Loss of first second of media
I've been trying out a few new termination providers, and am finding that the first second of media is being lost. I get early media (ringing), then when the call is answered the first half a second to a second is lost, then the rest of the call is fine. The engineers at one of the providers ran a sip trace and a full packet trace, and it looked like everything was being delivered from the pstn to my server. They said that none of their other customers have reported this problem before. I've tested connecting to pstn's in several different countries, with the same results. I've used a couple of other termination providers in the past (1 still in current use) which have been fine, but now have 2 that are losing the first part of the media. So it seems like it's specific to Freeswitch, with some subset of conditions for some providers. This is running on Ubuntu on a Amazon AWS server, in case that helps. Any ideas how to narrow it down further? Cheers, Fraser
Where is the other side of this call coming from? [ ( ) ] -> FS -> (PSTN via SIP) What goes in the empty space above? On Thu, Oct 21, 2010 at 10:10 AM, Fraser Redmond <fraserredmond@gmail.com> wrote:
I've been trying out a few new termination providers, and am finding that the first second of media is being lost.
I get early media (ringing), then when the call is answered the first half a second to a second is lost, then the rest of the call is fine.
The engineers at one of the providers ran a sip trace and a full packet trace, and it looked like everything was being delivered from the pstn to my server. They said that none of their other customers have reported this problem before.
I've tested connecting to pstn's in several different countries, with the same results. I've used a couple of other termination providers in the past (1 still in current use) which have been fine, but now have 2 that are losing the first part of the media.
So it seems like it's specific to Freeswitch, with some subset of conditions for some providers.
This is running on Ubuntu on a Amazon AWS server, in case that helps.
Any ideas how to narrow it down further?
Cheers, Fraser
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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 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
The call is originated from Freeswitch (via CLI) to a softphone, then when that is connected it bridges out to the gateway. Cheers, Fraser On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Where is the other side of this call coming from?
[ ( ) ] -> FS -> (PSTN via SIP)
What goes in the empty space above?
how are you accomplishing that? by which technique? On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond <fraserredmond@gmail.com> wrote:
The call is originated from Freeswitch (via CLI) to a softphone, then when that is connected it bridges out to the gateway.
Cheers, Fraser
On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Where is the other side of this call coming from?
[ ( ) ] -> FS -> (PSTN via SIP)
What goes in the empty space above?
_______________________________________________ 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
event_socket: api originate {vars=values}user/$fromExtn@Domain'set:bLegVars=values,transfer:$toNum xml outbound_call' inline then dialplan: <action application="set" data="effective_caller_id_number=+1800number"/> <action application="set" data="effective_caller_id_name="/> (set and/or export a bunch of other vars too) <action application="set" data="dial_string=sofia/gateway/ gatewayname.com/00${destination_number}"/> <action application="bridge" data="${dial_string}"/> Cheers, Fraser On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
how are you accomplishing that? by which technique?
On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond <fraserredmond@gmail.com> wrote:
The call is originated from Freeswitch (via CLI) to a softphone, then when that is connected it bridges out to the gateway.
Cheers, Fraser
On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Where is the other side of this call coming from?
[ ( ) ] -> FS -> (PSTN via SIP)
What goes in the empty space above?
are you setting ignore_early_media=true in the first vars=values area? This looks like you could be calling one leg who is still not answered and then bridging it to another dest. The bridge app will wait for the first leg to answer before bridging. Also if you have any NAT anywhere, look for an "auto-changing port" type message which can also be attributed to this due to a detection period for incorrect ports. On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
event_socket: api originate {vars=values}user/$fromExtn@Domain 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
then
dialplan: <action application="set" data="effective_caller_id_number=+1800number"/> <action application="set" data="effective_caller_id_name="/> (set and/or export a bunch of other vars too) <action application="set" data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}"/> <action application="bridge" data="${dial_string}"/>
Cheers, Fraser
On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
how are you accomplishing that? by which technique?
On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond <fraserredmond@gmail.com> wrote:
The call is originated from Freeswitch (via CLI) to a softphone, then when that is connected it bridges out to the gateway.
Cheers, Fraser
On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Where is the other side of this call coming from?
[ ( ) ] -> FS -> (PSTN via SIP)
What goes in the empty space above?
