Re: [Freeswitch-users] Adding H263 Video to Existing Call Fails First Time
Just a question, do you use Freeswitch in bypass-media-mode in this scenario? Then media negociation should be handled outside Freeswitch. Best regards Peter Jerry Richards schrieb: After establishing an audio call between two Bria softphones, and then starting video at the caller phone, FS replies to the re-INVITE with a 200 OK with only the PCMU codec. This looks incorrect. The audio call previously negotiated to the speex/16000 codec, and the re-INVITE from the caller added the H263-1998 codec. If I re-attempt to start video at the caller, then it is successful. I put a Freeswitch log 11596 into the pastebin that contains the complete scenario: establishing audio call, first failed start video attempt, and second successful start video attempt. Best Regards, Jerry ___ 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] Force endpoint to use rfc2833 for dtmf
Hello, in a bigger installation with some thousand endpoints in the field we see, that the endpoint equipment is always using INFO messages (standard setting is auto, so the endpoint decides which method to use). I have 2 questions to that scenario: 1. Is there a way that Freeswitch forces/restricts the endpoint to use rfc2833 or not to send to allow INFO in the invite message? 2. Currently INFO messages do not get forwarded from the caller through freeswitch to called endpoint. How can we enable that FS is fowarding the INFO messages? Best regards Peter ___ 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] Skypiax: not able to detect Inband dtmf tones from pstn call?
ubuntu-8.04.3-server-amd64.iso (update/upgrade)
FreeSWITCH Version 1.0.trunk (15787)
skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb
mod_skypiax
(POTS)--(PSTN)--(skypeIN)--(skype_client)--(skypiax)--(fs)
extension name=Indial_to_fs_via_skypeIN
condition field=destination_number expression=^501$
action application=start_dtmf /
action application=record_session
data=/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav/
action application=playback data=/root/Hello_16000.wav /
/condition
/extension
fsconsole loglevel 7
If I dial 501 from from a sip phone using inband dtmf I can see the
dtmf tones being detected and decoded by fs in the debug log.
If however I use a pstn phone and dial my skypeIN telephone number the
call comes into fs via skypiax but when I generate dtmf tones on the
phone they are not detected or decoded by fs.
If I take the record_session file and spectrum analyze the recorded
tones appear to be within spec.
Can anybody suggest why this is not working for me?
Is the correct sample rate being used in libteletone_detect.c?
Does the Goertzel algorithm work for other sample rates other than
8000hz?
I'm not sure why I can not get this to work?
regards,
Scott Torr
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Re: [Freeswitch-users] Adding H263 Video to Existing Call Fails First Time
Can you repeat that same trace with latest trunk? On Mon, Dec 21, 2009 at 6:44 PM, Jerry Richards jerry.richa...@teotech.comwrote: After establishing an audio call between two Bria softphones, and then starting video at the caller phone, FS replies to the re-INVITE with a 200 OK with only the PCMU codec. This looks incorrect. The audio call previously negotiated to the speex/16000 codec, and the re-INVITE from the caller added the H263-1998 codec. If I re-attempt to start video at the caller, then it is successful. I put a Freeswitch log 11596 into the pastebin that contains the complete scenario: establishing audio call, first failed start video attempt, and second successful start video attempt. Best Regards, Jerry ___ 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 msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com 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:+19193869900 ___ 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] Multitenant dialplans
The force-register-domain and force-register-db-domain are set in the defaults so you can only do one domain. Remove those and you'll be able to do multiple domains. /b On Dec 21, 2009, at 6:15 PM, j...@acsol.net wrote: I have Freeswitch setup and working as a single tenant system mostly using the default configuration. Trying to convert to a multitenant environment, I have used both the Multi-tenant and Multiple Companies wiki's. I get the phone to register, can call out using the external profile to a ITSP, can call music on hold; however I can not call other users in the company. It appears that when logged in with single company and default context it sucessfully calls other internal phones with bridge to sofia/internal/sip:exters...@public-ip:translated-port; however when I log into Company1 with the phones, it tries sofia/internal/dialed-extens...@company1 ... I also get User not Registered. The dialplans are the same either way. Any ideas? Thanks John ___ 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] Skypiax: not able to detect Inband dtmf tones from pstn call?
add start_dtmf app to your dialplan before bridge to start the inband dtmf
detector.
On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr
scott.torr...@letterboxes.orgwrote:
ubuntu-8.04.3-server-amd64.iso (update/upgrade)
FreeSWITCH Version 1.0.trunk (15787)
skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb
mod_skypiax
(POTS)--(PSTN)--(skypeIN)--(skype_client)--(skypiax)--(fs)
extension name=Indial_to_fs_via_skypeIN
condition field=destination_number expression=^501$
action application=start_dtmf /
action application=record_session
data=/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav/
action application=playback data=/root/Hello_16000.wav /
/condition
/extension
fsconsole loglevel 7
If I dial 501 from from a sip phone using inband dtmf I can see the
dtmf tones being detected and decoded by fs in the debug log.
If however I use a pstn phone and dial my skypeIN telephone number the
call comes into fs via skypiax but when I generate dtmf tones on the
phone they are not detected or decoded by fs.
If I take the record_session file and spectrum analyze the recorded
tones appear to be within spec.
Can anybody suggest why this is not working for me?
Is the correct sample rate being used in libteletone_detect.c?
Does the Goertzel algorithm work for other sample rates other than
8000hz?
I'm not sure why I can not get this to work?
regards,
Scott Torr
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--
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 msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
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:+19193869900
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Re: [Freeswitch-users] Skypiax: not able to detect Inband dtmf tones from pstn call?
do as anthm say :-)
On Tue, Dec 22, 2009 at 4:21 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
add start_dtmf app to your dialplan before bridge to start the inband dtmf
detector.
On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr scott.torr...@letterboxes.org
wrote:
ubuntu-8.04.3-server-amd64.iso (update/upgrade)
FreeSWITCH Version 1.0.trunk (15787)
skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb
mod_skypiax
(POTS)--(PSTN)--(skypeIN)--(skype_client)--(skypiax)--(fs)
extension name=Indial_to_fs_via_skypeIN
condition field=destination_number expression=^501$
action application=start_dtmf /
action application=record_session
data=/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav/
action application=playback data=/root/Hello_16000.wav /
/condition
/extension
fsconsole loglevel 7
If I dial 501 from from a sip phone using inband dtmf I can see the
dtmf tones being detected and decoded by fs in the debug log.
