Re: [Freeswitch-users] Does FS support STUN by default?
Michael Jerris wrote: we also support natpmp and static ip setting. What is static ip setting? Telling FS what the public IP is? If that's what it is, what about the UDP ports that must be open to allow incoming connections? So, in the case where the FS server is located in a private network, these are the ways to open up the ports it needs to allow remote SIP users to connect to it: - UPnP and NAT-PMP (FS asks the router for its public IP, and negotiates opening the required UDP ports dynamically) - STUN (to get the public IP address from a remote STUN server) + port-mapping (to permanently open required UDP ports on NAT firewall) - possibly this fourth solution above -- View this message in context: http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26740589.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
Re: [Freeswitch-users] embedded freeswitch compatable hardware
voyage linux is a stripped debian and i was using it on an alix board some time ago... Asterisk was compiling on that without any issue. I beleive FS will do the same. T. On Fri, Dec 11, 2009 at 2:57 AM, Brian May br...@microcomaustralia.com.auwrote: On Thu, Dec 10, 2009 at 03:53:32PM +1100, Brian May wrote: Lack of OpenZAP support might be an issue, I assume that would be required to connect to an onboard analogue port... I assume I could just install Debian or another distribution instead though. This is another distribution I found: http://linux.voyage.hk/ It comes with Asterisk out of the box, although I suspect it wouldn't be too hard to get Freeswitch working instead. -- Brian May br...@microcomaustralia.com.au ___ 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] Does FS support STUN by default?
Fred-145 wrote: What is static ip setting? Telling FS what the public IP is? If that's what it is, what about the UDP ports that must be open to allow incoming connections? Yes, static IP setting puts the (non-changing) IP addresses in the FS configuration. The ports must be manually opened/forwarded in the firewall. -- Russell Mosemann ___ 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
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 $6 /usr/local/freeswitch/scripts/log.log
However, if I uncomment the last line, I never see the user variables
being passed to the shell script. The email is sucessfully sent, but
the sms script doesnt work. If fact, the output of log.log is (for
example):
-f 1...@192.168.1.120 email_addr...@domain.com
Any ideas if it is possible to pass user variables via mod_voicemail
in this way?
Thanks
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Re: [Freeswitch-users] Passing user variables to mod_voicemail
Pennytel.com
On Sat, Dec 12, 2009 at 12:52 AM, Phillip Jones pjinthe...@gmail.com wrote:
Hi - sorry to go off topic - but we are looking for Voip supplier with SMS
capability. Would you mind telling me which Voip supplier you use?
On Thu, Dec 10, 2009 at 11:10 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
My voip provider provides a SOAP interface to be able to send SMS's,
so after a voicemail is left, I want to execute a 'send sms' script.
I don't want a separate statement in the dialplan after the voicemail
statement because I only want to send sms's when a voicemail is
actually left.
The way I was going to do this was to modify the mailer-app to point
to a shell script and modify the mailer-app-args to include some user
defined variables (in conf/directory/default/*.xml).