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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 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
Sorry, yes, I am setting ignore_early_media=true in the first area. (Or are you saying that should be off? I forget now why I needed it on, but there was a reason I added it.) Yes, the bridge doesn't start until after the A-leg has answered. Thanks for the suggestion about nat/auto-changing port, I'll have a look into that - would that be in the cli output or in a sip trace? I've already looked and it's not appearing in the CLI output (with loglevel=debug), haven't looked in the sip trace yet. Cheers, Fraser On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
are you setting ignore_early_media=true in the first vars=values area?
This looks like you could be calling one leg who is still not answered and then bridging it to another dest. The bridge app will wait for the first leg to answer before bridging.
Also if you have any NAT anywhere, look for an "auto-changing port" type message which can also be attributed to this due to a detection period for incorrect ports.
On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
event_socket: api originate {vars=values}user/$fromExtn@Domain 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
then
dialplan: <action application="set" data="effective_caller_id_number=+1800number"/> <action application="set" data="effective_caller_id_name="/> (set and/or export a bunch of other vars too) <action application="set" data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D> "/> <action application="bridge" data="${dial_string}"/>
Cheers, Fraser
On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
how are you accomplishing that? by which technique?
On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond <fraserredmond@gmail.com> wrote:
The call is originated from Freeswitch (via CLI) to a softphone, then when that is connected it bridges out to the gateway.
Cheers, Fraser
On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Where is the other side of this call coming from?
[ ( ) ] -> FS -> (PSTN via SIP)
What goes in the empty space above?
_______________________________________________ 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
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its a blue message on cli It could also be the other side expecting us to send media first or something silly. try getting a sip trace and figure out when the rtp starts arriving. On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Sorry, yes, I am setting ignore_early_media=true in the first area. (Or are you saying that should be off? I forget now why I needed it on, but there was a reason I added it.)
Yes, the bridge doesn't start until after the A-leg has answered.
Thanks for the suggestion about nat/auto-changing port, I'll have a look into that - would that be in the cli output or in a sip trace? I've already looked and it's not appearing in the CLI output (with loglevel=debug), haven't looked in the sip trace yet.
Cheers, Fraser
On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
are you setting ignore_early_media=true in the first vars=values area?
This looks like you could be calling one leg who is still not answered and then bridging it to another dest. The bridge app will wait for the first leg to answer before bridging.
Also if you have any NAT anywhere, look for an "auto-changing port" type message which can also be attributed to this due to a detection period for incorrect ports.
On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
event_socket: api originate {vars=values}user/$fromExtn@Domain 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
then
dialplan: <action application="set" data="effective_caller_id_number=+1800number"/> <action application="set" data="effective_caller_id_name="/> (set and/or export a bunch of other vars too) <action application="set"
data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}"/> <action application="bridge" data="${dial_string}"/>
Cheers, Fraser
On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
how are you accomplishing that? by which technique?
On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond <fraserredmond@gmail.com> wrote:
The call is originated from Freeswitch (via CLI) to a softphone, then when that is connected it bridges out to the gateway.
Cheers, Fraser
On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Where is the other side of this call coming from?
[ ( ) ] -> FS -> (PSTN via SIP)
What goes in the empty space above?
_______________________________________________ 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
_______________________________________________ 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 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
Thanks Anthony, Finally managed to get a sip trace - could you do me a favor and take a look and/or give me some ideas of what to look for? http://pastebin.freeswitch.org/14300 I've highlighted lines 168 and 193. In between these lines is where the number is dialed and rings once, then picks up, then theres silence for a second or two, and that second SIP message is when I start hearing audio. Thanks, Fraser On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
its a blue message on cli
It could also be the other side expecting us to send media first or something silly. try getting a sip trace and figure out when the rtp starts arriving.
On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Sorry, yes, I am setting ignore_early_media=true in the first area. (Or are you saying that should be off? I forget now why I needed it on, but there was a reason I added it.)
Yes, the bridge doesn't start until after the A-leg has answered.
Thanks for the suggestion about nat/auto-changing port, I'll have a look into that - would that be in the cli output or in a sip trace? I've already looked and it's not appearing in the CLI output (with loglevel=debug), haven't looked in the sip trace yet.
Cheers, Fraser
On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
are you setting ignore_early_media=true in the first vars=values area?