If however I use a pstn phone and dial my skypeIN telephone number the
call comes into fs via skypiax but when I generate dtmf tones on the
phone they are not detected or decoded by fs.
If I take the record_session file and spectrum analyze the recorded
tones appear to be within spec.
Can anybody suggest why this is not working for me?
Is the correct sample rate being used in libteletone_detect.c?
Does the Goertzel algorithm work for other sample rates other than
8000hz?
I'm not sure why I can not get this to work?
regards,
Scott Torr
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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:+19193869900
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Sincerely,
Giovanni Maruzzelli
Cell : +39-347-2665618
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Re: [Freeswitch-users] Skypiax: not able to detect Inband dtmf tones from pstn call?
It is probably because mod_skypiax does not analize incoming audio
looking for dtmf, because the normal call from a Skype client peer
sends *both* inband and out of band (signaling) dtmf.
So, I choose to only detect out of band (signaling) dtmfs, and ignore
possible inband dtmfs (in the audio flow), so to have the most
reliable source (signaling) and spare cpu (not analizing the incoming
audio).
Never tought you can receive calls from skypeIN with inband dtmfs...
Open a Jira for this, I'll think about.
Also, let me know your toughts...
-giovanni
On Tue, Dec 22, 2009 at 3:57 PM, Scott Torr
scott.torr...@letterboxes.org wrote:
ubuntu-8.04.3-server-amd64.iso (update/upgrade)
FreeSWITCH Version 1.0.trunk (15787)
skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb
mod_skypiax
(POTS)--(PSTN)--(skypeIN)--(skype_client)--(skypiax)--(fs)
extension name=Indial_to_fs_via_skypeIN
condition field=destination_number expression=^501$
action application=start_dtmf /
action application=record_session
data=/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav/
action application=playback data=/root/Hello_16000.wav /
/condition
/extension
fsconsole loglevel 7
If I dial 501 from from a sip phone using inband dtmf I can see the
dtmf tones being detected and decoded by fs in the debug log.
If however I use a pstn phone and dial my skypeIN telephone number the
call comes into fs via skypiax but when I generate dtmf tones on the
phone they are not detected or decoded by fs.
If I take the record_session file and spectrum analyze the recorded
tones appear to be within spec.
Can anybody suggest why this is not working for me?
Is the correct sample rate being used in libteletone_detect.c?
Does the Goertzel algorithm work for other sample rates other than
8000hz?
I'm not sure why I can not get this to work?
regards,
Scott Torr
___
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--
Sincerely,
Giovanni Maruzzelli
Cell : +39-347-2665618
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[Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed to Voice Mail
I have a Freeswitch PBX server with an installed Sangoma A101D card connected to a PRI. Most everything works okay, however when I get an inbound call from the PSTN, if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. How can I get the external call to route to voice mail after 30 seconds? I put a new 11595 log into the pastebin. Do you know any Freeswitch setting that might cause this? If this issue has been addressed before, what string should I use to search for it, because I can't find it. Thanks, Jerry ___ 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] Multitenant dialplans
Thanks Brian. I did have both force-register-domain and force-register-db-domain commented in both the internal.xml and internal-ipv6.xml. The phones appear to register to the company1 domain, as shown in sofia status profile company1; however I have noticed that when I try to make a call to another a phone in the same domain, the system is trying to call sofia/internal/1...@company1 -- this is when we get the message, user not registered. If I can the phones to just register to the IP address of the machine, they call fine and is shows sofia/internal/sip:1...@phonesgatewayipaddress. Is this a dialplan problem? In both cases I am just using the sample dialplan. On 12/22/2009 8:13 AM, Brian West wrote: The force-register-domain and force-register-db-domain are set in the defaults so you can only do one domain. Remove those and you'll be able to do multiple domains. /b On Dec 21, 2009, at 6:15 PM, j...@acsol.net wrote: I have Freeswitch setup and working as a single tenant system mostly using the default configuration. Trying to convert to a multitenant environment, I have used both the Multi-tenant and Multiple Companies wiki's. I get the phone to register, can call out using the external profile to a ITSP, can call music on hold; however I can not call other users in the company. It appears that when logged in with single company and default context it sucessfully calls other internal phones with bridge to sofia/internal/sip:exters...@public-ip:translated-port; however when I log into Company1 with the phones, it tries sofia/internal/dialed-extens...@company1 ... I also get User not Registered. The dialplans are the same either way. Any ideas? Thanks John ___ 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] Adding H263 Video to Existing Call FailsFirst Time
No. The following lines is commented out (internal.xml): !--param name=media-option value=bypass-media-after-att-xfer/-- !--param name=inbound-bypass-media value=true/-- Thanks, Jerry -Original Message- From: Peter P GMX [mailto:prometheus...@gmx.net] Sent: Tuesday, December 22, 2009 3:21 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Adding H263 Video to Existing Call FailsFirst Time Just a question, do you use Freeswitch in bypass-media-mode in this scenario? Then media negociation should be handled outside Freeswitch. Best regards Peter Jerry Richards schrieb: After establishing an audio call between two Bria softphones, and then starting video at the caller phone, FS replies to the re-INVITE with a 200 OK with only the PCMU codec. This looks incorrect. The audio call previously negotiated to the speex/16000 codec, and the re-INVITE from the caller added the H263-1998 codec. If I re-attempt to start video at the caller, then it is successful. I put a Freeswitch log 11596 into the pastebin that contains the complete scenario: establishing audio call, first failed start video attempt, and second successful start video attempt. Best Regards, Jerry ___ 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-use rs 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] [Freeswitch-dev] a1-has param in gateway setting
I'm not too sure you can put an a1-hash on outbound auth. /b On Dec 22, 2009, at 11:26 AM, Babak Karvandi wrote: Hi, Does any body know or has test the a1-hash parameter with gateway setting? I am not sure if it is even allowed. I have the following gateway setting but when the freeswitch starts up it simply ignores this provider without any error message or attempt to register in the log file. Thank you for your help in advance. include gateway name=iptel param name=username value=MY-USERNAME/ param name=realm value=iptel.org/ !-- param name=password value=MY_PASSWORD/ -- !-- replaced the password with MD5 encrypted -- !-- openssl dgst -md5 filename, or echo username:domain:password | openssldgst -md5 -- param name=a1-hash value=30f610a85e973f2b29b75ddc1ec3450e/ param name=proxy value=sip.iptel.org/ /gateway /include ___ 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] Multitenant dialplans
