.freeswitch.org. Use "FreeSWITCH Log" for syntax highlighting and then give us the pb URL in this thread. -MC On Thu, Mar 29, 2012 at 5:02 PM, Lesley Pervis wrote: > I'm using a small variation on the mad boss extension in the default > dialplan, much like this, except instead of using groups, I'm using > multiple user/100X@${domain} lines. > > http://wiki.freeswitch.org/wiki/Variable_conference_auto_outcall_prefix > > The problem is that the sip_exclude contact does not work as I expect. I > expect that the conference is NOT bridged to the caller, but it is. I can > see in the logs that a conference leg to the calling phone is set up, and > the phone answers and puts it on hold. This means there are two conference > legs up on the calling phone: the one that initiated the page, and another > deaf leg. When the first leg hangs up, the deaf one is on hold, and to end > the conference, I have to resume that leg and hang it up. > > Anyone know why this isn't working for me? > > FreeSWITCH version: 1.0.head (git-d8d4d20 2012-03-14 19-00-26 +0000) > > Thanks, > Les > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120329/006c09e5/attachment.html From sharad at coraltele.com Fri Mar 30 10:11:55 2012 From: sharad at coraltele.com (Sharad Garg) Date: Fri, 30 Mar 2012 11:41:55 +0530 Subject: [Freeswitch-users] Session-wise Language Setting References: <4CB4E6075A9F4F9BA1C8A57FF2A06735@sharad> <4F746EFF.3050305@communicatefreely.net> Message-ID: <00C2DD790B254A1BACD50CAA66B123B9@sharad> Dear Tim, Thanks for your kind reply. Just another curisity....For a live call, if I set the default_language say English, now that call will be entertained in english. Now while first call is in progreess, second call comes & this caller opts for Russian...so we have to set default language as russian. In this case, all the voice prompts which are being played to first caller, will be in russian. Is'nt it ? Actually first call should continue in english & second call should be in russian & so on. Plz clarify. Regards Sharad ----- Original Message ----- From: "Tim St. Pierre" To: "FreeSWITCH Users Help" Sent: Thursday, March 29, 2012 7:47 PM Subject: Re: [Freeswitch-users] Session-wise Language Setting > Yes! > > On incoming calls, I set default_language to either en or fr (I'm in > Canada), depending on the DID. I can also change that variable in an IVR > if the caller wants to change the language. On outgoing calls, I set the > same variable in the directory, so that any call that a phone makes will > have a language set. This is great in a multi-lingual environment, as > each user can have a language preference that will be used for voice mail > and other system prompts. We store that in a database and also have the > provisioning system look to that same variable, so the screen labels and > text on the phone is the same language. > > -Tim > > Sharad Garg wrote: >> Hi All >> >> Just wondering whether we can define the language from beginning of the >> call means when a call is originated or landed to Freeswitch, can we >> define the language of all the prompts whether the call should be >> processed in English or Russian or so on.? >> >> Regards >> Sharad >> >> >> ----- Original Message ----- >> From: >> To: >> Sent: Thursday, March 29, 2012 7:33 AM >> Subject: FreeSWITCH-users Digest, Vol 69, Issue 282 >> >> >> >>> Send FreeSWITCH-users mailing list submissions to >>> freeswitch-users at lists.freeswitch.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> or, via email, send a message with subject or body 'help' to >>> freeswitch-users-request at lists.freeswitch.org >>> >>> You can reach the person managing the list at >>> freeswitch-users-owner at lists.freeswitch.org >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of FreeSWITCH-users digest..." >>> >>> >> >> >> -------------------------------------------------------------------------------- >> >> >> >>> Today's Topics: >>> >>> 1. Re: Open Bugs on Jira and Call for help (Ken Rice) >>> 2. mod_callcenter and moh-sound (Vik Killa) >>> 3. Re: mod_callcenter and moh-sound (Vik Killa) >>> 4. Registration VIA TCP (Rob Moore) >>> 5. Forward calls from opensips to freeswitch (Sherif Omran) >>> 6. Re: Registration VIA TCP (Mitch Capper) >>> >>> >> >> >> -------------------------------------------------------------------------------- >> >> >> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > From peter.olsson at visionutveckling.se Fri Mar 30 10:46:53 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 30 Mar 2012 06:46:53 +0000 Subject: [Freeswitch-users] Session-wise Language Setting In-Reply-To: <00C2DD790B254A1BACD50CAA66B123B9@sharad> References: <4CB4E6075A9F4F9BA1C8A57FF2A06735@sharad> <4F746EFF.3050305@communicatefreely.net>, <00C2DD790B254A1BACD50CAA66B123B9@sharad> Message-ID: <1FFF97C269757C458224B7C895F35F15083A49@cantor.std.visionutv.se> It's a channel variable, it's unique per channel, so there is no risk that a variable on channel A overwrites the value of the variable in channel B. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Sharad Garg [sharad at coraltele.com] Skickat: den 30 mars 2012 08:11 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Session-wise Language Setting Dear Tim, Thanks for your kind reply. Just another curisity....For a live call, if I set the default_language say English, now that call will be entertained in english. Now while first call is in progreess, second call comes & this caller opts for Russian...so we have to set default language as russian. In this case, all the voice prompts which are being played to first caller, will be in russian. Is'nt it ? Actually first call should continue in english & second call should be in russian & so on. Plz clarify. Regards Sharad ----- Original Message ----- From: "Tim St. Pierre" To: "FreeSWITCH Users Help" Sent: Thursday, March 29, 2012 7:47 PM Subject: Re: [Freeswitch-users] Session-wise Language Setting > Yes! > > On incoming calls, I set default_language to either en or fr (I'm in > Canada), depending on the DID. I can also change that variable in an IVR > if the caller wants to change the language. On outgoing calls, I set the > same variable in the directory, so that any call that a phone makes will > have a language set. This is great in a multi-lingual environment, as > each user can have a language preference that will be used for voice mail > and other system prompts. We store that in a database and also have the > provisioning system look to that same variable, so the screen labels and > text on the phone is the same language. > > -Tim > > Sharad Garg wrote: >> Hi All >> >> Just wondering whether we can define the language from beginning of the >> call means when a call is originated or landed to Freeswitch, can we >> define the language of all the prompts whether the call should be >> processed in English or Russian or so on.? >> >> Regards >> Sharad >> >> >> ----- Original Message ----- >> From: >> To: >> Sent: Thursday, March 29, 2012 7:33 AM >> Subject: FreeSWITCH-users Digest, Vol 69, Issue 282 >> >> >> >>> Send FreeSWITCH-users mailing list submissions to >>> freeswitch-users at lists.freeswitch.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> or, via email, send a message with subject or body 'help' to >>> freeswitch-users-request at lists.freeswitch.org >>> >>> You can reach the person managing the list at >>> freeswitch-users-owner at lists.freeswitch.org >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of FreeSWITCH-users digest..." >>> >>> >> >> >> -------------------------------------------------------------------------------- >> >> >> >>> Today's Topics: >>> >>> 1. Re: Open Bugs on Jira and Call for help (Ken Rice) >>> 2. mod_callcenter and moh-sound (Vik Killa) >>> 3. Re: mod_callcenter and moh-sound (Vik Killa) >>> 4. Registration VIA TCP (Rob Moore) >>> 5. Forward calls from opensips to freeswitch (Sherif Omran) >>> 6. Re: Registration VIA TCP (Mitch Capper) >>> >>> >> >> >> -------------------------------------------------------------------------------- >> >> >> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f75538432768881816776! From Daniel.Knaggs at realitysolutions.co.uk Fri Mar 30 12:16:39 2012 From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs) Date: Fri, 30 Mar 2012 08:16:39 +0000 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls References: Message-ID: Any ideas, anyone? [cid:imagefb5387.PNG at ca5d49be.4fa92b00] Daniel Knaggs Software Developer Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE Tel: 01482 828000 / Fax: 01482 373100 Daniel.Knaggs at realitysolutions.co.uk www.realitysolutions.co.uk ________________________________ Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire [cid:image19a0f2.PNG at 39ea781d.48900adc] From: Daniel Knaggs Sent: 27 March 2012 10:22 To: 'FreeSWITCH Users Help' Subject: RE: [Freeswitch-users] Strange DTMF Tones On Inbound Calls OK, it does NOT appear in the recordings. Here is a spectrum analysis of the call when it happened: - Frequency (Hz) Level (dB) 15.625000 -59.445583 31.250000 -69.136353 46.875000 -71.569084 62.500000 -72.445877 78.125000 -67.142319 93.750000 -63.964447 109.375000 -61.157677 125.000000 -58.849174 140.625000 -62.083492 156.250000 -51.071819 171.875000 -48.957565 187.500000 -48.156109 203.125000 -48.908588 218.750000 -47.891460 234.375000 -46.257698 250.000000 -43.011139 265.625000 -48.706509 281.250000 -53.241238 296.875000 -52.513802 312.500000 -50.387054 328.125000 -53.275459 343.750000 -56.096489 359.375000 -54.788857 375.000000 -52.283993 390.625000 -50.975601 406.250000 -49.737530 421.875000 -49.870571 437.500000 -50.445431 453.125000 -50.647591 468.750000 -49.450035 484.375000 -48.767914 500.000000 -48.281902 515.625000 -49.419975 531.250000 -52.635048 546.875000 -52.856857 562.500000 -51.675240 578.125000 -50.667599 593.750000 -51.584862 609.375000 -48.992435 625.000000 -48.492191 640.625000 -49.701382 656.250000 -53.005512 671.875000 -54.269699 687.500000 -54.937397 703.125000 -53.922794 718.750000 -51.949009 734.375000 -52.763660 750.000000 -53.731155 765.625000 -54.977795 781.250000 -54.029835 796.875000 -54.375313 812.500000 -54.419693 828.125000 -54.796967 843.750000 -55.052872 859.375000 -53.625359 875.000000 -54.649513 890.625000 -56.736855 906.250000 -57.202946 921.875000 -56.587177 937.500000 -57.263260 953.125000 -56.744713 968.750000 -57.710808 984.375000 -58.118694 1000.000000 -60.595867 1015.625000 -60.736752 1031.250000 -57.966949 1046.875000 -58.101311 1062.500000 -59.093056 1078.125000 -59.889881 1093.750000 -59.972092 1109.375000 -59.728626 1125.000000 -58.060120 1140.625000 -56.506371 1156.250000 -55.109329 1171.875000 -54.278183 1187.500000 -56.115913 1203.125000 -57.535069 1218.750000 -58.304485 1234.375000 -58.614723 1250.000000 -58.873520 1265.625000 -61.268688 1281.250000 -62.702538 1296.875000 -63.828163 1312.500000 -63.712082 1328.125000 -65.020050 1343.750000 -64.481941 1359.375000 -62.652424 1375.000000 -61.640785 1390.625000 -61.719952 1406.250000 -61.451962 1421.875000 -58.490433 1437.500000 -57.193993 1453.125000 -56.973965 1468.750000 -57.940182 1484.375000 -59.935654 1500.000000 -61.829227 1515.625000 -63.259266 1531.250000 -62.169945 1546.875000 -61.958508 1562.500000 -62.018082 1578.125000 -60.562317 1593.750000 -59.146374 1609.375000 -58.868763 1625.000000 -60.028801 1640.625000 -59.241119 1656.250000 -60.101322 1671.875000 -60.429554 1687.500000 -60.538834 1703.125000 -61.073635 1718.750000 -60.375114 1734.375000 -60.152672 1750.000000 -59.529144 1765.625000 -59.783649 1781.250000 -61.850414 1796.875000 -63.506111 1812.500000 -63.869320 1828.125000 -62.442127 1843.750000 -62.519482 1859.375000 -63.814472 1875.000000 -63.300858 1890.625000 -61.998482 1906.250000 -62.199738 1921.875000 -61.745621 1937.500000 -61.611210 1953.125000 -61.174740 1968.750000 -60.504318 1984.375000 -60.277737 2000.000000 -61.696060 2015.625000 -61.693054 2031.250000 -62.755184 2046.875000 -63.983723 2062.500000 -63.584675 2078.125000 -62.943420 2093.750000 -63.238541 2109.375000 -62.929970 2125.000000 -62.243599 2140.625000 -60.967789 2156.250000 -60.915421 2171.875000 -61.997654 2187.500000 -63.702518 2203.125000 -64.316109 2218.750000 -63.577995 2234.375000 -63.250530 2250.000000 -63.107010 2265.625000 -62.953743 2281.250000 -62.933647 2296.875000 -61.956242 2312.500000 -62.948036 2328.125000 -64.375603 2343.750000 -64.296860 2359.375000 -63.552139 2375.000000 -62.556362 2390.625000 -62.979954 2406.250000 -64.505875 2421.875000 -65.626236 2437.500000 -65.732819 2453.125000 -66.130798 2468.750000 -65.920502 2484.375000 -64.183731 2500.000000 -63.387459 2515.625000 -63.110027 2531.250000 -64.085106 2546.875000 -64.269905 2562.500000 -64.181808 2578.125000 -64.600060 2593.750000 -63.998692 2609.375000 -63.854473 2625.000000 -65.015961 2640.625000 -65.751480 2656.250000 -66.293800 2671.875000 -66.494102 2687.500000 -66.300240 2703.125000 -66.383118 2718.750000 -66.466385 2734.375000 -65.733604 2750.000000 -65.110283 2765.625000 -65.537567 2781.250000 -66.125465 2796.875000 -65.979088 2812.500000 -64.833984 2828.125000 -63.773678 2843.750000 -64.419113 2859.375000 -64.800369 2875.000000 -64.710480 2890.625000 -64.088387 2906.250000 -64.790306 2921.875000 -65.160469 2937.500000 -65.285408 2953.125000 -66.030342 2968.750000 -65.027481 2984.375000 -64.623558 3000.000000 -65.082748 3015.625000 -63.680820 3031.250000 -62.836716 3046.875000 -62.210663 3062.500000 -61.578278 3078.125000 -62.397720 3093.750000 -63.185940 3109.375000 -62.439983 3125.000000 -62.382778 3140.625000 -63.123928 3156.250000 -64.276588 3171.875000 -65.444725 3187.500000 -65.891289 3203.125000 -65.480240 3218.750000 -64.761063 3234.375000 -65.140015 3250.000000 -66.010643 3265.625000 -66.964401 3281.250000 -67.296051 3296.875000 -66.430000 3312.500000 -66.564758 3328.125000 -67.878830 3343.750000 -67.748436 3359.375000 -68.965981 3375.000000 -70.426888 3390.625000 -71.400375 3406.250000 -72.067627 3421.875000 -71.944176 3437.500000 -72.285637 3453.125000 -71.983047 3468.750000 -72.565109 3484.375000 -72.350845 3500.000000 -72.335533 3515.625000 -72.608849 3531.250000 -72.417786 3546.875000 -73.100441 3562.500000 -73.461548 3578.125000 -73.558250 3593.750000 -73.218422 3609.375000 -73.994888 3625.000000 -74.379204 3640.625000 -74.896202 3656.250000 -74.944405 3671.875000 -74.958710 3687.500000 -75.029655 3703.125000 -74.314133 3718.750000 -74.855209 3734.375000 -76.021591 3750.000000 -76.441444 3765.625000 -77.122787 3781.250000 -77.733589 3796.875000 -79.138847 3812.500000 -80.206360 3828.125000 -80.391960 3843.750000 -81.214249 3859.375000 -81.434105 3875.000000 -82.234749 3890.625000 -82.529884 3906.250000 -82.929169 3921.875000 -83.735237 3937.500000 -84.770660 3953.125000 -84.203438 3968.750000 -84.455025 3984.375000 -85.116302 2012-03-27 09:50:40.412101 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=320/320/2000 seq=38825 lw=28160 2012-03-27 09:50:40.433101 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=480/480/2000 seq=38826 lw=28320 2012-03-27 09:50:40.454100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=640/640/2000 seq=38827 lw=28480 2012-03-27 09:50:40.475100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=800/800/2000 seq=38828 lw=28640 2012-03-27 09:50:40.496100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=960/960/2000 seq=38829 lw=28800 2012-03-27 09:50:40.517100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1120/1120/2000 seq=38830 lw=28960 2012-03-27 09:50:40.538100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1280/1280/2000 seq=38831 lw=29120 2012-03-27 09:50:40.559099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1440/1440/2000 seq=38832 lw=29280 2012-03-27 09:50:40.580099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1600/1600/2000 seq=38833 lw=29440 2012-03-27 09:50:40.601099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1760/1760/2000 seq=38834 lw=29600 2012-03-27 09:50:40.622098 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1920/1920/2000 seq=38835 lw=29760 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38836 lw=29760 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38837 lw=29760 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38838 lw=29760 Any ideas? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 26 March 2012 18:39 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls On Mon, Mar 26, 2012 at 3:01 AM, Daniel Knaggs > wrote: OK, call recording has been setup ? waiting for it to happen now. Interestingly, before I issue the ?record_session? application (and of course the ?RECORD_*? variables) I had to execute ?ring_ready? then ?pre_answer? otherwise the caller gets silence (changing the order of those two commands results in silence). I find it odd that just doing a pre_answer wouldn't be sufficient. A pre_answer will send a 183 w/SDP whereas ring_ready simply sends a 180. In any case, I'm glad you got your recordings. I also find it curious that only the "letter" DTMFs are being detected. Let us know if you actually hear those tones in the audio stream. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/d51cc4e3/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: imagefb5387.PNG Type: image/png Size: 22463 bytes Desc: imagefb5387.PNG Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/d51cc4e3/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: image19a0f2.PNG Type: image/png Size: 106263 bytes Desc: image19a0f2.PNG Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/d51cc4e3/attachment-0003.png From anton.vazir at gmail.com Fri Mar 30 12:59:44 2012 From: anton.vazir at gmail.com (Anton VG) Date: Fri, 30 Mar 2012 13:59:44 +0500 Subject: [Freeswitch-users] SIP trace specific call or uuid Message-ID: HI! Is there a way to enable sip-trace on the specific call or uuid? I've been only able to find how to enable it per profile, but on a busy situation, the information amount is too high to be useful, so it's often necessary to trace a specific call or call to specific destination. From anton.vazir at gmail.com Fri Mar 30 14:16:58 2012 From: anton.vazir at gmail.com (Anton VG) Date: Fri, 30 Mar 2012 15:16:58 +0500 Subject: [Freeswitch-users] When using nimbuzz, call disconnects after 90 seconds with RECOVERY_ON_TIMER_EXPIRE Message-ID: Please help, While other SIP clients works well, there is a problem with nimbuzz SIP client, which quite frequently used by 'not-advanced' users. seems It which works through their services (asterisk 1.6) This problem was not happening while I have been using Asterisk too, but after recent migration to FS - it does. nimbuzz client stays like online, just sound disappears, but there is a confirmation from nimbuzz server side, regarding the BYE most likely a NAT issue, but if the same time using Asterisk on the same server - there is not problems. So might be there any options to tune? there are some SIP activity like this before calls disconnects, no replies: send 977 bytes to udp/[195.211.48.6]:12621 at 10:00:02.535856: ------------------------------------------------------------------------ INVITE sip:anton at 192.168.0.20:5061;transport=udp SIP/2.0 Via: SIP/2.0/UDP 62.122.137.32;rport;branch=z9hG4bKaQm6tK1aeZ7Fe Max-Forwards: 70 From: "3777077" ;tag=eKv1Nej7ZrNUS To: "anton at 62.122.137.32" ;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 26214721 INVITE Contact: User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 272 v=0 o=FreeSWITCH 1333081454 1333081455 IN IP4 62.122.137.32 s=FreeSWITCH c=IN IP4 62.122.137.32 t=0 0 m=audio 20062 RTP/AVP 18 106 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=silenceSupp:off - - - - a=ptime:20 and than 2012-03-30 15:00:03.554460 [NOTICE] sofia.c:6301 Hangup sofia/internal/anton at 62.122.137.32 [CS_SOFT_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960518: ------------------------------------------------------------------------ BYE sip:3777077 at 62.122.137.32:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: ;tag=eKv1Nej7ZrNUS From: ;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Nimbuzz Single Content-Length: 0 ------------------------------------------------------------------------ send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960672: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 192.168.0.20:5061;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 From: ;tag=sip1331131792771 To: ;tag=eKv1Nej7ZrNUS Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960734: ------------------------------------------------------------------------ BYE sip:3777077 at 62.122.137.32:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: ;tag=eKv1Nej7ZrNUS From: ;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Nimbuzz Single Content-Length: 0 ------------------------------------------------------------------------ send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960781: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 192.168.0.20:5061;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 From: ;tag=sip1331131792771 To: ;tag=eKv1Nej7ZrNUS Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079094: ------------------------------------------------------------------------ ACK sip:anton at 62.122.137.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: ;tag=eKv1Nej7ZrNUS From: ;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 ACK Contact: Expires: 3600 User-Agent: Nimbuzz Single Content-Length: 0 ------------------------------------------------------------------------ recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079293: ------------------------------------------------------------------------ ACK sip:anton at 62.122.137.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: ;tag=eKv1Nej7ZrNUS From: ;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 ACK Contact: Expires: 3600 User-Agent: Nimbuzz Single Content-Length: 0 ------------------------------------------------------------------------ From bdfoster at endigotech.com Fri Mar 30 14:27:11 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 30 Mar 2012 06:27:11 -0400 Subject: [Freeswitch-users] Trouble setting "caller_id_name", "caller_id_number" on an inbound call Message-ID: As the subject says, I'm having issues setting the caller id name and number on an inbound call when doing some cleanups and a lookup. These dialplans are based on what information is listed for Mod_callcenter, and I have a feeling that they are wrong. I'll volunteer myself to change them if they are. Alright, so here's the dialplan: Call log here: http://pastebin.freeswitch.org/18786 I've tried setting the effective_caller_id_(name/number) but no luck there. Is there something I'm doing wrong here? -BDF -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/a1725642/attachment.html From avi at avimarcus.net Fri Mar 30 14:42:23 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 30 Mar 2012 13:42:23 +0300 Subject: [Freeswitch-users] SIP trace specific call or uuid In-Reply-To: References: Message-ID: Not within FS, yet. Try ngrep - http://wiki.freeswitch.org/wiki/Packet_Capture#ngrep -Avi On Fri, Mar 30, 2012 at 11:59 AM, Anton VG wrote: > HI! > > Is there a way to enable sip-trace on the specific call or uuid? > > I've been only able to find how to enable it per profile, but on a > busy situation, the information amount > is too high to be useful, so it's often necessary to trace a specific > call or call to specific destination. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/3852d52f/attachment.html From covici at ccs.covici.com Fri Mar 30 15:10:45 2012 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Fri, 30 Mar 2012 07:10:45 -0400 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: References: Message-ID: <32544.1333105845@ccs.covici.com> Maybe its generated by an ata box if you have an analog phone? Daniel Knaggs wrote: > Any ideas, anyone? > > > [cid:imagefb5387.PNG at ca5d49be.4fa92b00] > > Daniel Knaggs > > Software Developer > > > Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE > Tel: 01482 828000 / Fax: 01482 373100 > Daniel.Knaggs at realitysolutions.co.uk > www.realitysolutions.co.uk > ________________________________ > > Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire > > > [cid:image19a0f2.PNG at 39ea781d.48900adc] > > > From: Daniel Knaggs > Sent: 27 March 2012 10:22 > To: 'FreeSWITCH Users Help' > Subject: RE: [Freeswitch-users] Strange DTMF Tones On Inbound Calls > > OK, it does NOT appear in the recordings. > > > Here is a spectrum analysis of the call when it happened: - > > Frequency (Hz) Level (dB) > 15.625000 -59.445583 > 31.250000 -69.136353 > 46.875000 -71.569084 > 62.500000 -72.445877 > 78.125000 -67.142319 > 93.750000 -63.964447 > 109.375000 -61.157677 > 125.000000 -58.849174 > 140.625000 -62.083492 > 156.250000 -51.071819 > 171.875000 -48.957565 > 187.500000 -48.156109 > 203.125000 -48.908588 > 218.750000 -47.891460 > 234.375000 -46.257698 > 250.000000 -43.011139 > 265.625000 -48.706509 > 281.250000 -53.241238 > 296.875000 -52.513802 > 312.500000 -50.387054 > 328.125000 -53.275459 > 343.750000 -56.096489 > 359.375000 -54.788857 > 375.000000 -52.283993 > 390.625000 -50.975601 > 406.250000 -49.737530 > 421.875000 -49.870571 > 437.500000 -50.445431 > 453.125000 -50.647591 > 468.750000 -49.450035 > 484.375000 -48.767914 > 500.000000 -48.281902 > 515.625000 -49.419975 > 531.250000 -52.635048 > 546.875000 -52.856857 > 562.500000 -51.675240 > 578.125000 -50.667599 > 593.750000 -51.584862 > 609.375000 -48.992435 > 625.000000 -48.492191 > 640.625000 -49.701382 > 656.250000 -53.005512 > 671.875000 -54.269699 > 687.500000 -54.937397 > 703.125000 -53.922794 > 718.750000 -51.949009 > 734.375000 -52.763660 > 750.000000 -53.731155 > 765.625000 -54.977795 > 781.250000 -54.029835 > 796.875000 -54.375313 > 812.500000 -54.419693 > 828.125000 -54.796967 > 843.750000 -55.052872 > 859.375000 -53.625359 > 875.000000 -54.649513 > 890.625000 -56.736855 > 906.250000 -57.202946 > 921.875000 -56.587177 > 937.500000 -57.263260 > 953.125000 -56.744713 > 968.750000 -57.710808 > 984.375000 -58.118694 > 1000.000000 -60.595867 > 1015.625000 -60.736752 > 1031.250000 -57.966949 > 1046.875000 -58.101311 > 1062.500000 -59.093056 > 1078.125000 -59.889881 > 1093.750000 -59.972092 > 1109.375000 -59.728626 > 1125.000000 -58.060120 > 1140.625000 -56.506371 > 1156.250000 -55.109329 > 1171.875000 -54.278183 > 1187.500000 -56.115913 > 1203.125000 -57.535069 > 1218.750000 -58.304485 > 1234.375000 -58.614723 > 1250.000000 -58.873520 > 1265.625000 -61.268688 > 1281.250000 -62.702538 > 1296.875000 -63.828163 > 1312.500000 -63.712082 > 1328.125000 -65.020050 > 1343.750000 -64.481941 > 1359.375000 -62.652424 > 1375.000000 -61.640785 > 1390.625000 -61.719952 > 1406.250000 -61.451962 > 1421.875000 -58.490433 > 1437.500000 -57.193993 > 1453.125000 -56.973965 > 1468.750000 -57.940182 > 1484.375000 -59.935654 > 1500.000000 -61.829227 > 1515.625000 -63.259266 > 1531.250000 -62.169945 > 1546.875000 -61.958508 > 1562.500000 -62.018082 > 1578.125000 -60.562317 > 1593.750000 -59.146374 > 1609.375000 -58.868763 > 1625.000000 -60.028801 > 1640.625000 -59.241119 > 1656.250000 -60.101322 > 1671.875000 -60.429554 > 1687.500000 -60.538834 > 1703.125000 -61.073635 > 1718.750000 -60.375114 > 1734.375000 -60.152672 > 1750.000000 -59.529144 > 1765.625000 -59.783649 > 1781.250000 -61.850414 > 1796.875000 -63.506111 > 1812.500000 -63.869320 > 1828.125000 -62.442127 > 1843.750000 -62.519482 > 1859.375000 -63.814472 > 1875.000000 -63.300858 > 1890.625000 -61.998482 > 1906.250000 -62.199738 > 1921.875000 -61.745621 > 1937.500000 -61.611210 > 1953.125000 -61.174740 > 1968.750000 -60.504318 > 1984.375000 -60.277737 > 2000.000000 -61.696060 > 2015.625000 -61.693054 > 2031.250000 -62.755184 > 2046.875000 -63.983723 > 2062.500000 -63.584675 > 2078.125000 -62.943420 > 2093.750000 -63.238541 > 2109.375000 -62.929970 > 2125.000000 -62.243599 > 2140.625000 -60.967789 > 2156.250000 -60.915421 > 2171.875000 -61.997654 > 2187.500000 -63.702518 > 2203.125000 -64.316109 > 2218.750000 -63.577995 > 2234.375000 -63.250530 > 2250.000000 -63.107010 > 2265.625000 -62.953743 > 2281.250000 -62.933647 > 2296.875000 -61.956242 > 2312.500000 -62.948036 > 2328.125000 -64.375603 > 2343.750000 -64.296860 > 2359.375000 -63.552139 > 2375.000000 -62.556362 > 2390.625000 -62.979954 > 2406.250000 -64.505875 > 2421.875000 -65.626236 > 2437.500000 -65.732819 > 2453.125000 -66.130798 > 2468.750000 -65.920502 > 2484.375000 -64.183731 > 2500.000000 -63.387459 > 2515.625000 -63.110027 > 2531.250000 -64.085106 > 2546.875000 -64.269905 > 2562.500000 -64.181808 > 2578.125000 -64.600060 > 2593.750000 -63.998692 > 2609.375000 -63.854473 > 2625.000000 -65.015961 > 2640.625000 -65.751480 > 2656.250000 -66.293800 > 2671.875000 -66.494102 > 2687.500000 -66.300240 > 2703.125000 -66.383118 > 2718.750000 -66.466385 > 2734.375000 -65.733604 > 2750.000000 -65.110283 > 2765.625000 -65.537567 > 2781.250000 -66.125465 > 2796.875000 -65.979088 > 2812.500000 -64.833984 > 2828.125000 -63.773678 > 2843.750000 -64.419113 > 2859.375000 -64.800369 > 2875.000000 -64.710480 > 2890.625000 -64.088387 > 2906.250000 -64.790306 > 2921.875000 -65.160469 > 2937.500000 -65.285408 > 2953.125000 -66.030342 > 2968.750000 -65.027481 > 2984.375000 -64.623558 > 3000.000000 -65.082748 > 3015.625000 -63.680820 > 3031.250000 -62.836716 > 3046.875000 -62.210663 > 3062.500000 -61.578278 > 3078.125000 -62.397720 > 3093.750000 -63.185940 > 3109.375000 -62.439983 > 3125.000000 -62.382778 > 3140.625000 -63.123928 > 3156.250000 -64.276588 > 3171.875000 -65.444725 > 3187.500000 -65.891289 > 3203.125000 -65.480240 > 3218.750000 -64.761063 > 3234.375000 -65.140015 > 3250.000000 -66.010643 > 3265.625000 -66.964401 > 3281.250000 -67.296051 > 3296.875000 -66.430000 > 3312.500000 -66.564758 > 3328.125000 -67.878830 > 3343.750000 -67.748436 > 3359.375000 -68.965981 > 3375.000000 -70.426888 > 3390.625000 -71.400375 > 3406.250000 -72.067627 > 3421.875000 -71.944176 > 3437.500000 -72.285637 > 3453.125000 -71.983047 > 3468.750000 -72.565109 > 3484.375000 -72.350845 > 3500.000000 -72.335533 > 3515.625000 -72.608849 > 3531.250000 -72.417786 > 3546.875000 -73.100441 > 3562.500000 -73.461548 > 3578.125000 -73.558250 > 3593.750000 -73.218422 > 3609.375000 -73.994888 > 3625.000000 -74.379204 > 3640.625000 -74.896202 > 3656.250000 -74.944405 > 3671.875000 -74.958710 > 3687.500000 -75.029655 > 3703.125000 -74.314133 > 3718.750000 -74.855209 > 3734.375000 -76.021591 > 3750.000000 -76.441444 > 3765.625000 -77.122787 > 3781.250000 -77.733589 > 3796.875000 -79.138847 > 3812.500000 -80.206360 > 3828.125000 -80.391960 > 3843.750000 -81.214249 > 3859.375000 -81.434105 > 3875.000000 -82.234749 > 3890.625000 -82.529884 > 3906.250000 -82.929169 > 3921.875000 -83.735237 > 3937.500000 -84.770660 > 3953.125000 -84.203438 > 3968.750000 -84.455025 > 3984.375000 -85.116302 > > > 2012-03-27 09:50:40.412101 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=320/320/2000 seq=38825 lw=28160 > 2012-03-27 09:50:40.433101 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=480/480/2000 seq=38826 lw=28320 > 2012-03-27 09:50:40.454100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=640/640/2000 seq=38827 lw=28480 > 2012-03-27 09:50:40.475100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=800/800/2000 seq=38828 lw=28640 > 2012-03-27 09:50:40.496100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=960/960/2000 seq=38829 lw=28800 > 2012-03-27 09:50:40.517100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1120/1120/2000 seq=38830 lw=28960 > 2012-03-27 09:50:40.538100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1280/1280/2000 seq=38831 lw=29120 > 2012-03-27 09:50:40.559099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1440/1440/2000 seq=38832 lw=29280 > 2012-03-27 09:50:40.580099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1600/1600/2000 seq=38833 lw=29440 > 2012-03-27 09:50:40.601099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1760/1760/2000 seq=38834 lw=29600 > 2012-03-27 09:50:40.622098 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1920/1920/2000 seq=38835 lw=29760 > 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38836 lw=29760 > 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38837 lw=29760 > 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38838 lw=29760 > > > Any ideas? > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: 26 March 2012 18:39 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls > > > On Mon, Mar 26, 2012 at 3:01 AM, Daniel Knaggs > wrote: > OK, call recording has been setup ? waiting for it to happen now. > > > Interestingly, before I issue the ?record_session? application (and of course the ?RECORD_*? variables) I had to execute ?ring_ready? then ?pre_answer? otherwise the caller gets silence (changing the order of those two commands results in silence). > > I find it odd that just doing a pre_answer wouldn't be sufficient. A pre_answer will send a 183 w/SDP whereas ring_ready simply sends a 180. In any case, I'm glad you got your recordings. I also find it curious that only the "letter" DTMFs are being detected. Let us know if you actually hear those tones in the audio stream. > > -MC > > > > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From Daniel.Knaggs at realitysolutions.co.uk Fri Mar 30 15:25:30 2012 From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs) Date: Fri, 30 Mar 2012 11:25:30 +0000 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: <32544.1333105845@ccs.covici.com> References: <32544.1333105845@ccs.covici.com> Message-ID: Afraid not, as per my first post we're using ISDN. Daniel Knaggs Software Developer Reality Solutions Ltd 1 Global Business Park Hamburg Road Kingston upon Hull East Yorkshire, HU7 0AE Tel: 01482 373104 Mobile: 07932 408313 Email: mailto:Daniel.Knaggs at realitysolutions.co.uk http:// -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com Sent: 30 March 2012 12:11 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls Maybe its generated by an ata box if you have an analog phone? Daniel Knaggs wrote: > Any ideas, anyone? > > > [cid:imagefb5387.PNG at ca5d49be.4fa92b00] > > Daniel Knaggs > > Software Developer > > > Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE > Tel: 01482 828000 / Fax: 01482 373100 > Daniel.Knaggs at realitysolutions.co.uk > www.realitysolutions.co.uk > ________________________________ > > Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire > > > [cid:image19a0f2.PNG at 39ea781d.48900adc] > > > From: Daniel Knaggs > Sent: 27 March 2012 10:22 > To: 'FreeSWITCH Users Help' > Subject: RE: [Freeswitch-users] Strange DTMF Tones On Inbound Calls > > OK, it does NOT appear