This looks like you could be calling one leg who is still not answered and then bridging it to another dest. The bridge app will wait for the first leg to answer before bridging.
Also if you have any NAT anywhere, look for an "auto-changing port" type message which can also be attributed to this due to a detection period for incorrect ports.
On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
event_socket: api originate {vars=values}user/$fromExtn@Domain 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
then
dialplan: <action application="set" data="effective_caller_id_number=+1800number"/> <action application="set" data="effective_caller_id_name="/> (set and/or export a bunch of other vars too) <action application="set"
data="dial_string=sofia/gateway/
gatewayname.com/00${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D> "/>
<action application="bridge" data="${dial_string}"/>
Cheers, Fraser
On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
how are you accomplishing that? by which technique?
On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond <fraserredmond@gmail.com> wrote:
The call is originated from Freeswitch (via CLI) to a softphone,
then
when that is connected it bridges out to the gateway.
Cheers, Fraser
On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: > > Where is the other side of this call coming from? > > [ ( ) ] -> FS -> (PSTN via SIP) > > What goes in the empty space above?
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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
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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
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Can you do this trace with debug level logging in addition to the sip trace console loglevel debug you also may want to get a pcap of it tshark udp and port 5060 -w test.pcap On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks Anthony,
Finally managed to get a sip trace - could you do me a favor and take a look and/or give me some ideas of what to look for?
http://pastebin.freeswitch.org/14300
I've highlighted lines 168 and 193. In between these lines is where the number is dialed and rings once, then picks up, then theres silence for a second or two, and that second SIP message is when I start hearing audio.
Thanks, Fraser
On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
its a blue message on cli
It could also be the other side expecting us to send media first or something silly. try getting a sip trace and figure out when the rtp starts arriving.
On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Sorry, yes, I am setting ignore_early_media=true in the first area. (Or are you saying that should be off? I forget now why I needed it on, but there was a reason I added it.)
Yes, the bridge doesn't start until after the A-leg has answered.
Thanks for the suggestion about nat/auto-changing port, I'll have a look into that - would that be in the cli output or in a sip trace? I've already looked and it's not appearing in the CLI output (with loglevel=debug), haven't looked in the sip trace yet.
Cheers, Fraser
On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
are you setting ignore_early_media=true in the first vars=values area?
This looks like you could be calling one leg who is still not answered and then bridging it to another dest. The bridge app will wait for the first leg to answer before bridging.
Also if you have any NAT anywhere, look for an "auto-changing port" type message which can also be attributed to this due to a detection period for incorrect ports.
On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
event_socket: api originate {vars=values}user/$fromExtn@Domain 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
then
dialplan: <action application="set" data="effective_caller_id_number=+1800number"/> <action application="set" data="effective_caller_id_name="/> (set and/or export a bunch of other vars too) <action application="set"
data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}"/> <action application="bridge" data="${dial_string}"/>
Cheers, Fraser
On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
how are you accomplishing that? by which technique?
On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond <fraserredmond@gmail.com> wrote: > The call is originated from Freeswitch (via CLI) to a softphone, > then > when > that is connected it bridges out to the gateway. > > Cheers, > Fraser > > > > > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale > <anthony.minessale@gmail.com> wrote: >> >> Where is the other side of this call coming from? >> >> [ ( ) ] -> FS -> (PSTN via SIP) >> >> What goes in the empty space above?
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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 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
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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 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
_______________________________________________ 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 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
Thanks Anthony, it's here: http://pastebin.freeswitch.org/14350 And pcap is attached. The call connects around (or just before) the 16:58:35 mark (line 558 is what I see in the terminal while waiting for it to connect - both early-media and the missing start of the media) Cheers, Fraser On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Can you do this trace with debug level logging in addition to the sip trace console loglevel debug
you also may want to get a pcap of it
tshark udp and port 5060 -w test.pcap
On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks Anthony,
Finally managed to get a sip trace - could you do me a favor and take a look and/or give me some ideas of what to look for?
http://pastebin.freeswitch.org/14300
I've highlighted lines 168 and 193. In between these lines is where the number is dialed and rings once, then picks up, then theres silence for a second or two, and that second SIP message is when I start hearing audio.