param name=mailer-app
value=/usr/local/freeswitch/scripts/emailvm.sh/
param name=mailer-app-args value=${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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[Freeswitch-users] Still cant find it
Ok, So I have looked around a lot now, think I have read everything carefully, and don't see an answer to my questions anywhere, but apologize if it is already somewhere. SO I need the sphinx and tts modules, and don't see their src on the site with the freeswitch stuff. Do I just download from CMU? Any certain versions? Would be nice if somebody already compiled for windows I am very impressed with freeswitch, and thank you for your efforts!!! ___ 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] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. I tested with both Polycom IP650 and Bria 2.5.4, compared against port audio and googletalk endpoints in the same network. all SIP end points (Polycom and Bria) behind NAT but in the same subnet 192.168.0, I tried to change the settings below: param name=local-network-acl value=localnet.auto/ param name=apply-nat-acl value=nat.auto/ in /conf/sip_profiles/internal.xml using different combinations of either enabling or disabling them. the results are all the same, the audios on sip endpoints always got cut about 31 seconds, no issues at all with either port audio or gtalk, Could anyone point me to the right direction for the sofia_sip profile setup? Your helps are greatly appreciated Thanks, Chris ___ 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] Still cant find it
Download MSVC and compile it yourself is usually the best bet. /b On Dec 11, 2009, at 9:47 AM, Kendall Stauffer wrote: Ok, So I have looked around a lot now, think I have read everything carefully, and don’t see an answer to my questions anywhere, but apologize if it is already somewhere. SO I need the sphinx and tts modules, and don’t see their src on the site with the freeswitch stuff. Do I just download from CMU? Any certain versions? Would be nice if somebody already compiled for windows I am very impressed with freeswitch, and thank you for your efforts!!! ___ ___ 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] Still cant find it
The source tarballs are downloaded by the vs2008 project files when you build the solution Kendall Stauffer wrote: Ok, So I have looked around a lot now, think I have read everything carefully, and don't see an answer to my questions anywhere, but apologize if it is already somewhere. SO I need the sphinx and tts modules, and don't see their src on the site with the freeswitch stuff. Do I just download from CMU? Any certain versions? Would be nice if somebody already compiled for windows I am very impressed with freeswitch, and thank you for your efforts!!! ___ 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 -- View this message in context: http://n2.nabble.com/Still-cant-find-it-tp4152031p4152195.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
Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
On Fri, Dec 11, Chris Chen wrote: Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. --FC ___ 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] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly I noticed that the acl automatically having 192.168.0.0 set as deny, that's why I tried to changed the settings regarding nat acl and localnet acl. Chris On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle fr...@carmickle.comwrote: On Fri, Dec 11, Chris Chen wrote: Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. --FC ___ 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] FreeSWITCH Weekly Conference Call Starting!
Come one, come all! http://bit.ly/8KzHCZ Talk to you soon! -Michael ___ 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] windows pre compiled asr
It hasn't been included as of late since I'm getting an unresolved link error during the build. I'll need someone experienced in pocketsphinx to assist with this issue: 13ngram_search.obj : error LNK2001: unresolved external symbol _ngram_model_flush 13G:\freeswitch_dev\Release\pocketsphinx.dll : fatal error LNK1120: 1 unresolved externals regards, Carlos On Thu, Dec 10, 2009 at 6:30 PM, Kendall Stauffer k...@ksac.com wrote: I downloaded yesterdays latest pre compiled and seems to works great, but I get invalid Asr module when trying to run pizza app. It seemed to come pre configured with pocketsphynx, anything I should know before I spend a boat load of time on it? Rest seems real good,. thatks!!! ___ 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] windows pre compiled asr
