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Re: [Freeswitch-users] freeswitch in vmware UPDATE
by Anthony Minessale 12 Mar '08

12 Mar '08

I spoke too soon, I mean try it now =D revision 7857! Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: John Skopis (Lists) <jlists(a)skopis.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 11, 2008 7:35:00 AM Subject: Re: [Freeswitch-users] freeswitch in vmware UPDATE Anthony Minessale wrote: > can everyone who has any audio problems please > retest with after doing "make current" or a fresh checkout. > [snip] using Revision: 7855 no dice...same as before. I could try contacting vmware but I doubt we would get very far _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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woomera specific purpose (or maybe opal)
by Josip Djuricic 11 Mar '08

11 Mar '08

Hi there, I come to you again with very specific question. Is there a possibility to make woomera or opal work with freeswitch, cause I need it for specific purpose only. I don't need the call to work at all, what I would need would be if a call comes to woomera channel, just drop the call and generate specific release code. It is required for specific billing purpose in combination with mvts. Thank you very much. btw. about that other problem I had with incoming calls rtp not working, it was a nat problem. Josip

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freeswitch in vmware
by Cavalera Claudio Luigi 11 Mar '08

11 Mar '08

Hello guys, at the moment the only linux I've got handy is a slackware 12.0 under vmware server 1.04 I've installed freeswitch to start a sort of comparison with asterisk. I've registered a SIP 1000 user and dialed 9999 (I could not wait :-). The config is default. The music is choppy, I mean silence - music - silence within a second! Sometimes it sounds good for ten seconds or so. Could this be due to clock skew problems of virtual machines? 008-03-04 12:43:24 [CRIT] switch_time.c:321 softtimer_runtime() Forward Clock Skew Detected! Best Regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable
by Anthony Minessale 11 Mar '08

11 Mar '08

nonetheless, can you please see my bugnote in jira. ===== can you edit /usr/local/freeswitch/conf/autoload_configs/iax.conf.xml and set the debug param to 1. Then reproduce the case where it's not working. We can see if the mod_iax is even getting the digits. ===== also be sure to sync to the current release by entering "make current" from the build root just to ensure your problem still exists in the latest revision. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Birgit Arkesteijn <birgit(a)westhawk.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 11, 2008 9:36:22 AM Subject: Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable Hi Anthony, I'm fairly sure that it's not an IAX problem. IAX2 messages are properly acked, sound (between IAX2 and SIP) is ok, etc, etc. The problem is (only) with DTMF, as far as I can tell. It's more likely to be in core or sofia_sip? Thanks, Birgit On 11/03/08 13:20, Anthony Minessale wrote: > Just to warn you we do not use mod_iax all that much. It was used as a > tool to test the core of FreeSWITCH in the early stages of development > and we just kinda left it hanging around since it does at least work. > We had to add a bunch of code to the stock iaxclient lib to make it > threadsafe etc because the lib is designed to handle 1 or 2 calls and we > are using it in a b2bua type env where hundreds of concurrent calls may > be possible. That said, since we more or less forked the code to avoid > generating support issues for the other soft-phones that use iaxclient, > you may want to check if there were any bugs in the real version of the > library that match your issue. > > http://sourceforge.net/projects/iaxclient > > > We've had little to no demand for IAX2 so we have not focused very much > on it. We do have strategies for creating a more robust and perhaps even > an "improved" implementation of the IAX2 protocol. Alas, it's pretty > much purely an academic exercise because there is little reason to > implement it. > > > > > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Birgit Arkesteijn <birgit(a)westhawk.co.uk> > To: Freeswitch-users(a)lists.freeswitch.org > Sent: Monday, March 10, 2008 10:52:40 AM > Subject: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable > > Hi, > > We're running FreeSWITCH svn Trunk 7697 on SuSE Linux 9.3 (x86-64). > > I'm experiencing problems with DTMF: > The IAX2 DTMF frames aren't always transferred into RTP packets. > Sometimes none of the frames are translated, sometimes only the first > one is omitted. All the IAX2 DTMF packets are ack-ed. > > Our setup: > PhoneFromHere.com <http://PhoneFromHere.com> applet (IAX2 protocol) > -> FreeSWITCH (ns2.westhawk.co.uk <http://ns2.westhawk.co.uk>) > -> SIP to SystemX (sofia/default/123(a)66.212.134.192) > > > I searched on jira.freeswitch.org <http://jira.freeswitch.org> for a bug > that could explain this, but > failed, so I submitted this as a bug, see > http://jira.freeswitch.org/browse/FSCORE-109 > > Has anyone else experienced the same or a similar problem? > > Thanks, Birgit > -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk> _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable
by Anthony Minessale 11 Mar '08