Thanks, Fraser
On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
its a blue message on cli
It could also be the other side expecting us to send media first or something silly. try getting a sip trace and figure out when the rtp starts arriving.
On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Sorry, yes, I am setting ignore_early_media=true in the first area.
(Or
are you saying that should be off? I forget now why I needed it on, but there was a reason I added it.)
Yes, the bridge doesn't start until after the A-leg has answered.
Thanks for the suggestion about nat/auto-changing port, I'll have a look into that - would that be in the cli output or in a sip trace? I've already looked and it's not appearing in the CLI output (with loglevel=debug), haven't looked in the sip trace yet.
Cheers, Fraser
On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
are you setting ignore_early_media=true in the first vars=values
area?
This looks like you could be calling one leg who is still not
answered
and then bridging it to another dest. The bridge app will wait for the first leg to answer before bridging.
Also if you have any NAT anywhere, look for an "auto-changing port" type message which can also be attributed to this due to a detection period for incorrect ports.
On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
event_socket: api originate {vars=values}user/$fromExtn@Domain 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
then
dialplan: <action application="set" data="effective_caller_id_number=+1800number"/> <action application="set" data="effective_caller_id_name="/> (set and/or export a bunch of other vars too) <action application="set"
data="dial_string=sofia/gateway/ gatewayname.com/00${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D> "/> <action application="bridge" data="${dial_string}"/>
Cheers, Fraser
On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: > > how are you accomplishing that? by which technique? > > On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond > <fraserredmond@gmail.com> wrote: > > The call is originated from Freeswitch (via CLI) to a softphone, > > then > > when > > that is connected it bridges out to the gateway. > > > > Cheers, > > Fraser > > > > > > > > > > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale > > <anthony.minessale@gmail.com> wrote: > >> > >> Where is the other side of this call coming from? > >> > >> [ ( ) ] -> FS -> (PSTN via SIP) > >> > >> What goes in the empty space above?
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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
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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
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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
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can you try another one with just udp and not "port 5060" so I can see the rtp too On Fri, Oct 29, 2010 at 12:28 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks Anthony, it's here: http://pastebin.freeswitch.org/14350
And pcap is attached.
The call connects around (or just before) the 16:58:35 mark (line 558 is what I see in the terminal while waiting for it to connect - both early-media and the missing start of the media)
Cheers, Fraser
On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Can you do this trace with debug level logging in addition to the sip trace console loglevel debug
you also may want to get a pcap of it
tshark udp and port 5060 -w test.pcap
On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks Anthony,
Finally managed to get a sip trace - could you do me a favor and take a look and/or give me some ideas of what to look for?
http://pastebin.freeswitch.org/14300
I've highlighted lines 168 and 193. In between these lines is where the number is dialed and rings once, then picks up, then theres silence for a second or two, and that second SIP message is when I start hearing audio.
Thanks, Fraser
On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
its a blue message on cli
It could also be the other side expecting us to send media first or something silly. try getting a sip trace and figure out when the rtp starts arriving.
On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Sorry, yes, I am setting ignore_early_media=true in the first area. (Or are you saying that should be off? I forget now why I needed it on, but there was a reason I added it.)
Yes, the bridge doesn't start until after the A-leg has answered.
Thanks for the suggestion about nat/auto-changing port, I'll have a look into that - would that be in the cli output or in a sip trace? I've already looked and it's not appearing in the CLI output (with loglevel=debug), haven't looked in the sip trace yet.
Cheers, Fraser
On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
are you setting ignore_early_media=true in the first vars=values area?
This looks like you could be calling one leg who is still not answered and then bridging it to another dest. The bridge app will wait for the first leg to answer before bridging.
Also if you have any NAT anywhere, look for an "auto-changing port" type message which can also be attributed to this due to a detection period for incorrect ports.