in the recordings. > > > Here is a spectrum analysis of the call when it happened: - > > Frequency (Hz) Level (dB) > 15.625000 -59.445583 > 31.250000 -69.136353 > 46.875000 -71.569084 > 62.500000 -72.445877 > 78.125000 -67.142319 > 93.750000 -63.964447 > 109.375000 -61.157677 > 125.000000 -58.849174 > 140.625000 -62.083492 > 156.250000 -51.071819 > 171.875000 -48.957565 > 187.500000 -48.156109 > 203.125000 -48.908588 > 218.750000 -47.891460 > 234.375000 -46.257698 > 250.000000 -43.011139 > 265.625000 -48.706509 > 281.250000 -53.241238 > 296.875000 -52.513802 > 312.500000 -50.387054 > 328.125000 -53.275459 > 343.750000 -56.096489 > 359.375000 -54.788857 > 375.000000 -52.283993 > 390.625000 -50.975601 > 406.250000 -49.737530 > 421.875000 -49.870571 > 437.500000 -50.445431 > 453.125000 -50.647591 > 468.750000 -49.450035 > 484.375000 -48.767914 > 500.000000 -48.281902 > 515.625000 -49.419975 > 531.250000 -52.635048 > 546.875000 -52.856857 > 562.500000 -51.675240 > 578.125000 -50.667599 > 593.750000 -51.584862 > 609.375000 -48.992435 > 625.000000 -48.492191 > 640.625000 -49.701382 > 656.250000 -53.005512 > 671.875000 -54.269699 > 687.500000 -54.937397 > 703.125000 -53.922794 > 718.750000 -51.949009 > 734.375000 -52.763660 > 750.000000 -53.731155 > 765.625000 -54.977795 > 781.250000 -54.029835 > 796.875000 -54.375313 > 812.500000 -54.419693 > 828.125000 -54.796967 > 843.750000 -55.052872 > 859.375000 -53.625359 > 875.000000 -54.649513 > 890.625000 -56.736855 > 906.250000 -57.202946 > 921.875000 -56.587177 > 937.500000 -57.263260 > 953.125000 -56.744713 > 968.750000 -57.710808 > 984.375000 -58.118694 > 1000.000000 -60.595867 > 1015.625000 -60.736752 > 1031.250000 -57.966949 > 1046.875000 -58.101311 > 1062.500000 -59.093056 > 1078.125000 -59.889881 > 1093.750000 -59.972092 > 1109.375000 -59.728626 > 1125.000000 -58.060120 > 1140.625000 -56.506371 > 1156.250000 -55.109329 > 1171.875000 -54.278183 > 1187.500000 -56.115913 > 1203.125000 -57.535069 > 1218.750000 -58.304485 > 1234.375000 -58.614723 > 1250.000000 -58.873520 > 1265.625000 -61.268688 > 1281.250000 -62.702538 > 1296.875000 -63.828163 > 1312.500000 -63.712082 > 1328.125000 -65.020050 > 1343.750000 -64.481941 > 1359.375000 -62.652424 > 1375.000000 -61.640785 > 1390.625000 -61.719952 > 1406.250000 -61.451962 > 1421.875000 -58.490433 > 1437.500000 -57.193993 > 1453.125000 -56.973965 > 1468.750000 -57.940182 > 1484.375000 -59.935654 > 1500.000000 -61.829227 > 1515.625000 -63.259266 > 1531.250000 -62.169945 > 1546.875000 -61.958508 > 1562.500000 -62.018082 > 1578.125000 -60.562317 > 1593.750000 -59.146374 > 1609.375000 -58.868763 > 1625.000000 -60.028801 > 1640.625000 -59.241119 > 1656.250000 -60.101322 > 1671.875000 -60.429554 > 1687.500000 -60.538834 > 1703.125000 -61.073635 > 1718.750000 -60.375114 > 1734.375000 -60.152672 > 1750.000000 -59.529144 > 1765.625000 -59.783649 > 1781.250000 -61.850414 > 1796.875000 -63.506111 > 1812.500000 -63.869320 > 1828.125000 -62.442127 > 1843.750000 -62.519482 > 1859.375000 -63.814472 > 1875.000000 -63.300858 > 1890.625000 -61.998482 > 1906.250000 -62.199738 > 1921.875000 -61.745621 > 1937.500000 -61.611210 > 1953.125000 -61.174740 > 1968.750000 -60.504318 > 1984.375000 -60.277737 > 2000.000000 -61.696060 > 2015.625000 -61.693054 > 2031.250000 -62.755184 > 2046.875000 -63.983723 > 2062.500000 -63.584675 > 2078.125000 -62.943420 > 2093.750000 -63.238541 > 2109.375000 -62.929970 > 2125.000000 -62.243599 > 2140.625000 -60.967789 > 2156.250000 -60.915421 > 2171.875000 -61.997654 > 2187.500000 -63.702518 > 2203.125000 -64.316109 > 2218.750000 -63.577995 > 2234.375000 -63.250530 > 2250.000000 -63.107010 > 2265.625000 -62.953743 > 2281.250000 -62.933647 > 2296.875000 -61.956242 > 2312.500000 -62.948036 > 2328.125000 -64.375603 > 2343.750000 -64.296860 > 2359.375000 -63.552139 > 2375.000000 -62.556362 > 2390.625000 -62.979954 > 2406.250000 -64.505875 > 2421.875000 -65.626236 > 2437.500000 -65.732819 > 2453.125000 -66.130798 > 2468.750000 -65.920502 > 2484.375000 -64.183731 > 2500.000000 -63.387459 > 2515.625000 -63.110027 > 2531.250000 -64.085106 > 2546.875000 -64.269905 > 2562.500000 -64.181808 > 2578.125000 -64.600060 > 2593.750000 -63.998692 > 2609.375000 -63.854473 > 2625.000000 -65.015961 > 2640.625000 -65.751480 > 2656.250000 -66.293800 > 2671.875000 -66.494102 > 2687.500000 -66.300240 > 2703.125000 -66.383118 > 2718.750000 -66.466385 > 2734.375000 -65.733604 > 2750.000000 -65.110283 > 2765.625000 -65.537567 > 2781.250000 -66.125465 > 2796.875000 -65.979088 > 2812.500000 -64.833984 > 2828.125000 -63.773678 > 2843.750000 -64.419113 > 2859.375000 -64.800369 > 2875.000000 -64.710480 > 2890.625000 -64.088387 > 2906.250000 -64.790306 > 2921.875000 -65.160469 > 2937.500000 -65.285408 > 2953.125000 -66.030342 > 2968.750000 -65.027481 > 2984.375000 -64.623558 > 3000.000000 -65.082748 > 3015.625000 -63.680820 > 3031.250000 -62.836716 > 3046.875000 -62.210663 > 3062.500000 -61.578278 > 3078.125000 -62.397720 > 3093.750000 -63.185940 > 3109.375000 -62.439983 > 3125.000000 -62.382778 > 3140.625000 -63.123928 > 3156.250000 -64.276588 > 3171.875000 -65.444725 > 3187.500000 -65.891289 > 3203.125000 -65.480240 > 3218.750000 -64.761063 > 3234.375000 -65.140015 > 3250.000000 -66.010643 > 3265.625000 -66.964401 > 3281.250000 -67.296051 > 3296.875000 -66.430000 > 3312.500000 -66.564758 > 3328.125000 -67.878830 > 3343.750000 -67.748436 > 3359.375000 -68.965981 > 3375.000000 -70.426888 > 3390.625000 -71.400375 > 3406.250000 -72.067627 > 3421.875000 -71.944176 > 3437.500000 -72.285637 > 3453.125000 -71.983047 > 3468.750000 -72.565109 > 3484.375000 -72.350845 > 3500.000000 -72.335533 > 3515.625000 -72.608849 > 3531.250000 -72.417786 > 3546.875000 -73.100441 > 3562.500000 -73.461548 > 3578.125000 -73.558250 > 3593.750000 -73.218422 > 3609.375000 -73.994888 > 3625.000000 -74.379204 > 3640.625000 -74.896202 > 3656.250000 -74.944405 > 3671.875000 -74.958710 > 3687.500000 -75.029655 > 3703.125000 -74.314133 > 3718.750000 -74.855209 > 3734.375000 -76.021591 > 3750.000000 -76.441444 > 3765.625000 -77.122787 > 3781.250000 -77.733589 > 3796.875000 -79.138847 > 3812.500000 -80.206360 > 3828.125000 -80.391960 > 3843.750000 -81.214249 > 3859.375000 -81.434105 > 3875.000000 -82.234749 > 3890.625000 -82.529884 > 3906.250000 -82.929169 > 3921.875000 -83.735237 > 3937.500000 -84.770660 > 3953.125000 -84.203438 > 3968.750000 -84.455025 > 3984.375000 -85.116302 > > > 2012-03-27 09:50:40.412101 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=320/320/2000 seq=38825 lw=28160 > 2012-03-27 09:50:40.433101 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=480/480/2000 seq=38826 lw=28320 > 2012-03-27 09:50:40.454100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=640/640/2000 seq=38827 lw=28480 > 2012-03-27 09:50:40.475100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=800/800/2000 seq=38828 lw=28640 > 2012-03-27 09:50:40.496100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=960/960/2000 seq=38829 lw=28800 > 2012-03-27 09:50:40.517100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1120/1120/2000 seq=38830 lw=28960 > 2012-03-27 09:50:40.538100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1280/1280/2000 seq=38831 lw=29120 > 2012-03-27 09:50:40.559099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1440/1440/2000 seq=38832 lw=29280 > 2012-03-27 09:50:40.580099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1600/1600/2000 seq=38833 lw=29440 > 2012-03-27 09:50:40.601099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1760/1760/2000 seq=38834 lw=29600 > 2012-03-27 09:50:40.622098 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1920/1920/2000 seq=38835 lw=29760 > 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38836 lw=29760 > 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38837 lw=29760 > 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38838 lw=29760 > > > Any ideas? > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: 26 March 2012 18:39 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls > > > On Mon, Mar 26, 2012 at 3:01 AM, Daniel Knaggs > wrote: > OK, call recording has been setup ? waiting for it to happen now. > > > Interestingly, before I issue the ?record_session? application (and of course the ?RECORD_*? variables) I had to execute ?ring_ready? then ?pre_answer? otherwise the caller gets silence (changing the order of those two commands results in silence). > > I find it odd that just doing a pre_answer wouldn't be sufficient. A pre_answer will send a 183 w/SDP whereas ring_ready simply sends a 180. In any case, I'm glad you got your recordings. I also find it curious that only the "letter" DTMFs are being detected. Let us know if you actually hear those tones in the audio stream. > > -MC > > > > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From covici at ccs.covici.com Fri Mar 30 15:58:45 2012 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Fri, 30 Mar 2012 07:58:45 -0400 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: References: <32544.1333105845@ccs.covici.com> Message-ID: <6364.1333108725@ccs.covici.com> but how are your phones connected to fs? Is it through any kind of equipment? Daniel Knaggs wrote: > Afraid not, as per my first post we're using ISDN. > > > Daniel Knaggs > Software Developer > > Reality Solutions Ltd > 1 Global Business Park > Hamburg Road > Kingston upon Hull > East Yorkshire, HU7 0AE > > Tel: 01482 373104 > Mobile: 07932 408313 > Email: mailto:Daniel.Knaggs at realitysolutions.co.uk > http:// > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com > Sent: 30 March 2012 12:11 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls > > Maybe its generated by an ata box if you have an analog phone? > > Daniel Knaggs wrote: > > > Any ideas, anyone? > > > > > > [cid:imagefb5387.PNG at ca5d49be.4fa92b00] > > > > Daniel Knaggs > > > > Software Developer > > > > > > Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE > > Tel: 01482 828000 / Fax: 01482 373100 > > Daniel.Knaggs at realitysolutions.co.uk > > www.realitysolutions.co.uk > > ________________________________ > > > > Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire > > > > > > [cid:image19a0f2.PNG at 39ea781d.48900adc] > > > > > > From: Daniel Knaggs > > Sent: 27 March 2012 10:22 > > To: 'FreeSWITCH Users Help' > > Subject: RE: [Freeswitch-users] Strange DTMF Tones On Inbound Calls > > > > OK, it does NOT appear in the recordings. > > > > > > Here is a spectrum analysis of the call when it happened: - > > > > Frequency (Hz) Level (dB) > > 15.625000 -59.445583 > > 31.250000 -69.136353 > > 46.875000 -71.569084 > > 62.500000 -72.445877 > > 78.125000 -67.142319 > > 93.750000 -63.964447 > > 109.375000 -61.157677 > > 125.000000 -58.849174 > > 140.625000 -62.083492 > > 156.250000 -51.071819 > > 171.875000 -48.957565 > > 187.500000 -48.156109 > > 203.125000 -48.908588 > > 218.750000 -47.891460 > > 234.375000 -46.257698 > > 250.000000 -43.011139 > > 265.625000 -48.706509 > > 281.250000 -53.241238 > > 296.875000 -52.513802 > > 312.500000 -50.387054 > > 328.125000 -53.275459 > > 343.750000 -56.096489 > > 359.375000 -54.788857 > > 375.000000 -52.283993 > > 390.625000 -50.975601 > > 406.250000 -49.737530 > > 421.875000 -49.870571 > > 437.500000 -50.445431 > > 453.125000 -50.647591 > > 468.750000 -49.450035 > > 484.375000 -48.767914 > > 500.000000 -48.281902 > > 515.625000 -49.419975 > > 531.250000 -52.635048 > > 546.875000 -52.856857 > > 562.500000 -51.675240 > > 578.125000 -50.667599 > > 593.750000 -51.584862 > > 609.375000 -48.992435 > > 625.000000 -48.492191 > > 640.625000 -49.701382 > > 656.250000 -53.005512 > > 671.875000 -54.269699 > > 687.500000 -54.937397 > > 703.125000 -53.922794 > > 718.750000 -51.949009 > > 734.375000 -52.763660 > > 750.000000 -53.731155 > > 765.625000 -54.977795 > > 781.250000 -54.029835 > > 796.875000 -54.375313 > > 812.500000 -54.419693 > > 828.125000 -54.796967 > > 843.750000 -55.052872 > > 859.375000 -53.625359 > > 875.000000 -54.649513 > > 890.625000 -56.736855 > > 906.250000 -57.202946 > > 921.875000 -56.587177 > > 937.500000 -57.263260 > > 953.125000 -56.744713 > > 968.750000 -57.710808 > > 984.375000 -58.118694 > > 1000.000000 -60.595867 > > 1015.625000 -60.736752 > > 1031.250000 -57.966949 > > 1046.875000 -58.101311 > > 1062.500000 -59.093056 > > 1078.125000 -59.889881 > > 1093.750000 -59.972092 > > 1109.375000 -59.728626 > > 1125.000000 -58.060120 > > 1140.625000 -56.506371 > > 1156.250000 -55.109329 > > 1171.875000 -54.278183 > > 1187.500000 -56.115913 > > 1203.125000 -57.535069 > > 1218.750000 -58.304485 > > 1234.375000 -58.614723 > > 1250.000000 -58.873520 > > 1265.625000 -61.268688 > > 1281.250000 -62.702538 > > 1296.875000 -63.828163 > > 1312.500000 -63.712082 > > 1328.125000 -65.020050 > > 1343.750000 -64.481941 > > 1359.375000 -62.652424 > > 1375.000000 -61.640785 > > 1390.625000 -61.719952 > > 1406.250000 -61.451962 > > 1421.875000 -58.490433 > > 1437.500000 -57.193993 > > 1453.125000 -56.973965 > > 1468.750000 -57.940182 > > 1484.375000 -59.935654 > > 1500.000000 -61.829227 > > 1515.625000 -63.259266 > > 1531.250000 -62.169945 > > 1546.875000 -61.958508 > > 1562.500000 -62.018082 > > 1578.125000 -60.562317 > > 1593.750000 -59.146374 > > 1609.375000 -58.868763 > > 1625.000000 -60.028801 > > 1640.625000 -59.241119 > > 1656.250000 -60.101322 > > 1671.875000 -60.429554 > > 1687.500000 -60.538834 > > 1703.125000 -61.073635 > > 1718.750000 -60.375114 > > 1734.375000 -60.152672 > > 1750.000000 -59.529144 > > 1765.625000 -59.783649 > > 1781.250000 -61.850414 > > 1796.875000 -63.506111 > > 1812.500000 -63.869320 > > 1828.125000 -62.442127 > > 1843.750000 -62.519482 > > 1859.375000 -63.814472 > > 1875.000000 -63.300858 > > 1890.625000 -61.998482 > > 1906.250000 -62.199738 > > 1921.875000 -61.745621 > > 1937.500000 -61.611210 > > 1953.125000 -61.174740 > > 1968.750000 -60.504318 > > 1984.375000 -60.277737 > > 2000.000000 -61.696060 > > 2015.625000 -61.693054 > > 2031.250000 -62.755184 > > 2046.875000 -63.983723 > > 2062.500000 -63.584675 > > 2078.125000 -62.943420 > > 2093.750000 -63.238541 > > 2109.375000 -62.929970 > > 2125.000000 -62.243599 > > 2140.625000 -60.967789 > > 2156.250000 -60.915421 > > 2171.875000 -61.997654 > > 2187.500000 -63.702518 > > 2203.125000 -64.316109 > > 2218.750000 -63.577995 > > 2234.375000 -63.250530 > > 2250.000000 -63.107010 > > 2265.625000 -62.953743 > > 2281.250000 -62.933647 > > 2296.875000 -61.956242 > > 2312.500000 -62.948036 > > 2328.125000 -64.375603 > > 2343.750000 -64.296860 > > 2359.375000 -63.552139 > > 2375.000000 -62.556362 > > 2390.625000 -62.979954 > > 2406.250000 -64.505875 > > 2421.875000 -65.626236 > > 2437.500000 -65.732819 > > 2453.125000 -66.130798 > > 2468.750000 -65.920502 > > 2484.375000 -64.183731 > > 2500.000000 -63.387459 > > 2515.625000 -63.110027 > > 2531.250000 -64.085106 > > 2546.875000 -64.269905 > > 2562.500000 -64.181808 > > 2578.125000 -64.600060 > > 2593.750000 -63.998692 > > 2609.375000 -63.854473 > > 2625.000000 -65.015961 > > 2640.625000 -65.751480 > > 2656.250000 -66.293800 > > 2671.875000 -66.494102 > > 2687.500000 -66.300240 > > 2703.125000 -66.383118 > > 2718.750000 -66.466385 > > 2734.375000 -65.733604 > > 2750.000000 -65.110283 > > 2765.625000 -65.537567 > > 2781.250000 -66.125465 > > 2796.875000 -65.979088 > > 2812.500000 -64.833984 > > 2828.125000 -63.773678 > > 2843.750000 -64.419113 > > 2859.375000 -64.800369 > > 2875.000000 -64.710480 > > 2890.625000 -64.088387 > > 2906.250000 -64.790306 > > 2921.875000 -65.160469 > > 2937.500000 -65.285408 > > 2953.125000 -66.030342 > > 2968.750000 -65.027481 > > 2984.375000 -64.623558 > > 3000.000000 -65.082748 > > 3015.625000 -63.680820 > > 3031.250000 -62.836716 > > 3046.875000 -62.210663 > > 3062.500000 -61.578278 > > 3078.125000 -62.397720 > > 3093.750000 -63.185940 > > 3109.375000 -62.439983 > > 3125.000000 -62.382778 > > 3140.625000 -63.123928 > > 3156.250000 -64.276588 > > 3171.875000 -65.444725 > > 3187.500000 -65.891289 > > 3203.125000 -65.480240 > > 3218.750000 -64.761063 > > 3234.375000 -65.140015 > > 3250.000000 -66.010643 > > 3265.625000 -66.964401 > > 3281.250000 -67.296051 > > 3296.875000 -66.430000 > > 3312.500000 -66.564758 > > 3328.125000 -67.878830 > > 3343.750000 -67.748436 > > 3359.375000 -68.965981 > > 3375.000000 -70.426888 > > 3390.625000 -71.400375 > > 3406.250000 -72.067627 > > 3421.875000 -71.944176 > > 3437.500000 -72.285637 > > 3453.125000 -71.983047 > > 3468.750000 -72.565109 > > 3484.375000 -72.350845 > > 3500.000000 -72.335533 > > 3515.625000 -72.608849 > > 3531.250000 -72.417786 > > 3546.875000 -73.100441 > > 3562.500000 -73.461548 > > 3578.125000 -73.558250 > > 3593.750000 -73.218422 > > 3609.375000 -73.994888 > > 3625.000000 -74.379204 > > 3640.625000 -74.896202 > > 3656.250000 -74.944405 > > 3671.875000 -74.958710 > > 3687.500000 -75.029655 > > 3703.125000 -74.314133 > > 3718.750000 -74.855209 > > 3734.375000 -76.021591 > > 3750.000000 -76.441444 > > 3765.625000 -77.122787 > > 3781.250000 -77.733589 > > 3796.875000 -79.138847 > > 3812.500000 -80.206360 > > 3828.125000 -80.391960 > > 3843.750000 -81.214249 > > 3859.375000 -81.434105 > > 3875.000000 -82.234749 > > 3890.625000 -82.529884 > > 3906.250000 -82.929169 > > 3921.875000 -83.735237 > > 3937.500000 -84.770660 > > 3953.125000 -84.203438 > > 3968.750000 -84.455025 > > 3984.375000 -85.116302 > > > > > > 2012-03-27 