One point of clarification, currently all the phones are behind NAT, so it appears that when the phones are in a Non-multitenant scenario, they use SIP:dialed_num...@ip-address-of-their-gateway. On 12/22/2009 9:16 AM, John wrote: Thanks Brian. I did have both force-register-domain and force-register-db-domain commented in both the internal.xml and internal-ipv6.xml. The phones appear to register to the company1 domain, as shown in sofia status profile company1; however I have noticed that when I try to make a call to another a phone in the same domain, the system is trying to call sofia/internal/1...@company1 -- this is when we get the message, user not registered. If I can the phones to just register to the IP address of the machine, they call fine and is shows sofia/internal/sip:1...@phonesgatewayipaddress. Is this a dialplan problem? In both cases I am just using the sample dialplan. On 12/22/2009 8:13 AM, Brian West wrote: The force-register-domain and force-register-db-domain are set in the defaults so you can only do one domain. Remove those and you'll be able to do multiple domains. /b On Dec 21, 2009, at 6:15 PM, j...@acsol.net wrote: I have Freeswitch setup and working as a single tenant system mostly using the default configuration. Trying to convert to a multitenant environment, I have used both the Multi-tenant and Multiple Companies wiki's. I get the phone to register, can call out using the external profile to a ITSP, can call music on hold; however I can not call other users in the company. It appears that when logged in with single company and default context it sucessfully calls other internal phones with bridge to sofia/internal/sip:exters...@public-ip:translated-port; however when I log into Company1 with the phones, it tries sofia/internal/dialed-extens...@company1 ... I also get User not Registered. The dialplans are the same either way. Any ideas? Thanks John ___ 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] Freeswitch not seeing Register requests
I have set up a second FreeSWITCH box on the same LAN. I have v16018 installed on it and have changed nothing. I configured a Polycom phone to register one of its four lines to this second box, but it does not register. When looking at the console, there is no activity. However, there is SIP activity on the box which I have captured via ngrep. It looks like the phone is sending out REGISTER requests but there is no response. The request on the pastebin repeats forever, with only the timestamp varying. Is the problem that there are two FreeSWITCHes? Any suggestions on how I can make it work? On the original and the new box in vars.xml external_sip_ip=stun:stun.freeswitch.org On the original box in vars.xml external_sip_port=5090 but in the new it is 5080. Do I need to hardcode the external_sip_ip addresses in both boxes? http://pastebin.freeswitch.org/11600 Thanks Lars ___ 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 not seeing Register requests
On Tue, Dec 22, 2009 at 10:50 AM, Larry Marshall l...@marshap.com wrote: I have set up a second FreeSWITCH box on the same LAN. I have v16018 installed on it and have changed nothing. I configured a Polycom phone to register one of its four lines to this second box, but it does not register. When looking at the console, there is no activity. However, there is SIP activity on the box which I have captured via ngrep. It looks like the phone is sending out REGISTER requests but there is no response. The request on the pastebin repeats forever, with only the timestamp varying. On the new box do sofia status - does the internal profile exist? Is the problem that there are two FreeSWITCHes? Any suggestions on how I can make it work? On the original and the new box in vars.xml external_sip_ip=stun: stun.freeswitch.org On the original box in vars.xml external_sip_port=5090 but in the new it is 5080. Do I need to hardcode the external_sip_ip addresses in both boxes? http://pastebin.freeswitch.org/11600 Thanks Lars ___ 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?] Call either sent to VM or phone doesn't ring at all
Hello I'm running 1.0.trunk (15841) on Linux CentOS with a the default settings. After succesfully connecting a Windows PC running XLite (EyeBeam, really) and a GrandStream IP phone to Freeswitch, I try to make calls, but am having the following issues: 1. When calling XLite from GS, XLite rings, but when I pick up the call, the caller is sent to voice-mail right away (the person on extension 1001 is not available) 2. When calling GS from XLite, the GS phone doesn't even ring. FWIW, the phones seem to have registered OK: freeswi...@internal sofia status profile internal Registrations: Call-ID:Yzc2MzFiMjVhNGQwNjE5YWU1OGZjNGMxMTg0NDIwNDA. User: 1...@192.168.0.7 Contact:Freeswitch sip:1...@192.168.0.1:41380;rinstance=0516dddfe24deef4 Agent: eyeBeam release 1104a stamp 54437 Status: Registered(UDP)(unknown) EXP(2008-01-01 03:34:00) Host: centos.workgroup IP: 192.168.0.1 Port: 41380 Auth-User: 1001 Auth-Realm: 192.168.0.7 MWI-Account:1...@192.168.0.7 Call-ID:3f6d4ebebd5e8...@192.168.0.9 User: 1...@192.168.0.7 Contact:user sip:1...@82.237.75.54 Agent: Grandstream BT120 1.1.0.3 Status: Registered(UDP)(unknown) EXP(2008-01-01 03:44:02) Host: centos.workgroup IP: 192.168.0.9 Port: 5060 Auth-User: 1003 Auth-Realm: 192.168.0.7 MWI-Account:1...@192.168.0.7 Has someone seen this type of behavior? Thanks for any hint. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26892767.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] [Dialplan?] Call either sent to VM or phone doesn't ring at all
I found the cause for #2: The GS phone was still configured to use NAT, even though both XLite and GS are located in the same, private LAN. Unchecking this on the GS phone solved the issue. But I'm still having issue #1, regardless of which phone is calling or being called: When the phone answers the call, I'm sent automatically to voice-mail. Could it be codec-related, or something like that? Thank you. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26893059.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] [Dialplan?] Call either sent to VM or phone doesn't ring at all
It is usually CODEC related. probably the SIP messages has the cause inside. __Yehavi: 2009/12/22 Fred-145 codecompl...@free.fr I found the cause for #2: The GS phone was still configured to use NAT, even though both XLite and GS are located in the same, private LAN. Unchecking this on the GS phone solved the issue. But I'm still having issue #1, regardless of which phone is calling or being called: When the phone answers the call, I'm sent automatically to voice-mail. Could it be codec-related, or something like that? Thank you. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26893059.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
Re: [Freeswitch-users] How to debug TLS handshake errors?