On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond <fraserredmond@gmail.com> wrote: > event_socket: > api originate {vars=values}user/$fromExtn@Domain > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline > > then > > dialplan: > <action application="set" > data="effective_caller_id_number=+1800number"/> > <action application="set" data="effective_caller_id_name="/> > (set and/or export a bunch of other vars too) > <action application="set" > > > > data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}"/> > <action application="bridge" data="${dial_string}"/> > > Cheers, > Fraser > > > > > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale > <anthony.minessale@gmail.com> wrote: >> >> how are you accomplishing that? by which technique? >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond >> <fraserredmond@gmail.com> wrote: >> > The call is originated from Freeswitch (via CLI) to a >> > softphone, >> > then >> > when >> > that is connected it bridges out to the gateway. >> > >> > Cheers, >> > Fraser >> > >> > >> > >> > >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale >> > <anthony.minessale@gmail.com> wrote: >> >> >> >> Where is the other side of this call coming from? >> >> >> >> [ ( ) ] -> FS -> (PSTN via SIP) >> >> >> >> What goes in the empty space above? > > > _______________________________________________ > 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
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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
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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 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
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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 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
OK so, The phone sends a 180 ringing with NO SDP then it starts sending RTP That's is not right. It's a bug in the phone. On Fri, Oct 29, 2010 at 1:35 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you try another one with just udp and not "port 5060" so I can see the rtp too
On Fri, Oct 29, 2010 at 12:28 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks Anthony, it's here: http://pastebin.freeswitch.org/14350
And pcap is attached.
The call connects around (or just before) the 16:58:35 mark (line 558 is what I see in the terminal while waiting for it to connect - both early-media and the missing start of the media)
Cheers, Fraser
On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Can you do this trace with debug level logging in addition to the sip trace console loglevel debug
you also may want to get a pcap of it
tshark udp and port 5060 -w test.pcap
On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks Anthony,
Finally managed to get a sip trace - could you do me a favor and take a look and/or give me some ideas of what to look for?
http://pastebin.freeswitch.org/14300
I've highlighted lines 168 and 193. In between these lines is where the number is dialed and rings once, then picks up, then theres silence for a second or two, and that second SIP message is when I start hearing audio.
Thanks, Fraser
On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
its a blue message on cli
It could also be the other side expecting us to send media first or something silly. try getting a sip trace and figure out when the rtp starts arriving.
On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Sorry, yes, I am setting ignore_early_media=true in the first area. (Or are you saying that should be off? I forget now why I needed it on, but there was a reason I added it.)
Yes, the bridge doesn't start until after the A-leg has answered.
Thanks for the suggestion about nat/auto-changing port, I'll have a look into that - would that be in the cli output or in a sip trace? I've already looked and it's not appearing in the CLI output (with loglevel=debug), haven't looked in the sip trace yet.
Cheers, Fraser
On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: > > are you setting ignore_early_media=true in the first vars=values > area? > > This looks like you could be calling one leg who is still not > answered > and then bridging it to another dest. The bridge app will wait for > the first leg to answer before bridging. > > Also if you have any NAT anywhere, look for an "auto-changing port" > type message which can also be attributed to this due to a detection > period for incorrect ports. > > > > On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond > <fraserredmond@gmail.com> wrote: > > event_socket: > > api originate {vars=values}user/$fromExtn@Domain > > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline > > > > then > > > > dialplan: > > <action application="set" > > data="effective_caller_id_number=+1800number"/> > > <action application="set" data="effective_caller_id_name="/> > > (set and/or export a bunch of other vars too) > > <action application="set" > > > > > > > > data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}"/> > > <action application="bridge" data="${dial_string}"/> > > > > Cheers, > > Fraser > > > > > > > > > > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale > > <anthony.minessale@gmail.com> wrote: > >> > >> how are you accomplishing that? by which technique? > >> > >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond > >> <fraserredmond@gmail.com> wrote: > >> > The call is originated from Freeswitch (via CLI) to a > >> > softphone, > >> > then > >> > when > >> > that is connected it bridges out to the gateway. > >> > > >> > Cheers, > >> > Fraser > >> > > >> > > >> > > >> > > >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale > >> > <anthony.minessale@gmail.com> wrote: > >> >> > >> >> Where is the other side of this call coming from? > >> >> > >> >> [ ( ) ] -> FS -> (PSTN via SIP) > >> >> > >> >> What goes in the empty space above? > > > > > > _______________________________________________ > > 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 > > _______________________________________________ > 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 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
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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 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
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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 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
-- 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
Thanks, but that doesn't fit - I've tried it with Bria, and with the recent version of XLite 4, and its happened on both. They're both made by Counterpath, so I've also tried 3CXphone, and it's happening on that too. More suspicious though is that it only happens with some gateways, and not others. Or did you mean that you think it's a bug with the gateway's systems? The two I've tried that have this problem are both big outfits, not some guy operating out of his garage :-) Also, both were already listed in the Gateways on the wiki, so I'm guessing someone else is already using them with Freeswitch. My server is running on Amazon AWS - could it be a timing/virtualization type problem, or something like that? Any other ideas? Cheers, Fraser On Fri, Oct 29, 2010 at 9:39 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
OK so, The phone sends a 180 ringing with NO SDP then it starts sending RTP That's is not right. It's a bug in the phone.