09:50:40.412101 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=320/320/2000 seq=38825 lw=28160 > > 2012-03-27 09:50:40.433101 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=480/480/2000 seq=38826 lw=28320 > > 2012-03-27 09:50:40.454100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=640/640/2000 seq=38827 lw=28480 > > 2012-03-27 09:50:40.475100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=800/800/2000 seq=38828 lw=28640 > > 2012-03-27 09:50:40.496100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=960/960/2000 seq=38829 lw=28800 > > 2012-03-27 09:50:40.517100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1120/1120/2000 seq=38830 lw=28960 > > 2012-03-27 09:50:40.538100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1280/1280/2000 seq=38831 lw=29120 > > 2012-03-27 09:50:40.559099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1440/1440/2000 seq=38832 lw=29280 > > 2012-03-27 09:50:40.580099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1600/1600/2000 seq=38833 lw=29440 > > 2012-03-27 09:50:40.601099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1760/1760/2000 seq=38834 lw=29600 > > 2012-03-27 09:50:40.622098 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1920/1920/2000 seq=38835 lw=29760 > > 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38836 lw=29760 > > 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38837 lw=29760 > > 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38838 lw=29760 > > > > > > Any ideas? > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > > Sent: 26 March 2012 18:39 > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls > > > > > > On Mon, Mar 26, 2012 at 3:01 AM, Daniel Knaggs > wrote: > > OK, call recording has been setup ? waiting for it to happen now. > > > > > > Interestingly, before I issue the ?record_session? application (and of course the ?RECORD_*? variables) I had to execute ?ring_ready? then ?pre_answer? otherwise the caller gets silence (changing the order of those two commands results in silence). > > > > I find it odd that just doing a pre_answer wouldn't be sufficient. A pre_answer will send a 183 w/SDP whereas ring_ready simply sends a 180. In any case, I'm glad you got your recordings. I also find it curious that only the "letter" DTMFs are being detected. Let us know if you actually hear those tones in the audio stream. > > > > -MC > > > > > > > > > > ---------------------------------------------------- > > Alternatives: > > > > ---------------------------------------------------- > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From Daniel.Knaggs at realitysolutions.co.uk Fri Mar 30 16:20:22 2012 From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs) Date: Fri, 30 Mar 2012 12:20:22 +0000 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: <6364.1333108725@ccs.covici.com> References: <32544.1333105845@ccs.covici.com> <6364.1333108725@ccs.covici.com> Message-ID: ISDN -> TE121 -> FS -> PoE -> VoIP phone (LinkSys SPA942/962). Log states its coming IN from the card, not generated by the phones here. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com Sent: 30 March 2012 12:59 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls but how are your phones connected to fs? Is it through any kind of equipment? Daniel Knaggs wrote: > Afraid not, as per my first post we're using ISDN. > > > Daniel Knaggs > Software Developer > > Reality Solutions Ltd > 1 Global Business Park > Hamburg Road > Kingston upon Hull > East Yorkshire, HU7 0AE > > Tel: 01482 373104 > Mobile: 07932 408313 > Email: mailto:Daniel.Knaggs at realitysolutions.co.uk > http:// > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com > Sent: 30 March 2012 12:11 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls > > Maybe its generated by an ata box if you have an analog phone? > > Daniel Knaggs wrote: > > > Any ideas, anyone? > > > > > > [cid:imagefb5387.PNG at ca5d49be.4fa92b00] > > > > Daniel Knaggs > > > > Software Developer > > > > > > Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE > > Tel: 01482 828000 / Fax: 01482 373100 > > Daniel.Knaggs at realitysolutions.co.uk > > www.realitysolutions.co.uk > > ________________________________ > > > > Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire > > > > > > [cid:image19a0f2.PNG at 39ea781d.48900adc] > > > > > > From: Daniel Knaggs > > Sent: 27 March 2012 10:22 > > To: 'FreeSWITCH Users Help' > > Subject: RE: [Freeswitch-users] Strange DTMF Tones On Inbound Calls > > > > OK, it does NOT appear in the recordings. > > > > > > Here is a spectrum analysis of the call when it happened: - > > > > Frequency (Hz) Level (dB) > > 15.625000 -59.445583 > > 31.250000 -69.136353 > > 46.875000 -71.569084 > > 62.500000 -72.445877 > > 78.125000 -67.142319 > > 93.750000 -63.964447 > > 109.375000 -61.157677 > > 125.000000 -58.849174 > > 140.625000 -62.083492 > > 156.250000 -51.071819 > > 171.875000 -48.957565 > > 187.500000 -48.156109 > > 203.125000 -48.908588 > > 218.750000 -47.891460 > > 234.375000 -46.257698 > > 250.000000 -43.011139 > > 265.625000 -48.706509 > > 281.250000 -53.241238 > > 296.875000 -52.513802 > > 312.500000 -50.387054 > > 328.125000 -53.275459 > > 343.750000 -56.096489 > > 359.375000 -54.788857 > > 375.000000 -52.283993 > > 390.625000 -50.975601 > > 406.250000 -49.737530 > > 421.875000 -49.870571 > > 437.500000 -50.445431 > > 453.125000 -50.647591 > > 468.750000 -49.450035 > > 484.375000 -48.767914 > > 500.000000 -48.281902 > > 515.625000 -49.419975 > > 531.250000 -52.635048 > > 546.875000 -52.856857 > > 562.500000 -51.675240 > > 578.125000 -50.667599 > > 593.750000 -51.584862 > > 609.375000 -48.992435 > > 625.000000 -48.492191 > > 640.625000 -49.701382 > > 656.250000 -53.005512 > > 671.875000 -54.269699 > > 687.500000 -54.937397 > > 703.125000 -53.922794 > > 718.750000 -51.949009 > > 734.375000 -52.763660 > > 750.000000 -53.731155 > > 765.625000 -54.977795 > > 781.250000 -54.029835 > > 796.875000 -54.375313 > > 812.500000 -54.419693 > > 828.125000 -54.796967 > > 843.750000 -55.052872 > > 859.375000 -53.625359 > > 875.000000 -54.649513 > > 890.625000 -56.736855 > > 906.250000 -57.202946 > > 921.875000 -56.587177 > > 937.500000 -57.263260 > > 953.125000 -56.744713 > > 968.750000 -57.710808 > > 984.375000 -58.118694 > > 1000.000000 -60.595867 > > 1015.625000 -60.736752 > > 1031.250000 -57.966949 > > 1046.875000 -58.101311 > > 1062.500000 -59.093056 > > 1078.125000 -59.889881 > > 1093.750000 -59.972092 > > 1109.375000 -59.728626 > > 1125.000000 -58.060120 > > 1140.625000 -56.506371 > > 1156.250000 -55.109329 > > 1171.875000 -54.278183 > > 1187.500000 -56.115913 > > 1203.125000 -57.535069 > > 1218.750000 -58.304485 > > 1234.375000 -58.614723 > > 1250.000000 -58.873520 > > 1265.625000 -61.268688 > > 1281.250000 -62.702538 > > 1296.875000 -63.828163 > > 1312.500000 -63.712082 > > 1328.125000 -65.020050 > > 1343.750000 -64.481941 > > 1359.375000 -62.652424 > > 1375.000000 -61.640785 > > 1390.625000 -61.719952 > > 1406.250000 -61.451962 > > 1421.875000 -58.490433 > > 1437.500000 -57.193993 > > 1453.125000 -56.973965 > > 1468.750000 -57.940182 > > 1484.375000 -59.935654 > > 1500.000000 -61.829227 > > 1515.625000 -63.259266 > > 1531.250000 -62.169945 > > 1546.875000 -61.958508 > > 1562.500000 -62.018082 > > 1578.125000 -60.562317 > > 1593.750000 -59.146374 > > 1609.375000 -58.868763 > > 1625.000000 -60.028801 > > 1640.625000 -59.241119 > > 1656.250000 -60.101322 > > 1671.875000 -60.429554 > > 1687.500000 -60.538834 > > 1703.125000 -61.073635 > > 1718.750000 -60.375114 > > 1734.375000 -60.152672 > > 1750.000000 -59.529144 > > 1765.625000 -59.783649 > > 1781.250000 -61.850414 > > 1796.875000 -63.506111 > > 1812.500000 -63.869320 > > 1828.125000 -62.442127 > > 1843.750000 -62.519482 > > 1859.375000 -63.814472 > > 1875.000000 -63.300858 > > 1890.625000 -61.998482 > > 1906.250000 -62.199738 > > 1921.875000 -61.745621 > > 1937.500000 -61.611210 > > 1953.125000 -61.174740 > > 1968.750000 -60.504318 > > 1984.375000 -60.277737 > > 2000.000000 -61.696060 > > 2015.625000 -61.693054 > > 2031.250000 -62.755184 > > 2046.875000 -63.983723 > > 2062.500000 -63.584675 > > 2078.125000 -62.943420 > > 2093.750000 -63.238541 > > 2109.375000 -62.929970 > > 2125.000000 -62.243599 > > 2140.625000 -60.967789 > > 2156.250000 -60.915421 > > 2171.875000 -61.997654 > > 2187.500000 -63.702518 > > 2203.125000 -64.316109 > > 2218.750000 -63.577995 > > 2234.375000 -63.250530 > > 2250.000000 -63.107010 > > 2265.625000 -62.953743 > > 2281.250000 -62.933647 > > 2296.875000 -61.956242 > > 2312.500000 -62.948036 > > 2328.125000 -64.375603 > > 2343.750000 -64.296860 > > 2359.375000 -63.552139 > > 2375.000000 -62.556362 > > 2390.625000 -62.979954 > > 2406.250000 -64.505875 > > 2421.875000 -65.626236 > > 2437.500000 -65.732819 > > 2453.125000 -66.130798 > > 2468.750000 -65.920502 > > 2484.375000 -64.183731 > > 2500.000000 -63.387459 > > 2515.625000 -63.110027 > > 2531.250000 -64.085106 > > 2546.875000 -64.269905 > > 2562.500000 -64.181808 > > 2578.125000 -64.600060 > > 2593.750000 -63.998692 > > 2609.375000 -63.854473 > > 2625.000000 -65.015961 > > 2640.625000 -65.751480 > > 2656.250000 -66.293800 > > 2671.875000 -66.494102 > > 2687.500000 -66.300240 > > 2703.125000 -66.383118 > > 2718.750000 -66.466385 > > 2734.375000 -65.733604 > > 2750.000000 -65.110283 > > 2765.625000 -65.537567 > > 2781.250000 -66.125465 > > 2796.875000 -65.979088 > > 2812.500000 -64.833984 > > 2828.125000 -63.773678 > > 2843.750000 -64.419113 > > 2859.375000 -64.800369 > > 2875.000000 -64.710480 > > 2890.625000 -64.088387 > > 2906.250000 -64.790306 > > 2921.875000 -65.160469 > > 2937.500000 -65.285408 > > 2953.125000 -66.030342 > > 2968.750000 -65.027481 > > 2984.375000 -64.623558 > > 3000.000000 -65.082748 > > 3015.625000 -63.680820 > > 3031.250000 -62.836716 > > 3046.875000 -62.210663 > > 3062.500000 -61.578278 > > 3078.125000 -62.397720 > > 3093.750000 -63.185940 > > 3109.375000 -62.439983 > > 3125.000000 -62.382778 > > 3140.625000 -63.123928 > > 3156.250000 -64.276588 > > 3171.875000 -65.444725 > > 3187.500000 -65.891289 > > 3203.125000 -65.480240 > > 3218.750000 -64.761063 > > 3234.375000 -65.140015 > > 3250.000000 -66.010643 > > 3265.625000 -66.964401 > > 3281.250000 -67.296051 > > 3296.875000 -66.430000 > > 3312.500000 -66.564758 > > 3328.125000 -67.878830 > > 3343.750000 -67.748436 > > 3359.375000 -68.965981 > > 3375.000000 -70.426888 > > 3390.625000 -71.400375 > > 3406.250000 -72.067627 > > 3421.875000 -71.944176 > > 3437.500000 -72.285637 > > 3453.125000 -71.983047 > > 3468.750000 -72.565109 > > 3484.375000 -72.350845 > > 3500.000000 -72.335533 > > 3515.625000 -72.608849 > > 3531.250000 -72.417786 > > 3546.875000 -73.100441 > > 3562.500000 -73.461548 > > 3578.125000 -73.558250 > > 3593.750000 -73.218422 > > 3609.375000 -73.994888 > > 3625.000000 -74.379204 > > 3640.625000 -74.896202 > > 3656.250000 -74.944405 > > 3671.875000 -74.958710 > > 3687.500000 -75.029655 > > 3703.125000 -74.314133 > > 3718.750000 -74.855209 > > 3734.375000 -76.021591 > > 3750.000000 -76.441444 > > 3765.625000 -77.122787 > > 3781.250000 -77.733589 > > 3796.875000 -79.138847 > > 3812.500000 -80.206360 > > 3828.125000 -80.391960 > > 3843.750000 -81.214249 > > 3859.375000 -81.434105 > > 3875.000000 -82.234749 > > 3890.625000 -82.529884 > > 3906.250000 -82.929169 > > 3921.875000 -83.735237 > > 3937.500000 -84.770660 > > 3953.125000 -84.203438 > > 3968.750000 -84.455025 > > 3984.375000 -85.116302 > > > > > > 2012-03-27 09:50:40.412101 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=320/320/2000 seq=38825 lw=28160 > > 2012-03-27 09:50:40.433101 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=480/480/2000 seq=38826 lw=28320 > > 2012-03-27 09:50:40.454100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=640/640/2000 seq=38827 lw=28480 > > 2012-03-27 09:50:40.475100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=800/800/2000 seq=38828 lw=28640 > > 2012-03-27 09:50:40.496100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=960/960/2000 seq=38829 lw=28800 > > 2012-03-27 09:50:40.517100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1120/1120/2000 seq=38830 lw=28960 > > 2012-03-27 09:50:40.538100 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1280/1280/2000 seq=38831 lw=29120 > > 2012-03-27 09:50:40.559099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1440/1440/2000 seq=38832 lw=29280 > > 2012-03-27 09:50:40.580099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1600/1600/2000 seq=38833 lw=29440 > > 2012-03-27 09:50:40.601099 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1760/1760/2000 seq=38834 lw=29600 > > 2012-03-27 09:50:40.622098 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=28000 dur=1920/1920/2000 seq=38835 lw=29760 > > 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38836 lw=29760 > > 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38837 lw=29760 > > 2012-03-27 09:50:40.643098 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=28000 dur=2080/2080/2000 seq=38838 lw=29760 > > > > > > Any ideas? > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > > Sent: 26 March 2012 18:39 > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls > > > > > > On Mon, Mar 26, 2012 at 3:01 AM, Daniel Knaggs > wrote: > > OK, call recording has been setup ? waiting for it to happen now. > > > > > > Interestingly, before I issue the ?record_session? application (and of course the ?RECORD_*? variables) I had to execute ?ring_ready? then ?pre_answer? otherwise the caller gets silence (changing the order of those two commands results in silence). > > > > I find it odd that just doing a pre_answer wouldn't be sufficient. A pre_answer will send a 183 w/SDP whereas ring_ready simply sends a 180. In any case, I'm glad you got your recordings. I also find it curious that only the "letter" DTMFs are being detected. Let us know if you actually hear those tones in the audio stream. > > > > -MC > > > > > > > > > > ---------------------------------------------------- > > Alternatives: > > > > ---------------------------------------------------- > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sharad at coraltele.com Fri Mar 30 16:13:12 2012 From: sharad at coraltele.com (Sharad Garg) Date: Fri, 30 Mar 2012 17:43:12 +0530 Subject: [Freeswitch-users] Session-wise Language Setting References: <4CB4E6075A9F4F9BA1C8A57FF2A06735@sharad><4F746EFF.3050305@communicatefreely.net>, <00C2DD790B254A1BACD50CAA66B123B9@sharad> <1FFF97C269757C458224B7C895F35F15083A49@cantor.std.visionutv.se> Message-ID: <12BA6FAF92674DEE9650CF4829B95DF0@sharad> Thanks Peter...yes it works. Just a last query - Is there any API for setting the same so that it can be used from Javascript. Best Regards Sharad ----- Original Message ----- From: "Peter Olsson" To: "FreeSWITCH Users Help" Sent: Friday, March 30, 2012 12:16 PM Subject: Re: [Freeswitch-users] Session-wise Language Setting It's a channel variable, it's unique per channel, so there is no risk that a variable on channel A overwrites the value of the variable in channel B. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Sharad Garg [sharad at coraltele.com] Skickat: den 30 mars 2012 08:11 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Session-wise Language Setting Dear Tim, Thanks for your kind reply. Just another curisity....For a live call, if I set the default_language say English, now that call will be entertained in english. Now while first call is in progreess, second call comes & this caller opts for Russian...so we have to set default language as russian. In this case, all the voice prompts which are being played to first caller, will be in russian. Is'nt it ? Actually first call should continue in english & second call should be in russian & so on. Plz clarify. Regards Sharad ----- Original Message ----- From: "Tim St. Pierre" To: "FreeSWITCH Users Help" Sent: Thursday, March 29, 2012 7:47 PM Subject: Re: [Freeswitch-users] Session-wise Language Setting > Yes! > > On incoming calls, I set default_language to either en or fr (I'm in > Canada), depending on the DID. I can also change that variable in an IVR > if the caller wants to change the language. On outgoing calls, I set the > same variable in the directory, so that any call that a phone makes will > have a language set. This is great in a multi-lingual environment, as > each user can have a language preference that will be used for voice mail > and other system prompts. We store that in a database and also have the > provisioning system look to that same variable, so the screen labels and > text on the phone is the same language. > > -Tim > > Sharad Garg wrote: >> Hi All >> >> Just wondering whether we can define the language from beginning of the >> call means when a call is originated or landed to Freeswitch, can we >> define the language of all the prompts whether the call should be >> processed in English or Russian or so on.? >> >> Regards >> Sharad >> >> >> ----- Original Message ----- >> From: >> To: >> Sent: Thursday, March 29, 2012 7:33 AM >> Subject: FreeSWITCH-users Digest, Vol 69, Issue 282 >> >> >> >>> Send FreeSWITCH-users mailing list submissions to >>> freeswitch-users at lists.freeswitch.