My distro is fedora 10 with all the current patches. SSLwatch fails to build and it seems more than a trivial change to make it work; however, it seems that the error message from Freeswitch tells it all... Is there any special debug statement in Freeswitch to see more about its TLS negotations? Thanks, __Yehavi: 2009/12/21 Brian West br...@freeswitch.org You have to watch it with TLS. Make sure your distro didn't mess up your SSL libs due to the recent vulnerability found. I havn't tested with my polycom in a few weeks but it was working on my Polycom after I uploaded the ca cert and marked it as trusted/used on the phone. /b On Dec 20, 2009, at 8:26 AM, Yehavi Bourvine wrote: I am trying now to set a Polycom to work with FreeSwitch and TLS. I have a Polycom-501 which does not have an internal certificate, thus only one-way certificate validation is needed. I've downloaded the root certificate to he Polyciom, and Freeswitch gives me the following error: Peer did not provide X.509 Certificate I understand that it tries to do mutual authentication which is not possible in this case. How can I tell FreeSwitch to ignore the client's certificate? BTW, I am running 1.0.5pre9, and it works ok using TLS with SNOM and Yealink. Thanks! __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
Re: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all
On Tue, Dec 22, 2009 at 11:35 AM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: Try tracing the calls from both sides with TCPDUMP or enable siptrace on FreeSwitch. I guess this will give you some clue. __Yehavi: Additionally, turn on debugging on the console and capture that output. If you use fs_cli it has debug output turned on by default. Pastebin that output and post the link in this thread. If you happen to look at the traces and figure it out then please let us know. :) -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] Freeswitch not seeing Register requests
Yes, the internal profile exists. Name Type Data State = internal profile sip:mod_so...@192.168.10.25:5060 RUNNING (0) internal-ipv6 profile sip:mod_so...@[::1]:5060 RUNNING (0) external profile sip:mod_so...@192.168.10.25:5080 RUNNING (0) example.com gatewaysip:joeu...@example.com NOREG 192.168.10.25 alias internal ALIASED = 3 profiles 1 alias From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, December 22, 2009 11:15 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch not seeing Register requests On Tue, Dec 22, 2009 at 10:50 AM, Larry Marshall l...@marshap.com wrote: I have set up a second FreeSWITCH box on the same LAN. I have v16018 installed on it and have changed nothing. I configured a Polycom phone to register one of its four lines to this second box, but it does not register. When looking at the console, there is no activity. However, there is SIP activity on the box which I have captured via ngrep. It looks like the phone is sending out REGISTER requests but there is no response. The request on the pastebin repeats forever, with only the timestamp varying. On the new box do sofia status - does the internal profile exist? Is the problem that there are two FreeSWITCHes? Any suggestions on how I can make it work? On the original and the new box in vars.xml external_sip_ip=stun:stun.freeswitch.org On the original box in vars.xml external_sip_port=5090 but in the new it is 5080. Do I need to hardcode the external_sip_ip addresses in both boxes? http://pastebin.freeswitch.org/11600 Thanks Lars ___ 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] BLF on Grandstream GXP2020
Yuriy, The FS wiki has examples of how to control the BLF/MWI using events. I had no problem getting to work with my GXP2020. Let me know if you want some direct code examples. -- MM. On Thu, Dec 17, 2009 at 6:05 AM, Yuriy Ivzhenko yivzhe...@mksat.net wrote: Hallo All! I need information about setup BLF on GXP2010/2020 phones with Freeswitch. I search in Freeswitch Wiki and maillist archives but find no usable information. ___ 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] Authenticating end points by IP
Excellent work and answers. Thanks gentlemen. I'm firing off a new thread re: codecs et. al. Have a great Christmas and a wonderful, prosperous New Year. Regards, Ahmed. 2009/12/21 Bill W freeswi...@aastral.net I recently added an overview to this wiki page to help make things more clear as to which ACL you need for different purposes. http://wiki.freeswitch.org/wiki/ACL#Overview Thanks, Bill W. Mathieu Rene wrote: Check out: http://wiki.freeswitch.org/wiki/ACL#Users It'll automatically add users with a cidr= attribute to the ACL list. This way you can set channel variables in the users and use them through your dialplan, all authenticated by ip address. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca mailto:mr...@avgs.ca ___ 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 -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS ___ 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] Codecs and things
Hello people, Can someone please clear the following ambiguities with codecs: 1. Are we definitively able to run pass-through codecs (e.g. G.729) in Proxy Media mode, or does FS need to be running in bypass-media ? the Wiki is not clear in this regard 2. When an A-leg has negotiated a pass-through media codec, can the B-leg be transcoded into a non-pass-through codec, and vice-versa ? think A-leg incoming with a G.729 codec, and target for B-leg needs to be setup with a GSM-codec, say 3. Where in the developer's set of documentation are codecs discussed ? I would like to start porting some code of mine for G.729a/b/ab form a ti DSP platform to FS. FS lacking full G.729 support is proving quite a hindrance, and there is no clear direction from the dev community as to when the same will be available. Incidentally, any news on this effort ? where are we with code, and what's an ETA for a Beta ? 4. On the same lines as (3) above, there is a codec dev template in the source tree. Again, where can I find documentation relating to this ? the template has hardly any docs at all. Best regards and warm wishes for a Merry Christmas and a great New Year to one and all. Ahmed. -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS ___ 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 1.0.5pre10 is now available
It's upgrade-and-test time! The new release announcement is on the main FreeSWITCH page: http://www.freeswitch.org/node/224 Please update, test, and report back bugs and questions. Thanks! -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] Freeswitch not seeing Register requests
On Tue, Dec 22, 2009 at 11:46 AM, Lars Zeb larc...@yahoo.com wrote: Yes, the internal profile exists. Name Type Data State = internal profile sip:mod_so...@192.168.10.25:5060 RUNNING (0) internal-ipv6 profile sip:mod_so...@[::1]:5060 RUNNING (0) external profile sip:mod_so...@192.168.10.25:5080 RUNNING (0) example.com gateway sip:joeu...@example.comsip%3ajoeu...@example.com NOREG 192.168.10.25 alias internal ALIASED = 3 profiles 1 alias I would do a sanity check at this point: put this box and one phone on a completely separate network with nothing else and see what happens. -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] Choosing a Codec.