Thats being fixed today! ;) /b On Dec 11, 2009, at 11:04 AM, Carlos Talbot wrote: It hasn't been included as of late since I'm getting an unresolved link error during the build. I'll need someone experienced in pocketsphinx to assist with this issue: 13ngram_search.obj : error LNK2001: unresolved external symbol _ngram_model_flush 13G:\freeswitch_dev\Release\pocketsphinx.dll : fatal error LNK1120: 1 unresolved externals regards, Carlos ___ 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] bridge doesn't respect bypass_media=true over the socket
Hello everyone,
PB here:
http://pastebin.freeswitch.org/11482
FS rev 15909. The relevant bits from the log are here (starting
around line 135):
#
2009-12-11 12:06:12.875158 [DEBUG] switch_ivr.c:548
sofia/pjsip/nob...@192.168.4.253 Command Execute
bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=,origination_caller_id_name=Tara
Ousley}sofia/voalte/hut...@192.168.4.17)
#
EXECUTE sofia/pjsip/nob...@192.168.4.253
bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=,origination_caller_id_name=Tara
Ousley}sofia/voalte/hut...@192.168.4.17)
#
2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735
variable string 0 = [originate_timeout=30]
#
2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735
variable string 1 = [bypass_media=true]
#
2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735
variable string 2 = [origination_caller_id_number=]
#
2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735
variable string 3 = [origination_caller_id_name=Tara Ousley]
bypass_media=true yet the SDP of the outgoing INVITE looks like this:
#
send 1032 bytes to udp/[192.168.4.17]:5060 at 12:06:12.876994:
#
#
INVITE sip:hut...@192.168.4.17 SIP/2.0
#
Via: SIP/2.0/UDP 192.168.2.10:5062;rport;branch=z9hG4bK1K6mc3NcmmaNr
#
Max-Forwards: 69
#
From: Tara Ousley sip:1...@192.168.2.10;tag=0tU8SN9pvejNK
#
To: sip:hut...@192.168.4.17
#
Call-ID: 802f4045-4215-42a2-91a6-ff9cf18b1aa8
#
CSeq: 124148250 INVITE
#
Contact: sip:mod_so...@192.168.2.10:5062
#
User-Agent: Voalte Voice
#
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
#
Supported: timer, precondition, path, replaces
#
Allow-Events: talk, refer
#
Content-Type: application/sdp
#
Content-Disposition: session
#
Content-Length: 271
#
X-voalte-call-id: 898ef33c-50f2-487c-9e8d-8c6fcee15ab8
#
Remote-Party-ID: Tara Ousley
sip:1...@192.168.2.10;party=calling;screen=yes;privacy=off
#
#
v=0
#
o=FreeSWITCH 1260508000 1260508001 IN IP4 192.168.2.10
#
s=FreeSWITCH
#
c=IN IP4 192.168.2.10
#
t=0 0
#
m=audio 25172 RTP/AVP 9 0 101
#
a=rtpmap:9 G722/8000
#
a=rtpmap:0 PCMU/8000
#
a=rtpmap:101 telephone-event/8000
#
a=fmtp:101 0-16
#
a=silenceSupp:off - - - -
#
a=ptime:20
192.168.2.10 is the address of my FS box...
--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com
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Re: [Freeswitch-users] bridge doesn't respect bypass_media=true over the socket
Hey,
You can't set bypass_media=true in {} or it will not take effect unless that
b leg itself becomes an a leg some day.
you need to execute set on bypass_media=true on the leg before you call
bridge to trigger it.
Alternatively you could set {bypass_media_after_bridge=true} or set it on A
leg as described above on either leg and it will do the bypass once the
audio is flowing.
On Fri, Dec 11, 2009 at 11:14 AM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
Hello everyone,
PB here:
http://pastebin.freeswitch.org/11482
FS rev 15909. The relevant bits from the log are here (starting
around line 135):
#
2009-12-11 12:06:12.875158 [DEBUG] switch_ivr.c:548
sofia/pjsip/nob...@192.168.4.253 Command Execute
bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=,origination_caller_id_name=Tara
Ousley}sofia/voalte/hut...@192.168.4.17)
#
EXECUTE sofia/pjsip/nob...@192.168.4.253
bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=,origination_caller_id_name=Tara
Ousley}sofia/voalte/hut...@192.168.4.17)
#
2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735
variable string 0 = [originate_timeout=30]
#
2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735
variable string 1 = [bypass_media=true]
#
2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735
variable string 2 = [origination_caller_id_number=]
#
2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735
variable string 3 = [origination_caller_id_name=Tara Ousley]
bypass_media=true yet the SDP of the outgoing INVITE looks like this:
#
send 1032 bytes to udp/[192.168.4.17]:5060 at 12:06:12.876994:
#
#
INVITE sip:hut...@192.168.4.17 sip%3ahut...@192.168.4.17 SIP/2.0
#
Via: SIP/2.0/UDP 192.168.2.10:5062;rport;branch=z9hG4bK1K6mc3NcmmaNr
