[Freeswitch-users] No audio after Remote SDP:
Hi! I'm sure this is a NAT issue, but I'm not sure what options to use. I have a Linksys SPA3102, NAT'd on the internet (remotely) and connected to my FS on the otherside of the world, which is also natted. A PAP2T is connected on the same subnet as the FS. The 3102 registers successfully and a call can be set up from the PAP2 to the 3102. However, after FS receives the Remote SDP the audio stops (ring tone stops in my case) 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3646 Channel sofia/internal/sip:2...@192.168.1.3:56885 entering state [completing][200] 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3657 Remote SDP: v=0 o=- 18490612 18490612 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 16432 RTP/AVP 2 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 I notice that the ip address in the o and c fields indicate a local IP address. Should this IP address be an external IP address of the 3102 instead? Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] No audio after Remote SDP:
Thanks Brian and Gad, I have stun set and if I do a 'sofia status profile internal', I see the external IP address of the 3102 ATA, so I assume that stun is working correctly on the SPA3102. These are the options that I have set (according to the 3102 manual). • Handle VIA received: yes • Handle VIA rport: yes • Insert VIA received: yes • Insert VIA rport: yes • Substitute VIA Addr: yes • Send Resp To Src Port: yes • STUN Enable: Choose yes. • STUN Server: stun.freeswitch.org I assume that is all is needed? On Mon, Dec 21, 2009 at 9:36 AM, Brian West br...@freeswitch.org wrote: On Dec 20, 2009, at 2:53 PM, Gad Bentolila wrote: DISCLAIMER: I'm REALLY new to FreeSwitch, so please take my advice with a grain of salt. Welcome to the community. I have a similar setup (and problem) - the wiki documentation refers to it as double nat. Like you, my FS and client are behind different NATs and I can register my remote endpoint and make calls (in my case, to the the FS demo ivr at 5000). Since your external endpoint (spa3102) is registering, you've likely setup your sip profile correctly (ext-sip-ip, ext-rtp-ip, nat settings, etc). Your endpoint need only insert rport and FreeSWITCH will do the right thing. 1) Setup stun on your remote endpoint (spa3102 in your case) 2) Add variable name=sip-force-contact value=NDLB-connectile-dysfunction/ to the directory xml file that describes your spa3102 endpoint The device supports STUN also its highly recommended your device know how to overcome its own NAT. I personally do not believe its the registrars place to overcome an endpoints nat... puts undue burden on the registar. Option 1 worked for me right away (eyebeam in my case) and, as expected, the remote sdp had the correct (remote) IP address, since the endpoint is using stun to correctly identify its IP address to FS. However, option 2 has not made a difference (for me). Is it just me or is it strange that SIP works without stun, but RTP doesn't? I guess I've been spoiled by the way Asterisk handles NAT and was hopeful that NDLB-connectile-dysfunction would behave similarly, so I wouldn't have to tell users to setup stun on their clients. Maybe a FS user with some experience with this type of NAT setup and these settings can help. I'd be interested in knowing how to correctly setup remote NATted endpoints without stun - or, at least, hear from someone that this setting works for them without stun. Anyway, hope this helps you with your SPA3102. Bottom line is enable rport and use stun on the SPA and it'll just work. /b ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS
Thanks Yehavi... I posted a question on the Cisco Forum and am waiting a response from 'engineering' to tell us if they plan to implement standard SRTP support in the Linksys ATA's. TLS is working fine. On Thu, Dec 17, 2009 at 4:39 PM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: An interim update: Audiocodes: No success yet. I am working with the manufacturer to debug it. VegaStream: Got the necessary license, configured TLS but it doesn't work. I am working with the local representatives on it. Regards, __Yehavi: 2009/12/10 Brian West br...@freeswitch.org I have confirmed it works with Polycom, Snom and a few others polycom is the hardest to set due to having to put the ca cert into the phone... but other than that its good. /b On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote: An intermediate report: Audiocodes: TLS works only on outgoing requests, incoming ones are ignored. I am waiting for Audiocodes' help in order to debug it. SRTP: worked when no TLS is active. When TLS is active the call is disconnected when the remote party answers. Still debugging it. VegaStream Europa-50: SRTP works. Waiting for Vega for instructions how to enable TLS from the WEB interface. Regards, __Yehavi: ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Passing user variables to mod_voicemail
Pennytel.com
On Sat, Dec 12, 2009 at 12:52 AM, Phillip Jones pjinthe...@gmail.com wrote:
Hi - sorry to go off topic - but we are looking for Voip supplier with SMS
capability. Would you mind telling me which Voip supplier you use?
On Thu, Dec 10, 2009 at 11:10 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
My voip provider provides a SOAP interface to be able to send SMS's,
so after a voicemail is left, I want to execute a 'send sms' script.
I don't want a separate statement in the dialplan after the voicemail
statement because I only want to send sms's when a voicemail is
actually left.
The way I was going to do this was to modify the mailer-app to point
to a shell script and modify the mailer-app-args to include some user
defined variables (in conf/directory/default/*.xml).
param name=mailer-app
value=/usr/local/freeswitch/scripts/emailvm.sh/
param name=mailer-app-args value=${smsaccount} ${smspassword}
${smsnumber}/
The shell script would do the following:
emailvm.sh
#$1 $2 $3 = smsaccount smspassword textmessage
tee /tmp/vmmail | /usr/sbin/sendmail -t
exec /usr/local/freeswitch/scripts/sendsms.pl $1 $2 $3
#echo $1 $2 $3 $4 $5 $6 /usr/local/freeswitch/scripts/log.log
However, if I uncomment the last line, I never see the user variables
being passed to the shell script. The email is sucessfully sent, but
the sms script doesnt work. If fact, the output of log.log is (for
example):
-f 1...@192.168.1.120 email_addr...@domain.com
Any ideas if it is possible to pass user variables via mod_voicemail
in this way?
Thanks
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Re: [Freeswitch-users] embedded freeswitch compatable hardware
I use a dectop by Data Evolution... Its cheap at ~$100. I have it running debian lenny and FS... works well for me. http://www.dataevolution.com/dectop%20info%202.htm http://www.gadgettastic.com/2007/08/18/dectop-the-100-pc/ On Thu, Dec 10, 2009 at 10:45 PM, Fred-145 codecompl...@free.fr wrote: Frank Carmickle wrote: A board with an atom 330 on it would probably do the trick for you. There are a few made by Intel and Supermicro that look pretty nice. There were some other people on the list looking to use them. Maybe we can get a report from someone. Intel came up with the D945GSEJT, which is totally fanless and has an embedded DC/DC, so all you have to add is an external AC/DC power brick, some RAM, a PCI riser to save space, and either a hard-disk or a CompactFlash + IDE adaptor. I'm thinking of building one with a Digium-compatible PCI card. www.intel.com/products/desktop/motherboards/D945GSEJT/D945GSEJT-overview.htm -- View this message in context: http://old.nabble.com/embedded-freeswitch-compatable-hardware-tp26721589p26725918.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] Passing user variables to mod_voicemail
Hi!
My voip provider provides a SOAP interface to be able to send SMS's,
so after a voicemail is left, I want to execute a 'send sms' script.
I don't want a separate statement in the dialplan after the voicemail
statement because I only want to send sms's when a voicemail is
actually left.
The way I was going to do this was to modify the mailer-app to point
to a shell script and modify the mailer-app-args to include some user
defined variables (in conf/directory/default/*.xml).
param name=mailer-app value=/usr/local/freeswitch/scripts/emailvm.sh/
param name=mailer-app-args value=${smsaccount} ${smspassword}
${smsnumber}/
The shell script would do the following:
emailvm.sh
#$1 $2 $3 = smsaccount smspassword textmessage
tee /tmp/vmmail | /usr/sbin/sendmail -t
exec /usr/local/freeswitch/scripts/sendsms.pl $1 $2 $3
#echo $1 $2 $3 $4 $5 $6 /usr/local/freeswitch/scripts/log.log
However, if I uncomment the last line, I never see the user variables
being passed to the shell script. The email is sucessfully sent, but
the sms script doesnt work. If fact, the output of log.log is (for
example):
-f 1...@192.168.1.120 email_addr...@domain.com
Any ideas if it is possible to pass user variables via mod_voicemail
in this way?
Thanks
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[Freeswitch-users] Access to users variables
Hi!
How can I access the variables that are defined in a users xml file?
For example, say user 1000 has a variable called smsnumber, as defined below:
include
user id=1000 mailbox=1000
params
param name=password value=1000/
/params
variables
variable name=smsnumber value=12345/
/variables
/user
/include
How can I use variable ${smsnumber} in a dialplan to run a perl script
using action application=system data=sms.pl/ ?
Thanks
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Re: [Freeswitch-users] Access to users variables
Hi!
That's exactly what I want to do and that was the first thing I tried,
but nothing is passed to the script.
In a case like this, what defines if variable smsnumber is taken from
the A path or B path? (The A path does not have smsnumber defined)
On Tue, Dec 8, 2009 at 5:25 AM, Michael Collins m...@freeswitch.org wrote:
On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
How can I access the variables that are defined in a users xml file?
For example, say user 1000 has a variable called smsnumber, as defined
below:
include
user id=1000 mailbox=1000
params
param name=password value=1000/
/params
variables
variable name=smsnumber value=12345/
/variables
/user
/include
How can I use variable ${smsnumber} in a dialplan to run a perl script
using action application=system data=sms.pl/ ?
Do you just want to pass the value in smsnumber to the sms.pl script? Have
you tried this?
action application=system data=sms.pl ${smsnumber}/
-MC
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Re: [Freeswitch-users] Access to users variables
Perfect...
action application=set_user data=${dialed_extensi...@${domain}/
works like a charm.
Thanks Mike.
On Tue, Dec 8, 2009 at 5:56 AM, Michael Collins m...@freeswitch.org wrote:
Check out
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user
you might just need to set the user so that the vars become available on the
leg you're processing.
-MC
On Mon, Dec 7, 2009 at 10:37 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
That's exactly what I want to do and that was the first thing I tried,
but nothing is passed to the script.
In a case like this, what defines if variable smsnumber is taken from
the A path or B path? (The A path does not have smsnumber defined)
On Tue, Dec 8, 2009 at 5:25 AM, Michael Collins m...@freeswitch.org
wrote:
On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
How can I access the variables that are defined in a users xml file?
For example, say user 1000 has a variable called smsnumber, as defined
below:
include
user id=1000 mailbox=1000
params
param name=password value=1000/
/params
variables
variable name=smsnumber value=12345/
/variables
/user
/include
How can I use variable ${smsnumber} in a dialplan to run a perl script
using action application=system data=sms.pl/ ?
Do you just want to pass the value in smsnumber to the sms.pl script?
Have
you tried this?
