Hi I see the following logs repeating fast during a call sometimes: 2022-12-17 10:00:28.551240 99.20% [DEBUG] mod_opus.c:740 Opus encoder stats: Frames[0] Bytes encoded[0] Encoded length ms[0] Average encoded bitrate bps[0] 2022-12-17 10:00:28.551240 99.20% [DEBUG] mod_opus.c:619 Opus encoder: set bitrate to local settings [72000bps] 2022-12-17 10:00:28.631241 99.20% [DEBUG] mod_opus.c:725 Opus decoder stats: Frames[0] PLC[0] FEC[0] 2022-12-17 10:00:28.631241 99.20% [DEBUG] mod_opus.c:740 Opus encoder stats: Frames[0] Bytes encoded[0] Encoded length ms[0] Average encoded bitrate bps[0] 2022-12-17 10:00:28.631241 99.20% [DEBUG] mod_opus.c:619 Opus encoder: set bitrate to local settings [72000bps] 2022-12-17 10:00:28.651240 99.20% [DEBUG] mod_opus.c:725 Opus decoder stats: Frames[0] PLC[0] FEC[0] 2022-12-17 10:00:28.651240 99.20% [DEBUG] mod_opus.c:740 Opus encoder stats: Frames[0] Bytes encoded[0] Encoded length ms[0] Average encoded bitrate bps[0] 2022-12-17 10:00:28.651240 99.20% [DEBUG] mod_opus.c:619 Opus encoder: set bitrate to local settings [72000bps] What do they mean? Is something wrong? thanks
No, nothing wrong. You have a DEBUG level of logging set so you'll see more verbose information. More details can be found here https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack#SofiaSI... ~Len On Tue, Dec 27, 2022 at 10:35 AM Babak Yakhchali <babak.yakhchali@gmail.com> wrote:
Hi I see the following logs repeating fast during a call sometimes:
2022-12-17 10:00:28.551240 99.20% [DEBUG] mod_opus.c:740 Opus encoder stats: Frames[0] Bytes encoded[0] Encoded length ms[0] Average encoded bitrate bps[0] 2022-12-17 10:00:28.551240 99.20% [DEBUG] mod_opus.c:619 Opus encoder: set bitrate to local settings [72000bps] 2022-12-17 10:00:28.631241 99.20% [DEBUG] mod_opus.c:725 Opus decoder stats: Frames[0] PLC[0] FEC[0] 2022-12-17 10:00:28.631241 99.20% [DEBUG] mod_opus.c:740 Opus encoder stats: Frames[0] Bytes encoded[0] Encoded length ms[0] Average encoded bitrate bps[0] 2022-12-17 10:00:28.631241 99.20% [DEBUG] mod_opus.c:619 Opus encoder: set bitrate to local settings [72000bps] 2022-12-17 10:00:28.651240 99.20% [DEBUG] mod_opus.c:725 Opus decoder stats: Frames[0] PLC[0] FEC[0] 2022-12-17 10:00:28.651240 99.20% [DEBUG] mod_opus.c:740 Opus encoder stats: Frames[0] Bytes encoded[0] Encoded length ms[0] Average encoded bitrate bps[0] 2022-12-17 10:00:28.651240 99.20% [DEBUG] mod_opus.c:619 Opus encoder: set bitrate to local settings [72000bps]
What do they mean? Is something wrong? thanks _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
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Hey there.... Does anyone have experience with ring groups that show a preview (on all video capable phones) of a door phone? I have Fanvil (V67/I53W) and Grandstream video phones (GXV3480/GXV3240) connected to Fanvil(I64) and Axis(A8207) door phones. the door phones are able to send preview and the phones are able to receive previdw but the screens only show a black image - no live video. Jerry
On Tuesday 27 December 2022 at 18:32:58, Jerry Kendall wrote:
Does anyone have experience with ring groups that show a preview (on all video capable phones) of a door phone?
I somehow doubt that it is possible to send a preview to a ring group. That's verging on video-conferencing, after all.
I have Fanvil (V67/I53W) and Grandstream video phones (GXV3480/GXV3240) connected to Fanvil(I64) and Axis(A8207) door phones.
the door phones are able to send preview and the phones are able to receive previdw but the screens only show a black image - no live video.