#
Max-Forwards: 69
#
From: Tara Ousley sip:1...@192.168.2.10 sip%3a1...@192.168.2.10
;tag=0tU8SN9pvejNK
#
To: sip:hut...@192.168.4.17 sip%3ahut...@192.168.4.17
#
Call-ID: 802f4045-4215-42a2-91a6-ff9cf18b1aa8
#
CSeq: 124148250 INVITE
#
Contact: sip:mod_so...@192.168.2.10:5062
#
User-Agent: Voalte Voice
#
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
#
Supported: timer, precondition, path, replaces
#
Allow-Events: talk, refer
#
Content-Type: application/sdp
#
Content-Disposition: session
#
Content-Length: 271
#
X-voalte-call-id: 898ef33c-50f2-487c-9e8d-8c6fcee15ab8
#
Remote-Party-ID: Tara Ousley
sip:1...@192.168.2.10 sip%3a1...@192.168.2.10
;party=calling;screen=yes;privacy=off
#
#
v=0
#
o=FreeSWITCH 1260508000 1260508001 IN IP4 192.168.2.10
#
s=FreeSWITCH
#
c=IN IP4 192.168.2.10
#
t=0 0
#
m=audio 25172 RTP/AVP 9 0 101
#
a=rtpmap:9 G722/8000
#
a=rtpmap:0 PCMU/8000
#
a=rtpmap:101 telephone-event/8000
#
a=fmtp:101 0-16
#
a=silenceSupp:off - - - -
#
a=ptime:20
192.168.2.10 is the address of my FS box...
--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com
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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
As i said multiple times on irc last night, we need to see debug logs with sip trace to see what is going on. Mike On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly I noticed that the acl automatically having 192.168.0.0 set as deny, that's why I tried to changed the settings regarding nat acl and localnet acl. Chris On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle fr...@carmickle.com wrote: On Fri, Dec 11, Chris Chen wrote: Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. --FC ___ 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] Still cant find it
http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code On Dec 11, 2009, at 11:25 AM, Kendall Stauffer wrote: Yes, I can do that , I don’t see where I download the source, Sorry to bug you. ___ 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] gtalk dingaling G723
Can't use G723. /b On Dec 11, 2009, at 5:02 AM, zendel fernandez wrote: hi! Pls shed some light to the below dingaling/gtalk issue. ___ 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] Does FS support STUN by default?
One last question: Does someone know of a utility for Windows that can check that a NAT router supports either UPnP or NAT-PMP? I guess it's no big deal to write a small diagnostic by connecting to free firewall checkers to see if the relevant ports are open, but if it's already available... Thank you. -- View this message in context: http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26748901.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
Re: [Freeswitch-users] Does FS support STUN by default?
FreeSWITCH on windows will already poke holes in the windows firewall using upnp. Just start FS and it works. Your outer nat is a larger issue... /b On Dec 11, 2009, at 12:09 PM, Fred-145 wrote: One last question: Does someone know of a utility for Windows that can check that a NAT router supports either UPnP or NAT-PMP? I guess it's no big deal to write a small diagnostic by connecting to free firewall checkers to see if the relevant ports are open, but if it's already available... Thank you. -- ___ 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] The Building Freeswitch blog
Doing the building thing, seem to have come across a bug. Have a look at Part 2 of http://makingfs.blogspot.com/ If make crashes out, it states that it was successfully built ;) Julian ___ 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] The Building Freeswitch blog
well mod_alas.c is for the N800 Please open a jira. /b On Dec 11, 2009, at 12:19 PM, Julian Lyndon-Smith wrote: Doing the building thing, seem to have come across a bug. Have a look at Part 2 of http://makingfs.blogspot.com/ If make crashes out, it states that it was successfully built ;) Julian ___ 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] The Building Freeswitch blog