On Fri, Oct 29, 2010 at 1:35 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you try another one with just udp and not "port 5060" so I can see the rtp too
On Fri, Oct 29, 2010 at 12:28 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks Anthony, it's here: http://pastebin.freeswitch.org/14350
And pcap is attached.
The call connects around (or just before) the 16:58:35 mark (line 558 is what I see in the terminal while waiting for it to connect - both early-media and the missing start of the media)
Cheers, Fraser
On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Can you do this trace with debug level logging in addition to the sip trace console loglevel debug
you also may want to get a pcap of it
tshark udp and port 5060 -w test.pcap
On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks Anthony,
Finally managed to get a sip trace - could you do me a favor and take
a
look and/or give me some ideas of what to look for?
http://pastebin.freeswitch.org/14300
I've highlighted lines 168 and 193. In between these lines is where the number is dialed and rings once, then picks up, then theres silence for a second or two, and that second SIP message is when I start hearing audio.
Thanks, Fraser
On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
its a blue message on cli
It could also be the other side expecting us to send media first or something silly. try getting a sip trace and figure out when the rtp starts arriving.
On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond <fraserredmond@gmail.com> wrote: > Sorry, yes, I am setting ignore_early_media=true in the first
area.
> (Or > are > you saying that should be off? I forget now why I needed it on, but > there > was a reason I added it.) > > Yes, the bridge doesn't start until after the A-leg has answered. > > Thanks for the suggestion about nat/auto-changing port, I'll have a > look > into that - would that be in the cli output or in a sip trace? I've > already > looked and it's not appearing in the CLI output (with > loglevel=debug), > haven't looked in the sip trace yet. > > Cheers, > Fraser > > > > > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale > <anthony.minessale@gmail.com> wrote: >> >> are you setting ignore_early_media=true in the first vars=values >> area? >> >> This looks like you could be calling one leg who is still not >> answered >> and then bridging it to another dest. The bridge app will wait for >> the first leg to answer before bridging. >> >> Also if you have any NAT anywhere, look for an "auto-changing port" >> type message which can also be attributed to this due to a detection >> period for incorrect ports. >> >> >> >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond >> <fraserredmond@gmail.com> wrote: >> > event_socket: >> > api originate {vars=values}user/$fromExtn@Domain >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline >> > >> > then >> > >> > dialplan: >> > <action application="set" >> > data="effective_caller_id_number=+1800number"/> >> > <action application="set" data="effective_caller_id_name="/> >> > (set and/or export a bunch of other vars too) >> > <action application="set" >> > >> > >> > >> > data="dial_string=sofia/gateway/ gatewayname.com/00${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D> "/> >> > <action application="bridge" data="${dial_string}"/> >> > >> > Cheers, >> > Fraser >> > >> > >> > >> > >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale >> > <anthony.minessale@gmail.com> wrote: >> >> >> >> how are you accomplishing that? by which technique? >> >> >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond >> >> <fraserredmond@gmail.com> wrote: >> >> > The call is originated from Freeswitch (via CLI) to a >> >> > softphone, >> >> > then >> >> > when >> >> > that is connected it bridges out to the gateway. >> >> > >> >> > Cheers, >> >> > Fraser >> >> > >> >> > >> >> > >> >> > >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale >> >> > <anthony.minessale@gmail.com> wrote: >> >> >> >> >> >> Where is the other side of this call coming from? >> >> >> >> >> >> [ ( ) ] -> FS -> (PSTN via SIP) >> >> >> >> >> >> What goes in the empty space above? >> > >> > >> > _______________________________________________ >> > 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 >> >> _______________________________________________ >> 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 > >
-- 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
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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
-- 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
_______________________________________________ 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
-- 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
-- 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
_______________________________________________ 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
I'm not sure what you mean it doesn't fit. Did you look at your trace you sent me in wireshark ? (see bad_call.png) You can see the bria phone answer 180 ringing with no sdp in packet 7 Then it starts sending RTP .. ??? That is completely wrong. It never told us it's sending RTP how can we receive it? We don't have a SDP. Then in bad_call2.png it sends a 200+sdp (yes that's what we wanted) Now you have 2 way audio. On Fri, Oct 29, 2010 at 4:23 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks, but that doesn't fit - I've tried it with Bria, and with the recent version of XLite 4, and its happened on both. They're both made by Counterpath, so I've also tried 3CXphone, and it's happening on that too.