action application=system data=sms.pl ${smsnumber}/
-MC
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Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS
Hi All, I managed to borrow a SPA3102 with the latest firmware and have got it to register using TLS, but I am still struggling with SRTP. Has anyone managed to get SRTP working with the Linksys devices and if so, can they direct me on how to do this. I have generated a mini-certificates and SRTP Private Key using the gen-mc tool found at http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. However, when ever I initiate a call from the SPA, I can see that the call is not encrypted. Help appreciated. Thanks! On Sat, Nov 28, 2009 at 6:31 AM, eman e...@chabotel.com wrote: Check out the Linksys SPA2102 On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: The only ATA mentioned on the WIKI that supports TLS/SRTP is the Grandstream HandyTone 503. But, again according to the wiki, that doesn't seem to behave to well with TLS ... On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote: Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Does the SPA3102 support TLS or only SRTP? I don't know, but supporting only SRTP would be ridiculous, since the keys would then be transmitted in the clear and therefore amenable to interception. SRTP requires the SIP channel to be encrypted by TLS in order to be secure. ZRTP, on the other hand, doesn't have this limitation: it works entirely in RTP. I would be rather surprised were a hardware manufacturer to implement SRTP without TLS for the SIP traffic. On the other hand, we've seen often in this forum that some manufacturers are really clueless... ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS
Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH I'll check with Cisco regarding their implementation then and try to find out when/if they will support standard SRTP encryption. So, back to my origianal question then. Are there any ATA's that support TLS AND SRTP with FreeSwitch? On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri gk...@ieee.org wrote: AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH. They do their proprietary Sipura key exchange only, not sure if Cisco plans on upgrading the firmware to ever support SDES on the ATAs. They added support for SDES to their IP Phones about 1 year ago, but nothing has happened with the ATAs as of yet. Gabe On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi All, I managed to borrow a SPA3102 with the latest firmware and have got it to register using TLS, but I am still struggling with SRTP. Has anyone managed to get SRTP working with the Linksys devices and if so, can they direct me on how to do this. I have generated a mini-certificates and SRTP Private Key using the gen-mc tool found at http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. However, when ever I initiate a call from the SPA, I can see that the call is not encrypted. Help appreciated. Thanks! On Sat, Nov 28, 2009 at 6:31 AM, eman e...@chabotel.com wrote: Check out the Linksys SPA2102 On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: The only ATA mentioned on the WIKI that supports TLS/SRTP is the Grandstream HandyTone 503. But, again according to the wiki, that doesn't seem to behave to well with TLS ... On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote: Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Does the SPA3102 support TLS or only SRTP? I don't know, but supporting only SRTP would be ridiculous, since the keys would then be transmitted in the clear and therefore amenable to interception. SRTP requires the SIP channel to be encrypted by TLS in order to be secure. ZRTP, on the other hand, doesn't have this limitation: it works entirely in RTP. I would be rather surprised were a hardware manufacturer to implement SRTP without TLS for the SIP traffic. On the other hand, we've seen often in this forum that some manufacturers are really clueless... ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS
Cheers Gabriel.. thanks for the information. I'll look at the Mediatrix ATA's as an alternative - has anyone had experience with those and TLS/SRTP? On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri gk...@ieee.org wrote: The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the Grandstream and Mediatrix devices (although I've never tried either one with FreeSWITCH). I've personally never had any good experience with the Grandstream ATAs. The Mediatrix ATAs are OK devices, but I've never personally tested them with SRTP w/SDES and FreeSWITCH, but supposedly they support it (so says their marketing material and docs). I'd see if Cisco has any plans to add support for it to the ATAs. Next time I see our Cisco SE, I'll try to poke him about it. Gabe On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH I'll check with Cisco regarding their implementation then and try to find out when/if they will support standard SRTP encryption. So, back to my origianal question then. Are there any ATA's that support TLS AND SRTP with FreeSwitch? On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri gk...@ieee.org wrote: AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH. They do their proprietary Sipura key exchange only, not sure if Cisco plans on upgrading the firmware to ever support SDES on the ATAs. They added support for SDES to their IP Phones about 1 year ago, but nothing has happened with the ATAs as of yet. Gabe On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi All, I managed to borrow a SPA3102 with the latest firmware and have got it to register using TLS, but I am still struggling with SRTP. Has anyone managed to get SRTP working with the Linksys devices and if so, can they direct me on how to do this. I have generated a mini-certificates and SRTP Private Key using the gen-mc tool found at http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. However, when ever I initiate a call from the SPA, I can see that the call is not encrypted. Help appreciated. Thanks! On Sat, Nov 28, 2009 at 6:31 AM, eman e...@chabotel.com wrote: Check out the Linksys SPA2102 On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: The only ATA mentioned on the WIKI that supports TLS/SRTP is the Grandstream HandyTone 503. But, again according to the wiki, that doesn't seem to behave to well with TLS ... On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote: Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Does the SPA3102 support TLS or only SRTP? I don't know, but supporting only SRTP would be ridiculous, since the keys would then be transmitted in the clear and therefore amenable to interception. SRTP requires the SIP channel to be encrypted by TLS in order to be secure. ZRTP, on the other hand, doesn't have this limitation: it works entirely in RTP. I would be rather surprised were a hardware manufacturer to implement SRTP without TLS for the SIP traffic. On the other hand, we've seen often in this forum that some manufacturers are really clueless... ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@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@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@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@lists.freeswitch.org http
Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS
Thanks Yehavi, I would be very interested to find out how your test goes... can you report back after you have tested it? Thanks! On Fri, Dec 4, 2009 at 3:38 PM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: Hello, I have AudioCodes MP and Vega ATA adapters. They both support SRTP; they should support TLS also (will try it next week; up to now I preffered to not use TLS so I can sniff the traffic and debug things). Regards, __Yehavi: 2009/12/4 Mark Campbell-Smith mcampbellsm...@gmail.com Cheers Gabriel.. thanks for the information. I'll look at the Mediatrix ATA's as an alternative - has anyone had experience with those and TLS/SRTP? On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri gk...@ieee.org wrote: The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the Grandstream and Mediatrix devices (although I've never tried either one with FreeSWITCH). I've personally never had any good experience with the Grandstream ATAs. The Mediatrix ATAs are OK devices, but I've never personally tested them with SRTP w/SDES and FreeSWITCH, but supposedly they support it (so says their marketing material and docs). I'd see if Cisco has any plans to add support for it to the ATAs. Next time I see our Cisco SE, I'll try to poke him about it. Gabe On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH I'll check with Cisco regarding their implementation then and try to find out when/if they will support standard SRTP encryption. So, back to my origianal question then. Are there any ATA's that support TLS AND SRTP with FreeSwitch? On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri gk...@ieee.org wrote: AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH. They do their proprietary Sipura key exchange only, not sure if Cisco plans on upgrading the firmware to ever support SDES on the ATAs. They added support for SDES to their IP Phones about 1 year ago, but nothing has happened with the ATAs as of yet. Gabe On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi All, I managed to borrow a SPA3102 with the latest firmware and have got it to register using TLS, but I am still struggling with SRTP. Has anyone managed to get SRTP working with the Linksys devices and if so, can they direct me on how to do this. I have generated a mini-certificates and SRTP Private Key using the gen-mc tool found at http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. However, when ever I initiate a call from the SPA, I can see that the call is not encrypted. Help appreciated. Thanks! On Sat, Nov 28, 2009 at 6:31 AM, eman e...@chabotel.com wrote: Check out the Linksys SPA2102 On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: The only ATA mentioned on the WIKI that supports TLS/SRTP is the Grandstream HandyTone 503. But, again according to the wiki, that doesn't seem to behave to well with TLS ... On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote: Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Does the SPA3102 support TLS or only SRTP? I don't know, but supporting only SRTP would be ridiculous, since the keys would then be transmitted in the clear and therefore amenable to interception. SRTP requires the SIP channel to be encrypted by TLS in order to be secure. ZRTP, on the other hand, doesn't have this limitation: it works entirely in RTP. I would be rather surprised were a hardware manufacturer to implement SRTP without TLS for the SIP traffic. On the other hand, we've seen often in this forum that some manufacturers are really clueless... ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS
The only ATA mentioned on the WIKI that supports TLS/SRTP is the Grandstream HandyTone 503. But, again according to the wiki, that doesn't seem to behave to well with TLS ... On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote: Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Does the SPA3102 support TLS or only SRTP? I don't know, but supporting only SRTP would be ridiculous, since the keys would then be transmitted in the clear and therefore amenable to interception. SRTP requires the SIP channel to be encrypted by TLS in order to be secure. ZRTP, on the other hand, doesn't have this limitation: it works entirely in RTP. I would be rather surprised were a hardware manufacturer to implement SRTP without TLS for the SIP traffic. On the other hand, we've seen often in this forum that some manufacturers are really clueless... ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How Register soft sip phones to FreeSWITCH with extension number.
Didn't Michael already answer this? Best read the FS wiki and the softphone user guide for help with this. http://wiki.freeswitch.org/wiki/Getting_Started_Guide http://wiki.freeswitch.org/wiki/Interop_List On Wed, Nov 25, 2009 at 7:29 PM, ovvenkat ovvenkate...@gmail.com wrote: Hi to All, Any one please tell me , How to configure soft sip phone to freeswitch with extension number. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS
Hi there Itamar, Does the SPA3102 support TLS or only SRTP? And what about Brians comments that 'It uses a sick twisted method of doing SRTP'. Do you have it working using SRTP together with FS? What about TLS? Otherwise are there any other ATA's that support TLS SRTP? On Sun, Nov 22, 2009 at 8:41 PM, Itamar Reis Peixoto ita...@ispbrasil.com.br wrote: it's support SRTP On Sun, Nov 22, 2009 at 7:21 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Do LInksys devices support TLS and SRTP that FS supports? 3102 at least doesn't according to this post -- Itamar Reis Peixoto e-mail/msn/google talk/sip: ita...@ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS
Do LInksys devices support TLS and SRTP that FS supports? 3102 at least doesn't according to this post http://osdir.com/ml/telephony.freeswitch.user/2008-08/msg00904.html On Sun, Nov 22, 2009 at 7:20 PM, Itamar Reis Peixoto ita...@ispbrasil.com.br wrote: sipura/linksys look in ebay. On Sun, Nov 22, 2009 at 5:35 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: HI All, Has anyone got some recommendations on which ATA to buy that supports TLS and SRTP? Thanks! -- Itamar Reis Peixoto e-mail/msn/google talk/sip: ita...@ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] ATA that supports TLS/SRTP w FS
HI All, Has anyone got some recommendations on which ATA to buy that supports TLS and SRTP? Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Call from Secure RTP to non-secure RTP
Hi!
How do I setup FS so that placing a call from an extension that only
support SRTP (1002) to an extension that only supports RTP (1000)?
I put this dialstring, from the wiki
http://wiki.freeswitch.org/wiki/Tls, into the users xml file under
directory/default
param name=dial-string
value={sip_secure_media=${regex(${sofia_contact(${dialed_us...@${dialed_domain})}|transport=tls)},
presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}
/
I have also put a action application=export
data=sip_secure_media=true/ when 1000 is dialing 1002.
condition field=destination_number expression=^(1002)$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=export data=sip_secure_media=true/
action application=bridge data=user/${dialed_extensi...@${domain}/
However I never see crytpo sent in the RTP to 1002 and it responds
with Bad Security Level
What have I missed?
Thanks
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Re: [Freeswitch-users] Call from Secure RTP to non-secure RTP
Thanks Jim,
Yep, 1002 does TLS and SRTP, 1000 does UDP and RTP.
Cheers
On Thu, Nov 19, 2009 at 4:12 PM, Jim Burke j...@evolutiontel.net wrote:
Does 1002 use TLS to transport SIP signalling? My experience is that
TLS is required on some phones otherwise they will not do srtp and
will reply with the responce you have mentioned.
Sent from my iPhone
On 19/11/2009, at 1:36 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
How do I setup FS so that placing a call from an extension that only
support SRTP (1002) to an extension that only supports RTP (1000)?
I put this dialstring, from the wiki
http://wiki.freeswitch.org/wiki/Tls, into the users xml file under
directory/default
param name=dial-string
value={sip_secure_media=${regex(${sofia_contact(${dialed_us...@$
{dialed_domain})}|transport=tls)},
presence_id=${dialed_us...@${dialed_domain}}${sofia_contact($
{dialed_us...@${dialed_domain})}
/
I have also put a action application=export
data=sip_secure_media=true/ when 1000 is dialing 1002.
condition field=destination_number expression=^(1002)$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=export data=sip_secure_media=true/
action application=bridge data=user/${dialed_extensi...@$
{domain}/
However I never see crytpo sent in the RTP to 1002 and it responds
with Bad Security Level
What have I missed?
Thanks
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[Freeswitch-users] application=info
HI All, pretty basic question and I feel a bit stupid asking this, but what are the prerequisites for the INFO to be displayed when action application=info/ is called in a dialplan? ie are there requirements on the loglevel, does the INFO command have to be put at a certain place in the dialplan etc? The reason i ask is that I have a dialplan and the action application=info/ is not getting triggered on the fs_cli output. Is there some other debbugging level that needs to be set? Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] application=info
I had console loglevel set to DEBUG, so that should be fine. And I do see that FS is executing the exact extension where I have put the INFO application - still no info on the console I'm using FreeSWITCH Version 1.0.trunk (15490) On Wed, Nov 18, 2009 at 4:56 PM, Mathieu Rene mrene_li...@avgs.ca wrote: If you press F8 (or do /log 7), you will see what the dialplan is executing, try to see if you see the info app in there And, your global loglevel has to be = INFO too... fsctl loglevel debug Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 17-Nov-09, at 9:53 PM, Mark Campbell-Smith wrote: HI All, pretty basic question and I feel a bit stupid asking this, but what are the prerequisites for the INFO to be displayed when action application=info/ is called in a dialplan? ie are there requirements on the loglevel, does the INFO command have to be put at a certain place in the dialplan etc? The reason i ask is that I have a dialplan and the action application=info/ is not getting triggered on the fs_cli output. Is there some other debbugging level that needs to be set? Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] application=info
ahha... great. Thanks Mathieu. On Wed, Nov 18, 2009 at 5:08 PM, Mathieu Rene mrene_li...@avgs.ca wrote: Console loglevel only sets the loglevel on the console, not on fs_cli or other event_socket client programs. You have to do /log 7 on fs_cli. fsctl loglevel is the global system loglevel, if its at warning, you wont see anything below warning ANYWHERE (console/event socket log files) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 17-Nov-09, at 10:05 PM, Mark Campbell-Smith wrote: I had console loglevel set to DEBUG, so that should be fine. And I do see that FS is executing the exact extension where I have put the INFO application - still no info on the console I'm using FreeSWITCH Version 1.0.trunk (15490) On Wed, Nov 18, 2009 at 4:56 PM, Mathieu Rene mrene_li...@avgs.ca wrote: If you press F8 (or do /log 7), you will see what the dialplan is executing, try to see if you see the info app in there And, your global loglevel has to be = INFO too... fsctl loglevel debug Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 17-Nov-09, at 9:53 PM, Mark Campbell-Smith wrote: HI All, pretty basic question and I feel a bit stupid asking this, but what are the prerequisites for the INFO to be displayed when action application=info/ is called in a dialplan? ie are there requirements on the loglevel, does the INFO command have to be put at a certain place in the dialplan etc? The reason i ask is that I have a dialplan and the action application=info/ is not getting triggered on the fs_cli output. Is there some other debbugging level that needs to be set? Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@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@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@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] TLS support on debian lenny
Hi! I am trying to enable SSL support in FS. I have followed the wiki at http://wiki.freeswitch.org/wiki/SIP_TLS I already had libssl-dev installed, so I thought support should already have been compiled into FS, however enabling Internal_ssl_enable=true in vars.xml results in FS internal profile to not start: 2009-11-17 09:31:48.593240 [NOTICE] sofia.c:3016 Started Profile internal [sofia_reg_internal] 2009-11-17 09:31:48.907740 [ERR] sofia.c:1006 Error Creating SIP UA for profile: internal Checking freeswitch/libs/sofia-sip/config.log I see the following, which I assume means TLS has not been compiled with support: configure:27892: checking openssl/tls1.h usability configure:27909: gcc -c -DSU_DEBUG=0 -g -ggdb conftest.c 5 conftest.c:156:26: error: openssl/tls1.h: No such file or directory What package should I have installed prior to compiling FS on debian? There is no OpenSSL-Dev. Is it libcurl4-openssl-dev? Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] TLS support on debian lenny
I installed libcurl4-openssl-dev, but this automatically removed libcurl4-gnutls-dev, which is required by mod_dingaling. Now mod_dingaling fails to build with: Compiling mod_dingaling.c ... mod_dingaling.c:309:78: error: macro switch_odbc_handle_callback_exec requires 5 arguments, but only 4 given mod_dingaling.c: In function ‘mdl_execute_sql_callback’: mod_dingaling.c:309: error: ‘switch_odbc_handle_callback_exec’ undeclared (first use in this function) mod_dingaling.c:309: error: (Each undeclared identifier is reported only once mod_dingaling.c:309: error: for each function it appears in.) make[6]: *** [mod_dingaling.lo] Error 1 Anyone know which package should be installed so that TLS works on Debian? On Tue, Nov 17, 2009 at 10:05 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi! I am trying to enable SSL support in FS. I have followed the wiki at http://wiki.freeswitch.org/wiki/SIP_TLS I already had libssl-dev installed, so I thought support should already have been compiled into FS, however enabling Internal_ssl_enable=true in vars.xml results in FS internal profile to not start: 2009-11-17 09:31:48.593240 [NOTICE] sofia.c:3016 Started Profile internal [sofia_reg_internal] 2009-11-17 09:31:48.907740 [ERR] sofia.c:1006 Error Creating SIP UA for profile: internal Checking freeswitch/libs/sofia-sip/config.log I see the following, which I assume means TLS has not been compiled with support: configure:27892: checking openssl/tls1.h usability configure:27909: gcc -c -DSU_DEBUG=0 -g -ggdb conftest.c 5 conftest.c:156:26: error: openssl/tls1.h: No such file or directory What package should I have installed prior to compiling FS on debian? There is no OpenSSL-Dev. Is it libcurl4-openssl-dev? Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Extension: No audio
Hi! I have FS natted and am connecting with an 'external' extension that is registered to FS. ie the extension 2000 is registered on the internet with a public IP through my router to FS (192.168.1.120 IP address). uPnP works and I see that the extension is registered successfully. The problem is that I do not get any audio When looking at the SIP trace, I see the INVITE but do not see a TRYING or RINGING message. The extension is actually ringing. I modified the RTP port range on the remote end to match the RTP ports of freeswitch. I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 If anyone has an idea what needs to be set to get audio, help appreciated. Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Extension: No audio
Hi Mike, I should have put that in also. I do have external_rtp_ip set in my config. I have it set to my domain name: X-PRE-PROCESS cmd=set data=external_rtp_ip=host:mydomainname/ I should also mention that if I use flaphone.com (which registers with an external IP address), then I get audio. In sofia, I see my IP addresses: = Nameinternal Domain Name N/A DBName sofia_reg_internal Pres Hosts DialplanXML Context public Challenge Realm auto_from RTP-IP 192.168.1.120 Ext-RTP-IP 124.xxx.xxx.xxx SIP-IP 192.168.1.120 Ext-SIP-IP 124.xxx.xxx.x URL sip:mod_so...@192.168.1.120:5060 BIND-URLsip:mod_so...@192.168.1.120:5060 HOLD-MUSIC silence OUTBOUND-PROXY N/A CODECS G726-32,G722,PCMU,PCMA TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEGfalse PROXY-MEDIA false AGGRESSIVENAT true STUN-ENABLEDtrue STUN-AUTO-DISABLE false On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris m...@jerris.com wrote: You don't have ext-rtp-ip set in your config. Mike On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: Hi! I have FS natted and am connecting with an 'external' extension that is registered to FS. ie the extension 2000 is registered on the internet with a public IP through my router to FS (192.168.1.120 IP address). uPnP works and I see that the extension is registered successfully. The problem is that I do not get any audio When looking at the SIP trace, I see the INVITE but do not see a TRYING or RINGING message. The extension is actually ringing. I modified the RTP port range on the remote end to match the RTP ports of freeswitch. I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 If anyone has an idea what needs to be set to get audio, help appreciated. Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Extension: No audio
OK.. thanks Mike.
