Hi, I'm quite new to freeswitch and voip, so this may be in some way a noob question but I've dug through a lot of the freeswitch docs and done quite a bit of searching and haven't figured it out yet. We're creating a relatively straightforward VoIP system to be used entirely internally (ex: users can only connect with other registered users within our system) and we'd like to be able to combine combine two calls into a single audio stream to the user without the two calls hearing each other. For example, if I'm talking to Mike on line 1 and I'm talking to Erin on line 2, is there a way for me to hear both Erin and Mike simultaneously but for them not to hear each other? Alternatively, is there a straightforward way to simply merge the calls into a conference-type experience where we all hear each other without the user explicitly setting up a conference call? Any and all input is greatly appreciated, as I'm up to my ears in freeswitch but have approximately zero experience with it! Ben -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Merge-two-calls-tp5188763p5188... Sent from the freeswitch-users mailing list archive at Nabble.com.
2010. június 17. 02.24.08 dátummal benxmy az alábbiakat írta:
Hi, I'm quite new to freeswitch and voip, so this may be in some way a noob question but I've dug through a lot of the freeswitch docs and done quite a bit of searching and haven't figured it out yet.
We're creating a relatively straightforward VoIP system to be used entirely internally (ex: users can only connect with other registered users within our system) and we'd like to be able to combine combine two calls into a single audio stream to the user without the two calls hearing each other. For example, if I'm talking to Mike on line 1 and I'm talking to Erin on line 2, is there a way for me to hear both Erin and Mike simultaneously but for them not to hear each other?
Alternatively, is there a straightforward way to simply merge the calls into a conference-type experience where we all hear each other without the user explicitly setting up a conference call?
Any and all input is greatly appreciated, as I'm up to my ears in freeswitch but have approximately zero experience with it!
Ben
Hi. What is the point of talking to 2 person while they can't hear each other ? For example when you say a sentence to person "A" and he replies back with lots of sentences how person "B" knows when he can talk if he can't hear person "A"? Person "B" starts talking while person "A" so you can't understand a word. This would only confuse people...
I am going to guess 911 callcenter where Mike is bleeding somewhere and Erin is a police dispatcher. On Thu, Jun 17, 2010 at 9:56 AM, Puskás Zsolt <errotan@elder.hu> wrote:
2010. június 17. 02.24.08 dátummal benxmy az alábbiakat írta:
Hi, I'm quite new to freeswitch and voip, so this may be in some way a noob question but I've dug through a lot of the freeswitch docs and done quite a bit of searching and haven't figured it out yet.
We're creating a relatively straightforward VoIP system to be used entirely internally (ex: users can only connect with other registered users within our system) and we'd like to be able to combine combine two calls into a single audio stream to the user without the two calls hearing each other. For example, if I'm talking to Mike on line 1 and I'm talking to Erin on line 2, is there a way for me to hear both Erin and Mike simultaneously but for them not to hear each other?
Alternatively, is there a straightforward way to simply merge the calls into a conference-type experience where we all hear each other without the user explicitly setting up a conference call?
Any and all input is greatly appreciated, as I'm up to my ears in freeswitch but have approximately zero experience with it!
Ben
Hi.
What is the point of talking to 2 person while they can't hear each other ? For example when you say a sentence to person "A" and he replies back with lots of sentences how person "B" knows when he can talk if he can't hear person "A"? Person "B" starts talking while person "A" so you can't understand a word. This would only confuse people...
_______________________________________________ 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_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
It's a good idea. 911@freeswitch.org SIP URI with $1 per minute rate :-D I want to thank you again for providing a really good software. Sometime very buggy, but overall it's very good. I will never return to asterisk. On Thursday 17 June 2010, Anthony Minessale wrote:
I am going to guess 911 callcenter where Mike is bleeding somewhere and Erin is a police dispatcher.
On Thu, Jun 17, 2010 at 9:56 AM, Puskás Zsolt <errotan@elder.hu> wrote:
2010. június 17. 02.24.08 dátummal benxmy az alábbiakat írta:
Hi, I'm quite new to freeswitch and voip, so this may be in some way a noob question but I've dug through a lot of the freeswitch docs and done quite
a
bit of searching and haven't figured it out yet.