More suspicious though is that it only happens with some gateways, and not others. Or did you mean that you think it's a bug with the gateway's systems? The two I've tried that have this problem are both big outfits, not some guy operating out of his garage :-) Also, both were already listed in the Gateways on the wiki, so I'm guessing someone else is already using them with Freeswitch.
My server is running on Amazon AWS - could it be a timing/virtualization type problem, or something like that?
Any other ideas?
Cheers, Fraser
On Fri, Oct 29, 2010 at 9:39 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
OK so, The phone sends a 180 ringing with NO SDP then it starts sending RTP That's is not right. It's a bug in the phone.
On Fri, Oct 29, 2010 at 1:35 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you try another one with just udp and not "port 5060" so I can see the rtp too
On Fri, Oct 29, 2010 at 12:28 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks Anthony, it's here: http://pastebin.freeswitch.org/14350
And pcap is attached.
The call connects around (or just before) the 16:58:35 mark (line 558 is what I see in the terminal while waiting for it to connect - both early-media and the missing start of the media)
Cheers, Fraser
On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Can you do this trace with debug level logging in addition to the sip trace console loglevel debug
you also may want to get a pcap of it
tshark udp and port 5060 -w test.pcap
On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks Anthony,
Finally managed to get a sip trace - could you do me a favor and take a look and/or give me some ideas of what to look for?
http://pastebin.freeswitch.org/14300
I've highlighted lines 168 and 193. In between these lines is where the number is dialed and rings once, then picks up, then theres silence for a second or two, and that second SIP message is when I start hearing audio.
Thanks, Fraser
On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: > > its a blue message on cli > > It could also be the other side expecting us to send media first or > something silly. > try getting a sip trace and figure out when the rtp starts > arriving. > > > On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond > <fraserredmond@gmail.com> wrote: > > Sorry, yes, I am setting ignore_early_media=true in the first > > area. > > (Or > > are > > you saying that should be off? I forget now why I needed it on, > > but > > there > > was a reason I added it.) > > > > Yes, the bridge doesn't start until after the A-leg has answered. > > > > Thanks for the suggestion about nat/auto-changing port, I'll have > > a > > look > > into that - would that be in the cli output or in a sip trace? > > I've > > already > > looked and it's not appearing in the CLI output (with > > loglevel=debug), > > haven't looked in the sip trace yet. > > > > Cheers, > > Fraser > > > > > > > > > > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale > > <anthony.minessale@gmail.com> wrote: > >> > >> are you setting ignore_early_media=true in the first vars=values > >> area? > >> > >> This looks like you could be calling one leg who is still not > >> answered > >> and then bridging it to another dest. The bridge app will wait > >> for > >> the first leg to answer before bridging. > >> > >> Also if you have any NAT anywhere, look for an "auto-changing > >> port" > >> type message which can also be attributed to this due to a > >> detection > >> period for incorrect ports. > >> > >> > >> > >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond > >> <fraserredmond@gmail.com> wrote: > >> > event_socket: > >> > api originate {vars=values}user/$fromExtn@Domain > >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline > >> > > >> > then > >> > > >> > dialplan: > >> > <action application="set" > >> > data="effective_caller_id_number=+1800number"/> > >> > <action application="set" data="effective_caller_id_name="/> > >> > (set and/or export a bunch of other vars too) > >> > <action application="set" > >> > > >> > > >> > > >> > > >> > data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}"/> > >> > <action application="bridge" data="${dial_string}"/> > >> > > >> > Cheers, > >> > Fraser > >> > > >> > > >> > > >> > > >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale > >> > <anthony.minessale@gmail.com> wrote: > >> >> > >> >> how are you accomplishing that? by which technique? > >> >> > >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond > >> >> <fraserredmond@gmail.com> wrote: > >> >> > The call is originated from Freeswitch (via CLI) to a > >> >> > softphone, > >> >> > then > >> >> > when > >> >> > that is connected it bridges out to the