Have you tried reducing the configuration to a simple setup of one door phone and one video phone, to see whether a preview can be sent/received like that? If that works, do a packet capture and look at the SDP part of the SIP communication, and then compare with the same thing using a ring group. Antony. -- How does an Israeli man make beer? Hebrews it. Please reply to the list; please *don't* CC me.
Hi Antony, Yes, it does work with one device in the ring grouop. In fact, if I toogle or change the devices within a ring group it works regardless of which device - as long as it's only one device in the ring group. Once there are more than a single device, the the video preview only works on the first device to reply with a '183 progress...' All the other devices get a cancel as FS gets the 183 and cancels all other devices... even if they send a '183 progres...' if I set 'bridge_early_media, then all devices will accept the video however, the video is simply a blank screen with on content Jerry On 12/27/2022 12:40 PM, Antony Stone wrote:
On Tuesday 27 December 2022 at 18:32:58, Jerry Kendall wrote:
Does anyone have experience with ring groups that show a preview (on all video capable phones) of a door phone? I somehow doubt that it is possible to send a preview to a ring group.
That's verging on video-conferencing, after all.
I have Fanvil (V67/I53W) and Grandstream video phones (GXV3480/GXV3240) connected to Fanvil(I64) and Axis(A8207) door phones.
the door phones are able to send preview and the phones are able to receive previdw but the screens only show a black image - no live video. Have you tried reducing the configuration to a simple setup of one door phone and one video phone, to see whether a preview can be sent/received like that?
If that works, do a packet capture and look at the SDP part of the SIP communication, and then compare with the same thing using a ring group.
Antony.
Typo... the last line should be if I set 'bridge_early_media, then all devices will accept the video however, the video is simply a blank screen with *no* content- basically a black screen On 12/27/2022 1:58 PM, Jerry Kendall wrote:
Hi Antony,
Yes, it does work with one device in the ring grouop.
In fact, if I toogle or change the devices within a ring group it works regardless of which device - as long as it's only one device in the ring group. Once there are more than a single device, the the video preview only works on the first device to reply with a '183 progress...' All the other devices get a cancel as FS gets the 183 and cancels all other devices... even if they send a '183 progres...'
if I set 'bridge_early_media, then all devices will accept the video however, the video is simply a blank screen with on content
Jerry
On 12/27/2022 12:40 PM, Antony Stone wrote:
On Tuesday 27 December 2022 at 18:32:58, Jerry Kendall wrote:
Does anyone have experience with ring groups that show a preview (on all video capable phones) of a door phone? I somehow doubt that it is possible to send a preview to a ring group.
That's verging on video-conferencing, after all.
I have Fanvil (V67/I53W) and Grandstream video phones (GXV3480/GXV3240) connected to Fanvil(I64) and Axis(A8207) door phones.
the door phones are able to send preview and the phones are able to receive previdw but the screens only show a black image - no live video. Have you tried reducing the configuration to a simple setup of one door phone and one video phone, to see whether a preview can be sent/received like that?
If that works, do a packet capture and look at the SDP part of the SIP communication, and then compare with the same thing using a ring group.
Antony.
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.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 https://freeswitch.com
On Wednesday 28 December 2022 at 02:19:30, Jerry Kendall wrote:
Typo...
the last line should be
if I set 'bridge_early_media, then all devices will accept the video howeve r, the video is simply a blank screen with *no* content- basically a black screen
You're basically expecting to be able to convert unicast traffic into multicast (unicast means one device sending IP to one other device, multicast means one device sending IP to many devices) and although this may be possible, it's (a) not automatic, and (b) not something that FreeSwitch will do for you, to the best of my knowledge. Antony.
On 12/27/2022 1:58 PM, Jerry Kendall wrote:
Hi Antony,
Yes, it does work with one device in the ring grouop.