Hi, I am playing a file to a landline number. the format of the file is as follows: [r...@static-host var]# file message.wav message.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz In my vars.xml file I have used the following codec prefs: X-PRE-PROCESS cmd=set data=global_codec_prefs=PCMU,PCMA,GSM/ X-PRE-PROCESS cmd=set data=outbound_codec_prefs=PCMU,G722,GSM/ However, when freeswitch plays it, it always chooses the l...@8000hz codec. I'm not understanding why this is so. EXECUTE sofia/external/5135692...@208.78.161.197 playback(/var/message.wav) 2009-12-22 17:16:57.357048 [DEBUG] switch_ivr_play_say.c:1135 Codec Activated l...@8000hz 1 channels 20ms 2009-12-22 17:17:30.777182 [DEBUG] switch_ivr_play_say.c:1429 done playing file My basic intent is to avoid on-the-fly transcoding, while having a high quality audio playing on PSTN. Have I configured it wrong or does this transcoding always happen? Thanks, Vinuth. ___ 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] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch
Hi all, This weekend, I got the chance to buy a profoon IP-150 RJ11-to-USB device for just 15 euro. This is a device which has on one side a USB-connector and on the other side 2 RJ-11 connectors (one FXO and one FSX). Internally, the device seams to contain a tigerjet 560C chipset. (see here: http://www.tjnet.com/chips/tiger560C.htm) What is interesting on this device is that is uses standard USB device-classes that are by default supported by most operating-systems: usb-sound and usb-hid. When I connect it to my server (mac mini 3G running debian), the system automatically recognises these two classes [168391.922479] usbcore: registered new interface driver hiddev [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID 06e6:c31c] on usb-0001:10:1b.1-1 [168391.939548] usbcore: registered new interface driver usbhid [168391.943984] usbhid: v2.6:USB HID core driver [168392.154596] usbcore: registered new interface driver snd-usb-audio And -behold- when I connect a handset in one of the port, I even get a dialtone and I can sent out DTMF-dialtone which are somehow partly (But I have no idea what program actually generates this dialtone !!!) Now, the question: Any idea if / how this can incorperated into freeswitch? Is there a way to use this device to connect a phone to freeswitch without having to go throu a SIP-client first. Cheerio! Kr. Bonne. -- jabber/gtalk: krist...@krbonne.net attachment: kristoff_bonne.vcf signature.asc Description: OpenPGP digital signature ___ 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] Authenticating end points by IP
Hello Lars, You can apply any acl to any profile. What you should do really depends on what you want to accomplish. But let's take a simple example. Let's say you want to allow any phone on your internal network (192.168.0.0/24) to connect to your internal profile and make calls without having to provide a password. Then you could simply put these entries in your internal sofia profile. param name=apply-inbound-acl value=192.168.0.0/24/ param name=apply-register-acl value=192.168.0.0/24/ In that case, you do not need to include anything in the directory. The cidr entries in the directory are for providing additional control for each user id and what IPs they are allowed to make calls from. For your external profile, you may not want to have any ACLs at all, as you may not want to limit which IPs can connect to your switch to send you incoming calls. BUT, you need to make sure the dialplan connected to that external profile doesn't allow anyone to dial numbers that are not hosted on your system without proper authentication or controls. And believe me, people WILL try to do that. I've set up my system to email me whenever this happens and I have logged over 100 attempts to dial international numbers just since December 3rd. Hope this helps, Bill Lars Zeb wrote: Bill, Thanks for your ACL Overview. Perhaps you can help me understand more clearly. If you include the local-network-acl and apply-inbound-acl params in the sip_profiles and setup the list for localnet.auto in acl.conf.xml, does this mean you do not have to include the cidr attribute for individual extensions in the directory/default folder? Is apply-inbound-acl supposed to exist in both internal and external profiles while apply-inbound-acl is only in the internal? Thanks, Lars ___ 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] Choosing a Codec.
Why? You don't have to avoid it... why bother? /b On Dec 22, 2009, at 4:28 PM, Vinuth Madinur wrote: My basic intent is to avoid on-the-fly transcoding, while having a high quality audio playing on PSTN. ___ 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] Choosing a Codec.
The audio quality is a lot different when it plays on the landline. And the quality degrades a bit when the message played is lengthy 30s. So I thought it would be better if I have the file in mu-law and play it as is.. Thanks, Vinuth. On Wed, Dec 23, 2009 at 4:09 AM, Brian West br...@freeswitch.org wrote: Why? You don't have to avoid it... why bother? /b On Dec 22, 2009, at 4:28 PM, Vinuth Madinur wrote: My basic intent is to avoid on-the-fly transcoding, while having a high quality audio playing on PSTN. ___ 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] Choosing a Codec.
If its degrading like that you have bigger issues... the sound files played from wav files vs raw PCM files is NO different on a land line and I speak from very many years of experience... your wav files are ulaw in wav containers thus will never play native which might just be part of your problem. You would have to have raw headerless data in a .PCMU file for it to play native. Can you elaborate on your setup a bit more? /b On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote: The audio quality is a lot different when it plays on the landline. And the quality degrades a bit when the message played is lengthy 30s. So I thought it would be better if I have the file in mu-law and play it as is.. Thanks, Vinuth. ___ 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] Make error...
Hi all, I just downloaded the newest trunk about 5 minutes ago and I got the following make error on Ubuntu 8.04: gcc -E /usr/src/freeswitch/src/include/switch_cpp.h -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x 2/dev/null | grep -v ^# src/include/switch_swigable_cpp.h make OUR_MODULES=$(if test -z ; then tmp_mods=$(grep -v # /usr/src/freeswitch/modules.conf | sed -e s|^.*/|| | sort | uniq ); else tmp_mods= ; fi ; mods=$(for i in $tmp_mods ; do echo $i-all ; done ); echo $mods ) OUR_CLEAN_MODULES=$(if test -z ; then tmp_mods=$(grep -v # /usr/src/freeswitch/modules.conf | sed -e s|^.*/|| | sort | uniq ); else tmp_mods= ; fi ; mods=$(for i in $tmp_mods ; do echo $i-clean ; done ); echo $mods ) OUR_INSTALL_MODULES=$(if test -z ; then tmp_mods=$(grep -v # /usr/src/freeswitch/modules.conf | sed -e s|^.*/|| | sort | uniq ); else tmp_mods= ; fi ; mods=$(for i in $tmp_mods ; do echo $i-install ; done); echo $mods ) OUR_UNINSTALL_MODULES=$(if test -z ; then tmp_mods=$(grep -v # /usr/src/freeswitch/modules.conf | sed -e s|^.*/|| | sort | uniq ); else tmp_mods= ; fi ; mods=$(for i in $tmp_mods ; do echo $i-uninstall ; done); echo $mods ) OUR_DISABLED_MODULES=$(tmp_mods=$(grep # /usr/src/freeswitch/modules.conf | grep -v ## | sed -e s|^.*/|| | sort | uniq ); mods=$(for i in $tmp_mods ; do echo $i-all ; done ); echo $mods ) OUR_DISABLED_CLEAN_MODULES=$(tmp_mods=$(grep # /usr/src/freeswitch/modules.conf | grep -v ## | sed -e s|^.*/|| | sort | uniq ); mods=$(for i in $tmp_mods ; do echo $i-clean ; done ); echo $mods ) OUR_DISABLED_INSTALL_MODULES=$(tmp_mods=$(grep # /usr/src/freeswitch/modules.conf | grep -v ## | sed -e s|^.*/|| | sort | uniq ); mods=$(for i in $tmp_mods ; do echo $i-install ; done); echo $mods ) OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods=$(grep # /usr/src/freeswitch/modules.conf | grep -v ## | sed -e s|^.*/|| | sort | uniq ); mods=$(for i in $tmp_mods ; do echo $i-uninstall ; done); echo $mods ) `test -n || echo -s` all-recursive mkdir .libs Compiling src/switch_apr.c ... cc1: warnings being treated as errors src/switch_apr.c: In function 'switch_uuid_parse': src/switch_apr.c:899: warning: control reaches end of non-void function make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make: *** [all] Error 2 Regards, Klaus ___ 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] Make error...
