LUA session:Bridge not actually bridging calls ~
Hello, I wanted to make a lua script which just dials out two different numbers via some external gateway and when both calls are answered they are just bridged. For this a very impressive Lua example http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is copied and all I had to do was change the dialA and dialB strings and its working great as far as the SIP signalling is concerned. execute this string and I get calls on two different number but things get interesting when Freeswitch bridge() the two legs. No AUDIO..not even one-way. I could see on my own gateway that RTPs for both the legs are actually forwarded to Freeswitch ! On my sip pcap traces analyzing on wireshark I could actually hear the two persons saying Hello but neither could hear anything. The above example lua call_control script is used as it is. Please suggest. Regards Sammy Go.
Here are the FS console logs: http://pastebin.freeswitch.org/19595 Please suggest what am I missing here. On Sat, Jul 28, 2012 at 7:59 PM, SamyGo <govoiper@gmail.com> wrote:
Hello, I wanted to make a lua script which just dials out two different numbers via some external gateway and when both calls are answered they are just bridged. For this a very impressive Lua example http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is copied and all I had to do was change the dialA and dialB strings and its working great as far as the SIP signalling is concerned.
execute this string and I get calls on two different number but things get interesting when Freeswitch bridge() the two legs. No AUDIO..not even one-way. I could see on my own gateway that RTPs for both the legs are actually forwarded to Freeswitch !
On my sip pcap traces analyzing on wireshark I could actually hear the two persons saying Hello but neither could hear anything.
The above example lua call_control script is used as it is. Please suggest.
Regards Sammy Go.
Hi again, So, It was a very minor change in configuration and it was working. Basically FreeSwicth was bridging the two legs BUT there was a codec issue. All I had to do was in Asterisk (serving as my gateway) to allow only ulaw and alaw i.e disallow=all allow=ulaw allow=alaw What I really really wish to know is that why there was no indication of codec mismatch or ptime mismatch or sample rate mismatch while transcoding or anything. It will be fine if none replies but it will be great to know the real reason behind this and from where in logs can I verify this !! Thanks Sammy On Sat, Jul 28, 2012 at 8:18 PM, SamyGo <govoiper@gmail.com> wrote:
Here are the FS console logs: http://pastebin.freeswitch.org/19595
Please suggest what am I missing here.
On Sat, Jul 28, 2012 at 7:59 PM, SamyGo <govoiper@gmail.com> wrote:
Hello, I wanted to make a lua script which just dials out two different numbers via some external gateway and when both calls are answered they are just bridged. For this a very impressive Lua example http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is copied and all I had to do was change the dialA and dialB strings and its working great as far as the SIP signalling is concerned.
execute this string and I get calls on two different number but things get interesting when Freeswitch bridge() the two legs. No AUDIO..not even one-way. I could see on my own gateway that RTPs for both the legs are actually forwarded to Freeswitch !
On my sip pcap traces analyzing on wireshark I could actually hear the two persons saying Hello but neither could hear anything.
The above example lua call_control script is used as it is. Please suggest.
Regards Sammy Go.
This script is actually a bit over-complicated dialA = "sofia/gateway/fs1/9903" dialB = "user/1001" legA = freeswitch.Session(dialA) if (legA:ready()) { legA:execute("bridge", dialB) } I think the problem is that we don't support dealing with many codecs in the 200ok They send gsm and ulaw and then choose ulaw and we go with the first one gsm On Sat, Jul 28, 2012 at 11:23 AM, SamyGo <govoiper@gmail.com> wrote:
Hi again,
So, It was a very minor change in configuration and it was working. Basically FreeSwicth was bridging the two legs BUT there was a codec issue. All I had to do was in Asterisk (serving as my gateway) to allow only ulaw and alaw i.e
disallow=all allow=ulaw allow=alaw
What I really really wish to know is that why there was no indication of codec mismatch or ptime mismatch or sample rate mismatch while transcoding or anything.
It will be fine if none replies but it will be great to know the real reason behind this and from where in logs can I verify this !!