11 Mar '08

Just to warn you we do not use mod_iax all that much. It was used as a tool to test the core of FreeSWITCH in the early stages of development and we just kinda left it hanging around since it does at least work. We had to add a bunch of code to the stock iaxclient lib to make it threadsafe etc because the lib is designed to handle 1 or 2 calls and we are using it in a b2bua type env where hundreds of concurrent calls may be possible. That said, since we more or less forked the code to avoid generating support issues for the other soft-phones that use iaxclient, you may want to check if there were any bugs in the real version of the library that match your issue. http://sourceforge.net/projects/iaxclient We've had little to no demand for IAX2 so we have not focused very much on it. We do have strategies for creating a more robust and perhaps even an "improved" implementation of the IAX2 protocol. Alas, it's pretty much purely an academic exercise because there is little reason to implement it. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Birgit Arkesteijn <birgit(a)westhawk.co.uk> To: Freeswitch-users(a)lists.freeswitch.org Sent: Monday, March 10, 2008 10:52:40 AM Subject: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable Hi, We're running FreeSWITCH svn Trunk 7697 on SuSE Linux 9.3 (x86-64). I'm experiencing problems with DTMF: The IAX2 DTMF frames aren't always transferred into RTP packets. Sometimes none of the frames are translated, sometimes only the first one is omitted. All the IAX2 DTMF packets are ack-ed. Our setup: PhoneFromHere.com applet (IAX2 protocol) -> FreeSWITCH (ns2.westhawk.co.uk) -> SIP to SystemX (sofia/default/123(a)66.212.134.192) I searched on jira.freeswitch.org for a bug that could explain this, but failed, so I submitted this as a bug, see http://jira.freeswitch.org/browse/FSCORE-109 Has anyone else experienced the same or a similar problem? Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk> _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable
by Anthony Minessale 11 Mar '08

11 Mar '08

Just to warn you we do not use mod_iax all that much. It was used as a tool to test the core of FreeSWITCH in the early stages of development and we just kinda left it hanging around since it does at least work. We had to add a bunch of code to the stock iaxclient lib to make it threadsafe etc because the lib is designed to handle 1 or 2 calls and we are using it in a b2bua type env where hundreds of concurrent calls may be possible. That said, since we more or less forked the code to avoid generating support issues for the other soft-phones that use iaxclient, you may want to check if there were any bugs in the real version of the library that match your issue. http://sourceforge.net/projects/iaxclient We've had little to no demand for IAX2 so we have not focused very much on it. We do have strategies for creating a more robust and perhaps even an "improved" implementation of the IAX2 protocol. Alas, it's pretty much purely an academic exercise because there is little reason to implement it. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Birgit Arkesteijn <birgit(a)westhawk.co.uk> To: Freeswitch-users(a)lists.freeswitch.org Sent: Monday, March 10, 2008 10:52:40 AM Subject: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable Hi, We're running FreeSWITCH svn Trunk 7697 on SuSE Linux 9.3 (x86-64). I'm experiencing problems with DTMF: The IAX2 DTMF frames aren't always transferred into RTP packets. Sometimes none of the frames are translated, sometimes only the first one is omitted. All the IAX2 DTMF packets are ack-ed. Our setup: PhoneFromHere.com applet (IAX2 protocol) -> FreeSWITCH (ns2.westhawk.co.uk) -> SIP to SystemX (sofia/default/123(a)66.212.134.192) I searched on jira.freeswitch.org for a bug that could explain this, but failed, so I submitted this as a bug, see http://jira.freeswitch.org/browse/FSCORE-109 Has anyone else experienced the same or a similar problem? Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk> _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Re: [Freeswitch-users] freeswitch in vmware UPDATE
by Anthony Minessale 11 Mar '08