Thanks Mike. I understand why you don't want all to be built. However, there are things that I would like - such as mod_java. However, that fails to compile, I presume because of some missing dependency or requirement. Is there any tool to tell me what is needed in order to build a module ? Julian 2009/12/11 Michael Jerris m...@jerris.com: It just so happens I was looking at this same bug last night and having troubles chasing down a solution, if anyone comes up with anything good please let me know. The basics of this is that automake continues on to other subdirs if build in one subdir fails. Mike p..s. a note on the blog, I generally do not recommend just building everything, for example, mod_alsa is a module written specifically for the n800 due to mod_portaudio not working there. This module is barely touched and I would not use it unless you have a good reason to. On Dec 11, 2009, at 1:19 PM, Julian Lyndon-Smith wrote: Doing the building thing, seem to have come across a bug. Have a look at Part 2 of http://makingfs.blogspot.com/ If make crashes out, it states that it was successfully built ;) Julian ___ 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] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Hi Mike, the fs console log with sip trace on the internal profile is attached in the pastebin below, http://pastebin.freeswitch.org/11483 could you please take a look at it? Thanks, Chris On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris m...@jerris.com wrote: As i said multiple times on irc last night, we need to see debug logs with sip trace to see what is going on. Mike On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly I noticed that the acl automatically having 192.168.0.0 set as deny, that's why I tried to changed the settings regarding nat acl and localnet acl. Chris On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle fr...@carmickle.comwrote: On Fri, Dec 11, Chris Chen wrote: Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. --FC ___ 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] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Its not sending to the right Contact: header in the 200 OK packet. This was fixed in r15870, you have to update. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 11-Dec-09, at 2:03 PM, Chris Chen wrote: Hi Mike, the fs console log with sip trace on the internal profile is attached in the pastebin below, http://pastebin.freeswitch.org/11483 could you please take a look at it? Thanks, Chris On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris m...@jerris.com wrote: As i said multiple times on irc last night, we need to see debug logs with sip trace to see what is going on. Mike On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly I noticed that the acl automatically having 192.168.0.0 set as deny, that's why I tried to changed the settings regarding nat acl and localnet acl. Chris On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle fr...@carmickle.com wrote: On Fri, Dec 11, Chris Chen wrote: Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. --FC ___ 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] The Building Freeswitch blog
Probably the best list is: http://wiki.freeswitch.org/wiki/FreeSwitch_Dependencies Due to the fact that we allow you to change modules after configure there is no great way to have it error out when you don't have the right deps other than to just have the compile errors when you try to build. Its probably time for a tool like make menuconfig but we do not have that as of yet. Mike On Dec 11, 2009, at 1:47 PM, Julian Lyndon-Smith wrote: Thanks Mike. I understand why you don't want all to be built. However, there are things that I would like - such as mod_java. However, that fails to compile, I presume because of some missing dependency or requirement. Is there any tool to tell me what is needed in order to build a module ? Julian 2009/12/11 Michael Jerris m...@jerris.com: It just so happens I was looking at this same bug last night and having troubles chasing down a solution, if anyone comes up with anything good please let me know. The basics of this is that automake continues on to other subdirs if build in one subdir fails. Mike p..s. a note on the blog, I generally do not recommend just building everything, for example, mod_alsa is a module written specifically for the n800 due to mod_portaudio not working there. This module is barely touched and I would not use it unless you have a good reason to. On Dec 11, 2009, at 1:19 PM, Julian Lyndon-Smith wrote: Doing the building thing, seem to have come across a bug. Have a look at Part 2 of http://makingfs.blogspot.com/ If make crashes out, it states that it was successfully built ;) Julian ___ 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] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Thanks Mathieu, but I am on SVN r15912 now. Chris On Fri, Dec 11, 2009 at 2:09 PM, Mathieu Rene mrene_li...@avgs.ca wrote: Its not sending to the right Contact: header in the 200 OK packet. This was fixed in r15870, you have to update. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 11-Dec-09, at 2:03 PM, Chris Chen wrote: Hi Mike, the fs console log with sip trace on the internal profile is attached in the pastebin below, http://pastebin.freeswitch.org/11483 could you please take a look at it? Thanks, Chris On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris m...@jerris.com wrote: As i said multiple times on irc last night, we need to see debug logs with sip trace to see what is going on. Mike On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly I noticed that the acl automatically having 192.168.0.0 set as deny, that's why I tried to changed the settings regarding nat acl and localnet acl. Chris On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle fr...@carmickle.comwrote: On Fri, Dec 11, Chris Chen wrote: Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. --FC ___ 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] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
You set the extrtp ip to an IP exactly.. this is the issue we are fixing soon.. if you have natpmp or upnp set it to auto-nat and let it figure it out. The issue is we have restored the behavior in 1.0.4 that lies about the IP all the time... I'm going to commit a patch shortly that'll fix this. /b On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: Thanks Mathieu, but I am on SVN r15912 now. Chris ___ 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] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Thanks Brian for your explanation, could we still keep the option to set the extrip ip, as my DLINK DIR-655 UPNP is not working reliably, and I believe many other routers have similar issue. Chris On Fri, Dec 11, 2009 at 2:45 PM, Brian West br...@freeswitch.org wrote: You set the extrtp ip to an IP exactly.. this is the issue we are fixing soon.. if you have natpmp or upnp set it to auto-nat and let it figure it out. The issue is we have restored the behavior in 1.0.4 that lies about the IP all the time... I'm going to commit a patch shortly that'll fix this. /b On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: Thanks Mathieu, but I am on SVN r15912 now. Chris ___ 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] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Please test www.bkw.org/sofia_autonat_static_ip.diff /b On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: Thanks Mathieu, but I am on SVN r15912 now. Chris ___ 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] Problems with Freeswitch setup - Outbound
I am very new to Freeswitch so please accept my appologies if these questions
seem to be trivial
I am trying to setup FreeSwitch to work with Microsoft Speech Serevr 2007. I
have been successful in getting Freeswitch to pass an incomming PSTN call to
Speech Server. But I cannot get Freeswitch to dial out a call or transfer a
call that is sent from Speech Server
I have a file setup in C:\FreeSWITCH\conf\sip_profiles\external\ called
voipms.xml which contains the following..(I have an account with voip.ms)
include
gateway name=voipms
/gateway
/include
And I have a file setup in C:\FreeSWITCH\conf\dialplan\public\ called
Outbound.xml which contains the following
extension name=Outbound
condition field=destination_number expression=^(1{0,1}\d{10})$
action application=set
data=effective_caller_id_number=1222333/
action application=bridge data=sofia/gateway/voipms/$1/
/condition
/extension
When my Speech Server application tries to get FreeSwitch to transfer to
another number, the console shows the following
2009-12-11 17:54:31.863512 [NOTICE] switch_ivr.c:1349 Transfer
sofia/external/+1
9059183...@199.173.95.16:5060 to xml[%23904161...@public]
2009-12-11 17:54:31.879138 [NOTICE] switch_ivr_bridge.c:503 Hangup
sofia/interna
l/2482578...@127.0.0.1:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2009-12-11 17:54:31.879138 [INFO] mod_dialplan_xml.c:315 Processing
+14165551212
-%23904161234 in context public
2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1086 Session 2
(sofia/
internal/2482578...@127.0.0.1:5060) Ended
2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1088 Close Channel
sof
ia/internal/2482578...@127.0.0.1:5060 [CS_DESTROY]
2009-12-11 17:54:31.879138 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia
/external/+19059183...@199.173.95.16:5060 [CS_EXECUTE] [NORMAL_CLEARING]
2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1086 Session 1
(sofia/
external/+19059183...@199.173.95.16:5060) Ended
2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1088 Close Channel
sof
ia/external/+19059183...@199.173.95.16:5060 [CS_DESTROY]
I really dont understand the line above that I have in Bold Italic
The number being transfered to was 4161234567...so I would have thought the
line should read..