In fact, if I toogle or change the devices within a ring group it works regardless of which device - as long as it's only one device in the ring group. Once there are more than a single device, the the video preview only works on the first device to reply with a '183 progress...' All the other devices get a cancel as FS gets the 183 and cancels all other devices... even if they send a '183 progres...'
if I set 'bridge_early_media, then all devices will accept the video however, the video is simply a blank screen with on content
Jerry
On 12/27/2022 12:40 PM, Antony Stone wrote:
On Tuesday 27 December 2022 at 18:32:58, Jerry Kendall wrote:
Does anyone have experience with ring groups that show a preview (on all video capable phones) of a door phone?
I somehow doubt that it is possible to send a preview to a ring group.
That's verging on video-conferencing, after all.
I have Fanvil (V67/I53W) and Grandstream video phones (GXV3480/GXV3240) connected to Fanvil(I64) and Axis(A8207) door phones.
the door phones are able to send preview and the phones are able to receive previdw but the screens only show a black image - no live video.
Have you tried reducing the configuration to a simple setup of one door phone and one video phone, to see whether a preview can be sent/received like that?
If that works, do a packet capture and look at the SDP part of the SIP communication, and then compare with the same thing using a ring group.
-- Late in 1972 President Richard Nixon announced that the rate of increase of inflation was decreasing. This was the first time a sitting president used a third derivative to advance his case for re-election. - Hugo Rossi, Notices of the American Mathematical Society Please reply to the list; please *don't* CC me.
Hi all... I would like to create a dialplan conference. I would like the have this dialplan conference make outcalls to a group of maybe 20 devices This, I know is all basic stuff... I can already do this - usually use it for Paging Groups... Works great. However, I'd like to set the max_members at 2 and still have the outcalls happen and have the 20 or so devices ring as per normal. Now, assuming someone dials the conference number that hits this dialplan - this being member number 1 Here's the catch, Is there a way to have mod_conference, once the first ringing device (out of the 20 or so..) answers (hence the max_members = 2), send a CANCEL to all the other INVITEes..???? Thanks in advance for all your ideas... Jerry
Hey there... I've come across an odd issue with the outbound media from FS... I have a phone, calls a ring group at say 22. Now if the phones and FS are on the same network, the early media is sent to all phones properly... If the phones are on different networks connected via VPN, again, the early media is sent to all phones properly... not if the phones are on different networks and a firewall is between without vpn tunneling.... The early media is NOT delivered properly... It seems as thought he SDP payload is not fully processed - again, it seems.... So, how do I tell FS to send the early media to the phones when the phones are on private IP space behind a firewall.??? It almost looks like the private IPs and not being mapped properly when sending early media. (the phones are replying to the INVITE with 183 progress status that includes SDP) I have NAT working properly and calls, even confererce calls with and without video all work fine - ONCE the call has been answered. I have been trying to get early media(video) and works but, only when there is no firewall between FS and the phones. This, of course is not acceptable for customer based setups. Anyu ideas on where to look for the cause? Jerry
So I did a quick check, I ran 'tcpdump net 10.0.0.0/8' I got thousands of packets immediately on dialing a ring group. So, it seems that FS is indeed NOT processing the NAT mapping correctly... I could be wrong... I'm not above that... to draw a sort of image... [phones on 10.0.0.x net] ----------- [ FW ] --------------- { internet } ------------- [ FS Server on public IP ] The traffic to/from the phones goes correctly unless I have the group setup to send early media to allow ringing phones to show early media video... If the group is setup to send early media, the ringing phones are NOT getting the earling media traffic.. The phones have the video screen active AND showing but NO video is displayed as it is sent from FS on the public IP to the phones PRIVATE IP... I am doing this using a hard coded test dialplan conference call like so... ================== <context name="registered-ext"> <extension name="dial-videocom"> <condition> <action application="set" data="conference_auto_outcall_timeout=5"/> <action application="set" data="conference_auto_outcall_caller_id_name=Intercom"/> <action application="set" data="conference_auto_outcall_caller_id_number=350"/> <action application="set" data="conference_auto_outcall_skip_member_beep=false"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=false}"/> <action application="set" data="conference_auto_outcall_profile=pager"/> <action application="set" data="conference_utils_auto_outcall_flags=mute"/> <action application="export" data="bridge_early_media=true"/> <action application="export" data="ignore_early_media=false"/> <action application="set" data="ringback=%(2000,4000,440.0,480.0)"/>\n"; <action application="conference_set_auto_outcall" data="sofia/customer/500%domain"/> <action application="conference_set_auto_outcall" data="sofia/customer/501%domain"/> <action application="conference_set_auto_outcall" data="sofia/customer/502%domain"/> <action application="conference_set_auto_outcall" data="sofia/customer/503%domain"/> <action application="conference" data="vid_intercom_808@domain++flags{mute|endconf|moderator}"/> </condition> </extension> <context> ================== Does anyone have any ideas where I need to change things...?? Regards, Jerry On 12/31/2022 9:47 PM, Jerry Kendall wrote:
Hey there...