11 Mar '08

can everyone who has any audio problems please retest with after doing "make current" or a fresh checkout. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: John Skopis (Lists) <jlists(a)skopis.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, March 7, 2008 8:00:31 AM Subject: Re: [Freeswitch-users] freeswitch in vmware Hello, Please see my comments inline. Cavalera Claudio Luigi wrote: >> The problem is that in bridged network mode RTP packets from my sip >> phone (on the host system) were seen twice by the guest (same >> sequence >> number, same packet really). MikeJ pointed this out so I >> switched to NAT >> mode, which fixed it for me. To keep your clock in sync just >> boot with >> 'clock=pit' and run ntpd. > > > I have done as you suggested to fix the clock (ntpd plus clock=pit) and > it seems to work. > The problem of choppiness remain, I'm not sure if bridge network is the > problem because I've tried to connect from other machine to the guest > running freeswitch and the problem is the same. > > Thanks to everyone for suggestions so far! > Claudio > I found this link, perhaps it is helpful: http://www.vmware.com/support/ws5/doc/ws_net_advanced_2hostonly_routing.html I am working on setting up NAT on the host system using IPFW. I am sure you could do the same (or similar w/ netfilter). I haven't quite figured out what exactly I want to do yet but I would imagine that it involves natting both 1:1, and regular NAT. > >>>> MikeJ helped to figured out that the culprit is actually >> the vmware >>>> bridge. Change the network type to NAT and it will "just >> work". Mike >>>> says that if the soft phone was not on the same box as the >>>> host system >>>> that it would work a little better. I will test with a >> native linux >>>> bridge and xen in a few days. =] >>>> >>>> -John >>> >>> But if I changed network type to NAT how can I reach the freeswitch >>> server inside vmware? >>> Isn't that the same as reaching a private host behind a home nat dsl >>> router? >>> >>> Best regards, >>> Claudio >>> >> [snip] >> >> Sorry, but I can't be much of a help here. I believe there is a file >> /etc/vmware/vmnet8 or so that allows you to configure port forwarding >> inside the VM. I agree dealing with nat even once (let alone >> twice) is a >> big fat PITA. Perhaps you can figure out how to run VMware in >> "routed" >> mode. I would prefer this as I could then run NAT on the host system >> doing 1:1 NAT or on the router itself. >> >> Also, and anthm can explain better but if you add something like: >> <action application="set" data="timer_name=soft" /> >> >> to extension 9999 for example (before the playback). The >> choppyness will >> disappear. From what I understood freeswitch gets its timing >> information >> from the client and if the client is sending every packet twice it >> confuses freeswitch. >> >> Actually another solution/work-around (without modifying the >> rtp code) >> would be to run ebtables on your linux box and filter the erronous >> packet, on my system it comes from some cisco mac address. >> >> >> HTH, >> John >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >> itch-users >> http://www.freeswitch.org > > > Internet Email Confidentiality Footer > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)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(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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hi there
by Josip Djuricic 10 Mar '08

10 Mar '08

Hello, I'm testing freeswitch with AS5300 for inbound and outbound call's. Does anyone have any info on how to disable Proxy authentication, because when the AS5300 sends the call to freeswitch, debug on AS said: Received: SIP/2.0 407 Proxy Authentication Required. I tried to turn off gateway authorization, and setting authless authorization, but it just doesn't work. Thanks

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IAX2 DTMF -> SIP RTP not reliable
by Birgit Arkesteijn 10 Mar '08

10 Mar '08

Hi, We're running FreeSWITCH svn Trunk 7697 on SuSE Linux 9.3 (x86-64). I'm experiencing problems with DTMF: The IAX2 DTMF frames aren't always transferred into RTP packets. Sometimes none of the frames are translated, sometimes only the first one is omitted. All the IAX2 DTMF packets are ack-ed. Our setup: PhoneFromHere.com applet (IAX2 protocol) -> FreeSWITCH (ns2.westhawk.co.uk) -> SIP to SystemX (sofia/default/123(a)66.212.134.192) I searched on jira.freeswitch.org for a bug that could explain this, but failed, so I submitted this as a bug, see http://jira.freeswitch.org/browse/FSCORE-109 Has anyone else experienced the same or a similar problem? Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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Latest FS - problems with allocating ports
by David Knell 10 Mar '08

10 Mar '08

Hi all - Upgraded to the latest FS a day or so ago, and I've had a couple of issues with the thing not being able to allocate ports for RTP. Symptom is that a call comes in and is immediately dropped; console error is: [CRIT] sofia_glue.c:498 sofia_glue_tech_choose_port() No RTP ports available! netstat -an | grep udp shows just a few ports in use. sofia status shows nothing untowards. This has happened for all calls, effectively taking the switch down completely, and has happened twice, each time after the box has been up for a while. Anyone else seen this, or is it just me? --Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk

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