I assume I am using the Internal profile. I have defined user 2000
in the 'directory' using a context called family: switch_ivr.c:1367
Transfer sofia/internal/1...@192.168.1.120 to xml[2...@family]
This is an extract from sofia:
sofia status profile internal
=
Nameinternal
Domain Name N/A
DBName sofia_reg_internal
Pres Hosts
DialplanXML
Context public
Challenge Realm auto_from
RTP-IP 192.168.1.120
Ext-RTP-IP 124.xxx.xxx.xxx
SIP-IP 192.168.1.120
Ext-SIP-IP 124.xxx.xxx.xxx
URL sip:mod_so...@192.168.1.120:5060
BIND-URLsip:mod_so...@192.168.1.120:5060
HOLD-MUSIC silence
OUTBOUND-PROXY N/A
CODECS G726-32,G722,PCMU,PCMA
TEL-EVENT 101
DTMF-MODE rfc2833
CNG 13
SESSION-TO 0
MAX-DIALOG 0
NOMEDIA false
LATE-NEGfalse
PROXY-MEDIA false
AGGRESSIVENAT true
STUN-ENABLEDtrue
STUN-AUTO-DISABLE false
CALLS-IN100
FAILED-CALLS-IN 25
CALLS-OUT 38
FAILED-CALLS-OUT31
Registrations:
=
Call-ID:68534bba9b461...@58.169.138.53
User: 2...@192.168.1.120
Contact:user sip:2...@58.xxx.xxx.xxx:5060
Agent: dunno
Status: Registered(UDP)(unknown) EXP(2009-11-09 14:58:30)
Host: freeswitch
IP: 58.xxx.xxx.xxx
Port: 5060
Auth-User: 2000
Auth-Realm: markcs.dyndns.org
MWI-Account:2...@192.168.1.120
The internal.xml file has a lot in it, but I guess these are the
important things for this profile:
param name=ext-rtp-ip value=auto-nat/
param name=ext-sip-ip value=auto-nat/
param name=sip-port value=$${internal_sip_port}/
param name=rtp-ip value=auto/
I will try to change auto-nat to be $${external_sip_ip}
One question though: Any idea why I never see the TRYING or RINGING
messages? Are tehse related to the RTP IP address or not? Without
these I assume something is incorrect and I do not hear ringback
Thanks!
On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris m...@jerris.com wrote:
Your packet traces would disagree with the statements below. It is
sending your internal address in rtp, so its not set correctly on
whatever profile your using to call out,
MIke
On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote:
Hi Mike,
I should have put that in also.
I do have external_rtp_ip set in my config. I have it set to my
domain name:
X-PRE-PROCESS cmd=set data=external_rtp_ip=host:mydomainname/
I should also mention that if I use flaphone.com (which registers with
an external IP address), then I get audio. In sofia, I see my IP
addresses:
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
==
Name internal
Domain Name N/A
DBName sofia_reg_internal
Pres Hosts
Dialplan XML
Context public
Challenge Realm auto_from
RTP-IP 192.168.1.120
Ext-RTP-IP 124.xxx.xxx.xxx
SIP-IP 192.168.1.120
Ext-SIP-IP 124.xxx.xxx.x
URL sip:mod_so...@192.168.1.120:5060
BIND-URL sip:mod_so...@192.168.1.120:5060
HOLD-MUSIC silence
OUTBOUND-PROXY N/A
CODECS G726-32,G722,PCMU,PCMA
TEL-EVENT 101
DTMF-MODE rfc2833
CNG 13
SESSION-TO 0
MAX-DIALOG 0
NOMEDIA false
LATE-NEG false
PROXY-MEDIA false
AGGRESSIVENAT true
STUN-ENABLED true
STUN-AUTO-DISABLE false
On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris m...@jerris.com
wrote:
You don't have ext-rtp-ip set in your config.
Mike
On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote:
Hi!
I have FS natted and am connecting with an 'external' extension that
is registered to FS. ie the extension 2000 is registered on the
internet with a public IP through my router to FS (192.168.1.120 IP
address). uPnP works and I see that the extension is registered
successfully.
The problem is that I do not get any audio
When looking at the SIP trace, I see the INVITE but do not see a
TRYING or RINGING message. The extension is actually ringing. I
modified the RTP port range on the remote end to match the RTP ports
of freeswitch.
I have put a sip trace in the pastebin at
http://pastebin.freeswitch.org
Re: [Freeswitch-users] Extension: No audio
Hi again,
Actually, changing the param name=ext-rtp-ip value=auto-nat/ to
param name=ext-rtp-ip value=$${external_sip_ip}/ means that I
now see the IP address in the INVITE message:
v=0
o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 124.xxx.xxx.xxx
t=0 0
m=audio 21234 RTP/AVP 0 2 9 8 101 13
Why would this be? I thought auto-nat was meant to solve these issues?
However, I still do not see the TRYING or RINGING messages ideas
appreciated.
Thanks!
On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
OK.. thanks Mike.
I assume I am using the Internal profile. I have defined user 2000
in the 'directory' using a context called family: switch_ivr.c:1367
Transfer sofia/internal/1...@192.168.1.120 to xml[2...@family]
This is an extract from sofia:
sofia status profile internal
=
Name internal
Domain Name N/A
DBName sofia_reg_internal
Pres Hosts
Dialplan XML
Context public
Challenge Realm auto_from
RTP-IP 192.168.1.120
Ext-RTP-IP 124.xxx.xxx.xxx
SIP-IP 192.168.1.120
Ext-SIP-IP 124.xxx.xxx.xxx
URL sip:mod_so...@192.168.1.120:5060
BIND-URL sip:mod_so...@192.168.1.120:5060
HOLD-MUSIC silence
OUTBOUND-PROXY N/A
CODECS G726-32,G722,PCMU,PCMA
TEL-EVENT 101
DTMF-MODE rfc2833
CNG 13
SESSION-TO 0
MAX-DIALOG 0
NOMEDIA false
LATE-NEG false
PROXY-MEDIA false
AGGRESSIVENAT true
STUN-ENABLED true
STUN-AUTO-DISABLE false
CALLS-IN 100
FAILED-CALLS-IN 25
CALLS-OUT 38
FAILED-CALLS-OUT 31
Registrations:
=
Call-ID: 68534bba9b461...@58.169.138.53
User: 2...@192.168.1.120
Contact: user sip:2...@58.xxx.xxx.xxx:5060
Agent: dunno
Status: Registered(UDP)(unknown) EXP(2009-11-09 14:58:30)
Host: freeswitch
IP: 58.xxx.xxx.xxx
Port: 5060
Auth-User: 2000
Auth-Realm: markcs.dyndns.org
MWI-Account: 2...@192.168.1.120
The internal.xml file has a lot in it, but I guess these are the
important things for this profile:
param name=ext-rtp-ip value=auto-nat/
param name=ext-sip-ip value=auto-nat/
param name=sip-port value=$${internal_sip_port}/
param name=rtp-ip value=auto/
I will try to change auto-nat to be $${external_sip_ip}
One question though: Any idea why I never see the TRYING or RINGING
messages? Are tehse related to the RTP IP address or not? Without
these I assume something is incorrect and I do not hear ringback
Thanks!
On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris m...@jerris.com wrote:
Your packet traces would disagree with the statements below. It is
sending your internal address in rtp, so its not set correctly on
whatever profile your using to call out,
MIke
On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote:
Hi Mike,
I should have put that in also.
I do have external_rtp_ip set in my config. I have it set to my
domain name:
X-PRE-PROCESS cmd=set data=external_rtp_ip=host:mydomainname/
I should also mention that if I use flaphone.com (which registers with
an external IP address), then I get audio. In sofia, I see my IP
addresses:
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
==
Name internal
Domain Name N/A
DBName sofia_reg_internal
Pres Hosts
Dialplan XML
Context public
Challenge Realm auto_from
RTP-IP 192.168.1.120
Ext-RTP-IP 124.xxx.xxx.xxx
SIP-IP 192.168.1.120
Ext-SIP-IP 124.xxx.xxx.x
URL sip:mod_so...@192.168.1.120:5060
BIND-URL sip:mod_so...@192.168.1.120:5060
HOLD-MUSIC silence
OUTBOUND-PROXY N/A
CODECS G726-32,G722,PCMU,PCMA
TEL-EVENT 101
DTMF-MODE rfc2833
CNG 13
SESSION-TO 0
MAX-DIALOG 0
NOMEDIA false
LATE-NEG false
PROXY-MEDIA false
AGGRESSIVENAT true
STUN-ENABLED true
STUN-AUTO-DISABLE false
On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris m...@jerris.com
wrote:
You don't have ext-rtp-ip set in your config.
Mike
On Nov 8, 2009, at 6:59 AM, Mark
Re: [Freeswitch-users] Extension: No audio
Is there a way to determine if FS has detected nat? I am behind UPnP
and I can see on the router the mappings for Freeswitch.
2009/11/9 João Mesquita jmesqu...@freeswitch.org:
It is if FS was able to detect NAT. Are you behind PMP or UPnP? Otherwise,
no go
Have you changed the ext-sip-ip too?
Regards,
JM
On Mon, Nov 9, 2009 at 12:32 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi again,
Actually, changing the param name=ext-rtp-ip value=auto-nat/ to
param name=ext-rtp-ip value=$${external_sip_ip}/ means that I
now see the IP address in the INVITE message:
v=0
o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 124.xxx.xxx.xxx
t=0 0
m=audio 21234 RTP/AVP 0 2 9 8 101 13
Why would this be? I thought auto-nat was meant to solve these issues?
However, I still do not see the TRYING or RINGING messages ideas
appreciated.
Thanks!
On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
OK.. thanks Mike.
