March 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Re: [Freeswitch-users] freeswitch in vmware
by Anthony Minessale 10 Mar '08

10 Mar '08

It was confirmed by 2 people that using it in bridged mode is the problem. Can you change vmware networking to NAT mode and tell me if it works for you? 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: Cavalera Claudio Luigi <Claudio.Cavalera(a)italtel.it> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, March 7, 2008 2:39:45 AM Subject: Re: [Freeswitch-users] freeswitch in vmware > 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 > >> 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 ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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Scalable voip providers
by Daniel Swarbrick 10 Mar '08

10 Mar '08

I'm working on a project that requires pstn "break-in" (ie, forwarding pstn numbers to a sip trunk) in the UK, to a Freeswitch box that will also use a (possibly different) sip trunk provider to dial numbers in the US. In the UK, we will eventually need contiguous blocks of perhaps a few hundred numbers (although most numbers will just step to an available channel), and may handle up to a few hundred simultaneous calls. Likewise, the sip trunk provider for outbound dialling needs to support potentially up to a few hundred calls, although these call volumes won't be there initially - more like probably about 20-30 simultaneous calls. I've done a fair bit of searching, but most voip providers seem to cater for home users who just want to replace their landline. What I need is probably bordering on wholesale providers. If the UK provider could arrange the number blocks too, that would be even better. Can anyone recommend a provider or providers that would fit the requirements above, and that plays nicely with Freeswitch? If you're wondering if this is a toll-bypass voip provider project that may infringe your own business, don't worry. It's a distributed call centre business.

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Re: [Freeswitch-users] Intercom
by Anthony Minessale 09 Mar '08

09 Mar '08

file a bug in jira i know how to fix it but i keep forgetting 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: andrew <andrew_nh(a)yahoo.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, March 6, 2008 1:51:33 PM Subject: [Freeswitch-users] Intercom I want to get intercom working on FreeSWITCH with an Aastra phone. There is some code that does this already in the default config files. ---------------------------------- <action application="export"><![CDATA[sip_h_Call-Info=<sip:$${domain}>;answer-after=0]]></action> <action application="export" data="sip_auto_answer=true"/> <action application="bridge" data="user/${dialed_ext}@$${domain}"/> ---------------------------------- But it doesn’t seem to work on the Aastra. I have this code in Asterisk that does intercom my Aastra phone ---------------------------------- exten => s,n,Set(ALERTINFO=Alert-Info: Ring Answer) exten => s,n,Set(CALLINFO=Call-Info: <uri>\;answer-after=0) exten => s,n,Set(SIPURI=intercom=true) exten => s,n,ExecIf($["${ALERTINFO}" != ""],SipAddHeader,${ALERTINFO}) exten => s,n,ExecIf($["${CALLINFO}" != ""],SipAddHeader,${CALLINFO}) exten => s,n,ExecIf($["${SIPURI}" != ""],Set,__SIP_URI_OPTIONS=${SIPURI}) ---------------------------------- Looks like I need to get the "Alert-Info: Ring Answer" and the "set sip uri options intercom=true" parts working in FreeSWITCH. Any idea how to do this? -Andrew ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ _______________________________________________ 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] Application needed
by Anthony Minessale 08 Mar '08

08 Mar '08

Find me monday on IRC. 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: David Knell <dave(a)3c.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, March 8, 2008 2:22:49 PM Subject: [Freeswitch-users] Application needed Hi - I'm after a FreeSWITCH app which will do the following: 1. Call comes in. 2. Outbound call is originated to the same number via a gateway 3. If we don't get ringback (a SIP 183) from the outbound leg within a timeout, return a 480 to the original inbound caller 4. Having got a 183 from the outbound leg, start tone detection to detect ringtone (425Hz single tone in this case, but it'd be good to be able to detect dual tones as well) 5. If we hear ringtone within a timeout period, then send a 183 to the original caller and bridge the calls together; the calls then proceed as normal 6. If we don't, then return a 480 to the original caller and dump the outbound call. If, at any time, the outbound call is answered, it needs to be immediately bridged to the inbound one. I'm able to offer a reasonably substantial bribe for this - if you'd like to have a go at claiming it, please let me know how large a bribe you'd need, how long it'll take you and don't start until it's agreed..! Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.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] Application needed
by Anthony Minessale 08 Mar '08

08 Mar '08

Find me monday on IRC. 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: David Knell <dave(a)3c.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, March 8, 2008 2:22:49 PM Subject: [Freeswitch-users] Application needed Hi - I'm after a FreeSWITCH app which will do the following: 1. Call comes in. 2. Outbound call is originated to the same number via a gateway 3. If we don't get ringback (a SIP 183) from the outbound leg within a timeout, return a 480 to the original inbound caller 4. Having got a 183 from the outbound leg, start tone detection to detect ringtone (425Hz single tone in this case, but it'd be good to be able to detect dual tones as well) 5. If we hear ringtone within a timeout period, then send a 183 to the original caller and bridge the calls together; the calls then proceed as normal 6. If we don't, then return a 480 to the original caller and dump the outbound call. If, at any time, the outbound call is answered, it needs to be immediately bridged to the inbound one. I'm able to offer a reasonably substantial bribe for this - if you'd like to have a go at claiming it, please let me know how large a bribe you'd need, how long it'll take you and don't start until it's agreed..! Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.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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