I've come across an odd issue with the outbound media from FS...
I have a phone, calls a ring group at say 22. Now if the phones and FS are on the same network, the early media is sent to all phones properly... If the phones are on different networks connected via VPN, again, the early media is sent to all phones properly...
not if the phones are on different networks and a firewall is between without vpn tunneling.... The early media is NOT delivered properly...
It seems as thought he SDP payload is not fully processed - again, it seems....
So, how do I tell FS to send the early media to the phones when the phones are on private IP space behind a firewall.??? It almost looks like the private IPs and not being mapped properly when sending early media. (the phones are replying to the INVITE with 183 progress status that includes SDP)
I have NAT working properly and calls, even confererce calls with and without video all work fine - ONCE the call has been answered. I have been trying to get early media(video) and works but, only when there is no firewall between FS and the phones.
This, of course is not acceptable for customer based setups.
Anyu ideas on where to look for the cause?
Jerry
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.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 https://freeswitch.com
Another bit of hopefully useful info - the video shows up on all phones when they are answered. Jerry On 12/31/2022 10:47 PM, Jerry Kendall wrote:
So I did a quick check,
I ran 'tcpdump net 10.0.0.0/8'
I got thousands of packets immediately on dialing a ring group. So, it seems that FS is indeed NOT processing the NAT mapping correctly... I could be wrong... I'm not above that...
to draw a sort of image...
[phones on 10.0.0.x net] ----------- [ FW ] --------------- { internet } ------------- [ FS Server on public IP ]
The traffic to/from the phones goes correctly unless I have the group setup to send early media to allow ringing phones to show early media video...
If the group is setup to send early media, the ringing phones are NOT getting the earling media traffic.. The phones have the video screen active AND showing but NO video is displayed as it is sent from FS on the public IP to the phones PRIVATE IP...
I am doing this using a hard coded test dialplan conference call like so... ================== <context name="registered-ext"> <extension name="dial-videocom"> <condition> <action application="set" data="conference_auto_outcall_timeout=5"/> <action application="set" data="conference_auto_outcall_caller_id_name=Intercom"/> <action application="set" data="conference_auto_outcall_caller_id_number=350"/> <action application="set" data="conference_auto_outcall_skip_member_beep=false"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=false}"/> <action application="set" data="conference_auto_outcall_profile=pager"/> <action application="set" data="conference_utils_auto_outcall_flags=mute"/>
<action application="export" data="bridge_early_media=true"/> <action application="export" data="ignore_early_media=false"/>
<action application="set" data="ringback=%(2000,4000,440.0,480.0)"/>\n";
<action application="conference_set_auto_outcall" data="sofia/customer/500%domain"/> <action application="conference_set_auto_outcall" data="sofia/customer/501%domain"/> <action application="conference_set_auto_outcall" data="sofia/customer/502%domain"/> <action application="conference_set_auto_outcall" data="sofia/customer/503%domain"/>
<action application="conference" data="vid_intercom_808@domain++flags{mute|endconf|moderator}"/> </condition> </extension> <context> ==================
Does anyone have any ideas where I need to change things...??
Regards, Jerry
On 12/31/2022 9:47 PM, Jerry Kendall wrote:
Hey there...
I've come across an odd issue with the outbound media from FS...
I have a phone, calls a ring group at say 22. Now if the phones and FS are on the same network, the early media is sent to all phones properly... If the phones are on different networks connected via VPN, again, the early media is sent to all phones properly...
not if the phones are on different networks and a firewall is between without vpn tunneling.... The early media is NOT delivered properly...