Thanks Sammy
On Sat, Jul 28, 2012 at 8:18 PM, SamyGo <govoiper@gmail.com> wrote:
Here are the FS console logs: http://pastebin.freeswitch.org/19595
Please suggest what am I missing here.
On Sat, Jul 28, 2012 at 7:59 PM, SamyGo <govoiper@gmail.com> wrote:
Hello, I wanted to make a lua script which just dials out two different numbers via some external gateway and when both calls are answered they are just bridged. For this a very impressive Lua example http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is copied and all I had to do was change the dialA and dialB strings and its working great as far as the SIP signalling is concerned.
execute this string and I get calls on two different number but things get interesting when Freeswitch bridge() the two legs. No AUDIO..not even one-way. I could see on my own gateway that RTPs for both the legs are actually forwarded to Freeswitch !
On my sip pcap traces analyzing on wireshark I could actually hear the two persons saying Hello but neither could hear anything.
The above example lua call_control script is used as it is. Please suggest.
Regards Sammy Go.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Though the problem is solved but is there any corresponding WARN or anything in the logs so that anyone can pick the problem right away. This fixing of codecs took alot of time and I was never expecting that this could be the cause Thinking that its Bridge() which is not doing its part. Thanks a lot for your time on this. Regards, Sammy On Mon, Jul 30, 2012 at 6:22 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
This script is actually a bit over-complicated
dialA = "sofia/gateway/fs1/9903" dialB = "user/1001" legA = freeswitch.Session(dialA)
if (legA:ready()) { legA:execute("bridge", dialB) }
I think the problem is that we don't support dealing with many codecs in the 200ok They send gsm and ulaw and then choose ulaw and we go with the first one gsm
On Sat, Jul 28, 2012 at 11:23 AM, SamyGo <govoiper@gmail.com> wrote:
Hi again,
So, It was a very minor change in configuration and it was working. Basically FreeSwicth was bridging the two legs BUT there was a codec issue. All I had to do was in Asterisk (serving as my gateway) to allow only ulaw and alaw i.e
disallow=all allow=ulaw allow=alaw
What I really really wish to know is that why there was no indication of codec mismatch or ptime mismatch or sample rate mismatch while transcoding or anything.
It will be fine if none replies but it will be great to know the real reason behind this and from where in logs can I verify this !!
Thanks Sammy
On Sat, Jul 28, 2012 at 8:18 PM, SamyGo <govoiper@gmail.com> wrote:
Here are the FS console logs: http://pastebin.freeswitch.org/19595
Please suggest what am I missing here.
On Sat, Jul 28, 2012 at 7:59 PM, SamyGo <govoiper@gmail.com> wrote:
Hello, I wanted to make a lua script which just dials out two different
numbers
via some external gateway and when both calls are answered they are just bridged. For this a very impressive Lua example http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is copied and all I had to do was change the dialA and dialB strings and its working great as far as the SIP signalling is concerned.
execute this string and I get calls on two different number but things get interesting when Freeswitch bridge() the two legs. No AUDIO..not even one-way. I could see on my own gateway that RTPs for both the legs are actually forwarded to Freeswitch !
On my sip pcap traces analyzing on wireshark I could actually hear the two persons saying Hello but neither could hear anything.
The above example lua call_control script is used as it is. Please suggest.
Regards Sammy Go.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
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
Not at the moment. If everyone is going to start doing this we may have to change something. On Jul 31, 2012 12:41 AM, "SamyGo" <govoiper@gmail.com> wrote:
Though the problem is solved but is there any corresponding WARN or anything in the logs so that anyone can pick the problem right away. This fixing of codecs took alot of time and I was never expecting that this could be the cause Thinking that its Bridge() which is not doing its part.
Thanks a lot for your time on this.
Regards, Sammy
On Mon, Jul 30, 2012 at 6:22 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
This script is actually a bit over-complicated
dialA = "sofia/gateway/fs1/9903" dialB = "user/1001" legA = freeswitch.Session(dialA)
if (legA:ready()) { legA:execute("bridge", dialB) }
I think the problem is that we don't support dealing with many codecs in the 200ok They send gsm and ulaw and then choose ulaw and we go with the first one gsm
On Sat, Jul 28, 2012 at 11:23 AM, SamyGo <govoiper@gmail.com> wrote:
Hi again,
So, It was a very minor change in configuration and it was working. Basically FreeSwicth was bridging the two legs BUT there was a codec issue. All I had to do was in Asterisk (serving as my gateway) to allow only ulaw and alaw i.e
disallow=all allow=ulaw allow=alaw
What I really really wish to know is that why there was no indication of codec mismatch or ptime mismatch or sample rate mismatch while transcoding or anything.