Klaus Hochlehnert maili...@kh-dev.de wrote: src/switch_apr.c:899: warning: control reaches end of non-void function Are you on rev. 16032? As of 16032, this function shouldn't generate any such warning unless there's a compiler bug. ___ 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] Make error...
I was on 16031. Now I downloaded 16032 and currently the make is running. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jason White Sent: Wednesday, December 23, 2009 12:38 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Make error... Klaus Hochlehnert maili...@kh-dev.de wrote: src/switch_apr.c:899: warning: control reaches end of non-void function Are you on rev. 16032? As of 16032, this function shouldn't generate any such warning unless there's a compiler bug. ___ 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] Codecs and things
On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji a.alalo...@gmail.com wrote: Hello people, Can someone please clear the following ambiguities with codecs: Are we definitively able to run pass-through codecs (e.g. G.729) in Proxy Media mode, or does FS need to be running in bypass-media ? the Wiki is not clear in this regard Yes, you can use proxy media, bypass media, or even regular mode if you don't transcode (special for g729). Proxy media is really a special hack that should only be used for T38 passthrough. If you are using it for other purposes, think about it some more When an A-leg has negotiated a pass-through media codec, can the B-leg be transcoded into a non-pass-through codec, and vice-versa ? think A-leg incoming with a G.729 codec, and target for B-leg needs to be setup with a GSM-codec, say That would require transcoding - which can't be done if the codec is pass-through. Where in the developer's set of documentation are codecs discussed ? I would like to start porting some code of mine for G.729a/b/ab form a ti DSP platform to FS. FS lacking full G.729 support is proving quite a hindrance, and there is no clear direction from the dev community as to when the same will be available. Incidentally, any news on this effort ? where are we with code, and what's an ETA for a Beta ? I'd say look at the broadvoice or other simple self-contained codecs are done. Currently the only supported g729 solution is to use a digium board with mod_dahdi_codec. I don't have any info on a software based g729 solution. On the same lines as (3) above, there is a codec dev template in the source tree. Again, where can I find documentation relating to this ? the template has hardly any docs at all. Best regards and warm wishes for a Merry Christmas and a great New Year to one and all. Ahmed. -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS ___ 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 -- -Rupa ___ 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] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch
Interesting. It would have to do more than just dialtone/dtmf though. Need call control, caller id, etc. What do they ship with it as far as drivers go? On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne kristoff.bo...@skypro.be wrote: Hi all, This weekend, I got the chance to buy a profoon IP-150 RJ11-to-USB device for just 15 euro. This is a device which has on one side a USB-connector and on the other side 2 RJ-11 connectors (one FXO and one FSX). Internally, the device seams to contain a tigerjet 560C chipset. (see here: http://www.tjnet.com/chips/tiger560C.htm) What is interesting on this device is that is uses standard USB device-classes that are by default supported by most operating-systems: usb-sound and usb-hid. When I connect it to my server (mac mini 3G running debian), the system automatically recognises these two classes [168391.922479] usbcore: registered new interface driver hiddev [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID 06e6:c31c] on usb-0001:10:1b.1-1 [168391.939548] usbcore: registered new interface driver usbhid [168391.943984] usbhid: v2.6:USB HID core driver [168392.154596] usbcore: registered new interface driver snd-usb-audio And -behold- when I connect a handset in one of the port, I even get a dialtone and I can sent out DTMF-dialtone which are somehow partly (But I have no idea what program actually generates this dialtone !!!) Now, the question: Any idea if / how this can incorperated into freeswitch? Is there a way to use this device to connect a phone to freeswitch without having to go throu a SIP-client first. Cheerio! Kr. Bonne. -- jabber/gtalk: krist...@krbonne.net ___ 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 -- -Rupa ___ 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] Choosing a Codec.
On the other hand, a u-law WAV turned into L16 and then back to u-law to be sent down the line shouldn't suffer any alteration at all - if it does, the there's something wrong with the translation. The quality dropping over time is almost certainly down to something else. Vinuth -can you get a recording to compare with the original? --Dave If its degrading like that you have bigger issues... the sound files played from wav files vs raw PCM files is NO different on a land line and I speak from very many years of experience... your wav files are ulaw in wav containers thus will never play native which might just be part of your problem. You would have to have raw headerless data in a .PCMU file for it to play native. Can you elaborate on your setup a bit more? /b On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote: The audio quality is a lot different when it plays on the landline. And the quality degrades a bit when the message played is lengthy 30s. So I thought it would be better if I have the file in mu-law and play it as is.. Thanks, Vinuth. ___ 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] Choosing a Codec.
Have you considered GIPS http://www.gipscorp.com/products/overview.php ? -E ___ 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] Faxing Advice
Am I correct in presuming that Freeswitch will answer a fax from a local zap
based user
just like it does from an FXO port connected to a POTS line? What I hope to
do here is
catch any call made from that extension (the zap based fax machine/user) and
push its
call into the fax module.
Yes, when a device (phone/fax/modem/whatever) is plugged into the FXS it gets
dialtone
and dials. Whatever it dials is put into ${destination_number} just like any
SIP phone that
dials. This extension looks ok. Try it out and let us know how it goes.
-MC
Michael,
It worked well, there was however a humorous moment: I was testing with my own
shell script
that simply emailed me directly to my postfix gateway, my exchange server and
mua understood the
uuencoded attachment so once it started working I modified the script to send
to our fax service.