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> or, via email, send a message with subject or body 'help' to >>> freeswitch-users-request at lists.freeswitch.org >>> >>> You can reach the person managing the list at >>> freeswitch-users-owner at lists.freeswitch.org >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of FreeSWITCH-users digest..." >>> >>> >> >> >> -------------------------------------------------------------------------------- >> >> >> >>> Today's Topics: >>> >>> 1. Re: Open Bugs on Jira and Call for help (Ken Rice) >>> 2. mod_callcenter and moh-sound (Vik Killa) >>> 3. Re: mod_callcenter and moh-sound (Vik Killa) >>> 4. Registration VIA TCP (Rob Moore) >>> 5. Forward calls from opensips to freeswitch (Sherif Omran) >>> 6. Re: Registration VIA TCP (Mitch Capper) >>> >>> >> >> >> -------------------------------------------------------------------------------- >> >> >> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f75538432768881816776! From andrew at cassidywebservices.co.uk Fri Mar 30 17:18:36 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Fri, 30 Mar 2012 14:18:36 +0100 Subject: [Freeswitch-users] NAT issues - Outbound Call drops after~30seconds In-Reply-To: References: <1363342dbcd.1079533506949741633.-8314378552953017400@zoho.com> <4F6970A2.000019.15156@FLIGHTPC> Message-ID: We got it, the problem appears to have been that even though it was a 1:1 NAT the firewall was natting outbound sip traffic and the carrier was responding on that port. We just switched off the NAT option at the carrier and all seems to be working fine now. On 29 March 2012 21:03, Michael Collins wrote: > > > On Thu, Mar 29, 2012 at 8:14 AM, Andrew Cassidy < > andrew at cassidywebservices.co.uk> wrote: > >> I have some more information about absolutely's problem. >> >> The call is established fine, there is two-way audio, all loooks fine, >> until about 60 seconds into the call when FreeSWITCH issues what appears to >> be unsolicited INVITES to which it recieves no response from the carrier >> and as such, it drops the call. >> >> Any suggestions? >> >> > I'd get the console log and sip trace and throw it up on pb so we could > see that in action. > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS* Managing Director *T* 03300 100 960 *F* 03300 100 961 *E* andrew at cassidywebservices.co.uk *W* www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/c7d4ef48/attachment.html From chrisbware at yahoo.it Fri Mar 30 14:22:58 2012 From: chrisbware at yahoo.it (Chris B. Ware) Date: Fri, 30 Mar 2012 11:22:58 +0100 (BST) Subject: [Freeswitch-users] Day of week Message-ID: <1333102978.12464.YahooMailNeo@web132301.mail.ird.yahoo.com> Hi all, I'm playing with voicemail, to do an italian version of templates, sound files and tts.xml. By default in voicemail.conf.xml we can find %A is evaluated in english. Is there a way to customize it according to a different language? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/71c2a6f4/attachment.html From d.campbell at ampersand.com Fri Mar 30 18:47:26 2012 From: d.campbell at ampersand.com (Doug Campbell) Date: Fri, 30 Mar 2012 10:47:26 -0400 Subject: [Freeswitch-users] Unexpected behavior with hostname in socket application In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD022D77D397@NY1-EXMB-01.ip-soft.net> References: <4F749B07.6050700@ampersand.com> <6A6B4C284AD15042B429EB9D904544AD022D77D397@NY1-EXMB-01.ip-soft.net> Message-ID: <4F75C77E.7010003@ampersand.com> > The obvious answer would be: fix the DNS settings :) If I could, I would. (Machine not under my admin). But it's a larger problem than that. See below. > (or use the ip address of the network, not the loopback ip) No address or hostname ever resolves to anything but "localhost" in freeswitch. Very strange, because it picks up the port number, just not the hostname. > What happens if you try to do a "telnet localhost 4574" and then a "telnet 127.0.0.1 4574" on the same machine? Works in one case and not in the other? I've forced the host to use only local files for name resolution, fixed "localhost" to point to 127.0.0.1, and when I telnet or nc to localhost:4574 it works. nslookup localhost is the only thing that returns the wrong address. Earlier versions of freeswitch used to work fine in this way. Something about the current versions is not using the host-configured name lookup, but seems to be going directly to DNS lookups. (In addition to the problem of not picking up the correct hostname from the dialplan). Again thanks for any help, Doug On 03/29/2012 03:26 PM, Hector Geraldino wrote: > The obvious answer would be: fix the DNS settings :) (or use the ip address of the network, not the loopback ip) > > Anyway, I'm curious about this one. What happens if you try to do a "telnet localhost 4574" and then a "telnet 127.0.0.1 4574" on the same machine? Works in one case and not in the other? > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Doug Campbell > Sent: Thursday, March 29, 2012 1:25 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Unexpected behavior with hostname in socket application > > I am getting unexpected behavior on the hostname used in the socket application. > > I have a trivial dialplan: > > > > > > > > > > This outbound event socket gets connected fine on my CentOS 6.2 machine. > Though the freeswitch.log file seems to change "127.0.0.1" to "localhost": > > EXECUTE sofia/phoneglue/9789735994 socket(localhost:4574 async) > > However, with the identical software (including fs built from git today), on my RHEL 6.2 machine that has a DNS glitch resolving "localhost" badly (don't ask), it fails. > The freeswitch.log file says: > > EXECUTE sofia/phoneglue/9789735994 socket(localhost:4574 async) > 2012-03-29 10:05:08.200547 [ERR] mod_event_socket.c:458 Socket Error! > > Why is it changing "127.0.0.1" to "localhost"? > Shouldn't it be using the raw IP when it is so specified? > Without it doing this, my RHEL box is doomed. > > Much thanks for any help, > Doug > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Fri Mar 30 19:29:48 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 30 Mar 2012 10:29:48 -0500 Subject: [Freeswitch-users] Unexpected behavior with hostname in socket application In-Reply-To: <4F749B07.6050700@ampersand.com> References: <4F749B07.6050700@ampersand.com> Message-ID: There is nothing on Fs that would do this. Search your configs for localhost..... On Mar 29, 2012 1:36 PM, "Doug Campbell" wrote: > I am getting unexpected behavior on the hostname used in the socket > application. > > I have a trivial dialplan: > > > > > > > > > > This outbound event socket gets connected fine on my CentOS 6.2 machine. > Though the freeswitch.log file seems to change "127.0.0.1" to "localhost": > > EXECUTE sofia/phoneglue/9789735994 socket(localhost:4574 async) > > However, with the identical software (including fs built from git today), > on my RHEL 6.2 machine that has a DNS glitch resolving "localhost" badly > (don't ask), it fails. > The freeswitch.log file says: > > EXECUTE sofia/phoneglue/9789735994 socket(localhost:4574 async) > 2012-03-29 10:05:08.200547 [ERR] mod_event_socket.c:458 Socket Error! > > Why is it changing "127.0.0.1" to "localhost"? > Shouldn't it be using the raw IP when it is so specified? > Without it doing this, my RHEL box is doomed. > > Much thanks for any help, > Doug > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/912c93de/attachment.html From djbinter at gmail.com Fri Mar 30 19:33:55 2012 From: djbinter at gmail.com (DJB International) Date: Fri, 30 Mar 2012 08:33:55 -0700 Subject: [Freeswitch-users] Trouble setting "caller_id_name", "caller_id_number" on an inbound call In-Reply-To: References: Message-ID: In general, caller_id_name and caller_id_number are read-only values. In this case, if you want to set them for inline use purpose, you can set them to other variables like in_cid_name and in_cid_num, then you can call that variables by using it as {in_cid_name} and/or {in_cid_num} to run to dialplan matching later. -djbinter On Fri, Mar 30, 2012 at 3:27 AM, Brian Foster wrote: > As the subject says, I'm having issues setting the caller id name and > number on an inbound call when doing some cleanups and a lookup. These > dialplans are based on what information is listed for Mod_callcenter, and I > have a feeling that they are wrong. I'll volunteer myself to change them if > they are. > > Alright, so here's the dialplan: > > > > inline="true"/> > > > > > > > > > > > expression="true"/> > expression="^${caller_id_number}$|^$"/> > expression="^(?:/+1)?([2-9]\d\d[2-9]\d{6})$"> > > > > > Call log here: http://pastebin.freeswitch.org/18786 > > I've tried setting the effective_caller_id_(name/number) but no luck > there. Is there something I'm doing wrong here? > > -BDF > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/81d49103/attachment-0001.html From d.campbell at ampersand.com Fri Mar 30 20:09:23 2012 From: d.campbell at ampersand.com (Doug Campbell) Date: Fri, 30 Mar 2012 12:09:23 -0400 Subject: [Freeswitch-users] Unexpected behavior with hostname in socket application In-Reply-To: References: <4F749B07.6050700@ampersand.com> Message-ID: <4F75DAB3.5040003@ampersand.com> > There is nothing on Fs that would do this. Search your configs for localhost..... Thanks for the info. I suspected this, but this just makes it more puzzling. Here's a grep of localhost in any of my configs and the computed runtime xml. I don't see any smoking guns, do you? [root at warhel31 freeswitch]# pwd /usr/local/freeswitch [root at warhel31 freeswitch]# find conf -exec fgrep localhost {} \; -print conf/autoload_configs/redis.conf.xml conf/autoload_configs/cdr_pg_csv.conf.xml conf/autoload_configs/xml_cdr.conf.xml conf/autoload_configs/memcache.conf.xml [root at warhel31 freeswitch]# fgrep localhost log/freeswitch.xml.fsxml [root at warhel31 freeswitch]# From lesley.pervis at gmail.com Fri Mar 30 20:17:08 2012 From: lesley.pervis at gmail.com (Lesley Pervis) Date: Fri, 30 Mar 2012 10:17:08 -0600 Subject: [Freeswitch-users] sip_exclude_contact in mad boss scenario does not work In-Reply-To: References: Message-ID: Thanks Michael, The paste is at: http://pastebin.freeswitch.org/18788 Here is my extension: When I call from, for example, 1002, both 1003 and 1004 pick up and broadcast the page, and when I hang up, all legs get torn down. When I call from 1003, however, deaf legs to both 1003 and 1004 are set up, but I expect the sip_exclude_contact setting to ensure no leg is set up to 1003. When I hang up, the "page" leg hangs up, but the deaf legs are still up, and the phone at 1003 has its deaf leg on hold. I need to resume the leg and hang it up for the conference to end. The paste is from this second scenario. I looked in the code, and it looks src/mod/endpoints/mod_sofia/mod_sofia.c does an sql lookup and excludes anything "like" the network_addr in the contact header, but I couldn't trace back how this is actually used. Maybe I'm using this extension wrong. Does this work only with groups? (Note that I haven't figured out yet how to use groups properly, since the mad boss groups example in dialplan/default.xml does not work with the groups as defined in directory/default.xml. That "pointer" seems to break things.) In the meantime, I'll look more closely at that. Thanks for any insight you might have on this. Les On Thu, Mar 29, 2012 at 8:03 PM, Michael Collins wrote: > Grab a console debug log and throw it on pastebin.freeswitch.org. Use > "FreeSWITCH Log" for syntax highlighting and then give us the pb URL in > this thread. > -MC > > > On Thu, Mar 29, 2012 at 5:02 PM, Lesley Pervis wrote: > >> I'm using a small variation on the mad boss extension in the default >> dialplan, much like this, except instead of using groups, I'm using >> multiple user/100X@${domain} lines. >> >> http://wiki.freeswitch.org/wiki/Variable_conference_auto_outcall_prefix >> >> The problem is that the sip_exclude contact does not work as I expect. I >> expect that the conference is NOT bridged to the caller, but it is. I can >> see in the logs that a conference leg to the calling phone is set up, and >> the phone answers and puts it on hold. This means there are two conference >> legs up on the calling phone: the one that initiated the page, and another >> deaf leg. When the first leg hangs up, the deaf one is on hold, and to end >> the conference, I have to resume that leg and hang it up. >> >> Anyone know why this isn't working for me? >> >> FreeSWITCH version: 1.0.head (git-d8d4d20 2012-03-14 19-00-26 +0000) >> >> Thanks, >> Les >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/95114d71/attachment.html From bob.mccarthy at experient.com Fri Mar 30 20:37:20 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Fri, 30 Mar 2012 10:37:20 -0600 Subject: [Freeswitch-users] Setting Hangup Hook on api originate Message-ID: <017801cd0e93$621f4410$265dcc30$@mccarthy@experient.com> I am having trouble getting a hangup hook to execute after an api originate. bgapi expand originate {api_hangup_hook=confhanguphook.lua,session_in_hangup_hook=true}sofia/intern al/2001@${switchvox} &conference(25bce61d-fb32-44d7-8d99-7ceafc599f22 at sla) I get into the conference just fine but the script does not execute when I hang up. Any Idea's of what I am doing wrong? Bob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/85214d3b/attachment.html From bob.mccarthy at experient.com Fri Mar 30 20:42:31 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Fri, 30 Mar 2012 10:42:31 -0600 Subject: [Freeswitch-users] Setting CallerID for inbound SLA Barge in In-Reply-To: <00bb01cd05a3$563ae1c0$02b0a540$@com> References: <4f66b919.84d2e00a.7bc2.61f1SMTPIN_ADDED@mx.google.com> <00bb01cd05a3$563ae1c0$02b0a540$@com> Message-ID: <017d01cd0e94$1b1c4a80$5154df80$@mccarthy@experient.com> I am displaying the participants in an application and would like to have show who the "phone is" rather than the shared user From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bote Man Sent: Monday, March 19, 2012 1:39 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] Setting CallerID for inbound SLA Barge in That originates in the device itself and is contained in the caller_id_name and caller_id_number variables. I just tried the effective_ series of variables, but those are for the outbound B-leg calls and are empty at the time the conference is entered. But if a phone is barging into a conference how does Caller*ID have any impact, other than to test for it in the dialplan? Do you intend to say: the number or name? Bote From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob McCarthy Sent: Monday, 19 March, 2012 00:37 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Setting CallerID for inbound SLA Barge in I have read where you can set the outbound Callerid name and number for calling out of a conference, but how do you set the Callerid name and number for phones that barge in on a conference (SLA in particular) ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/f3b03037/attachment.html From ghostofbasho at gmail.com Fri Mar 30 19:44:48 2012 From: ghostofbasho at gmail.com (Thomas McCarthy-Howe) Date: Fri, 30 Mar 2012 11:44:48 -0400 Subject: [Freeswitch-users] Specifying Port on Record-Route header Message-ID: <485C943713C34D18909EF7560822BC17@gmail.com> Hi all - I'm trying to configure my FS box to run on the same server as an OpenSIPs instance, and I think I'm having a problem because, when FS adds a Record-Route header to the 200 OK, it does not specify the port. So, my endpoint is sending the ACK back to OpenSIPs, not to FS, and it keeps looping on me. The 200 OK from FS running on port 5070: 15:21:29.330683 IP 192.168.153.13.vtsas > 192.168.153.13.sip: SIP, length: 1044 E..0.N.. at . ........SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.153.13;branch=z9hG4bK36e1.59a34dc3.0 Via: SIP/2.0/UDP 192.168.153.28;branch=z9hG4bKebe2bd1a967399c3d Record-Route: From: ;tag=8739e6ff4f To: ;tag=F5yXe7FeFH6jg Call-ID: c0bd06789074500d CSeq: 498777857 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-62c6855 2012-02-27 13-28-27 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 217 Remote-Party-ID: "MEETME01" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1333103289 1333103290 IN IP4 192.168.153.13 s=FreeSWITCH c=IN IP4 192.168.153.13 t=0 0 m=audio 17600 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:10 m=audio 0 RTP/AVP 19 The ACK comes back to 5060, and loops: 15:21:29.338255 IP 192.168.153.13.sip > 192.168.153.13.sip: SIP, length: 599 ....._..ACK sip:192.168.153.13;lr=on SIP/2.0 Via: SIP/2.0/UDP 192.168.153.13;branch=z9hG4bK36e1.59a34dc3.2 Via: SIP/2.0/UDP 192.168.153.28;branch=z9hG4bK01e5c3b7e9a67e0dd Proxy-Authorization: Digest username="D05",realm="192.168.153.13",nonce="4f75cf9700005326420497c8c462fb80c60a857e0d0049ee",uri="sip:MEETME01 at 192.168.153.13",response="961b9d5a496f610d10fd136fecf0291c" Max-Forwards: 69 From: ;tag=8739e6ff4f To: ;tag=F5yXe7FeFH6jg Call-ID: c0bd06789074500d CSeq: 498777857 ACK User-Agent: ATC ION M5T SIP Stack/4.1.8.13 Content-Length: 0 Any thoughts? Thanks in advance. Thomas -- Thomas McCarthy-Howe Sent with Sparrow (http://www.sparrowmailapp.com/?sig) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/d3c5e4c0/attachment-0001.html From luis.daniel.lucio at gmail.com Fri Mar 30 22:55:35 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Fri, 30 Mar 2012 14:55:35 -0400 Subject: [Freeswitch-users] Could not locate channel type skinny