It will be fine if none replies but it will be great to know the real reason behind this and from where in logs can I verify this !!
Thanks Sammy
On Sat, Jul 28, 2012 at 8:18 PM, SamyGo <govoiper@gmail.com> wrote:
Here are the FS console logs: http://pastebin.freeswitch.org/19595
Please suggest what am I missing here.
On Sat, Jul 28, 2012 at 7:59 PM, SamyGo <govoiper@gmail.com> wrote:
Hello, I wanted to make a lua script which just dials out two different
numbers
via some external gateway and when both calls are answered they are just bridged. For this a very impressive Lua example http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is copied and all I had to do was change the dialA and dialB strings and its working great as far as the SIP signalling is concerned.
execute this string and I get calls on two different number but things get interesting when Freeswitch bridge() the two legs. No AUDIO..not even one-way. I could see on my own gateway that RTPs for both the legs are actually forwarded to Freeswitch !
On my sip pcap traces analyzing on wireshark I could actually hear the two persons saying Hello but neither could hear anything.
The above example lua call_control script is used as it is. Please suggest.
Regards Sammy Go.
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
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
Thats understandable. Im cool with it, afterall its working. :) On Jul 31, 2012 9:40 AM, "SamyGo" <govoiper@gmail.com> wrote:
Though the problem is solved but is there any corresponding WARN or anything in the logs so that anyone can pick the problem right away. This fixing of codecs took alot of time and I was never expecting that this could be the cause Thinking that its Bridge() which is not doing its part.
Thanks a lot for your time on this.
Regards, Sammy
On Mon, Jul 30, 2012 at 6:22 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
This script is actually a bit over-complicated
dialA = "sofia/gateway/fs1/9903" dialB = "user/1001" legA = freeswitch.Session(dialA)
if (legA:ready()) { legA:execute("bridge", dialB) }
I think the problem is that we don't support dealing with many codecs in the 200ok They send gsm and ulaw and then choose ulaw and we go with the first one gsm
On Sat, Jul 28, 2012 at 11:23 AM, SamyGo <govoiper@gmail.com> wrote:
Hi again,
So, It was a very minor change in configuration and it was working. Basically FreeSwicth was bridging the two legs BUT there was a codec issue. All I had to do was in Asterisk (serving as my gateway) to allow only ulaw and alaw i.e
disallow=all allow=ulaw allow=alaw
What I really really wish to know is that why there was no indication of codec mismatch or ptime mismatch or sample rate mismatch while transcoding or anything.
It will be fine if none replies but it will be great to know the real reason behind this and from where in logs can I verify this !!
Thanks Sammy
On Sat, Jul 28, 2012 at 8:18 PM, SamyGo <govoiper@gmail.com> wrote:
Here are the FS console logs: http://pastebin.freeswitch.org/19595
Please suggest what am I missing here.
On Sat, Jul 28, 2012 at 7:59 PM, SamyGo <govoiper@gmail.com> wrote:
Hello, I wanted to make a lua script which just dials out two different
numbers
via some external gateway and when both calls are answered they are just bridged. For this a very impressive Lua example http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is copied and all I had to do was change the dialA and dialB strings and its working great as far as the SIP signalling is concerned.
execute this string and I get calls on two different number but things get interesting when Freeswitch bridge() the two legs. No AUDIO..not even one-way. I could see on my own gateway that RTPs for both the legs are actually forwarded to Freeswitch !
On my sip pcap traces analyzing on wireshark I could actually hear the two persons saying Hello but neither could hear anything.
The above example lua call_control script is used as it is. Please suggest.
Regards Sammy Go.
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
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
participants (2)
-
Anthony Minessale
-
SamyGo