I assume I am using the Internal profile. I have defined user 2000
in the 'directory' using a context called family: switch_ivr.c:1367
Transfer sofia/internal/1...@192.168.1.120 to xml[2...@family]
This is an extract from sofia:
sofia status profile internal
=
Name internal
Domain Name N/A
DBName sofia_reg_internal
Pres Hosts
Dialplan XML
Context public
Challenge Realm auto_from
RTP-IP 192.168.1.120
Ext-RTP-IP 124.xxx.xxx.xxx
SIP-IP 192.168.1.120
Ext-SIP-IP 124.xxx.xxx.xxx
URL sip:mod_so...@192.168.1.120:5060
BIND-URL sip:mod_so...@192.168.1.120:5060
HOLD-MUSIC silence
OUTBOUND-PROXY N/A
CODECS G726-32,G722,PCMU,PCMA
TEL-EVENT 101
DTMF-MODE rfc2833
CNG 13
SESSION-TO 0
MAX-DIALOG 0
NOMEDIA false
LATE-NEG false
PROXY-MEDIA false
AGGRESSIVENAT true
STUN-ENABLED true
STUN-AUTO-DISABLE false
CALLS-IN 100
FAILED-CALLS-IN 25
CALLS-OUT 38
FAILED-CALLS-OUT 31
Registrations:
=
Call-ID: 68534bba9b461...@58.169.138.53
User: 2...@192.168.1.120
Contact: user sip:2...@58.xxx.xxx.xxx:5060
Agent: dunno
Status: Registered(UDP)(unknown) EXP(2009-11-09 14:58:30)
Host: freeswitch
IP: 58.xxx.xxx.xxx
Port: 5060
Auth-User: 2000
Auth-Realm: markcs.dyndns.org
MWI-Account: 2...@192.168.1.120
The internal.xml file has a lot in it, but I guess these are the
important things for this profile:
param name=ext-rtp-ip value=auto-nat/
param name=ext-sip-ip value=auto-nat/
param name=sip-port value=$${internal_sip_port}/
param name=rtp-ip value=auto/
I will try to change auto-nat to be $${external_sip_ip}
One question though: Any idea why I never see the TRYING or RINGING
messages? Are tehse related to the RTP IP address or not? Without
these I assume something is incorrect and I do not hear ringback
Thanks!
On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris m...@jerris.com wrote:
Your packet traces would disagree with the statements below. It is
sending your internal address in rtp, so its not set correctly on
whatever profile your using to call out,
MIke
On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote:
Hi Mike,
I should have put that in also.
I do have external_rtp_ip set in my config. I have it set to my
domain name:
X-PRE-PROCESS cmd=set data=external_rtp_ip=host:mydomainname/
I should also mention that if I use flaphone.com (which registers with
an external IP address), then I get audio. In sofia, I see my IP
addresses:
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
==
Name internal
Domain Name N/A
DBName sofia_reg_internal
Pres Hosts
Dialplan XML
Context public
Challenge Realm auto_from
RTP-IP 192.168.1.120
Ext-RTP-IP 124.xxx.xxx.xxx
SIP-IP 192.168.1.120
Ext-SIP-IP 124.xxx.xxx.x
URL sip:mod_so...@192.168.1.120:5060
BIND-URL sip:mod_so...@192.168.1.120:5060
HOLD-MUSIC silence
Re: [Freeswitch-users] Extension: No audio
I think I've fixed it, but I had to change a few things...
I had a host name set in vars.xml for external_rtp_ip and for
external_sip_ip. Having the external_rtp_ip set to a hostname, sofia
showed the
RTP-IP 192.168.1.120
Ext-RTP-IP host:myhostname
SIP-IP 192.168.1.120
Ext-SIP-IP 124.190.249.9
I think this caused some problems. Once this was changed back to
stun, I now get RINGING messages and I get audio.
I still have ext-rtp-ip and ext-sip-ip set to auto-nat in internal.xml.
Could this be the cause or is there something else that caused this
issue? I am using FreeSWITCH Version 1.0.trunk (15126)
On Mon, Nov 9, 2009 at 2:40 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Is there a way to determine if FS has detected nat? I am behind UPnP
and I can see on the router the mappings for Freeswitch.
2009/11/9 João Mesquita jmesqu...@freeswitch.org:
It is if FS was able to detect NAT. Are you behind PMP or UPnP? Otherwise,
no go
Have you changed the ext-sip-ip too?
Regards,
JM
On Mon, Nov 9, 2009 at 12:32 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi again,
Actually, changing the param name=ext-rtp-ip value=auto-nat/ to
param name=ext-rtp-ip value=$${external_sip_ip}/ means that I
now see the IP address in the INVITE message:
v=0
o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 124.xxx.xxx.xxx
t=0 0
m=audio 21234 RTP/AVP 0 2 9 8 101 13
Why would this be? I thought auto-nat was meant to solve these issues?
However, I still do not see the TRYING or RINGING messages ideas
appreciated.
Thanks!
On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
OK.. thanks Mike.
I assume I am using the Internal profile. I have defined user 2000
in the 'directory' using a context called family: switch_ivr.c:1367
Transfer sofia/internal/1...@192.168.1.120 to xml[2...@family]
This is an extract from sofia:
sofia status profile internal
=
Name internal
Domain Name N/A
DBName sofia_reg_internal
Pres Hosts
Dialplan XML
Context public
Challenge Realm auto_from
RTP-IP 192.168.1.120
Ext-RTP-IP 124.xxx.xxx.xxx
SIP-IP 192.168.1.120
Ext-SIP-IP 124.xxx.xxx.xxx
URL sip:mod_so...@192.168.1.120:5060
BIND-URL sip:mod_so...@192.168.1.120:5060
HOLD-MUSIC silence
OUTBOUND-PROXY N/A
CODECS G726-32,G722,PCMU,PCMA
TEL-EVENT 101
DTMF-MODE rfc2833
CNG 13
SESSION-TO 0
MAX-DIALOG 0
NOMEDIA false
LATE-NEG false
PROXY-MEDIA false
AGGRESSIVENAT true
STUN-ENABLED true
STUN-AUTO-DISABLE false
CALLS-IN 100
FAILED-CALLS-IN 25
CALLS-OUT 38
FAILED-CALLS-OUT 31
Registrations:
=
Call-ID: 68534bba9b461...@58.169.138.53
User: 2...@192.168.1.120
Contact: user sip:2...@58.xxx.xxx.xxx:5060
Agent: dunno
Status: Registered(UDP)(unknown) EXP(2009-11-09 14:58:30)
Host: freeswitch
IP: 58.xxx.xxx.xxx
Port: 5060
Auth-User: 2000
Auth-Realm: markcs.dyndns.org
MWI-Account: 2...@192.168.1.120
The internal.xml file has a lot in it, but I guess these are the
important things for this profile:
param name=ext-rtp-ip value=auto-nat/
param name=ext-sip-ip value=auto-nat/
param name=sip-port value=$${internal_sip_port}/
param name=rtp-ip value=auto/
I will try to change auto-nat to be $${external_sip_ip}
One question though: Any idea why I never see the TRYING or RINGING
messages? Are tehse related to the RTP IP address or not? Without
these I assume something is incorrect and I do not hear ringback
Thanks!
On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris m...@jerris.com wrote:
Your packet traces would disagree with the statements below. It is
sending your internal address in rtp, so its not set correctly on
whatever profile your using to call out,
MIke
On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote:
Hi Mike,
I should have put that in also.
I do have external_rtp_ip set in my config. I have it set to my
domain name:
X-PRE-PROCESS cmd=set data=external_rtp_ip=host:mydomainname/
I should also mention that if I use flaphone.com (which registers with
an external IP address), then I get audio
Re: [Freeswitch-users] Core Dump question!
I think the (exported) means you don't have the latest svn, but probably the officially released build 1.0.4 that can be downloaded from the FS page. I think you should see something like (the latest trunk is 15203): freeswi...@internal version FreeSWITCH Version 1.0.trunk (15126) I guess you need to checkout the latest FS trunk On Fri, Oct 23, 2009 at 2:25 PM, Ujjval Karihaloo ujj...@simplesignal.comwrote: freeswi...@internal version FreeSWITCH Version 1.0.4 (exported) freeswi...@internal Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 [image: bvoip] http://www.simplesignal.com/ *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Diego Viola *Sent:* Thursday, October 22, 2009 9:04 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Core Dump question! Type version on the CLI. On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo ujj...@simplesignal.com wrote: How do I tell if it’s the latest…I downloaded is yesterday..and installed it from freeswitch.org Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 [image: bvoip] http://www.simplesignal.com/ *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* Thursday, October 22, 2009 6:26 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Core Dump question! Yes, if this is latest SVN (after a make current) then open a jira. -MC On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo ujj...@simplesignal.com wrote: I do have the core dump, should I open a ticket. I am running latest Freeswitch 1.0.4 and had done a make current just before it happened. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org image001.jpg___ FreeSWITCH-users mailing list FreeSWITCH-users@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] NOT in dialplan expression
Hi!
How do I do a NOT equal to in a dialplan expression
Normaly in regex I would use the ! character. This doesn't seem to work in FS..
ie
condition field=${variable} expression=!^1
Shouldn't that match when the variable is not starting with one?
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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
Can't you use the inline statement to set a variable so that it can be
used directly in a condition?
http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
On Thu, Oct 22, 2009 at 3:08 PM, Ahmed Munir ahmedmunir...@gmail.com wrote:
Hi,
Thanks for reply, it really helped me. One more thing to ask, how can we
make decision against ,, =, = in condition header? Like we use == for
action and != for anti-action.
Kindly highlight it.
-- Forwarded message --
From: Ahmed Munir ahmedmunir...@gmail.com
To: FreeSwitch freeswitch-users@lists.freeswitch.org
Date: Wed, 21 Oct 2009 15:37:15 +0500
Subject: [Freeswitch-users] Call custom variable in condition
Hi,
I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
want to use it in condition, as I'm listing down the configuration below;
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS}
XML Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=AUTHENTICATION_STATUS expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
But unfortunately it is not working. Kindly advise me how to do implement
it(Note: I don't want to call script). And one more thing to ask how can I
transfer the values within the same context?
--
Regards,
Ahmed Munir
-- Forwarded message --
From: Ghulam Mustafa mustafa...@gmail.com
To: freeswitch-us...@lists.freeswitch.org
Date: Wed, 21 Oct 2009 15:52:24 +0500
Subject: Re: [Freeswitch-users] Call custom variable in condition
Ahmed,
you can't use variables set by set application within a condition,
though it doesn't make sense. wondering if there is any logic behind this or
it's just a simple missing feature. anyone?
-m
Ahmed Munir wrote:
Hi,
I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
want to use it in condition, as I'm listing down the configuration below;
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS}
XML Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=AUTHENTICATION_STATUS expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
But unfortunately it is not working. Kindly advise me how to do
implement it(Note: I don't want to call script). And one more thing to ask
how can I transfer the values within the same context?
--
Regards,
Ahmed Munir
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-- Forwarded message --
From: Tihomir Culjaga tculj...@gmail.com
To: freeswitch-us...@lists.freeswitch.org
Date: Wed, 21 Oct 2009 13:13:13 +0200
Subject: Re: [Freeswitch-users] Call custom variable in condition
consider this:
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS}
XML Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=${AUTHENTICATION_STATUS} expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
here is one of my dialplan. I'm using execute_extension but it is quite
the same...
extension name=ServiceLookup
condition field=destination_number expression=(^300030)(.*)
action application=lookup_service_destination data=in
${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
1, in ${network_addr}:5060, out red_contact, out authResult/
action application=log data=INFO
ServiceLookup \n/
action application=log data=INFO
contact = '${red_contact}' ##\n/
action application=log data=INFO
CallerNum
Re: [Freeswitch-users] 2 voicemail questions
1. Can I email the voicemail message to multiple email addresses?
I revisited this again after not requiring it for a while. A comma
separated list in the extension.xml file does work. The problem was
the template file. Once I removed and in the To: field, I can
send to multple emails no problems:
In 1000.xml (for example)
param name=vm-mailto value=em...@number1.com,em...@number2.com/
And then I editied voicemail.tpl and changed the second line from:
To: ${voicemail_email}
TO:
To: ${voicemail_email}
Not sure if this is a fault in the mail server or not. Maybe the
default templates should be changed to handle this?
On Fri, Jul 10, 2009 at 8:57 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
1. Can I email the voicemail message to multiple email addresses?
A comma separated list does not work in the extensions.xml file
(1000.xml), but it does work if I hard code the email addresses into
the notify-voicemail.tpl file.
Could this be added to the switch so that it can handle comma separated lists?
2. How can I make Freeswitch dial a number AFTER a voicemail is left?
api Hangup hook?
i want the 'voicemail' application to appear to call the extension to
notify the user that there is a waiting message. This is an extract
from my dialplan.xml:
condition field=destination_number expression=^(10[01][0-9])$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
!-- action application=set data=ringback=${us-ring}/ --
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=bridge
data=user/${dialed_extensi...@${domain_name}/
action application=answer/
action application=sleep data=1000/
action application=voicemail data=default ${domain_name}
${dialed_extension}/
/condition
condition
field=${vm_boxcount(${destination_numb...@${domain_name})}
expression=^(1)$
action application=log data=MSGBOX
${vm_boxcount(${dialed_extensi...@${domain_name})}/
action application=set data=api_hangup_hook=originate
sofia/internal_nat/${dialed_etension}%${domain_name} default default
Message 4000 4000 3/
This only works if the B leg (ie voicemail application) hangs up
first. This would be an unusual situation and does not achieve what I
want... is there any other way to achieve this?
Thanks
Hi!
I have 2 questions regarding voicemail ...
1. Can I email the voicemail message to multiple email addresses? If
so, what format is this in?
param name=vm-mailto value=m...@myemail.com/
Try a comma sep. list. Not sure if it will work.
2. How can I make Freeswitch dial a number AFTER a voicemail is left?
api Hangup hook?
I g
From: Brian West br...@freeswitch.org
On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote:
Hi!
I have 2 questions regarding voicemail ...
1. Can I email the voicemail message to multiple email addresses? If
so, what format is this in?
param name=vm-mailto value=m...@myemail.com/
Try a comma sep. list. Not sure if it will work.