Well they didn't understand uuencode so the attachment, a single page tiff, got
faxed as 23 pages
of binary :) I used mutt with a redirection to a specific muttrc which
understands mime encoding
which should work everywhere...
Thanks for the help, you've made an office full of people happy...
jlc
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Re: [Freeswitch-users] WARNING On Inbound Call Question
If this is using prid it also requires the latest drivers from sangoma. I am pretty sure these are just in dev snapshots not release drivers yet. Something 3.5.8.6 or later iirc. Mike On Dec 21, 2009, at 7:52 PM, Brian West br...@freeswitch.org wrote: You know that warning is meaningless. Search the archives we have talked about this to no end it seems. And I'm sure Moy fixed this. /b On Dec 21, 2009, at 6:24 PM, Jerry Richards wrote: Okay, I upgraded to 1.0.5pre9 and tried this test again and I do not see the WARNING in the Freeswitch log. However, it still behaves the same way. That is, the internal callee rings for about 12 seconds, then stops ringing, and the PSTN caller just hears ringback for about 60 seconds and is not given the opportunity to leave voice mail. In contrast, an internal-to-internal call will go to voice mail after 30 seconds. I put a new 11595 log into the pastebin. Is there some Sangoma Wanpipe driver (or Freeswitch) setting that would correct this? Best Regards, Jerry ___ 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] Variables for install directories
For the path in the dialplan I don't think we have any right now but
file a bug on jira and I can try to add them. As for something in the
script itself that is a bit more work but if anyone has a patch to
inject some vars into scripts like that it would be a nice addition.
Mike
On Dec 21, 2009, at 7:03 PM, Joseph L. Casale jcas...@activenetwerx.com
wrote:
Searching through the wiki for any indication as to what if any
variables exist
for the install location in that I can leverage in a script.
Can anyone point me along, I can’t seem to find anything. I want to
place a shell
script in /opt/freeswitch/scripts that needs a reference to a conf
file that a binary
it runs is calling.
So now I have in two places hardcoded paths that I was hoping to
avoid, in the dialplan
and in the shell script. When either of these is run, does there
exist something like
action application=system data=${freeswitch_install_dir}/scripts/
shell_script.sh/
and the same for use inside the shell script?
Thanks!
jlc
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Re: [Freeswitch-users] Force endpoint to use rfc2833 for dtmf
Not sure if we have an option to disable info. Even without this, dtmf should go across the bridge fine. Please open up a bug on jira about this Mike On Dec 22, 2009, at 6:40 AM, Peter P GMX prometheus...@gmx.net wrote: Hello, in a bigger installation with some thousand endpoints in the field we see, that the endpoint equipment is always using INFO messages (standard setting is auto, so the endpoint decides which method to use). I have 2 questions to that scenario: 1. Is there a way that Freeswitch forces/restricts the endpoint to use rfc2833 or not to send to allow INFO in the invite message? 2. Currently INFO messages do not get forwarded from the caller through freeswitch to called endpoint. How can we enable that FS is fowarding the INFO messages? Best regards Peter ___ 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] Freeswitch not seeing Register requests
If your seeing the trafic in ngrep bit not in sip trace in Sofia when enabled, your firewall is blocking the traffic Mike On Dec 22, 2009, at 5:20 PM, Michael Collins m...@freeswitch.org wrote: On Tue, Dec 22, 2009 at 11:46 AM, Lars Zeb larc...@yahoo.com wrote: Yes, the internal profile exists. Name Type Data State === === === === === === === === === == internal profile sip:mod_so...@192.168.10.25:5060 RUNNING (0) internal-ipv6 profile sip:mod_so...@[:: 1]:5060 RUNNING (0) external profile sip:mod_so...@192.168.10.25:5080 RUNNING (0) example.com gateway sip:joeu...@example.com NOREG 192.168.10.25 alias internal ALIASED === === === === === === === === === == 3 profiles 1 alias I would do a sanity check at this point: put this box and one phone on a completely separate network with nothing else and see what happens. -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 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] Codecs and things
We expect the g729 sometime very soon, weeks not months away. Mike On Dec 22, 2009, at 7:45 PM, Rupa Schomaker r...@rupa.com wrote: On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji a.alalo...@gmail.com wrote: Hello people, Can someone please clear the following ambiguities with codecs: Are we definitively able to run pass-through codecs (e.g. G.729) in Proxy Media mode, or does FS need to be running in bypass-media ? the Wiki is not clear in this regard Yes, you can use proxy media, bypass media, or even regular mode if you don't transcode (special for g729). Proxy media is really a special hack that should only be used for T38 passthrough. If you are using it for other purposes, think about it some more When an A-leg has negotiated a pass-through media codec, can the B- leg be transcoded into a non-pass-through codec, and vice-versa ? think A- leg incoming with a G.729 codec, and target for B-leg needs to be setup with a GSM-codec, say That would require transcoding - which can't be done if the codec is pass-through. Where in the developer's set of documentation are codecs discussed ? I would like to start porting some code of mine for G.729a/b/ab form a ti DSP platform to FS. FS lacking full G.729 support is proving quite a hindrance, and there is no clear direction from the dev community as to when the same will be available. Incidentally, any news on this effort ? where are we with code, and what's an ETA for a Beta ? I'd say look at the broadvoice or other simple self-contained codecs are done. Currently the only supported g729 solution is to use a digium board with mod_dahdi_codec. I don't have any info on a software based g729 solution. On the same lines as (3) above, there is a codec dev template in the source tree. Again, where can I find documentation relating to this ? the template has hardly any docs at all. Best regards and warm wishes for a Merry Christmas and a great New Year to one and all. Ahmed. -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS ___ 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 -- -Rupa ___ 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] Choosing a Codec.