It seems as thought he SDP payload is not fully processed - again, it seems....
So, how do I tell FS to send the early media to the phones when the phones are on private IP space behind a firewall.??? It almost looks like the private IPs and not being mapped properly when sending early media. (the phones are replying to the INVITE with 183 progress status that includes SDP)
I have NAT working properly and calls, even confererce calls with and without video all work fine - ONCE the call has been answered. I have been trying to get early media(video) and works but, only when there is no firewall between FS and the phones.
This, of course is not acceptable for customer based setups.
Anyu ideas on where to look for the cause?
Jerry
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.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 https://freeswitch.com
The issues has been resolved... On Grandstream phones, you need STUN setup and you need to have the NAT-Traversal set to STUN. Thanks, Jerry On 1/1/2023 12:50 AM, Jerry Kendall wrote:
Another bit of hopefully useful info - the video shows up on all phones when they are answered.
Jerry
On 12/31/2022 10:47 PM, Jerry Kendall wrote:
So I did a quick check,
I ran 'tcpdump net 10.0.0.0/8'
I got thousands of packets immediately on dialing a ring group. So, it seems that FS is indeed NOT processing the NAT mapping correctly... I could be wrong... I'm not above that...
to draw a sort of image...
[phones on 10.0.0.x net] ----------- [ FW ] --------------- { internet } ------------- [ FS Server on public IP ]
The traffic to/from the phones goes correctly unless I have the group setup to send early media to allow ringing phones to show early media video...
If the group is setup to send early media, the ringing phones are NOT getting the earling media traffic.. The phones have the video screen active AND showing but NO video is displayed as it is sent from FS on the public IP to the phones PRIVATE IP...
I am doing this using a hard coded test dialplan conference call like so... ================== <context name="registered-ext"> <extension name="dial-videocom"> <condition> <action application="set" data="conference_auto_outcall_timeout=5"/> <action application="set" data="conference_auto_outcall_caller_id_name=Intercom"/> <action application="set" data="conference_auto_outcall_caller_id_number=350"/> <action application="set" data="conference_auto_outcall_skip_member_beep=false"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=false}"/> <action application="set" data="conference_auto_outcall_profile=pager"/> <action application="set" data="conference_utils_auto_outcall_flags=mute"/>
<action application="export" data="bridge_early_media=true"/> <action application="export" data="ignore_early_media=false"/>
<action application="set" data="ringback=%(2000,4000,440.0,480.0)"/>\n";
<action application="conference_set_auto_outcall" data="sofia/customer/500%domain"/> <action application="conference_set_auto_outcall" data="sofia/customer/501%domain"/> <action application="conference_set_auto_outcall" data="sofia/customer/502%domain"/> <action application="conference_set_auto_outcall" data="sofia/customer/503%domain"/>
<action application="conference" data="vid_intercom_808@domain++flags{mute|endconf|moderator}"/> </condition> </extension> <context> ==================
Does anyone have any ideas where I need to change things...??
Regards, Jerry
On 12/31/2022 9:47 PM, Jerry Kendall wrote:
Hey there...
I've come across an odd issue with the outbound media from FS...
I have a phone, calls a ring group at say 22. Now if the phones and FS are on the same network, the early media is sent to all phones properly... If the phones are on different networks connected via VPN, again, the early media is sent to all phones properly...
not if the phones are on different networks and a firewall is between without vpn tunneling.... The early media is NOT delivered properly...
It seems as thought he SDP payload is not fully processed - again, it seems....
So, how do I tell FS to send the early media to the phones when the phones are on private IP space behind a firewall.??? It almost looks like the private IPs and not being mapped properly when sending early media. (the phones are replying to the INVITE with 183 progress status that includes SDP)
I have NAT working properly and calls, even confererce calls with and without video all work fine - ONCE the call has been answered. I have been trying to get early media(video) and works but, only when there is no firewall between FS and the phones.
This, of course is not acceptable for customer based setups.
Anyu ideas on where to look for the cause?
Jerry
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.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 https://freeswitch.com
participants (4)
-
Antony Stone
-
Babak Yakhchali
-
Jerry Kendall
-
Len Graham