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Re: [Freeswitch-users] validating dtmf digits received
Thanks Mike,
I have a lateish trunk and inline seems to work okay.
Does the inline statement below set variable ${code} to be used
directly or does it require transfer also? ie is digits_dialed
available for use right after a read statement (action
application=read data=1 10
ivr/ivr-please_enter_pin_followed_by_pound.wav res 1 9/ in my
case) or is it not 'set' until after the transfer?
action inline=true application=set data=code=${digits_dialed}
Thanks!
On Tue, Oct 20, 2009 at 12:32 AM, Michael Jerris m...@jerris.com wrote:
inline is new, it won't work unless your using recent trunk. That
being said, read is not being run inline, so the set is actually being
run before digits_dialed is set. You will most likely need to use
transfer in this situation.
Mike
On Oct 19, 2009, at 12:53 AM, Mark Campbell-Smith wrote:
Hi!
I simply want to validate the dtmf digits I read from a user. From
the wiki, it appears I need to use inline=true when setting the
variable so it can be used directly within the same extension.
What have I done wrong below? I have tried many different
alternatives, but the second condition field, which is meant to match
the dtmf digits received (in this case ) is never matched, and the
anti-action is called instead.
:
some code here
action application=read data=1 10
ivr/ivr-please_enter_pin_followed_by_pound.wav res 1 9/
action application=phrase data=spell,${res}/
action inline=true application=set data=code=$
{digits_dialed}/
!-- action inline=true application=set data=code=$
{res}/ --
/condition
condition field=digits_dialed expression=^$
!-- condition field=${code} expression=^$ --
!-- condition field=${res} expression=^$ --
some code here
anti-action application=hangup/
Thanks!
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[Freeswitch-users] validating dtmf digits received
Hi!
I simply want to validate the dtmf digits I read from a user.From
the wiki, it appears I need to use inline=true when setting the
variable so it can be used directly within the same extension.
What have I done wrong below? I have tried many different
alternatives, but the second condition field, which is meant to match
the dtmf digits received (in this case ) is never matched, and the
anti-action is called instead.
:
some code here
action application=read data=1 10
ivr/ivr-please_enter_pin_followed_by_pound.wav res 1 9/
action application=phrase data=spell,${res}/
action inline=true application=set data=code=${digits_dialed}/
!-- action inline=true application=set data=code=${res}/ --
/condition
condition field=digits_dialed expression=^$
!-- condition field=${code} expression=^$ --
!-- condition field=${res} expression=^$ --
some code here
anti-action application=hangup/
Thanks!
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Re: [Freeswitch-users] Dingaling: using a hostname instead of stun for rtp
Thanks for the response Mike and Brian. Using stun is not a problem and I have it working okay now. Is there something different in the implementation of the stun lookup between dingaling and sofia? I normally use stun.freeswitch.org and sofia never had a problem. It doesn't matter which stun server I use with dingaling (I have tried with numerous servers) and the same thing seems to happen. The first stun lookup fails with a timeout (on the first inbound call), and then if another dingaling call is tried directly, the second stun lookup works. I noticed a similar behaviour by issuing the stun command from fs_cli Thanks On Sun, Oct 18, 2009 at 8:04 AM, Michael Jerris m...@jerris.com wrote: If you don't have working stun, jingle is not going to work very well. It is a required part of the protocol. You need to be able to determine your external ports for media on each call, using a host name will not do this for you. Mike On Oct 16, 2009, at 10:48 AM, Brian West wrote: If you setup your own stun server it wouldn't do that But the hostlookup only solves half the problem .. getting the external IP vs poking holes for RTP which is what stun will do. /b On Oct 15, 2009, at 10:35 PM, Mark Campbell-Smith wrote: Thanks Brian. Is this something that is planned to be implemented? The workaround is to set the stun server also in the dingaling configuration, but as I said, for some reason the stun times for me out occasionally with dingaling. Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Dingaling / Jingle DTMF support?
Perfect... thanks for the + tip with googletalk... I had been using gtalk2voip but didn't want to rely on another service Wki for dingaling updated. On Sat, Oct 17, 2009 at 8:56 AM, Anthony Minessale anthony.miness...@gmail.com wrote: It already should support at least recv of rfc2833 any time. as a workaround in googletalk any string that starts with + typed in the chat box is treated as dtmf by FS e.g. +1 Once the jingle spec stops being a moving target we will re-investigate making sure its supported properly. On Thu, Oct 15, 2009 at 5:19 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi! I was wondering if the Dingaling implementation in FS supports DTMF? This is now supported in the jingle specs ( http://www.jabberforum.org/showthread.php?t=2709 ), even though Google Talk client does not currently support DTMF. If not, are there plans to implement this? Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@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_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] Dingaling / Jingle DTMF support?
Hi! I was wondering if the Dingaling implementation in FS supports DTMF? This is now supported in the jingle specs ( http://www.jabberforum.org/showthread.php?t=2709 ), even though Google Talk client does not currently support DTMF. If not, are there plans to implement this? Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Dingaling: using a hostname instead of stun for rtp
Thanks Brian. Is this something that is planned to be implemented? The workaround is to set the stun server also in the dingaling configuration, but as I said, for some reason the stun times for me out occasionally with dingaling. Thanks! On Wed, Oct 14, 2009 at 11:33 AM, Brian West br...@freeswitch.org wrote: I don't think mod_dingaling will do a lookup for host: like sofia will as it doesn't have the code for that last I checked... I could be wrong but I don't recall it doing that. /b On Oct 13, 2009, at 3:02 PM, Mark Campbell-Smith wrote: I have a hostname set in vars.conf.xml for the parameters external_rtp_ip and the external_sip_ip instead of the usual stun. I found that stun was timing out and was causing some problems. And as I have a hostname, it makes sense to use that instead of relying on stun. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] dingaling: Destination out of order
Hi! I am trying to call from FS to gtalk. This used to work, so not sure if there is a problem with my build (FreeSWITCH Version 1.0.trunk (15126)) freeswi...@internal dingaling status --DingaLing status-- login | connected mygmai...@gmail.com/gtalk| AUTHORIZED It looks okay and I also see FS registered and online in the GTALK client. When I dial (which is to call my gtalk user), I get the following in the console: 2009-10-13 20:49:13.458712 [INFO] mod_dialplan_xml.c:391 Processing 1000- in context default 2009-10-13 20:49:13.490719 [NOTICE] mod_dingaling.c:712 Close Channel N/A [CS_NEW] 2009-10-13 20:49:13.498706 [ERR] switch_ivr_originate.c:1667 Cannot create outgoing channel of type [dingaling] cause: [DESTINATION_OUT_OF_ORDER In dingaling.conf.xml, I only have the PCMU codec specified, and the ATA is requesting PCMU/8000. Any ideas why I am seeing this? Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dingaling: Destination out of order
This is all I see: console loglevel 9 +OK console log level set to DEBUG freeswi...@internal 2009-10-13 21:33:05.578863 [NOTICE] switch_channel.c:613 New Channel sofia/internal_nat/1...@192.168.1.120 [cad049fe-b7e3-11de-94a7-1dd4d003eac8] 2009-10-13 21:33:05.634924 [INFO] mod_dialplan_xml.c:391 Processing 1- in context default 2009-10-13 21:33:05.666835 [NOTICE] mod_dingaling.c:712 Close Channel N/A [CS_NEW] 2009-10-13 21:33:05.674929 [ERR] switch_ivr_originate.c:1667 Cannot create outgoing channel of type [dingaling] cause: [DESTINATION_OUT_OF_ORDER] 2009-10-13 21:33:05.674929 [INFO] mod_dptools.c:2133 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2009-10-13 21:33:05.674929 [NOTICE] mod_dptools.c:2166 Hangup sofia/internal_nat/1...@192.168.1.120 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2009-10-13 21:33:05.802716 [NOTICE] switch_core_session.c:1087 Session 16 (sofia/internal_nat/1...@192.168.1.120) Ended 2009-10-13 21:33:05.807529 [NOTICE] switch_core_session.c:1089 Close Channel sofia/internal_nat/1...@192.168.1.120 [CS_DESTROY] On Tue, Oct 13, 2009 at 9:16 PM, Jason White ja...@jasonjgw.net wrote: Mark Campbell-Smith mcampbellsm...@gmail.com wrote: When I dial (which is to call my gtalk user), I get the following in the console: [snip] Could you turn on debug logging in the console and post the output? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dingaling: Destination out of order
I've fixed the problem. My dialplan for outbound calling had a typo: action application=bridge data=dingaling/gtalk/mygmai...@gmail.com/ The gtalk was gtallk somehow . On Tue, Oct 13, 2009 at 8:56 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi! I am trying to call from FS to gtalk. This used to work, so not sure if there is a problem with my build (FreeSWITCH Version 1.0.trunk (15126)) freeswi...@internal dingaling status --DingaLing status-- login | connected mygmai...@gmail.com/gtalk | AUTHORIZED It looks okay and I also see FS registered and online in the GTALK client. When I dial (which is to call my gtalk user), I get the following in the console: 2009-10-13 20:49:13.458712 [INFO] mod_dialplan_xml.c:391 Processing 1000- in context default 2009-10-13 20:49:13.490719 [NOTICE] mod_dingaling.c:712 Close Channel N/A [CS_NEW] 2009-10-13 20:49:13.498706 [ERR] switch_ivr_originate.c:1667 Cannot create outgoing channel of type [dingaling] cause: [DESTINATION_OUT_OF_ORDER In dingaling.conf.xml, I only have the PCMU codec specified, and the ATA is requesting PCMU/8000. Any ideas why I am seeing this? Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Dingaling: using a hostname instead of stun for rtp
Hi! I have a hostname set in vars.conf.xml for the parameters external_rtp_ip and the external_sip_ip instead of the usual stun. I found that stun was timing out and was causing some problems. And as I have a hostname, it makes sense to use that instead of relying on stun. However, when I use Dingaling, I see the following error message in the dl_debug tracing: iq to=FS_EMAIL@gmail.com/gtalk77BBCB94 id=344 type=error from=email1@gmail.com/Talk.v10488E8134A ses:session type=candidates id=3697758817 initiator=email1@gmail.com/Talk.v10488D9134A xmlns:ses=http://www.google.com/session; ses:candidate name=rtp address=host:hostname port=30710 username=nVv7FqwNyWbiMuTt password=nVv7BqwQyVbiMuTt preference=1.0 protocol=udp type=local network=0 generation=0/ses:candidate /ses:session error type=modify sta:bad-request xmlns:sta=urn:ietf:params:xml:ns:xmpp-stanzas/sta:bad-request sta:text xml:lang=en xmlns:sta=urn:ietf:params:xml:ns:xmpp-stanzascandidate has address of zero/sta:text /error /iq Is Gtalk expecting an address in this field 'ses:candidate name=rtp address=host:hostname' instead of a hostname? Or should FS be sending the hostname only and not host:hostname? Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Question about fax tone detection
This is what I have in my dialplan and the fax is detected
beautifully. Note that in my case, extension 1000 will ring for a
second or two before the fax is detected. So in your example, the fax
does not have time to be detected, the dialplan exists and the call is
hungup.
When the fax is detected, the call is transferred to the receivefax
extension in context features. The extension 1000 does not have to be
answered for the transfer to occur.
extension name=Local_1000s
condition field=destination_number expression=^(10[01][0-9])$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=set data=ringback=${au-ring}/
action application=tone_detect data=fax 1100 r +5000
transfer 'receivefax XML features' 1 /
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=bridge data=user/${dialed_extensi...@${domain}/
action application=answer/
action application=sleep data=1000/
action application=voicemail data=default ${domain_name}
${dialed_extension}/
/condition
/extension
and in the features context I have
extension name=receivefax
condition field=destination_number expression=^receivefax$
action application=answer /
action application=playback data=silence_stream://2000/
action application=rxfax
data=//usr//local//freeswitch//storage//${caller_id_number}-${uuid}.rxfax.tiff/
action application=system
data=/usr/local/freeswitch/scripts/emailfax.sh
/usr/local/freeswitch/storage/${caller_id_number}-${uuid}.rxfax.tiff/
action application=hangup/
/condition
/extension
On Tue, Oct 13, 2009 at 5:00 AM, Michael Collins m...@freeswitch.org wrote:
On Mon, Oct 12, 2009 at 4:01 AM, homqua ngay01042...@gmail.com wrote:
Hi,
I have implemented the solution for tone detection in wiki, and also
answer
the channel before detecting the tone:
condition field=destination_number expression=^(055138419992)$
action application=answer/
action application=tone_detect data=fax 1100 r +15000 transfer fax
XML default/
extension name=fax
condition field=destination_number expression=^fax$
action application=answer /
action application=sleep data=1000/
action application=rxfax
data=/usr/local/freeswitch/storage/fax/${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.tiff/
action application=set data=fax_mode=recv/
action application=hangup/
/condition
/extension
But FS cannot recognize the tone, and therefore cannot move to fax
extension. Below are the error in FS:
2009-10-12 10:57:16.702287 [NOTICE] switch_channel.c:602 New Channel
sofia/external/anonym...@anonymous.invalid
[c431f0a3-9231-4724-ba39-9e4ef7edfca2]
2009-10-12 10:57:16.703413 [INFO] mod_dialplan_xml.c:315 Processing
Anonymous-055138419992 in context public
2009-10-12 10:57:16.719288 [NOTICE] switch_ivr.c:1349 Transfer
sofia/external/anonym...@anonymous.invalid to xml[055138419...@default]
2009-10-12 10:57:16.719288 [INFO] mod_dialplan_xml.c:315 Processing
Anonymous-055138419992 in context default
2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:649 Channel
[sofia/external/anonym...@anonymous.invalid] has been answered
2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:1324 Enabling tone
detection 'fax' '1100'
2009-10-12 10:57:16.723302 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia/external/anonym...@anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING]
2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1086 Session 1
(sofia/external/anonym...@anonymous.invalid) Ended
2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1088 Close
Channel
sofia/external/anonym...@anonymous.invalid [CS_DESTROY]
And the trace for SIP messages: http://pastebin.com/m4e47e7d9
If anyone has any idea, tell me please.