We're creating a relatively straightforward VoIP system to be used
entirely
internally (ex: users can only connect with other registered users within our system) and we'd like to be able to combine combine two calls into a single audio stream to the user without the two calls hearing each other. For example, if I'm talking to Mike on line 1 and I'm talking to Erin on line 2, is there a way for me to hear both Erin and Mike simultaneously
but
for them not to hear each other?
Alternatively, is there a straightforward way to simply merge the calls into a conference-type experience where we all hear each other without
the
user explicitly setting up a conference call?
Any and all input is greatly appreciated, as I'm up to my ears in freeswitch but have approximately zero experience with it!
Ben
Hi.
What is the point of talking to 2 person while they can't hear each other ? For example when you say a sentence to person "A" and he replies back with lots of sentences how person "B" knows when he can talk if he can't hear person "A"? Person "B" starts talking while person "A" so you can't understand a word. This would only confuse people...
_______________________________________________ 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
What do you mean with "Sometime very buggy" ? -- I personally find FreeSWITCH to be very stable. Diego On Thu, Jun 17, 2010 at 2:24 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
It's a good idea. 911@freeswitch.org SIP URI with $1 per minute rate :-D
I want to thank you again for providing a really good software. Sometime very buggy, but overall it's very good. I will never return to asterisk.
On Thursday 17 June 2010, Anthony Minessale wrote:
I am going to guess 911 callcenter where Mike is bleeding somewhere and Erin is a police dispatcher.
On Thu, Jun 17, 2010 at 9:56 AM, Puskás Zsolt <errotan@elder.hu> wrote:
2010. június 17. 02.24.08 dátummal benxmy az alábbiakat írta:
Hi, I'm quite new to freeswitch and voip, so this may be in some way a noob question but I've dug through a lot of the freeswitch docs and done quite
a
bit of searching and haven't figured it out yet.
We're creating a relatively straightforward VoIP system to be used
entirely
internally (ex: users can only connect with other registered users within our system) and we'd like to be able to combine combine two calls into a single audio stream to the user without the two calls hearing each other. For example, if I'm talking to Mike on line 1 and I'm talking to Erin on line 2, is there a way for me to hear both Erin and Mike simultaneously
but
for them not to hear each other?
Alternatively, is there a straightforward way to simply merge the calls into a conference-type experience where we all hear each other without
the
user explicitly setting up a conference call?
Any and all input is greatly appreciated, as I'm up to my ears in freeswitch but have approximately zero experience with it!
Ben
Hi.
What is the point of talking to 2 person while they can't hear each other ? For example when you say a sentence to person "A" and he replies back with lots of sentences how person "B" knows when he can talk if he can't hear person "A"? Person "B" starts talking while person "A" so you can't understand a word. This would only confuse people...
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See bugs MODSOFIA-72 and MODSOFIA-59. Proposed patch DP-16 is a workaround for MODSOFIA-59. On Friday 18 June 2010, Diego Viola wrote:
What do you mean with "Sometime very buggy" ? -- I personally find FreeSWITCH to be very stable.
Diego
On Thu, Jun 17, 2010 at 2:24 PM, Sergey Okhapkin
<sos@sokhapkin.dyndns.org>wrote:
It's a good idea. 911@freeswitch.org SIP URI with $1 per minute rate :-D
I want to thank you again for providing a really good software. Sometime very buggy, but overall it's very good. I will never return to asterisk.
On Thursday 17 June 2010, Anthony Minessale wrote:
I am going to guess 911 callcenter where Mike is bleeding somewhere and Erin is a police dispatcher.
On Thu, Jun 17, 2010 at 9:56 AM, Puskás Zsolt <errotan@elder.hu> wrote:
2010. június 17. 02.24.08 dátummal benxmy az alábbiakat írta:
Hi, I'm quite new to freeswitch and voip, so this may be in some way a
noob
question but I've dug through a lot of the freeswitch docs and done quite
a
bit of searching and haven't figured it out yet.
We're creating a relatively straightforward VoIP system to be used
entirely
internally (ex: users can only connect with other registered users within our system) and we'd like to be able to combine combine two calls into a single audio stream to the user without the two calls hearing each other. For example, if I'm talking to Mike on line 1 and I'm talking to Erin on line 2, is there a way for me to hear both
Erin
and Mike simultaneously
but
for them not to hear each other?
Alternatively, is there a straightforward way to simply merge the
calls
into a conference-type experience where we all hear each other
without
the
user explicitly setting up a conference call?