Message-ID: Helo, Strange thing. I'm mapping a DID to a extension. Using a softphone incomming call is working using a hwrdware atcome phone it fails with this log: EXECUTE sofia/external/18886498985 at voxbone.com bridge(skinny/internal/1100) 2012-03-30 14:47:36.320320 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-30 14:47:36.320320 [ERR] switch_core_session.c:427 Could not locate channel type skinny 2012-03-30 14:47:36.320320 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [skinny] cause: [CHAN_NOT_IMPLEMENTED] 2012-03-30 14:47:36.320320 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] 2012-03-30 14:47:36.320320 [INFO] mod_dptools.c:2916 Originate Failed. Cause: CHAN_NOT_IMPLEMENTED 2012-03-30 14:47:36.320320 [DEBUG] switch_channel.c:2848 (sofia/external/18886498985 at voxbone.com) Callstate Change RINGING -> HANGUP Is it a missing thing ? Thanks LD From kris at kriskinc.com Fri Mar 30 23:00:38 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 30 Mar 2012 15:00:38 -0400 Subject: [Freeswitch-users] Specifying Port on Record-Route header In-Reply-To: <485C943713C34D18909EF7560822BC17@gmail.com> References: <485C943713C34D18909EF7560822BC17@gmail.com> Message-ID: Do you have a full trace of the interaction between the various components? It's not completely clear what's going on from these two packets. On Fri, Mar 30, 2012 at 11:44 AM, Thomas McCarthy-Howe wrote: > Hi all - > > I'm trying to configure my FS box to run on the same server as an OpenSIPs > instance, and I think I'm having a problem because, when FS adds a > Record-Route header to the 200 OK, it does not specify the port. So, my > endpoint is sending the ACK back to OpenSIPs, not to FS, and it keeps > looping on me. > > The 200 OK from FS running on port 5070: > 15:21:29.330683 IP 192.168.153.13.vtsas > 192.168.153.13.sip: SIP, length: > 1044 > E..0.N.. at . > ........SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.153.13;branch=z9hG4bK36e1.59a34dc3.0 > Via: SIP/2.0/UDP 192.168.153.28;branch=z9hG4bKebe2bd1a967399c3d > Record-Route: > From: ;tag=8739e6ff4f > To: ;tag=F5yXe7FeFH6jg > Call-ID: c0bd06789074500d > CSeq: 498777857 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-62c6855 2012-02-27 13-28-27 > -0600 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 217 > Remote-Party-ID: "MEETME01" > ;party=calling;privacy=off;screen=no > > v=0 > o=FreeSWITCH 1333103289 1333103290 IN IP4 192.168.153.13 > s=FreeSWITCH > c=IN IP4 192.168.153.13 > t=0 0 > m=audio 17600 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > a=silenceSupp:off - - - - > a=ptime:10 > m=audio 0 RTP/AVP 19 > > > The ACK comes back to 5060, and loops: > > > 15:21:29.338255 IP 192.168.153.13.sip > 192.168.153.13.sip: SIP, length: 599 > ....._..ACK sip:192.168.153.13;lr=on SIP/2.0 > Via: SIP/2.0/UDP 192.168.153.13;branch=z9hG4bK36e1.59a34dc3.2 > Via: SIP/2.0/UDP 192.168.153.28;branch=z9hG4bK01e5c3b7e9a67e0dd > Proxy-Authorization: Digest > username="D05",realm="192.168.153.13",nonce="4f75cf9700005326420497c8c462fb80c60a857e0d0049ee",uri="sip:MEETME01 at 192.168.153.13",response="961b9d5a496f610d10fd136fecf0291c" > Max-Forwards: 69 > From: ;tag=8739e6ff4f > To: ;tag=F5yXe7FeFH6jg > Call-ID: c0bd06789074500d > CSeq: 498777857 ACK > User-Agent: ATC ION M5T SIP Stack/4.1.8.13 > Content-Length: 0 > > > Any thoughts? > > Thanks in advance. > Thomas > > > -- > Thomas McCarthy-Howe > Sent with Sparrow > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From bdfoster at endigotech.com Fri Mar 30 23:20:49 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 30 Mar 2012 15:20:49 -0400 Subject: [Freeswitch-users] Trouble setting "caller_id_name", "caller_id_number" on an inbound call In-Reply-To: References: Message-ID: I need to change the caller_id_name and caller_id_number. I've got ATA's that don't like the plus in front of the number. At the same time, I'm doing a cidlooup, so that's going to have to be changed as well. -BDF On Fri, Mar 30, 2012 at 11:33 AM, DJB International wrote: > In general, caller_id_name and caller_id_number are read-only values. In > this case, if you want to set them for inline use purpose, you can set them > to other variables like in_cid_name and in_cid_num, then you can call that > variables by using it as {in_cid_name} and/or {in_cid_num} to run to > dialplan matching later. > > -djbinter > > On Fri, Mar 30, 2012 at 3:27 AM, Brian Foster wrote: > >> As the subject says, I'm having issues setting the caller id name and >> number on an inbound call when doing some cleanups and a lookup. These >> dialplans are based on what information is listed for Mod_callcenter, and I >> have a feeling that they are wrong. I'll volunteer myself to change them if >> they are. >> >> Alright, so here's the dialplan: >> >> >> >> > inline="true"/> >> >> >> >> >> >> >> >> >> >> >> > expression="true"/> >> > expression="^${caller_id_number}$|^$"/> >> > expression="^(?:/+1)?([2-9]\d\d[2-9]\d{6})$"> >> >> >> >> >> Call log here: http://pastebin.freeswitch.org/18786 >> >> I've tried setting the effective_caller_id_(name/number) but no luck >> there. Is there something I'm doing wrong here? >> >> -BDF >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/02f69442/attachment.html From bdfoster at endigotech.com Fri Mar 30 23:24:30 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 30 Mar 2012 15:24:30 -0400 Subject: [Freeswitch-users] Could not locate channel type skinny In-Reply-To: References: Message-ID: Did you mean to use skinny? -BDF On Fri, Mar 30, 2012 at 2:55 PM, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > Helo, > > Strange thing. > > I'm mapping a DID to a extension. > > Using a softphone incomming call is working > using a hwrdware atcome phone it fails with this log: > > EXECUTE sofia/external/18886498985 at voxbone.combridge(skinny/internal/1100) > 2012-03-30 14:47:36.320320 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2012-03-30 14:47:36.320320 [ERR] switch_core_session.c:427 Could not > locate channel type skinny > 2012-03-30 14:47:36.320320 [NOTICE] switch_ivr_originate.c:2459 Cannot > create outgoing channel of type [skinny] cause: [CHAN_NOT_IMPLEMENTED] > 2012-03-30 14:47:36.320320 [DEBUG] switch_ivr_originate.c:3364 > Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] > 2012-03-30 14:47:36.320320 [INFO] mod_dptools.c:2916 Originate Failed. > Cause: CHAN_NOT_IMPLEMENTED > 2012-03-30 14:47:36.320320 [DEBUG] switch_channel.c:2848 > (sofia/external/18886498985 at voxbone.com) Callstate Change RINGING -> > HANGUP > > > Is it a missing thing ? > > Thanks > > LD > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/8143c032/attachment-0001.html From msc at freeswitch.org Fri Mar 30 23:37:17 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 30 Mar 2012 12:37:17 -0700 Subject: [Freeswitch-users] Trouble setting "caller_id_name", "caller_id_number" on an inbound call In-Reply-To: References: Message-ID: On Fri, Mar 30, 2012 at 12:20 PM, Brian Foster wrote: > I need to change the caller_id_name and caller_id_number. I've got ATA's > that don't like the plus in front of the number. At the same time, I'm > doing a cidlooup, so that's going to have to be changed as well. > > -BDF > > It looks like PB 18786 doesn't have very much information. I'd recommend getting the whole call from start to finish, and get the SIP trace as well. Also, if you're creating a new leg with a bridge then you'll need to export effective_caller_id_number or set it in the {} in front of the dialstring. Lastly, be sure to use the sip_cid_typevar to select which field has the caller ID info in it. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/54bf3223/attachment.html From msc at freeswitch.org Fri Mar 30 23:43:29 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 30 Mar 2012 12:43:29 -0700 Subject: [Freeswitch-users] When using nimbuzz, call disconnects after 90 seconds with RECOVERY_ON_TIMER_EXPIRE In-Reply-To: References: Message-ID: In a case like this I like to compare working siptraces against non-working ones so that I have a frame of reference. Put those on pastebin and the gang here will take a look. -MC On Fri, Mar 30, 2012 at 3:16 AM, Anton VG wrote: > Please help, > > While other SIP clients works well, there is a problem with nimbuzz > SIP client, which quite frequently used by 'not-advanced' users. seems > It which works through their services (asterisk 1.6) > > This problem was not happening while I have been using Asterisk too, > but after recent migration to FS - it does. > > nimbuzz client stays like online, just sound disappears, but there is > a confirmation from nimbuzz server side, regarding the BYE > > most likely a NAT issue, but if the same time using Asterisk on the > same server - there is not problems. So might be there any options to > tune? > > there are some SIP activity like this before calls disconnects, no replies: > > send 977 bytes to udp/[195.211.48.6]:12621 at 10:00:02.535856: > ------------------------------------------------------------------------ > INVITE sip:anton at 192.168.0.20:5061;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 62.122.137.32;rport;branch=z9hG4bKaQm6tK1aeZ7Fe > Max-Forwards: 70 > From: "3777077" ;tag=eKv1Nej7ZrNUS > To: "anton at 62.122.137.32" ;tag=sip1331131792771 > Call-ID: 3696882418 > CSeq: 26214721 INVITE > Contact: > User-Agent: Synaptic Switch v2.1 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uac > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 272 > > v=0 > o=FreeSWITCH 1333081454 1333081455 IN IP4 62.122.137.32 > s=FreeSWITCH > c=IN IP4 62.122.137.32 > t=0 0 > m=audio 20062 RTP/AVP 18 106 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:106 telephone-event/8000 > a=fmtp:106 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > and than > > 2012-03-30 15:00:03.554460 [NOTICE] sofia.c:6301 Hangup > sofia/internal/anton at 62.122.137.32 [CS_SOFT_EXECUTE] > [RECOVERY_ON_TIMER_EXPIRE] > > recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960518: > ------------------------------------------------------------------------ > BYE sip:3777077 at 62.122.137.32:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 > Max-Forwards: 70 > To: ;tag=eKv1Nej7ZrNUS > From: ;tag=sip1331131792771 > Call-ID: 3696882418 > CSeq: 3 BYE > User-Agent: Nimbuzz Single > Content-Length: 0 > > ------------------------------------------------------------------------ > send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960672: > ------------------------------------------------------------------------ > SIP/2.0 481 Call Does Not Exist > Via: SIP/2.0/UDP > 192.168.0.20:5061 > ;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 > From: ;tag=sip1331131792771 > To: ;tag=eKv1Nej7ZrNUS > Call-ID: 3696882418 > CSeq: 3 BYE > User-Agent: Synaptic Switch v2.1 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960734: > ------------------------------------------------------------------------ > BYE sip:3777077 at 62.122.137.32:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 > Max-Forwards: 70 > To: ;tag=eKv1Nej7ZrNUS > From: ;tag=sip1331131792771 > Call-ID: 3696882418 > CSeq: 3 BYE > User-Agent: Nimbuzz Single > Content-Length: 0 > > ------------------------------------------------------------------------ > send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960781: > ------------------------------------------------------------------------ > SIP/2.0 481 Call Does Not Exist > Via: SIP/2.0/UDP > 192.168.0.20:5061 > ;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 > From: ;tag=sip1331131792771 > To: ;tag=eKv1Nej7ZrNUS > Call-ID: 3696882418 > CSeq: 3 BYE > User-Agent: Synaptic Switch v2.1 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079094: > ------------------------------------------------------------------------ > ACK sip:anton at 62.122.137.32 SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 > Max-Forwards: 70 > To: ;tag=eKv1Nej7ZrNUS > From: ;tag=sip1331131792771 > Call-ID: 3696882418 > CSeq: 3 ACK > Contact: > Expires: 3600 > User-Agent: Nimbuzz Single > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079293: > ------------------------------------------------------------------------ > ACK sip:anton at 62.122.137.32 SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 > Max-Forwards: 70 > To: ;tag=eKv1Nej7ZrNUS > From: ;tag=sip1331131792771 > Call-ID: 3696882418 > CSeq: 3 ACK > Contact: > Expires: 3600 > User-Agent: Nimbuzz Single > Content-Length: 0 > > ------------------------------------------------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/96220079/attachment.html From msc at freeswitch.org Fri Mar 30 23:45:22 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 30 Mar 2012 12:45:22 -0700 Subject: [Freeswitch-users] Session-wise Language Setting In-Reply-To: <12BA6FAF92674DEE9650CF4829B95DF0@sharad> References: <4CB4E6075A9F4F9BA1C8A57FF2A06735@sharad> <4F746EFF.3050305@communicatefreely.net> <00C2DD790B254A1BACD50CAA66B123B9@sharad> <1FFF97C269757C458224B7C895F35F15083A49@cantor.std.visionutv.se> <12BA6FAF92674DEE9650CF4829B95DF0@sharad> Message-ID: On Fri, Mar 30, 2012 at 5:13 AM, Sharad Garg wrote: > Thanks Peter...yes it works. > > Just a last query - Is there any API for setting the same so that it can be > used from Javascript. > > Best Regards > Sharad > > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_setvar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/34d16621/attachment.html From msc at freeswitch.org Fri Mar 30 23:59:41 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 30 Mar 2012 12:59:41 -0700 Subject: [Freeswitch-users] Setting Hangup Hook on api originate In-Reply-To: <4f75e17e.084cec0a.594e.ffff9d3bSMTPIN_ADDED@mx.google.com> References: <4f75e17e.084cec0a.594e.ffff9d3bSMTPIN_ADDED@mx.google.com> Message-ID: On Fri, Mar 30, 2012 at 9:37 AM, Bob McCarthy wrote: > I am having trouble getting a hangup hook to execute after an api > originate.**** > > **** > > bgapi expand originate > {api_hangup_hook=confhanguphook.lua,session_in_hangup_hook=true}sofia/internal/2001@${switchvox} > &conference(25bce61d-fb32-44d7-8d99-7ceafc599f22 at sla)**** > > ** ** > > I get into the conference just fine but the script does not execute when I > hang up. Any Idea's of what I am doing wrong? > I have an idea: how about actually calling 'lua' and not just the name of your script? ;) bgapi expand originate {api_hangup_hook='lua confhanguphook.lua',session_in_hangup_hook=true}sofia/internal/2001@${switchvox} &conference(25bce61d-fb32-44d7-8d99-7ceafc599f22 at sla) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/dd67af53/attachment-0001.html From bdfoster at endigotech.com Sat Mar 31 00:06:55 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 30 Mar 2012 16:06:55 -0400 Subject: [Freeswitch-users] Trouble setting "caller_id_name", "caller_id_number" on an inbound call In-Reply-To: References: Message-ID: http://pastebin.freeswitch.org/18791 << Entire Call It seems like a pretty common question, and there doesn't seem to be a good answer to just cleaning up the caller id and doing a lookup. Even the documentation on the wiki is wrong. To me, if anything, I'd like to get something that's correct on the wiki in order to facilitate new users. I've gotten this to work as well, and I don't remember having to go through much to get it working. I wonder if something has changed? -BDF On Fri, Mar 30, 2012 at 3:37 PM, Michael Collins wrote: > > > On Fri, Mar 30, 2012 at 12:20 PM, Brian Foster wrote: > >> I need to change the caller_id_name and caller_id_number. I've got ATA's >> that don't like the plus in front of the number. At the same time, I'm >> doing a cidlooup, so that's going to have to be changed as well. >> >> -BDF >> >> > It looks like PB 18786 doesn't have very much information. I'd recommend > getting the whole call from start to finish, and get the SIP trace as well. > Also, if you're creating a new leg with a bridge then you'll need to export > effective_caller_id_number or set it in the {} in front of the dialstring. > Lastly, be sure to use the sip_cid_typevar to select which field has the caller ID info in it. > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/e8b3a135/attachment.html From luis.daniel.lucio at gmail.com Sat Mar 31 00:20:09 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Fri, 30 Mar 2012 16:20:09 -0400 Subject: [Freeswitch-users] Could not locate channel type skinny In-Reply-To: References: Message-ID: No, i've dissable 729 in freeswitch. My concert is why it fails only in incomming calls when using a DID->ext mapping and only with that hardware phone, softphone works nice. Wondering if you have a workarround. Le 30 mars 2012 15:24, Brian Foster a ?crit : > Did you mean to use skinny? > > -BDF > > > On Fri, Mar 30, 2012 at 2:55 PM, Luis Daniel Lucio Quiroz > wrote: >> >> Helo, >> >> Strange thing. >> >> I'm ?mapping a DID to a extension. >> >> Using a softphone ?incomming call is working >> using a hwrdware atcome phone it fails with this log: >> >> EXECUTE sofia/external/18886498985 at voxbone.com >> bridge(skinny/internal/1100) >> 2012-03-30 14:47:36.320320 [DEBUG] switch_ivr_originate.c:1884 Parsing >> global variables >> 2012-03-30 14:47:36.320320 [ERR] switch_core_session.c:427 Could not >> locate channel type skinny >> 2012-03-30 14:47:36.320320 [NOTICE] switch_ivr_originate.c:2459 Cannot >> create outgoing