Thanks.
I think the trouble here is that you don't have anything else in the
dialplan after the tone_detect. The tone_detect app is non-block, which
means that it doesn't sit there and wait for a tone. If you want the
dialplan to sit and wait then do a sleep app after your tone_detect. The
other question I would have is this: what happens if the incoming call is
not a fax? What do you want to do then?
-MC
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Re: [Freeswitch-users] [ERR] mod_dingaling.c:980 Stun Failed!
It was the svn revision I was using... I have updated now and it is working again. On Fri, Oct 2, 2009 at 9:00 PM, Muhammad Shahzad shaherya...@googlemail.com wrote: Yes, i had same problem, then i changed stun server to one of our own servers. You may try some of public stun servers listed on below link, http://www.voip-info.org/wiki/view/STUN Thank you. On Fri, Oct 2, 2009 at 2:58 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Anyone have this issue? On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi! I have just started to use dingaling again, and noticed I constantly get a stun error. 2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed! stun.fwdnet.net:3478 [Remote Address Error!] I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers and keep getting this error with dingaling. I have no problems with inbound sip calls, so I don't think its the actual stun server. Has anyone else seen this? I am using: FreeSWITCH Version 1.0.trunk (14952) Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- | | | FATAL ERROR --- O X | |___| | You have moved the mouse. | | Windows must be restarted for the changes to take effect. | | OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] mod_fax compile fails
HI all, I just tried to update to the latest svn and I get these errors right at the end after issuing a 'make current'. I am using Debian Lenny. making all mod_fax make[5]: Entering directory `/home/mark/freeswitch/src/mod/applications/mod_fax' make[6]: Entering directory `/home/mark/freeswitch/src/mod/applications/mod_fax' make[7]: Entering directory `/home/mark/freeswitch/libs/tiff-3.8.2' running /bin/sh ./configure --prefix=/usr/local/freeswitch --cache-file=/dev/null --srcdir=. --disable-shared --with-pic --no-create --no-recursion configure: error: cannot run /bin/sh config/config.sub make[7]: *** [config.status] Error 1 make[7]: Leaving directory `/home/mark/freeswitch/libs/tiff-3.8.2' make[6]: *** [../../../../libs/tiff-3.8.2/libtiff/libtiff.la] Error 2 make[6]: Leaving directory `/home/mark/freeswitch/src/mod/applications/mod_fax' make[5]: *** [all] Error 1 make[5]: Leaving directory `/home/mark/freeswitch/src/mod/applications/mod_fax' make[4]: *** [mod_fax-all] Error 1 make[4]: Leaving directory `/home/mark/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/home/mark/freeswitch/src' Making all in build make[3]: Entering directory `/home/mark/freeswitch/build' + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +--+ make[3]: Leaving directory `/home/mark/freeswitch/build' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/home/mark/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/home/mark/freeswitch' make: *** [current] Error 2 Also, are the 'Leaving directory / all-recursive' errors going to cause a problem? Thanks! Any ideas what the cause is? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_fax compile fails
Thanks Rob, Is this a fault in the svn update process? if so, should/has it been bug reported? On Fri, Oct 9, 2009 at 2:39 PM, Rob Forman rob4manh...@gmail.com wrote: I had that issue too where make current failed on mod_fax (under libs/ tiff). And yeah, it caused a problem where a bunch of modules wouldn't load. You'll want to get it resolved before installing. I ended up moving the existing source aside and re-checked out the trunk, which compiled fine. On Oct 8, 2009, at 10:06 PM, Mark Campbell-Smith wrote: HI all, I just tried to update to the latest svn and I get these errors right at the end after issuing a 'make current'. I am using Debian Lenny. making all mod_fax make[5]: Entering directory `/home/mark/freeswitch/src/mod/ applications/mod_fax' make[6]: Entering directory `/home/mark/freeswitch/src/mod/ applications/mod_fax' make[7]: Entering directory `/home/mark/freeswitch/libs/tiff-3.8.2' running /bin/sh ./configure --prefix=/usr/local/freeswitch --cache-file=/dev/null --srcdir=. --disable-shared --with-pic --no-create --no-recursion configure: error: cannot run /bin/sh config/config.sub make[7]: *** [config.status] Error 1 make[7]: Leaving directory `/home/mark/freeswitch/libs/tiff-3.8.2' make[6]: *** [../../../../libs/tiff-3.8.2/libtiff/libtiff.la] Error 2 make[6]: Leaving directory `/home/mark/freeswitch/src/mod/ applications/mod_fax' make[5]: *** [all] Error 1 make[5]: Leaving directory `/home/mark/freeswitch/src/mod/ applications/mod_fax' make[4]: *** [mod_fax-all] Error 1 make[4]: Leaving directory `/home/mark/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/home/mark/freeswitch/src' Making all in build make[3]: Entering directory `/home/mark/freeswitch/build' + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +--+ make[3]: Leaving directory `/home/mark/freeswitch/build' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/home/mark/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/home/mark/freeswitch' make: *** [current] Error 2 Also, are the 'Leaving directory / all-recursive' errors going to cause a problem? Thanks! Any ideas what the cause is? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Detecting a fax
Thanks for the response Mike,
I read that page and this one (among others)
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
I'm still lost. This is an extract of my dialplan
extension name=Local
condition field=destination_number expression=^(10[01][0-9])$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=set data=ringback=${au-ring}/
action application=fax_detect/
action application=tone_detect data=fax 1100 r +5000
transfer fax XML features /
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=bridge data=user/${dialed_extensi...@${domain}/
I would assume that on detecting a fax, the dialplan 'fax' is called
in context features. This never happens.
When is the fax tone detected? Is it while the call is ringing or
can it be detected after the call is answered? My goal is to be able
to have the same extension for a voice and fax call. i assume that
the fax 'tones' are standardised and the ones on the wiki are correct?
Also, I guess this doesn't work with media bypass (which I don't
use).
Thanks!
On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris m...@jerris.com wrote:
check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
Note, you can't just have tone_detect as your last iten in the
dialplan as the call will just get hung up.
Mike
On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
Hi
I was hoping someone could help me to setup the fax detection / tone
detection application.
I want to be able to transfer an incoming fax to a specific extension.
In my default.xml file, I have the following (extracted):
extension name=1000
condition field=destination_number expression=^(10[01][0-9])
$
action application=fax_detect/
action application=tone_detect data=fax 1100 r +5000
transfer fax XML features /
I can't get the fax to be detected and transferred. Is there any way
this can be done?
Thanks!
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Re: [Freeswitch-users] Detecting a fax
Thanks for your help Mike and Tihomir.
A little more playing around and I found that having action
application=fax_detect/ as well as action
application=tone_detect data=fax 1100 r +5000 transfer fax XML
features / do not work together.
Simply by removing fax_detect, the fax is detected beautifully.
My problem now is trying to email the fax. I followed the
instructions on the wiki at http://wiki.freeswitch.org/wiki/Mod_fax,
but the dialplan action application=system .../ is not executed
after the rxfax command. I know the script works because if I put the
system command in another part of the dialplan and hard code the
filename to attach, then the email is sent.
extension name=receivefax
condition field=destination_number expression=^receivefax$
action application=answer /
action application=playback data=silence_stream://2000/
action application=rxfax
data=//usr//local//freeswitch//storage//${caller_id_number}-${uuid}.rxfax.tiff/
action application=system
data=/usr/local/freeswitch/scripts/emailfax.sh
/usr/local/freeswitch/storage/${caller_id_number}-${uuid}.rxfax.tiff/
action application=hangup/
/condition
/extension
ideas?
Thanks!
On Tue, Oct 6, 2009 at 1:32 AM, Tihomir Culjaga tculj...@gmail.com wrote:
hi Mark,
This is an inbound call leg and media channel (so far) is open in reverse
direction only (application ringback). I'm afraid you have to answer the
call to be able to hear the fax tone.
T.
On Mon, Oct 5, 2009 at 2:32 PM, Michael Jerris m...@jerris.com wrote:
Fax tones are not played by the remote machine until after answer, the
tone_detect application starts a media bug that listens for the tone,
can you confirm the tone is happening at all. Maybe the issue here is
the timeout, try making that longer, or doing the tone_detect in
execute_on_answer
Mike
On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:
Thanks for the response Mike,
I read that page and this one (among others)
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
I'm still lost. This is an extract of my dialplan
extension name=Local
condition field=destination_number expression=^(10[01][0-9])
$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=set data=ringback=${au-ring}/
action application=fax_detect/
action application=tone_detect data=fax 1100 r +5000
transfer fax XML features /
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=bridge data=user/${dialed_extensi...@$
{domain}/
I would assume that on detecting a fax, the dialplan 'fax' is called
in context features. This never happens.
When is the fax tone detected? Is it while the call is ringing or
can it be detected after the call is answered? My goal is to be able
to have the same extension for a voice and fax call. i assume that
the fax 'tones' are standardised and the ones on the wiki are correct?
Also, I guess this doesn't work with media bypass (which I don't
use).
Thanks!
On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris m...@jerris.com
wrote:
check out
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
Note, you can't just have tone_detect as your last iten in the
dialplan as the call will just get hung up.
Mike
On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
Hi
I was hoping someone could help me to setup the fax detection / tone
detection application.
I want to be able to transfer an incoming fax to a specific
extension.
In my default.xml file, I have the following (extracted):
extension name=1000
condition field=destination_number expression=^(10[01]
[0-9])
$
action application=fax_detect/
action application=tone_detect data=fax 1100 r +5000
transfer fax XML features /
I can't get the fax to be detected and transferred. Is there any
way
this can be done?
Thanks!
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Re: [Freeswitch-users] Detecting a fax
Further playing around and everything is working fine (even the emailing). I'm not sure what I changed though to document it. cheers /M On Mon, Oct 5, 2009 at 12:03 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi I was hoping someone could help me to setup the fax detection / tone detection application. I want to be able to transfer an incoming fax to a specific extension. In my default.xml file, I have the following (extracted): extension name=1000 condition field=destination_number expression=^(10[01][0-9])$ action application=fax_detect/ action application=tone_detect data=fax 1100 r +5000 transfer fax XML features / I can't get the fax to be detected and transferred. Is there any way this can be done? Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Detecting a fax
Hi I was hoping someone could help me to setup the fax detection / tone detection application. I want to be able to transfer an incoming fax to a specific extension. In my default.xml file, I have the following (extracted): extension name=1000 condition field=destination_number expression=^(10[01][0-9])$ action application=fax_detect/ action application=tone_detect data=fax 1100 r +5000 transfer fax XML features / I can't get the fax to be detected and transferred. Is there any way this can be done? Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [ERR] mod_dingaling.c:980 Stun Failed!
Anyone have this issue? On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi! I have just started to use dingaling again, and noticed I constantly get a stun error. 2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed! stun.fwdnet.net:3478 [Remote Address Error!] I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers and keep getting this error with dingaling. I have no problems with inbound sip calls, so I don't think its the actual stun server. Has anyone else seen this? I am using: FreeSWITCH Version 1.0.trunk (14952) Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] [ERR] mod_dingaling.c:980 Stun Failed!
Hi! I have just started to use dingaling again, and noticed I constantly get a stun error. 2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed! stun.fwdnet.net:3478 [Remote Address Error!] I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers and keep getting this error with dingaling. I have no problems with inbound sip calls, so I don't think its the actual stun server. Has anyone else seen this? I am using: FreeSWITCH Version 1.0.trunk (14952) Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH Start up Script - Ubuntu Server
What about this one for Debian...
http://wiki.freeswitch.org/wiki/Freeswitch_init
On Mon, Sep 28, 2009 at 9:15 AM, Aloysius Thevarajah Lloyd
lloyd.aloys...@gmail.com wrote:
Yes. I have seen the scripts. But I could not find a suitable one for
Ubuntu.
Thank you.
LLoyd
2009/9/27 João Mesquita jmesqu...@freeswitch.org
Only 3 init scripts available on trunk today (${SVNROOT}/build) are for
archlinux, redhat or suse.
We would love to have more for other distros.
Regards,
jmesquita
On Sun, Sep 27, 2009 at 10:42 AM, Aloysius Thevarajah Lloyd
lloyd.aloys...@gmail.com wrote:
Hi All,
I am trying to setup FreeSwitch on a Ubuntu Server.
Where can I find the start up(boot time) script for FreeSwitch on a
Ubuntu Server?
Thank you .
Lloyd
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Re: [Freeswitch-users] FreeSWITCH Start up Script - Ubuntu Server
Yep.. it works for me. You will probably have to modify these lines
to match the user/group that FS normally run user on your system:
FS_USER=freeswitch
FS_GROUP=freeswitch
On Mon, Sep 28, 2009 at 10:37 AM, Aloysius Thevarajah Lloyd
lloyd.aloys...@gmail.com wrote:
I try earlier today this script ... but it is not working. Did you try ?
On Sun, Sep 27, 2009 at 8:12 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
What about this one for Debian...
http://wiki.freeswitch.org/wiki/Freeswitch_init
On Mon, Sep 28, 2009 at 9:15 AM, Aloysius Thevarajah Lloyd
lloyd.aloys...@gmail.com wrote:
Yes. I have seen the scripts. But I could not find a suitable one for
Ubuntu.
Thank you.
LLoyd
2009/9/27 João Mesquita jmesqu...@freeswitch.org
Only 3 init scripts available on trunk today (${SVNROOT}/build) are for
archlinux, redhat or suse.
We would love to have more for other distros.