Any and all input is greatly appreciated, as I'm up to my ears in freeswitch but have approximately zero experience with it!
Ben
Hi.
What is the point of talking to 2 person while they can't hear each
other
? For example when you say a sentence to person "A" and he replies back with lots of sentences how person "B" knows when he can talk if he
can't
hear person "A"? Person "B" starts talking while person "A" so you
can't
understand a word. This would only confuse people...
_______________________________________________ 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
_______________________________________________ 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
Lets go over your bugs. MODSOFIA-72 - You have yet to provide the offending sip packet and you said it happens rarely. MODSOFIA-59 - HAS already been fixed have you not even tried the latest git checkout? We worked on this yesterday too. DP-16 - You'll have to hunt mikej or anthony down to talk about that one. If you like I can issue you a refund if the software is that awful. /b On Jun 18, 2010, at 10:28 AM, Sergey Okhapkin wrote:
See bugs MODSOFIA-72 and MODSOFIA-59. Proposed patch DP-16 is a workaround for MODSOFIA-59.
On Friday 18 June 2010, Brian West wrote:
Lets go over your bugs.
MODSOFIA-72 - You have yet to provide the offending sip packet and you said it happens rarely.
The SIP packet is provided in the attachment to the bug.
MODSOFIA-59 - HAS already been fixed have you not even tried the latest git checkout? We worked on this yesterday too.
Good to know, will check right now.
DP-16 - You'll have to hunt mikej or anthony down to talk about that one.
If you like I can issue you a refund if the software is that awful.
:-)
/b
On Jun 18, 2010, at 10:28 AM, Sergey Okhapkin wrote:
See bugs MODSOFIA-72 and MODSOFIA-59. Proposed patch DP-16 is a workaround for MODSOFIA-59.
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Just tried latest git, the issue MODSOFIA-59 is still there, caller gets wrong media port in SDP. On Friday 18 June 2010, Brian West wrote:
Lets go over your bugs.
MODSOFIA-72 - You have yet to provide the offending sip packet and you said it happens rarely. MODSOFIA-59 - HAS already been fixed have you not even tried the latest git checkout? We worked on this yesterday too.
DP-16 - You'll have to hunt mikej or anthony down to talk about that one.
If you like I can issue you a refund if the software is that awful.
/b
On Jun 18, 2010, at 10:28 AM, Sergey Okhapkin wrote:
See bugs MODSOFIA-72 and MODSOFIA-59. Proposed patch DP-16 is a workaround for MODSOFIA-59.
_______________________________________________ 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
2 headsets On Fri, Jun 18, 2010 at 2:06 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
Just tried latest git, the issue MODSOFIA-59 is still there, caller gets wrong media port in SDP.
On Friday 18 June 2010, Brian West wrote:
Lets go over your bugs.
MODSOFIA-72 - You have yet to provide the offending sip packet and you said it happens rarely. MODSOFIA-59 - HAS already been fixed have you not even tried the latest git checkout? We worked on this yesterday too.
DP-16 - You'll have to hunt mikej or anthony down to talk about that one.
If you like I can issue you a refund if the software is that awful.
/b
On Jun 18, 2010, at 10:28 AM, Sergey Okhapkin wrote:
See bugs MODSOFIA-72 and MODSOFIA-59. Proposed patch DP-16 is a workaround for MODSOFIA-59.
_______________________________________________ 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
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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_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
please stop trolling on this list or face moderation. On Fri, Jun 18, 2010 at 2:06 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
Just tried latest git, the issue MODSOFIA-59 is still there, caller gets wrong media port in SDP.
On Friday 18 June 2010, Brian West wrote:
Lets go over your bugs.
MODSOFIA-72 - You have yet to provide the offending sip packet and you said it happens rarely. MODSOFIA-59 - HAS already been fixed have you not even tried the latest git checkout? We worked on this yesterday too.
DP-16 - You'll have to hunt mikej or anthony down to talk about that one.
If you like I can issue you a refund if the software is that awful.
/b
On Jun 18, 2010, at 10:28 AM, Sergey Okhapkin wrote:
See bugs MODSOFIA-72 and MODSOFIA-59. Proposed patch DP-16 is a workaround for MODSOFIA-59.