channel of type [skinny] cause: [CHAN_NOT_IMPLEMENTED] >> 2012-03-30 14:47:36.320320 [DEBUG] switch_ivr_originate.c:3364 >> Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] >> 2012-03-30 14:47:36.320320 [INFO] mod_dptools.c:2916 Originate Failed. >> ?Cause: CHAN_NOT_IMPLEMENTED >> 2012-03-30 14:47:36.320320 [DEBUG] switch_channel.c:2848 >> (sofia/external/18886498985 at voxbone.com) Callstate Change RINGING -> >> HANGUP >> >> >> Is it a missing thing ? >> >> Thanks >> >> LD >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From djbinter at gmail.com Sat Mar 31 00:46:12 2012 From: djbinter at gmail.com (DJB International) Date: Fri, 30 Mar 2012 13:46:12 -0700 Subject: [Freeswitch-users] Trouble setting "caller_id_name", "caller_id_number" on an inbound call In-Reply-To: References: Message-ID: If you want to pass to B-leg with a new number format, then you'll need to export effective_caller_id_number/effective_caller_id_name per MC. -djbinter On Fri, Mar 30, 2012 at 1:06 PM, Brian Foster wrote: > http://pastebin.freeswitch.org/18791 << Entire Call > > It seems like a pretty common question, and there doesn't seem to be a > good answer to just cleaning up the caller id and doing a lookup. Even the > documentation on the wiki is wrong. To me, if anything, I'd like to get > something that's correct on the wiki in order to facilitate new users. I've > gotten this to work as well, and I don't remember having to go through much > to get it working. I wonder if something has changed? > > -BDF > > On Fri, Mar 30, 2012 at 3:37 PM, Michael Collins wrote: > >> >> >> On Fri, Mar 30, 2012 at 12:20 PM, Brian Foster wrote: >> >>> I need to change the caller_id_name and caller_id_number. I've got ATA's >>> that don't like the plus in front of the number. At the same time, I'm >>> doing a cidlooup, so that's going to have to be changed as well. >>> >>> -BDF >>> >>> >> It looks like PB 18786 doesn't have very much information. I'd recommend >> getting the whole call from start to finish, and get the SIP trace as well. >> Also, if you're creating a new leg with a bridge then you'll need to export >> effective_caller_id_number or set it in the {} in front of the dialstring. >> Lastly, be sure to use the sip_cid_typevar to select which field has the caller ID info in it. >> >> -MC >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/d33ae9b2/attachment.html From bdfoster at endigotech.com Sat Mar 31 00:46:59 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 30 Mar 2012 16:46:59 -0400 Subject: [Freeswitch-users] Trouble setting "caller_id_name", "caller_id_number" on an inbound call In-Reply-To: References: Message-ID: Solution time! On Fri, Mar 30, 2012 at 4:06 PM, Brian Foster wrote: > http://pastebin.freeswitch.org/18791 << Entire Call > > It seems like a pretty common question, and there doesn't seem to be a > good answer to just cleaning up the caller id and doing a lookup. Even the > documentation on the wiki is wrong. To me, if anything, I'd like to get > something that's correct on the wiki in order to facilitate new users. I've > gotten this to work as well, and I don't remember having to go through much > to get it working. I wonder if something has changed? > > -BDF > > On Fri, Mar 30, 2012 at 3:37 PM, Michael Collins wrote: > >> >> >> On Fri, Mar 30, 2012 at 12:20 PM, Brian Foster wrote: >> >>> I need to change the caller_id_name and caller_id_number. I've got ATA's >>> that don't like the plus in front of the number. At the same time, I'm >>> doing a cidlooup, so that's going to have to be changed as well. >>> >>> -BDF >>> >>> >> It looks like PB 18786 doesn't have very much information. I'd recommend >> getting the whole call from start to finish, and get the SIP trace as well. >> Also, if you're creating a new leg with a bridge then you'll need to export >> effective_caller_id_number or set it in the {} in front of the dialstring. >> Lastly, be sure to use the sip_cid_typevar to select which field has the caller ID info in it. >> >> -MC >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/f3b4aa26/attachment-0001.html From bdfoster at endigotech.com Sat Mar 31 00:56:47 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 30 Mar 2012 16:56:47 -0400 Subject: [Freeswitch-users] Trouble setting "caller_id_name", "caller_id_number" on an inbound call In-Reply-To: References: Message-ID: On Mar 30, 2012 4:47 PM, "DJB International" wrote: > > If you want to pass to B-leg with a new number format, then you'll need to export effective_caller_id_number/effective_caller_id_name per MC. > > -djbinter Both of those variables are automatically exported by default. No need to export those. Just set and forget! I would like to point out that this: ...does not work. I think this is a bug. It rubs fine but it might just be setting the wrong variable. I'll take that to JIRA. -BDF > > On Fri, Mar 30, 2012 at 1:06 PM, Brian Foster wrote: >> >> http://pastebin.freeswitch.org/18791 << Entire Call >> >> It seems like a pretty common question, and there doesn't seem to be a good answer to just cleaning up the caller id and doing a lookup. Even the documentation on the wiki is wrong. To me, if anything, I'd like to get something that's correct on the wiki in order to facilitate new users. I've gotten this to work as well, and I don't remember having to go through much to get it working. I wonder if something has changed? >> >> -BDF >> >> On Fri, Mar 30, 2012 at 3:37 PM, Michael Collins wrote: >>> >>> >>> >>> On Fri, Mar 30, 2012 at 12:20 PM, Brian Foster wrote: >>>> >>>> I need to change the caller_id_name and caller_id_number. I've got ATA's that don't like the plus in front of the number. At the same time, I'm doing a cidlooup, so that's going to have to be changed as well. >>>> >>>> -BDF >>>> >>> >>> It looks like PB 18786 doesn't have very much information. I'd recommend getting the whole call from start to finish, and get the SIP trace as well. Also, if you're creating a new leg with a bridge then you'll need to export effective_caller_id_number or set it in the {} in front of the dialstring. Lastly, be sure to use the sip_cid_type var to select which field has the caller ID info in it. >>> >>> -MC >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/b4c9bc50/attachment.html From bdfoster at endigotech.com Sat Mar 31 01:06:09 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 30 Mar 2012 17:06:09 -0400 Subject: [Freeswitch-users] Trouble setting "caller_id_name", "caller_id_number" on an inbound call In-Reply-To: References: Message-ID: Paid the wiki tax. It is documented by how it should work, by doing the line posted in a previous email as opposed to doing an API call. -BDF On Fri, Mar 30, 2012 at 4:56 PM, Brian Foster wrote: > > On Mar 30, 2012 4:47 PM, "DJB International" wrote: > > > > If you want to pass to B-leg with a new number format, then you'll need > to export effective_caller_id_number/effective_caller_id_name per MC. > > > > -djbinter > > Both of those variables are automatically exported by default. No need to > export those. Just set and forget! > > I would like to point out that this: > > > > ...does not work. I think this is a bug. It rubs fine but it might just be > setting the wrong variable. I'll take that to JIRA. > > -BDF > > > > > On Fri, Mar 30, 2012 at 1:06 PM, Brian Foster > wrote: > >> > >> http://pastebin.freeswitch.org/18791 << Entire Call > >> > >> It seems like a pretty common question, and there doesn't seem to be a > good answer to just cleaning up the caller id and doing a lookup. Even the > documentation on the wiki is wrong. To me, if anything, I'd like to get > something that's correct on the wiki in order to facilitate new users. I've > gotten this to work as well, and I don't remember having to go through much > to get it working. I wonder if something has changed? > >> > >> -BDF > >> > >> On Fri, Mar 30, 2012 at 3:37 PM, Michael Collins > wrote: > >>> > >>> > >>> > >>> On Fri, Mar 30, 2012 at 12:20 PM, Brian Foster < > bdfoster at endigotech.com> wrote: > >>>> > >>>> I need to change the caller_id_name and caller_id_number. I've got > ATA's that don't like the plus in front of the number. At the same time, > I'm doing a cidlooup, so that's going to have to be changed as well. > >>>> > >>>> -BDF > >>>> > >>> > >>> It looks like PB 18786 doesn't have very much information. I'd > recommend getting the whole call from start to finish, and get the SIP > trace as well. Also, if you're creating a new leg with a bridge then you'll > need to export effective_caller_id_number or set it in the {} in front of > the dialstring. Lastly, be sure to use the sip_cid_type var to select which > field has the caller ID info in it. > >>> > >>> -MC > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> Brian D. Foster > >> Endigo Computer LLC > >> Email: bdfoster at endigotech.com > >> Phone: 317-800-7876 > >> Indianapolis, Indiana, USA > >> > >> This message contains confidential information and is intended for > those listed in the "To:", "CC:", and/or "BCC:" fields of the message > header. If you are not the intended recipient you are notified that > disclosing, copying, distributing or taking any action in reliance on the > contents of this information is strictly prohibited. E-mail transmission > cannot be guaranteed to be secure or error-free as information could be > intercepted, corrupted, lost, destroyed, arrive late or incomplete, or > contain viruses. The sender therefore does not accept liability for any > errors or omissions in the contents of this message, which arise as a > result of e-mail transmission. If verification is required please request a > hard-copy version. > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/a1a2e5ce/attachment-0001.html From msc at freeswitch.org Sat Mar 31 01:32:13 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 30 Mar 2012 14:32:13 -0700 Subject: [Freeswitch-users] Could not locate channel type skinny In-Reply-To: References: Message-ID: On Fri, Mar 30, 2012 at 1:20 PM, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > No, i've dissable 729 in freeswitch. > > My concert is why it fails only in incomming calls when using a > DID->ext mapping and only with that hardware phone, softphone works > nice. Wondering if you have a workarround. > > This has nothing to do with G729. You have something in your dialplan telling extension 1100 to use "skinny". Search for that in your dialplan and you'll probably find it. If 1100 is a registered user of your system then you want to do something like this: -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/649f11a6/attachment.html From kris at kriskinc.com Sat Mar 31 01:37:05 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 30 Mar 2012 17:37:05 -0400 Subject: [Freeswitch-users] When using nimbuzz, call disconnects after 90 seconds with RECOVERY_ON_TIMER_EXPIRE In-Reply-To: References: Message-ID: As a start try disabling session timers in your FreeSWITCH profile: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer On Fri, Mar 30, 2012 at 6:16 AM, Anton VG wrote: > Please help, > > While other SIP clients works well, there is a problem with nimbuzz > SIP client, which quite frequently used by 'not-advanced' users. seems > It which works through their services (asterisk 1.6) > > This problem was not happening while I have been using Asterisk too, > but after recent migration to FS - it does. > > nimbuzz client stays like online, just sound disappears, but there is > a confirmation from nimbuzz server side, regarding the BYE > > most likely a NAT issue, but if the same time using Asterisk on the > same server - there is not problems. So might be there any options to > tune? > > there are some SIP activity like this before calls disconnects, no replies: > > send 977 bytes to udp/[195.211.48.6]:12621 at 10:00:02.535856: > ? ------------------------------------------------------------------------ > ? INVITE sip:anton at 192.168.0.20:5061;transport=udp SIP/2.0 > ? Via: SIP/2.0/UDP 62.122.137.32;rport;branch=z9hG4bKaQm6tK1aeZ7Fe > ? Max-Forwards: 70 > ? From: "3777077" ;tag=eKv1Nej7ZrNUS > ? To: "anton at 62.122.137.32" ;tag=sip1331131792771 > ? Call-ID: 3696882418 > ? CSeq: 26214721 INVITE > ? Contact: > ? User-Agent: Synaptic Switch v2.1 > ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > ? Supported: timer, precondition, path, replaces > ? Session-Expires: 120;refresher=uac > ? Min-SE: 120 > ? Content-Type: application/sdp > ? Content-Disposition: session > ? Content-Length: 272 > > ? v=0 > ? o=FreeSWITCH 1333081454 1333081455 IN IP4 62.122.137.32 > ? s=FreeSWITCH > ? c=IN IP4 62.122.137.32 > ? t=0 0 > ? m=audio 20062 RTP/AVP 18 106 > ? a=rtpmap:18 G729/8000 > ? a=fmtp:18 annexb=no > ? a=rtpmap:106 telephone-event/8000 > ? a=fmtp:106 0-16 > ? a=silenceSupp:off - - - - > ? a=ptime:20 > > and than > > 2012-03-30 15:00:03.554460 [NOTICE] sofia.c:6301 Hangup > sofia/internal/anton at 62.122.137.32 [CS_SOFT_EXECUTE] > [RECOVERY_ON_TIMER_EXPIRE] > > recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960518: > ? ------------------------------------------------------------------------ > ? BYE sip:3777077 at 62.122.137.32:5060;transport=udp SIP/2.0 > ? Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 > ? Max-Forwards: 70 > ? To: ;tag=eKv1Nej7ZrNUS > ? From: ;tag=sip1331131792771 > ? Call-ID: 3696882418 > ? CSeq: 3 BYE > ? User-Agent: Nimbuzz Single > ? Content-Length: 0 > > ? ------------------------------------------------------------------------ > send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960672: > ? ------------------------------------------------------------------------ > ? SIP/2.0 481 Call Does Not Exist > ? Via: SIP/2.0/UDP > 192.168.0.20:5061;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 > ? From: ;tag=sip1331131792771 > ? To: ;tag=eKv1Nej7ZrNUS > ? Call-ID: 3696882418 > ? CSeq: 3 BYE > ? User-Agent: Synaptic Switch v2.1 > ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > ? Supported: timer, precondition, path, replaces > ? Content-Length: 0 > > ? ------------------------------------------------------------------------ > recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960734: > ? ------------------------------------------------------------------------ > ? BYE sip:3777077 at 62.122.137.32:5060;transport=udp SIP/2.0 > ? Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 > ? Max-Forwards: 70 > ? To: ;tag=eKv1Nej7ZrNUS > ? From: ;tag=sip1331131792771 > ? Call-ID: 3696882418 > ? CSeq: 3 BYE > ? User-Agent: Nimbuzz Single > ? Content-Length: 0 > > ? ------------------------------------------------------------------------ > send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960781: > ? ------------------------------------------------------------------------ > ? SIP/2.0 481 Call Does Not Exist > ? Via: SIP/2.0/UDP > 192.168.0.20:5061;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 > ? From: ;tag=sip1331131792771 > ? To: ;tag=eKv1Nej7ZrNUS > ? Call-ID: 3696882418 > ? CSeq: 3 BYE > ? User-Agent: Synaptic Switch v2.1 > ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > ? Supported: timer, precondition, path, replaces > ? Content-Length: 0 > > ? ------------------------------------------------------------------------ > recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079094: > ? ------------------------------------------------------------------------ > ? ACK sip:anton at 62.122.137.32 SIP/2.0 > ? Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 > ? Max-Forwards: 70 > ? To: ;tag=eKv1Nej7ZrNUS > ? From: ;tag=sip1331131792771 > ? Call-ID: 3696882418 > ? CSeq: 3 ACK > ? Contact: > ? Expires: 3600 > ? User-Agent: Nimbuzz Single > ? Content-Length: 0 > > ? ------------------------------------------------------------------------ > recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079293: > ? ------------------------------------------------------------------------ > ? ACK sip:anton at 62.122.137.32 SIP/2.0 > ? Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 > ? Max-Forwards: 70 > ? To: ;tag=eKv1Nej7ZrNUS > ? From: ;tag=sip1331131792771 > ? Call-ID: 3696882418 > ? CSeq: 3 ACK > ? Contact: > ? Expires: 3600 > ? User-Agent: Nimbuzz Single > ? Content-Length: 0 > > ? ------------------------------------------------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From djbinter at gmail.com Sat Mar 31 02:46:52 2012 From: djbinter at gmail.com (DJB International) Date: Fri, 30 Mar 2012 15:46:52 -0700 Subject: [Freeswitch-users] Trouble setting "caller_id_name", "caller_id_number" on an inbound call In-Reply-To: References: Message-ID: I don't think cidlookup would set anything. It's just use for looking up number to name mapping to the source of your preference. You just need to have it look up and return the variable that you need, then you can set or export to your need. -djbinter On Fri, Mar 30, 2012 at 1:56 PM, Brian Foster wrote: > > On Mar 30, 2012 4:47 PM, "DJB International" wrote: > > > > If you want to pass to B-leg with a new number format, then you'll need > to export effective_caller_id_number/effective_caller_id_name per MC. > > > > -djbinter > > Both of those variables are automatically exported by default. No need to > export those. Just set and forget! > > I would like to point out that this: > > > > ...does not work. I think this is a bug. It rubs fine but it might just be > setting the wrong variable. I'll take that to JIRA. > > -BDF > > > > > On Fri, Mar 30, 2012 at 1:06 PM, Brian Foster > wrote: > >> > >> http://pastebin.freeswitch.org/18791 << Entire Call > >> > >> It seems like a pretty common question, and there doesn't seem to be a > good answer to just cleaning up the caller id and doing a lookup. Even the > documentation on the wiki is wrong. To me, if anything, I'd like to get > something that's correct on the wiki in order to facilitate new users. I've > gotten this to work as well, and I don't remember having to go through much > to get it working. I wonder if something has changed? > >> > >> -BDF > >> > >> On Fri, Mar 30, 2012 at 3:37 PM, Michael Collins > wrote: > >>> > >>> > >>> > >>> On Fri, Mar 30, 2012 at 12:20 PM, Brian Foster < > bdfoster at endigotech.com> wrote: > >>>> > >>>> I need to change the caller_id_name and caller_id_number. I've got > ATA's that don't like the plus in front of the number. At the same time, > I'm doing a cidlooup, so that's going to have to be changed as well. > >>>> > >>>> -BDF > >>>> > >>> > >>> It looks like PB 18786 doesn't have very much information. I'd > recommend getting the whole call from start to finish, and get the SIP > trace as well. Also, if you're creating a new leg with a bridge then you'll > need to export effective_caller_id_number or set it in the {} in front of > the dialstring. Lastly, be sure to use the sip_cid_type var to select which > field has the caller ID info in it. > >>> > >>> -MC > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> Brian D. Foster > >> Endigo Computer LLC > >> Email: bdfoster at endigotech.com > >> Phone: 317-800-7876 > >> Indianapolis, Indiana, USA > >> > >> This message contains confidential information and is intended for > those listed in the "To:", "CC:", and/or "BCC:" fields of the message > header. If you are not the intended recipient you are notified that > disclosing, copying, distributing or taking any action in reliance on the > contents of this information is strictly prohibited. E-mail transmission > cannot be guaranteed to be secure or error-free as information could be > intercepted, corrupted, lost, destroyed, arrive late or incomplete, or > contain viruses. The sender therefore does not accept liability for any > errors or omissions in the contents of this message, which arise as a > result of e-mail transmission. If verification is required please request a > hard-copy version. > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/8456bf08/attachment-0001.html From bob.mccarthy at experient.com Sat Mar 31 03:24:01 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Fri, 30 Mar 2012 17:24:01 -0600 Subject: [Freeswitch-users] Setting Hangup Hook on api originate In-Reply-To: References: <4f75e17e.084cec0a.594e.ffff9d3bSMTPIN_ADDED@mx.google.com> Message-ID: <01eb01cd0ecc$321e82b0$965b8810$@mccarthy@experient.com> I just love it when it is simple :) thanks Michael . From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, March 30, 2012 2:00 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Setting Hangup Hook on api originate On Fri, Mar 30, 2012 at 9:37 AM, Bob McCarthy wrote: I am having trouble getting a hangup hook to execute after an api originate. bgapi expand originate {api_hangup_hook=confhanguphook.lua,session_in_hangup_hook=true}sofia/intern al/2001@${switchvox} &conference(25bce61d-fb32-44d7-8d99-7ceafc599f22 at sla) I get into the conference just fine but the script does not execute when I hang up. Any Idea's of what I am doing wrong? I have an idea: how about actually calling 'lua' and not just the name of your script? ;) bgapi expand originate {api_hangup_hook='lua confhanguphook.lua',session_in_hangup_hook=true}sofia/internal/2001@${switch vox} &conference(25bce61d-fb32-44d7-8d99-7ceafc599f22 at sla) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/9057c8b2/attachment.html From bdfoster at endigotech.com Sat Mar 31 06:03:18 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 30 Mar 2012 22:03:18 -0400 Subject: [Freeswitch-users] Trouble setting "caller_id_name", "caller_id_number" on an inbound call In-Reply-To: References: Message-ID: I can confirm this did work with an older version. I have since updated because of some changes to spandsp and forgot to grab the revision ID (stoopid me!). It would make sense that running cidlookup as an application would automatically set the variable, since I'm pretty sure it doesn't return anything to the dialplan so you can do what you want with the lookup UNLESS you run it as an API call. Anyway, bug was filed. -BDF On Mar 30, 2012 6:48 PM, "DJB International" wrote: > I don't think cidlookup would set anything. It's just use for looking up > number to name mapping to the source of your preference. > > You just need to have it look up and return the variable that you need, > then you can set or export to your need. > > -djbinter > > On Fri, Mar 30, 2012 at 1:56 PM, Brian Foster wrote: > >> >> On Mar 30, 2012 4:47 PM, "DJB International" wrote: >> > >> > If you want to pass to B-leg with a new number format, then you'll need >> to export effective_caller_id_number/effective_caller_id_name per MC. >> > >> > -djbinter >> >> Both of those variables are automatically exported by default. No need to >> export those. Just set and forget! >> >> I would like to point out that this: >> >> >> >> ...does not work. I think this is a bug. It rubs fine but it might just >> be setting the wrong variable. I'll take that to JIRA. >> >> -BDF >> >> > >> > On Fri, Mar 30, 2012 at 1:06 PM, Brian Foster >> wrote: >> >> >> >> http://pastebin.freeswitch.org/18791 << Entire Call >> >> >> >> It seems like a pretty common question, and there doesn't seem to be a >> good answer to just cleaning up the caller id and doing a lookup. Even the >> documentation on the wiki is wrong. To me, if anything, I'd like to get >> something that's correct on the wiki in order to facilitate new users. I've >> gotten this to work as well, and I don't remember having to go through much >> to get it working. I wonder if something has changed? >> >> >> >> -BDF >> >> >> >> On Fri, Mar 30, 2012 at 3:37 PM, Michael Collins >> wrote: >> >>> >> >>> >> >>> >> >>> On Fri, Mar 30, 2012 at 12:20 PM, Brian Foster < >> bdfoster at endigotech.com> wrote: >> >>>> >> >>>> I need to change the caller_id_name and caller_id_number. I've got >> ATA's that don't like the plus in front of the number. At the same time, >> I'm doing a cidlooup, so that's going to have to be changed as well. >> >>>> >> >>>> -BDF >> >>>> >> >>> >> >>> It looks like PB 18786 doesn't have very much information. I'd >> recommend getting the whole call from start to finish, and get the SIP >> trace as well. Also, if you're creating a new leg with a bridge then you'll >> need to export effective_caller_id_number or set it in the {} in front of >> the dialstring. Lastly, be sure to use the sip_cid_type var to select which >> field has the caller ID info in it. >> >>> >> >>> -MC >> >>> >> >>> >> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >> >> >> >> >> >> >> -- >> >> Brian D. Foster >> >> Endigo Computer LLC >> >> Email: bdfoster at endigotech.com >> >> Phone: 317-800-7876 >> >> Indianapolis, Indiana, USA >> >> >> >> This message contains confidential information and is intended for >> those listed in the "To:", "CC:", and/or "BCC:" fields of the message >> header. If you are not the intended recipient you are notified that >> disclosing, copying, distributing or taking any action in reliance on the >> contents of this information is strictly prohibited. E-mail transmission >> cannot be guaranteed to be secure or error-free as information could be >> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or >> contain viruses. The sender therefore does not accept liability for any >> errors or omissions in the contents of this message, which arise as a >> result of e-mail transmission. If verification is required please request a >> hard-copy version. >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120330/489b1f51/attachment-0001.html From anton.vazir at gmail.com Sat Mar 31 09:47:38 2012 From: anton.vazir at gmail.com (Anton VG) Date: Sat, 31 Mar 2012 10:47:38 +0500 Subject: [Freeswitch-users] When using nimbuzz, call disconnects after 90 seconds with RECOVERY_ON_TIMER_EXPIRE In-Reply-To: References: Message-ID: Thanks for help guys! Disabling SIP timers solves this problem. But I suppose it can raise other problems in other cases. Michael, do you think collecting of SIP headers still reasonable, or Asterisk just ignores or do not have any specific SIP session timer? Can I enable/disable timers per session rather per profile (say via the channel variable while 'originating'? in this case I can check if nimbuss is a UA and than disable timers? 2012/3/31 Kristian Kielhofner : > As a start try disabling session timers in your FreeSWITCH profile: > > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer > > On Fri, Mar 30, 2012 at 6:16 AM, Anton VG wrote: >> Please help, >> >> While other SIP clients works well, there is a problem with nimbuzz >> SIP client, which quite frequently used by 'not-advanced' users. seems >> It which works through their services (asterisk 1.6) >> >> This problem was not happening while I have been using Asterisk too, >> but after recent migration to FS - it does. >> >> nimbuzz client stays like online, just sound disappears, but there is >> a confirmation from nimbuzz server side, regarding the BYE >> >> most likely a NAT issue, but if the same time using Asterisk on the >> same server - there is not problems. So might be there any options to >> tune? >> >> there are some SIP activity like this before calls disconnects, no replies: >> >> send 977 bytes to udp/[195.211.48.6]:12621 at 10:00:02.535856: >> ? ------------------------------------------------------------------------ >> ? INVITE sip:anton at 192.168.0.20:5061;transport=udp SIP/2.0 >> ? Via: SIP/2.0/UDP 62.122.137.32;rport;branch=z9hG4bKaQm6tK1aeZ7Fe >> ? Max-Forwards: 70 >> ? From: "3777077" ;tag=eKv1Nej7ZrNUS >> ? To: "anton at 62.122.137.32" ;tag=sip1331131792771 >> ? Call-ID: 3696882418 >> ? CSeq: 26214721 INVITE >> ? Contact: >> ? User-Agent: Synaptic Switch v2.1 >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> ? Supported: timer, precondition, path, replaces >> ? Session-Expires: 120;refresher=uac >> ? Min-SE: 120 >> ? Content-Type: application/sdp >> ? Content-Disposition: session >> ? Content-Length: 272 >> >> ? v=0 >> ? o=FreeSWITCH 1333081454 1333081455 IN IP4 62.122.137.32 >> ? s=FreeSWITCH >> ? c=IN IP4 62.122.137.32 >> ? t=0 0 >> ? m=audio 20062 RTP/AVP 18 106 >> ? a=rtpmap:18 G729/8000 >> ? a=fmtp:18 annexb=no >> ? a=rtpmap:106 telephone-event/8000 >> ? a=fmtp:106 0-16 >> ? a=silenceSupp:off - - - - >> ? a=ptime:20 >> >> and than >> >> 2012-03-30 15:00:03.554460 [NOTICE] sofia.c:6301 Hangup >> sofia/internal/anton at 62.122.137.32 [CS_SOFT_EXECUTE] >> [RECOVERY_ON_TIMER_EXPIRE] >> >> recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960518: >> ? ------------------------------------------------------------------------ >> ? BYE sip:3777077 at 62.122.137.32:5060;transport=udp SIP/2.0 >> ? Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 >> ? Max-Forwards: 70 >> ? To: ;tag=eKv1Nej7ZrNUS >> ? From: ;tag=sip1331131792771 >> ? Call-ID: 3696882418 >> ? CSeq: 3 BYE >> ? User-Agent: Nimbuzz Single >> ? Content-Length: 0 >> >> ? ------------------------------------------------------------------------ >> send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960672: >> ? ------------------------------------------------------------------------ >> ? SIP/2.0 481 Call Does Not Exist >> ? Via: SIP/2.0/UDP >> 192.168.0.20:5061;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 >> ? From: ;tag=sip1331131792771 >> ? To: ;tag=eKv1Nej7ZrNUS >> ? Call-ID: 3696882418 >> ? CSeq: 3 BYE >> ? User-Agent: Synaptic Switch v2.1 >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> ? Supported: timer, precondition, path, replaces >> ? Content-Length: 0 >> >> ? ------------------------------------------------------------------------ >> recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960734: >> ? ------------------------------------------------------------------------ >> ? BYE sip:3777077 at 62.122.137.32:5060;transport=udp SIP/2.0 >> ? Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 >> ? Max-Forwards: 70 >> ? To: ;tag=eKv1Nej7ZrNUS >> ? From: ;tag=sip1331131792771 >> ? Call-ID: 3696882418 >> ? CSeq: 3 BYE >> ? User-Agent: Nimbuzz Single >> ? Content-Length: 0 >> >> ? ------------------------------------------------------------------------ >> send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960781: >> ? ------------------------------------------------------------------------ >> ? SIP/2.0 481 Call Does Not Exist >> ? Via: SIP/2.0/UDP >> 192.168.0.20:5061;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 >> ? From: ;tag=sip1331131792771 >> ? To: ;tag=eKv1Nej7ZrNUS >> ? Call-ID: 3696882418 >> ? CSeq: 3 BYE >> ? User-Agent: Synaptic Switch v2.1 >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> ? Supported: timer, precondition, path, replaces >> ? Content-Length: 0 >> >> ? ------------------------------------------------------------------------ >> recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079094: >> ? ------------------------------------------------------------------------ >> ? ACK sip:anton at 62.122.137.32 SIP/2.0 >> ? Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 >> ? Max-Forwards: 70 >> ? To: ;tag=eKv1Nej7ZrNUS >> ? From: ;tag=sip1331131792771 >> ? Call-ID: 3696882418 >> ? CSeq: 3 ACK >> ? Contact: >> ? Expires: 3600 >> ? User-Agent: Nimbuzz Single >> ? Content-Length: 0 >> >> ? ------------------------------------------------------------------------ >> recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079293: >> ? ------------------------------------------------------------------------ >> ? ACK sip:anton at 62.122.137.32 SIP/2.0 >> ? Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 >> ? Max-Forwards: 70 >> ? To: ;tag=eKv1Nej7ZrNUS >> ? From: ;tag=sip1331131792771 >> ? Call-ID: 3696882418 >> ? CSeq: 3 ACK >> ? Contact: >> ? Expires: 3600 >> ? User-Agent: Nimbuzz Single >> ? Content-Length: 0 >> >> ? ------------------------------------------------------------------------ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sharad at coraltele.com Sat Mar 31 10:21:21 2012 From: sharad at coraltele.com (Sharad Garg) Date: Sat, 31 Mar 2012 11:51:21 +0530 Subject: [Freeswitch-users] Session-wise Language Setting References: <4CB4E6075A9F4F9BA1C8A57FF2A06735@sharad><4F746EFF.3050305@communicatefreely.net><00C2DD790B254A1BACD50CAA66B123B9@sharad><1FFF97C269757C458224B7C895F35F15083A49@cantor.std.visionutv.se><12BA6FAF92674DEE9650CF4829B95DF0@sharad> Message-ID: <13D710C546A14C09801E1CC46A394BB4@sharad> Hello Mr. MC Thanks for the link.....seems working. But observed another point i.e. it works only for the speech phrase macros. When I write a simple xml dialplan like this - This dialplan still plays the test.wav which is in /sounds/en/us/callie/ivr/8000/test.wav. While it should play the test.wav from /sounds/ru/RU/elena/ivr/8000/test.wav. & at the same time, speech phrase macro use the 'ru' language. So when I did my googling, I found that the language settings is only for speech phrase macros. It is not implemented in Freeswitch to use the desired languages at all places. So need to know is there any latest development on this ? Thanks in advance. Regards Sharad ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Saturday, March 31, 2012 1:15 AM Subject: Re: [Freeswitch-users] Session-wise Language Setting On Fri, Mar 30, 2012 at 5:13 AM, Sharad Garg wrote: Thanks Peter...yes it works. Just a last query - Is there any API for setting the same so that it can be used from Javascript. Best Regards Sharad http://wiki.freeswitch.org/wiki/Mod_commands#uuid_setvar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120331/aca42c4b/attachment.html From gabe at gundy.org Sat Mar 31 12:11:37 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 31 Mar 2012 02:11:37 -0600 Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls In-Reply-To: References: Message-ID: On Fri, Mar 30, 2012 at 2:16 AM, Daniel Knaggs < Daniel.Knaggs at realitysolutions.co.uk> wrote: > Any ideas, anyone? I'll be honest. I haven't followed the thread because of the loud, in-your-face advertisement in the email footer. Nobody is going to make you remove it, but they might respond to more of your emails if you show them that you respect the list by not cluttering it with noise. Just my 2 cents. And of course good luck on the issue. Gabe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120331/62b0e76d/attachment.html From tang.du at hotmail.com Sat Mar 31 19:23:37 2012 From: tang.du at hotmail.com (tangdu) Date: Sat, 31 Mar 2012 08:23:37 -0700 (PDT) Subject: [Freeswitch-users] Skypopen problem on centos5.8 In-Reply-To: References: <1331439760882-7362245.post@n2.nabble.com> Message-ID: <1333207417157-7424987.post@n2.nabble.com> THANKS FOR YOUR HELP! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Skypopen-problem-on-centos5-8-tp7362245p7424987.html Sent from the freeswitch-users mailing list archive at Nabble.com.