Regards,
jmesquita
On Sun, Sep 27, 2009 at 10:42 AM, Aloysius Thevarajah Lloyd
lloyd.aloys...@gmail.com wrote:
Hi All,
I am trying to setup FreeSwitch on a Ubuntu Server.
Where can I find the start up(boot time) script for FreeSwitch on a
Ubuntu Server?
Thank you .
Lloyd
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Re: [Freeswitch-users] Unknown call drops.. INFO DTMF(3)
The problem hasn't been seen again, but the exact call case has not been performed again. Will update freeswitch and monitor next time. Cheers MCS On Sat, Sep 19, 2009 at 6:27 AM, Michael Collins m...@freeswitch.org wrote: Is this still happening? If so please make sure that you are on latest trunk and re-test. Get a pcap of the traffic (SIP and RTP) for review and then report back. Thanks, MC On Sun, Sep 13, 2009 at 2:34 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi! I have just experienced some call drops and each time the sequence is the same in the freeswitch.log file. Both parties are sure that they did not accidentally hit the 3 button to send the DTMF tone (and the same thing has happened four times already after ~5 minutes). 2009-09-13 19:28:23.835216 [DEBUG] sofia.c:4329 INFO DTMF(3) 2009-09-13 19:28:23.835216 [DEBUG] sofia.c:4450 dispatched freeswitch event for INFO 2009-09-13 19:28:23.859408 [DEBUG] switch_rtp.c:1624 Send start packet for [3] ts=2591120 dur=160/160/13120 seq=64923 2009-09-13 19:28:23.879439 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=320/320/13120 seq=64924 2009-09-13 19:28:23.899455 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=480/480/13120 seq=64925 : : 2009-09-13 19:28:25.439404 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=12800/12800/13120 seq=65002 2009-09-13 19:28:25.459312 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=12960/12960/13120 seq=65003 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65004 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65005 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65006 2009-09-13 19:28:33.879341 [NOTICE] sofia.c:322 Hangup sofia/external/number [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-09-13 19:28:33.879341 [DEBUG] switch_channel.c:1683 Send signal sofia/external/number [KILL] 2009-09-13 19:28:33.879341 [DEBUG] switch_core_session.c:932 Send signal sofia/external/number [BREAK] 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:371 sofia/external/number ending bridge by request from write function 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:426 sofia/internal_nat/1...@192.168.1.120 receive message [UNBRIDGE] 2009-09-13 19:28:33.900940 [DEBUG] switch_core_session.c:630 Send signal sofia/internal_nat/1...@192.168.1.120 [BREAK] 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:452 BRIDGE THREAD DONE [sofia/internal_nat/1...@192.168.1.120] 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:454 Send signal sofia/external/number [BREAK] 2009-09-13 19:28:33.912049 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal_nat/1...@192.168.1.120 [CS_EXECUTE] [NORMAL_CLEARING] Anyone have any idea what this sequence means and why I am getting this? Is it my sip provider or something in FreeSwitch? What does the 'Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65006' mean? Notice that dur (duration?) is increasing a lot until the call drops. Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Simple call waiting question
Thanks Brian, (I bought a dud phone.. and its a new DECT! - crazy) I am using the 5900 and 5901 for parking/unparking. That functionality works fine and I can park/unpark the B leg as I wish. The problem is that if I park the B-leg, the A-leg then gets a busy signal. If the A leg is then hung up, a user-busy signal is sent to the C-leg, so the call goes to voicemail. What I want to happen is park B and answer C directly. Is this possible? On Thu, Sep 17, 2009 at 11:21 PM, Brian West br...@freeswitch.org wrote: Personally I would throw the phone in the trash. :P In the default dialplan look at 5900 for park and 5901 for unpark. /b On Sep 17, 2009, at 7:58 AM, Mark Campbell-Smith wrote: I am trying to create a simple call waiting dialplan as my phone does not have Recall button. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] Unknown call drops.. INFO DTMF(3)
Hi! I have just experienced some call drops and each time the sequence is the same in the freeswitch.log file. Both parties are sure that they did not accidentally hit the 3 button to send the DTMF tone (and the same thing has happened four times already after ~5 minutes). 2009-09-13 19:28:23.835216 [DEBUG] sofia.c:4329 INFO DTMF(3) 2009-09-13 19:28:23.835216 [DEBUG] sofia.c:4450 dispatched freeswitch event for INFO 2009-09-13 19:28:23.859408 [DEBUG] switch_rtp.c:1624 Send start packet for [3] ts=2591120 dur=160/160/13120 seq=64923 2009-09-13 19:28:23.879439 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=320/320/13120 seq=64924 2009-09-13 19:28:23.899455 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=480/480/13120 seq=64925 : : 2009-09-13 19:28:25.439404 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=12800/12800/13120 seq=65002 2009-09-13 19:28:25.459312 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=12960/12960/13120 seq=65003 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65004 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65005 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65006 2009-09-13 19:28:33.879341 [NOTICE] sofia.c:322 Hangup sofia/external/number [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-09-13 19:28:33.879341 [DEBUG] switch_channel.c:1683 Send signal sofia/external/number [KILL] 2009-09-13 19:28:33.879341 [DEBUG] switch_core_session.c:932 Send signal sofia/external/number [BREAK] 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:371 sofia/external/number ending bridge by request from write function 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:426 sofia/internal_nat/1...@192.168.1.120 receive message [UNBRIDGE] 2009-09-13 19:28:33.900940 [DEBUG] switch_core_session.c:630 Send signal sofia/internal_nat/1...@192.168.1.120 [BREAK] 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:452 BRIDGE THREAD DONE [sofia/internal_nat/1...@192.168.1.120] 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:454 Send signal sofia/external/number [BREAK] 2009-09-13 19:28:33.912049 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal_nat/1...@192.168.1.120 [CS_EXECUTE] [NORMAL_CLEARING] Anyone have any idea what this sequence means and why I am getting this? Is it my sip provider or something in FreeSwitch? What does the 'Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65006' mean? Notice that dur (duration?) is increasing a lot until the call drops. Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Dialplan Context
Hi! Where in the dialplan does FS decide which context is used for processing.. I am dialing an outbound call but the call is being processed in context public and not default? mod_dialplan_xml.c:315 Processing Extension1000-dest number in context public Why is FS choosing the public context for an outbound call instead of the normal default context? Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Dialplan Context
Hi!
Actually I think the problem was with the acl list..
I had put commented line below in. How did this cause the internal
profile be executed in the public extension?
list name=domains default=deny
node type=allow domain=$${domain}/
!-- node type=allow cidr=192.168.0.0/16/ -- - when
removed it works again
Thanks
On Thu, Sep 10, 2009 at 10:09 PM, Tihomir Culjagatculj...@gmail.com wrote:
check your sip profiles
/usr/local/freeswitch/conf/sip_profiles/external.xml
param name=context value=public/
/usr/local/freeswitch/conf/sip_profiles/internal.xml
param name=context value=default/
/usr/local/freeswitch/conf/vars.xml
!-- Internal SIP Profile --
X-PRE-PROCESS cmd=set data=internal_auth_calls=true/
X-PRE-PROCESS cmd=set data=internal_sip_port=5080/
X-PRE-PROCESS cmd=set data=internal_tls_port=5081/
X-PRE-PROCESS cmd=set data=internal_ssl_enable=false/
X-PRE-PROCESS cmd=set data=internal_ssl_dir=$${base_dir}/conf/ssl/
!-- External SIP Profile --
X-PRE-PROCESS cmd=set data=external_auth_calls=false/
X-PRE-PROCESS cmd=set data=external_sip_port=5060/
X-PRE-PROCESS cmd=set data=external_tls_port=5061/
X-PRE-PROCESS cmd=set data=external_ssl_enable=false/
X-PRE-PROCESS cmd=set data=external_ssl_dir=$${base_dir}/conf/ssl/
It is simple :P
On Thu, Sep 10, 2009 at 1:39 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
Where in the dialplan does FS decide which context is used for
processing.. I am dialing an outbound call but the call is being
processed in context public and not default?
mod_dialplan_xml.c:315 Processing Extension1000-dest number in context
public
Why is FS choosing the public context for an outbound call instead of
the normal default context?
Thanks!
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Re: [Freeswitch-users] New to Freeswitch - some help needed
Hi Alan, I hope you find your answers here as these are the sort of things that are hard to find on the wiki, which is somewhat outdated in areas. If you do find your answers, please post them back here for everyone else. I am new to FS also, so my comments below may not be 100% correct! 1. Very similar to what I want to have setup as well. Do you have a static IP address at home. If not, get a dyndns account and setup an entry there so that your friends/family can register using your dns name instead of ip address 2. No idea. Maybe try another stun server? 3. Not sure if double-NAT is needed now with the newer builds of FreeSwitch. Download the latest 1.0.4 to be on the safeside and compile it again! (I have FS 1.0.4 pre9 and it works I think). As long as your clients can register remotely you should be okay. I think FS can work around most home NATs. Make sure you have auto-nat set in your internal.xml file (I think its this one) 4. SIP is the signaling. RTP is the payload, or voice in your case. Any transition is done via the SIP signaling. This is how FS can transfer calls etc or use the media bypass mode by specifying the IP address where the RTP should be sent, which does not have to be the same as the signaling. Make sure you enable tracing in the internal.xml file so you can debug the signaling. You don't need to take a laptop to your daughters to test this. Use an internet sip phone like flaphone.com, which works through your web browser. This will register with an external IP address exactly like your daughters and save you time traveling. Note that sound isn't so clear for me using this service, but it helps with debugging. I also would recommend a sip client on windows like Zoiper, or CounterPath's X-Lite.. both are free. X-Lite is well known, Zoiper allows for multiple SIP registrations and comes in a portable version. On Fri, Aug 7, 2009 at 6:18 PM, Alan Chandlera...@chandlerfamily.org.uk wrote: I apologize, as my first post to this list, that I ask a detailed set of questions, but I have spend some time looking at all the docs and can't get what I need to do completely sorted in my head. I am definitely one who likes to UNDERSTAND what is happening rather than follow blank recipies, so please bear with me as I try understand all the details. I do understand about networking, NAT etc - but I am new to SIP/RTP and in particular what I think is a double NAT problem Firstly - what am I trying to achieve: I am in the UK and have a small home network behind a D-Link DIR-100 Router/NAT/Firewall one of those machines, running Debian Lenny, acts as my main server for everything (and in an earlier incarnation was the firewall/router/nat box too - I only say this is because I had all this working using Asterisk a year or so ago, but with this important difference in configuration). Many of the ports on the firewall are port forwarded to this machine. I have set Freeswitch up on this server to act as a small voip pbx for the home - but MORE IMPORTANTLY - to enable my daughter from her house to talk to us. At my house locally I have a Linksys PAP2T two phone SIP box - and that is working with Freeswitch's default configuration (I set up to be 1000 and 1001 and used all the facilities). I will later add a Linksys SPA 3102 - although I DO NOT intend to use its facility to bridge to the normal phone network. My daughter, living in another house, also has a Nat box (unknown - its part of her ADSL modem/router/wireless access point) and also has a PAP2T which she will connect to the her network. This will be her phone. There is a family relation living in Australia who will load up a whatever softphone that we tell him to use. I expect, but don't know, that he will behind a NAT box too. Later, I have some friends in the USA that I might wish to add it too - especially so that we can hold some teleconferences. They will have a mixture of Windows and MACs, and I will need to recommend softphone clients for them. I want to set this up as a small private voice network, so anyone can ring anyone else. I will add fancy facilities such as conferencing and voicemail later - I just want to get the basics working first. Secondly I installed a stun client on my home machine and ran it against stun.freeswitch.org. It reported:- Primary: Independent Mapping, Independent Filter, preserves ports, no hairpin But I have no idea what this means - I can't find any clear statement via googling for it - how this set of answers maps to the different types of NAT that might be required to get this to all work. CAN SOMEONE ENLIGHTEN me please. Thirdly I have set up a sip profile called double nat from the recipe in the wiki. This defines the SIP port to be 5090. However, what I DO NOT UNDERSTAND is how the PAP2T box in my daughters house will initiate a connection to my server. Presumably, I have to port forward 5090 from the
[Freeswitch-users] NAT'd FS / publice softphone problems
Hi All, I know this question has come up before but I couldn't find the answer that I could understand! Sorry in advance. My setup is: Freeswtch NAT'd (192.168.x.x) - Router - Internet - Softphone with public IP I can easily get the softphones to register, but when I try to call from the softphone to voicemail (for example), I don't get any audio. I checked out this page: http://wiki.freeswitch.org/wiki/External_profile (section Switch with External SoftPhone) but I am not clear how I can get this to work. I have played around with the rtp-ip and external-rtp-ip but without success. Is it possible for someone to help me configure this so softphones that are outside the nat'd lan get audio correctly? Help appreciated! Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] 2 voicemail questions
Hi Mike,
This was my dialplan (extracted from my last email):
2. How can I make Freeswitch dial a number AFTER a voicemail is left?
api Hangup hook?
i want the 'voicemail' application to appear to call the extension to
notify the user that there is a waiting message. This is an extract
from my dialplan.xml:
condition field=destination_number expression=^(10[01][0-9])$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
!-- action application=set data=ringback=${us-ring}/ --
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=bridge
data=user/${dialed_extensi...@${domain_name}/
action application=answer/
action application=sleep data=1000/
action application=voicemail data=default ${domain_name}
${dialed_extension}/
/condition
condition
field=${vm_boxcount(${destination_numb...@${domain_name})}
expression=^(1)$
action application=log data=MSGBOX
${vm_boxcount(${dialed_extensi...@${domain_name})}/
action application=set data=api_hangup_hook=originate
sofia/internal_nat/${dialed_etension}%${domain_name} default default
Message 4000 4000 3/
This only works if the B leg (ie voicemail application) hangs up
first. This would be an unusual situation and does not achieve what I
want... is there any other way to achieve this?