Processing +14165551212-4161234567 in context public
Can anyone tell me what the %2390 means and also any problems with my XML
files that could be preventing the transfers from taking place
Thanks
Brian
--
View this message in context:
http://old.nabble.com/Problems-with-Freeswitch-setup---Outbound-tp26752894p26752894.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
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Re: [Freeswitch-users] Problems with Freeswitch setup - Outbound
On Fri, Dec 11, 2009 at 3:11 PM, bcxml bc...@hotmail.com wrote:
I am very new to Freeswitch so please accept my appologies if these
questions
seem to be trivial
I am trying to setup FreeSwitch to work with Microsoft Speech Serevr 2007.
I
have been successful in getting Freeswitch to pass an incomming PSTN call
to
Speech Server. But I cannot get Freeswitch to dial out a call or transfer a
call that is sent from Speech Server
I have a file setup in C:\FreeSWITCH\conf\sip_profiles\external\ called
voipms.xml which contains the following..(I have an account with voip.ms)
include
gateway name=voipms
/gateway
/include
And I have a file setup in C:\FreeSWITCH\conf\dialplan\public\ called
Outbound.xml which contains the following
extension name=Outbound
condition field=destination_number expression=^(1{0,1}\d{10})$
action application=set
data=effective_caller_id_number=1222333/
action application=bridge data=sofia/gateway/voipms/$1/
/condition
/extension
When my Speech Server application tries to get FreeSwitch to transfer to
another number, the console shows the following
2009-12-11 17:54:31.863512 [NOTICE] switch_ivr.c:1349 Transfer
sofia/external/+1
9059183...@199.173.95.16:5060 to xml[%23904161...@public]
2009-12-11 17:54:31.879138 [NOTICE] switch_ivr_bridge.c:503 Hangup
sofia/interna
l/2482578...@127.0.0.1:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2009-12-11 17:54:31.879138 [INFO] mod_dialplan_xml.c:315 Processing
+14165551212
-%23904161234 in context public
This line is basically saying that you have a call coming from 4165551212
and it's looking for a destination number of %23904161234. The key here is
that it is coming in the public context so you'll need to handle the routing
in conf/dialplan/public.xml
What should this call be doing once it comes in to FS?
-MC
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Re: [Freeswitch-users] Does FS support STUN by default?
You don't have to do that usually... /b On Dec 11, 2009, at 5:38 PM, Fred-145 wrote: I'll see if I can find a utility that checks that the ports are open after FS is up and running. ___ 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] Problems with Freeswitch setup - Outbound
%23 is # so the question is should we URL decode that before routing? I thought we did... what version are you using now? /b On Dec 11, 2009, at 5:34 PM, Michael Collins wrote: This line is basically saying that you have a call coming from 4165551212 and it's looking for a destination number of %23904161234. The key here is that it is coming in the public context so you'll need to handle the routing in conf/dialplan/ public.xml What should this call be doing once it comes in to FS? -MC ___ 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] Problems with Freeswitch setup - Outbound
The version is FreeSWITCH Version 1.0.4 (14460) Brian Brian West-3 wrote: %23 is # so the question is should we URL decode that before routing? I thought we did... what version are you using now? /b On Dec 11, 2009, at 5:34 PM, Michael Collins wrote: This line is basically saying that you have a call coming from 4165551212 and it's looking for a destination number of %23904161234. The key here is that it is coming in the public context so you'll need to handle the routing in conf/dialplan/ public.xml What should this call be doing once it comes in to FS? -MC ___ 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 -- View this message in context: http://old.nabble.com/Problems-with-Freeswitch-setup---Outbound-tp26752894p26753346.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
Re: [Freeswitch-users] Problems with Freeswitch setup - Outbound
On Dec 11, 2009, at 4:02 PM, bcxml bc...@hotmail.com wrote: The version is FreeSWITCH Version 1.0.4 (14460) Ouch. You are nearly 6 months and 1500 revs behind. You badly need to update to latest trunk. -MC ___ 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] Getting started on IVR Library
Dear all , I've seen the IVR library functions which are implemented in C language. Can any one please suggest how can I use that library or give idea to do the IVR programs in C through this library. Please help me... -- Regards, Thangappan.M ___ 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