That being said, ulaw l16 alaw will cause degredation and any other modifications such as volume adjustment in this path will make it worse. Tha being said that does not sound like what you are experiencing Mike On Dec 22, 2009, at 10:29 PM, David Knell d...@3c.co.uk wrote: On the other hand, a u-law WAV turned into L16 and then back to u- law to be sent down the line shouldn't suffer any alteration at all - if it does, the there's something wrong with the translation. The quality dropping over time is almost certainly down to something else. Vinuth -can you get a recording to compare with the original? --Dave If its degrading like that you have bigger issues... the sound files played from wav files vs raw PCM files is NO different on a land line and I speak from very many years of experience... your wav files are ulaw in wav containers thus will never play native which might just be part of your problem. You would have to have raw headerless data in a .PCMU file for it to play native. Can you elaborate on your setup a bit more? /b On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote: The audio quality is a lot different when it plays on the landline. And the quality degrades a bit when the message played is lengthy 30s. So I thought it would be better if I have the file in mu-law and play it as is.. Thanks, Vinuth. ___ 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] Variables for install directories
For the path in the dialplan I don't think we have any right now but file a bug on jira and I can try to add them. As for something in the script itself that is a bit more work but if anyone has a patch to inject some vars into scripts like that it would be a nice addition. Mike Ok, signed up for an account, where does the dialplan part go, FSCORE? Thanks for the help! jlc ___ 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_conference voice problems when two parties speaking
Hello. I've written an application using mod_conference which often has two parties speaking at once and one party listening. When only one party is speaking, the sound quality is fine, but when a second party starts speaking while the first party is still speaking, the second party's voice is cut-off at the beginning, and both parties voices seem to get choppy, like maybe all of the packets aren't getting delivered properly. I'm experiencing this with the latest trunk version (16012). I have the member-flags variable set to waste, and comfort-noise is set to true. I'm not sure where the problem is coming from; I think if it was a VOIP issue I'd hear the same problem when only one party is speaking. Is there something in mod_conference which would try to filter out other voices when one voice is speaking? I'd really appreciate any suggestions about where to look to find this problem. Thanks in advance, Marc___ 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] Variables for install directories
Sounds right to me, just assign it to me if it lets you Mike On Dec 23, 2009, at 12:03 AM, Joseph L. Casale jcas...@activenetwerx.com wrote: For the path in the dialplan I don't think we have any right now but file a bug on jira and I can try to add them. As for something in the script itself that is a bit more work but if anyone has a patch to inject some vars into scripts like that it would be a nice addition. Mike Ok, signed up for an account, where does the dialplan part go, FSCORE? Thanks for the help! jlc ___ 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] mod_conference voice problems when two parties speaking
Try setting your energy-level down, at 0 for instance. If it helps, then increase until you find a happy medium. On Dec 22, 2009, at 11:14 PM, Marc Orenberg wrote: Hello. I've written an application using mod_conference which often has two parties speaking at once and one party listening. When only one party is speaking, the sound quality is fine, but when a second party starts speaking while the first party is still speaking, the second party's voice is cut-off at the beginning, and both parties voices seem to get choppy, like maybe all of the packets aren't getting delivered properly. I'm experiencing this with the latest trunk version (16012). I have the member-flags variable set to waste, and comfort- noise is set to true. I'm not sure where the problem is coming from; I think if it was a VOIP issue I'd hear the same problem when only one party is speaking. Is there something in mod_conference which would try to filter out other voices when one voice is speaking? I'd really appreciate any suggestions about where to look to find this problem. Thanks in advance, Marc ___ 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] mod_conference voice problems when two parties speaking
Thanks Rob, thanks Jason. I'm going to try this first thing tomorrow. The energy-level paramter is described in the file as, Energy level required for audio to be sent to the other users, so one would think that this would have no effect if member-flags is set to waste, right? From: Jason White ja...@jasonjgw.net To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 23, 2009 12:29:58 AM Subject: Re: [Freeswitch-users] mod_conference voice problems when two parties speaking Marc Orenberg m...@kasteris.com wrote: Is there something in mod_conference which would try to filter out other voices when one voice is speaking? Try reducing the energy level parameter in case this is the issue. It's 7/8/9 on the key pad during the call, or via the conference command, or the settings in conference.conf.xml. ___ 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] mod_conference voice problems when two parties speaking
No, from my understanding that's not how it works. Waste just means it'll always send RTP packets, doesn't mean it will contain audio... so if you have audio that's under your energy threshold, you still won't hear it. Dan On Wed, Dec 23, 2009 at 12:45 AM, Marc Orenberg m...@kasteris.com wrote: Thanks Rob, thanks Jason. I'm going to try this first thing tomorrow. The energy-level paramter is described in the file as, Energy level required for audio to be sent to the other users, so one would think that this would have no effect if member-flags is set to waste, right? -- *From:* Jason White ja...@jasonjgw.net *To:* freeswitch-users@lists.freeswitch.org *Sent:* Wed, December 23, 2009 12:29:58 AM *Subject:* Re: [Freeswitch-users] mod_conference voice problems when two parties speaking Marc Orenberg m...@kasteris.com wrote: Is there something in mod_conference which would try to filter out other voices when one voice is speaking? Try reducing the energy level parameter in case this is the issue. It's 7/8/9 on the key pad during the call, or via the conference command, or the settings in conference.conf.xml. ___ 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] Choosing a Codec.
On Tue, 2009-12-22 at 23:52 -0500, Michael Jerris wrote: That being said, ulaw l16 alaw will cause degredation and any other modifications such as volume adjustment in this path will make it worse. Indeed. Storing prompts as 8k, 16-bit WAVs makes a lot of sense. [I am inordinately pleased with the above] --Dave ___ 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] Choosing a Codec.
My setup is as follows: FreeSWITCH - SIP Trunk - PSTN. From freeswitch, I'm making outbound calls using event socket via the external profile. Except for the ext_rtp_ip and ext_sip_ip, everything is default settings. Using playback application, I'm playing a mu-law audio. I'm also starting the vmd application, so that I can replay the message on beep. Thanks for your suggestion on native format. I'll try it. Thanks, Vinuth. On Wed, Dec 23, 2009 at 4:41 AM, Brian West br...@freeswitch.org wrote: If its degrading like that you have bigger issues... the sound files played from wav files vs raw PCM files is NO different on a land line and I speak from very many years of experience... your wav files are ulaw in wav containers thus will never play native which might just be part of your problem. You would have to have raw headerless data in a .PCMU file for it to play native. Can you elaborate on your setup a bit more? /b On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote: The audio quality is a lot different when it plays on the landline. And the quality degrades a bit when the message played is lengthy 30s. So I thought it would be better if I have the file in mu-law and play it as is.. Thanks, Vinuth. ___ 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] Choosing a Codec.
On Wed, Dec 23, 2009 at 12:17 PM, David Knell d...@3c.co.uk wrote: On Tue, 2009-12-22 at 23:52 -0500, Michael Jerris wrote: That being said, ulaw l16 alaw will cause degredation and any other modifications such as volume adjustment in this path will make it worse. Indeed. Storing prompts as 8k, 16-bit WAVs makes a lot of sense. [I am inordinately pleased with the above] --Dave Thanks, will try and get back. ___ 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