On Sat, Jul 11, 2009 at 2:04 AM, Michael Jerrism...@jerris.com wrote:
could you post how you tired to do it in dialplan that didn't work?
Mike
On Jul 10, 2009, at 5:57 AM, Mark Campbell-Smith wrote:
Hi!
1. Can I email the voicemail message to multiple email addresses?
A comma separated list does not work in the extensions.xml file
(1000.xml), but it does work if I hard code the email addresses into
the notify-voicemail.tpl file.
Could this be added to the switch so that it can handle comma
separated lists?
2. How can I make Freeswitch dial a number AFTER a voicemail is
left?
api Hangup hook?
i want the 'voicemail' application to appear to call the extension to
notify the user that there is a waiting message. This is an extract
from my dialplan.xml:
condition field=destination_number expression=^(10[01][0-9])
$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
!-- action application=set data=ringback=${us-ring}/ --
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=bridge
data=user/${dialed_extensi...@${domain_name}/
action application=answer/
action application=sleep data=1000/
action application=voicemail data=default ${domain_name}
${dialed_extension}/
/condition
condition
field=${vm_boxcount(${destination_numb...@${domain_name})}
expression=^(1)$
action application=log data=MSGBOX
${vm_boxcount(${dialed_extensi...@${domain_name})}/
action application=set data=api_hangup_hook=originate
sofia/internal_nat/${dialed_etension}%${domain_name} default default
Message 4000 4000 3/
This only works if the B leg (ie voicemail application) hangs up
first. This would be an unusual situation and does not achieve what I
want... is there any other way to achieve this?
Thanks
Hi!
I have 2 questions regarding voicemail ...
1. Can I email the voicemail message to multiple email addresses? If
so, what format is this in?
param name=vm-mailto value=m...@myemail.com/
Try a comma sep. list. Not sure if it will work.
2. How can I make Freeswitch dial a number AFTER a voicemail is left?
api Hangup hook?
I g
From: Brian West br...@freeswitch.org
On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote:
Hi!
I have 2 questions regarding voicemail ...
1. Can I email the voicemail message to multiple email addresses? If
so, what format is this in?
param name=vm-mailto value=m...@myemail.com/
Try a comma sep. list. Not sure if it will work.
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Re: [Freeswitch-users] 2 voicemail questions
Hi!
1. Can I email the voicemail message to multiple email addresses?
A comma separated list does not work in the extensions.xml file
(1000.xml), but it does work if I hard code the email addresses into
the notify-voicemail.tpl file.
Could this be added to the switch so that it can handle comma separated lists?
2. How can I make Freeswitch dial a number AFTER a voicemail is left?
api Hangup hook?
i want the 'voicemail' application to appear to call the extension to
notify the user that there is a waiting message. This is an extract
from my dialplan.xml:
condition field=destination_number expression=^(10[01][0-9])$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
!-- action application=set data=ringback=${us-ring}/ --
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=bridge
data=user/${dialed_extensi...@${domain_name}/
action application=answer/
action application=sleep data=1000/
action application=voicemail data=default ${domain_name}
${dialed_extension}/
/condition
condition
field=${vm_boxcount(${destination_numb...@${domain_name})}
expression=^(1)$
action application=log data=MSGBOX
${vm_boxcount(${dialed_extensi...@${domain_name})}/
action application=set data=api_hangup_hook=originate
sofia/internal_nat/${dialed_etension}%${domain_name} default default
Message 4000 4000 3/
This only works if the B leg (ie voicemail application) hangs up
first. This would be an unusual situation and does not achieve what I
want... is there any other way to achieve this?
Thanks
Hi!
I have 2 questions regarding voicemail ...
1. Can I email the voicemail message to multiple email addresses? If
so, what format is this in?
param name=vm-mailto value=m...@myemail.com/
Try a comma sep. list. Not sure if it will work.
2. How can I make Freeswitch dial a number AFTER a voicemail is left?
api Hangup hook?
I g
From: Brian West br...@freeswitch.org
On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote:
Hi!
I have 2 questions regarding voicemail ...
1. Can I email the voicemail message to multiple email addresses? If
so, what format is this in?
param name=vm-mailto value=m...@myemail.com/
Try a comma sep. list. Not sure if it will work.
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[Freeswitch-users] 2 voicemail questions
Hi! I have 2 questions regarding voicemail ... 1. Can I email the voicemail message to multiple email addresses? If so, what format is this in? param name=vm-mailto value=m...@myemail.com/ 2. How can I make Freeswitch dial a number AFTER a voicemail is left? Thanks! ___ Freeswitch-users mailing list Freeswitch-users@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] freeswitch segfault
Hi! My call dropped and I saw this error in the syslog: Jun 24 17:05:04 freeswitch kernel: [157531.309017] freeswitch[4621]: segfault at c ip b73b2a42 sp b72a3840 error 4 in mod_sofia.so[b7369000+16c000] How can I get more information on this fault to file a bug report? Thanks! ___ Freeswitch-users mailing list Freeswitch-users@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] mod_dingaling picking wrong IP address / no audio?
Hi! I am trying to call from my corporate network (firewalled) using Gtalk to Freeswitch. I am not getting any audio. In the logs I see that mod_dingaling is using my internal corporate IP address which is not publically addressable. 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates 146.xx.xx.xx:50320 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable Candidate 146.xx.xx.xx:50320 Further on in the log, I can see GTalk sending a new candidate IP address to use: 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 name=rtp type=local protocol=udp username=e+JTkVHT1xEkqXGD password=fAxU6Pr1oF9Zq48U address=192.168.1.102 port=50322 pref=1.00 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 name=rtp type=stun protocol=udp username=RBqyF2XNMYLfJNoU password=DQMjon1fSVoJIRTp address=124.xxx.xxx.xxx port=50323 pref=0.90 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=relay protocol=udp username=62L5zs2FHbcUdeCJ password=KxmNgkUmZsLfuX6S address=209.xx.xxx.xxx port=19295 pref=0.50 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] Because of this, I never get audio. Any ideas how to fix this? Thanks! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio?
Thanks Anthony.
I am getting closer. I had to put in the 146 address, which is the
firewalled address I get at work. The problem now is that when the
call is bridged, I do not hear audio.
2 scenarios:
1 - the local extension is not registered. There is two way audio -
I hear the voicemail in Gtalk and I can leave a message which can then
be played back.
2 - the local extension is registered. There is no audio
In my incoming dialplan I am doing this bridge: action
application=bridge data=user/1...@${domain}/
It bridges okay, the phone rings, but there is no audio.
On a side note: Isn't putting the candidate-acl list a temporary
measure? When I travel, I will most likely get a different internal
company IP address that does not start with 146. Isn't there a
smarter way for dingaling to know that there is no RTP packets being
received and then modify which candidate should be used?
Thanks!
On Tue, Jun 23, 2009 at 6:44 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
try adding this to your jingle profile in client.xml
param name=candidate-acl value=wan/
then edit acl.conf.xml and add this list
list name=wan default=allow
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
this tells mod_dingaling that it should only pick candidates that pass the
acl list given
the one we made called wan excludes all the private ranges.
If you update to latest trunk this list is created internally as wan.auto
so you can use that
instead of making one in your config.
On Tue, Jun 23, 2009 at 7:51 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
I am trying to call from my corporate network (firewalled) using Gtalk
to Freeswitch. I am not getting any audio.
In the logs I see that mod_dingaling is using my internal corporate IP
address which is not publically addressable.
2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing
session for 4085152502
2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates
2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates
146.xx.xx.xx:50320
2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable
Candidate 146.xx.xx.xx:50320
Further on in the log, I can see GTalk sending a new candidate IP
address to use:
2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1
name=rtp
type=local
protocol=udp
username=e+JTkVHT1xEkqXGD
password=fAxU6Pr1oF9Zq48U
address=192.168.1.102
port=50322
pref=1.00
2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing
session for 4085152502
2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked
an IP [146.xx.xx.xx]
and
2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2
name=rtp
type=stun
protocol=udp
username=RBqyF2XNMYLfJNoU
password=DQMjon1fSVoJIRTp
address=124.xxx.xxx.xxx
port=50323
pref=0.90
2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing
session for 4085152502
2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked
an IP [146.xx.xx.xx]
and
2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3
name=rtp
type=relay
protocol=udp
username=62L5zs2FHbcUdeCJ
password=KxmNgkUmZsLfuX6S
address=209.xx.xxx.xxx
port=19295
pref=0.50
2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing
session for 4085152502
2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked
an IP [146.xx.xx.xx]
Because of this, I never get audio. Any ideas how to fix this?
Thanks!
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Re: [Freeswitch-users] email core dump
Thanks Brian, but still no luck with the email.. I have configured exim4 so that I can send messages from the command line using 'mail' command and these are sent successfully. I still get a core dump in the log when freeswitch is trying to send the mail: /bin/cat: write error: Broken pipe sh: line 1: 4492 Done(1) /bin/cat /tmp/mail.1245811149abdc 4493 Segmentation fault (core dumped) | /usr/local/bin/eximcompat.sh -t x...@xx.com 2009-06-24 12:39:09.285351 [DEBUG] switch_utils.c:554 Emailed file [/tmp/mail.1245811149abdc] to [...@xx.com] 2009-06-24 12:39:09.285351 [DEBUG] mod_voicemail.c:2491 Sending message to x...@xx.com eximcompat.sh is as described on the wiki: freeswitch:/# cat /usr/local/bin/eximcompat.sh #!/bin/bash exec exim4 -t Any other thoughts? From: Brian West br...@freeswitch.org Subject: Re: [Freeswitch-users] email core dump To: freeswitch-users@lists.freeswitch.org Message-ID: 7c7a8ed9-eced-4100-87f6-0875c054e...@freeswitch.org Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings /b On Jun 21, 2009, at 6:27 AM, Mark Campbell-Smith wrote: Hi! I am trying to email from 2009-06-21 20:43:24.273895 [DEBUG] switch_core_codec.c:122 Restore original codec. 2009-06-21 20:43:24.273895 [DEBUG] mod_voicemail.c:2321 Deliver VM to 1...@192.168.0.20 /bin/cat: write error: Broken pipe sh: line 1: 11975 Done(1) /bin/cat /tmp/mail. 124558382500b1 11976 Segmentation fault (core dumped) | exim4 -t myem...@xx.com 2009-06-21 20:43:24.670899 [DEBUG] switch_utils.c:554 Emailed file [/tmp/mail.12455810042c7f] to [myem...@xx.com] I can manually send an email to myself with exim4, but freeswitch fails. Any ideas what I have configured incorrectly? Thanks ___ Freeswitch-users mailing list Freeswitch-users@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] email core dump
Hi! I am trying to email from 2009-06-21 20:43:24.273895 [DEBUG] switch_core_codec.c:122 Restore original codec. 2009-06-21 20:43:24.273895 [DEBUG] mod_voicemail.c:2321 Deliver VM to 1...@192.168.0.20 /bin/cat: write error: Broken pipe sh: line 1: 11975 Done(1) /bin/cat /tmp/mail.124558382500b1 11976 Segmentation fault (core dumped) | exim4 -t myem...@xx.com 2009-06-21 20:43:24.670899 [DEBUG] switch_utils.c:554 Emailed file [/tmp/mail.12455810042c7f] to [myem...@xx.com] I can manually send an email to myself with exim4, but freeswitch fails. Any ideas what I have configured incorrectly? Thanks ___ Freeswitch-users mailing list Freeswitch-users@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] voicemail problem
Hi! I have a problem with voicemail in that freeswitch fails to let users leave their message. Something wrong in the config I guess. I see this in the logs: 2009-06-21 11:28:32.202912 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-06-21 11:28:32.238744 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-record_message] (en:en) 2009-06-21 11:28:32.290756 [ERR] mod_native_file.c:68 Error opening /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.PCMU 2009-06-21 11:28:32.402826 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-06-21 11:28:32.434733 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-goodbye.wav] (en:en) I assume the vm-record_message.PCMU is the file that will be created to record the voicemail. Is that correct and how can I fix this? Thanks! ___ Freeswitch-users mailing list Freeswitch-users@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] Porta Billing?
Hi!
Does freeswitch support extracting the billing data (PortaBilling) in
SIP messages? If so, is there anyway I can get that information to an
extension?
03:36:00.245: //-1//SIP/Msg/ccsipDis playMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.mydomain.com:5060;branch=z9hG4 bK1FF90
From: {sip:61xx...@sip.pennytel.com}; tag=47E011-580
To: {sip:61xx...@sip.pennytel.com};
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Thanks!
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[Freeswitch-users] Remove voicemail prompts
Hi! How can I configure voicemail so that I do not get the options such as record your message at the tone and mark this message as urgent Thanks! ___ Freeswitch-users mailing list Freeswitch-users@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] gtalk text chat
Hi! Is it possible to configure freeswitch and mod_dingaling so that it sends a text chat to a gtalk client? I would plan to use this for certain debugging purposes. Thanks! /Mark ___ Freeswitch-users mailing list Freeswitch-users@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] How to enable debug in dingaling?
Hi! I need to enable debug mode in dingaling as I can't see that freeswitch is coming online in gtalk. I have changed the following: changed the loglevel to debug in console.conf.xml changed the debug level to 1 in dingaling.conf to 1 I do not see any xmpp logs in the console or in freeswitch.log file. All I can see in the window is: 2009-05-12 17:56:35 [DEBUG] mod_dingaling.c:1854 init_profile() Started Thread for myfreeswitchn...@gmail.com/gt...@xml 2009-05-12 17:56:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_dingaling] 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'dingaling' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_debug' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_pres' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_logout' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_login' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:354 switch_loadable_module_process() Adding Chat interface 'jingle' Where is the XMPP traces? Thanks ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