_______________________________________________ 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
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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_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
MODSOFIA-59 is not our bug as far as I can tell. You are using bypass media.. I'm pretty sure we just pass those SDP's thru as we receive them from one side to the other. I double checked and that appears to be what is taking place so the bug is not in FreeSWITCH as it doesn't even allocate ports or rtp for bypass calls. /b On Jun 18, 2010, at 2:06 PM, Sergey Okhapkin wrote:
Just tried latest git, the issue MODSOFIA-59 is still there, caller gets wrong media port in SDP.
On Friday 18 June 2010, Brian West wrote:
Lets go over your bugs.
MODSOFIA-72 - You have yet to provide the offending sip packet and you said it happens rarely. MODSOFIA-59 - HAS already been fixed have you not even tried the latest git checkout? We worked on this yesterday too.
DP-16 - You'll have to hunt mikej or anthony down to talk about that one.
If you like I can issue you a refund if the software is that awful.
/b
I see what you're doing... and I know why its doing it. Wish your descriptions and logs were as clear as when Anthony explained to me what exactly was going on. /b On Jun 18, 2010, at 2:06 PM, Sergey Okhapkin wrote:
Just tried latest git, the issue MODSOFIA-59 is still there, caller gets wrong media port in SDP.
On Friday 18 June 2010, Brian West wrote:
Lets go over your bugs.
MODSOFIA-72 - You have yet to provide the offending sip packet and you said it happens rarely. MODSOFIA-59 - HAS already been fixed have you not even tried the latest git checkout? We worked on this yesterday too.
DP-16 - You'll have to hunt mikej or anthony down to talk about that one.
If you like I can issue you a refund if the software is that awful.
/b
It seems that the original subject of the thread pretty quickly disappeared... Does anyone have any suggestions involving a user listening to two independent lines at once while both those lines can hear the user but not each other? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Merge-two-calls-tp5188763p5196... Sent from the freeswitch-users mailing list archive at Nabble.com.
also you could put them in a conference and use the relationships to make them not able to hear each other. On Fri, Jun 18, 2010 at 2:12 PM, benxmy <benxmy@gmail.com> wrote:
It seems that the original subject of the thread pretty quickly disappeared...
Does anyone have any suggestions involving a user listening to two independent lines at once while both those lines can hear the user but not each other? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Merge-two-calls-tp5188763p5196... Sent from the freeswitch-users mailing list archive at Nabble.com.
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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_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Not really good choice of words. "Very buggy" would indicate that the software keeps crashing and authors don't care at all. That is miles far from the real experience imho. On Thu, Jun 17, 2010 at 7:24 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
It's a good idea. 911@freeswitch.org SIP URI with $1 per minute rate :-D
I want to thank you again for providing a really good software. Sometime very buggy, but overall it's very good. I will never return to asterisk.
On Thursday 17 June 2010, Anthony Minessale wrote:
I am going to guess 911 callcenter where Mike is bleeding somewhere and Erin is a police dispatcher.
On Thu, Jun 17, 2010 at 9:56 AM, Puskás Zsolt <errotan@elder.hu> wrote:
2010. június 17. 02.24.08 dátummal benxmy az alábbiakat írta:
Hi, I'm quite new to freeswitch and voip, so this may be in some way a noob question but I've dug through a lot of the freeswitch docs and done quite
a
bit of searching and haven't figured it out yet.
We're creating a relatively straightforward VoIP system to be used
entirely
internally (ex: users can only connect with other registered users within our system) and we'd like to be able to combine combine two calls into a single audio stream to the user without the two calls hearing each other. For example, if I'm talking to Mike on line 1 and I'm talking to Erin on line 2, is there a way for me to hear both Erin and Mike simultaneously
but
for them not to hear each other?
Alternatively, is there a straightforward way to simply merge the calls into a conference-type experience where we all hear each other without
the
user explicitly setting up a conference call?
Any and all input is greatly appreciated, as I'm up to my ears in freeswitch but have approximately zero experience with it!
Ben
Hi.
What is the point of talking to 2 person while they can't hear each other ? For example when you say a sentence to person "A" and he replies back with lots of sentences how person "B" knows when he can talk if he can't hear person "A"? Person "B" starts talking while person "A" so you can't understand a word. This would only confuse people...
_______________________________________________ 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
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participants (7)
-
Anthony Minessale
-
benxmy
-
Brian West
-
Diego Viola
-
Ognjen Seslija
-
Puskás Zsolt
-
Sergey Okhapkin