gateway. > >> >> > > >> >> > Cheers, > >> >> > Fraser > >> >> > > >> >> > > >> >> > > >> >> > > >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale > >> >> > <anthony.minessale@gmail.com> wrote: > >> >> >> > >> >> >> Where is the other side of this call coming from? > >> >> >> > >> >> >> [ ( ) ] -> FS -> (PSTN via SIP) > >> >> >> > >> >> >> What goes in the empty space above? > >> > > >> > > >> > _______________________________________________ > >> > 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 > >> > >> _______________________________________________ > >> 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 > > > > > > > > -- > 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 > > _______________________________________________ > 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
-- 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
_______________________________________________ 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
-- 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
-- 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
_______________________________________________ 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
-- 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
Ah, ok, thanks, I'll do some more tests with the trace running, and see whats happening with different softphones and gateways. Thanks, Fraser On Fri, Oct 29, 2010 at 10:53 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I'm not sure what you mean it doesn't fit.
Did you look at your trace you sent me in wireshark ? (see bad_call.png) You can see the bria phone answer 180 ringing with no sdp in packet 7 Then it starts sending RTP .. ??? That is completely wrong.
It never told us it's sending RTP how can we receive it? We don't have a SDP.
Then in bad_call2.png it sends a 200+sdp (yes that's what we wanted) Now you have 2 way audio.
On Fri, Oct 29, 2010 at 4:23 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks, but that doesn't fit - I've tried it with Bria, and with the recent version of XLite 4, and its happened on both. They're both made by Counterpath, so I've also tried 3CXphone, and it's happening on that too.
More suspicious though is that it only happens with some gateways, and not others. Or did you mean that you think it's a bug with the gateway's systems? The two I've tried that have this problem are both big outfits, not some guy operating out of his garage :-) Also, both were already listed in the Gateways on the wiki, so I'm guessing someone else is already using them with Freeswitch.
My server is running on Amazon AWS - could it be a timing/virtualization type problem, or something like that?
Any other ideas?
Cheers, Fraser
On Fri, Oct 29, 2010 at 9:39 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
OK so, The phone sends a 180 ringing with NO SDP then it starts sending RTP That's is not right. It's a bug in the phone.
On Fri, Oct 29, 2010 at 1:35 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you try another one with just udp and not "port 5060" so I can see the rtp too
On Fri, Oct 29, 2010 at 12:28 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks Anthony, it's here: http://pastebin.freeswitch.org/14350
And pcap is attached.
The call connects around (or just before) the 16:58:35 mark (line 558 is what I see in the terminal while waiting for it to connect - both early-media and the missing start of the media)
Cheers, Fraser
On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Can you do this trace with debug level logging in addition to the
sip
trace console loglevel debug
you also may want to get a pcap of it
tshark udp and port 5060 -w test.pcap
On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond <fraserredmond@gmail.com> wrote: > Thanks Anthony, > > Finally managed to get a sip trace - could you do me a favor and > take a > look > and/or give me some ideas of what to look for? > > http://pastebin.freeswitch.org/14300 > > I've highlighted lines 168 and 193. In between these lines is where > the > number is dialed and rings once, then picks up, then theres silence > for > a > second or two, and that second SIP message is when I start hearing > audio. > > Thanks, > Fraser > > > > > On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale > <anthony.minessale@gmail.com> wrote: >> >> its a blue message on cli >> >> It could also be the other side expecting us to send media first or >> something silly. >> try getting a sip trace and figure out when the rtp starts >> arriving. >> >> >> On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond >> <fraserredmond@gmail.com> wrote: >> > Sorry, yes, I am setting ignore_early_media=true in the first >> > area. >> > (Or >> > are >> > you saying that should be off? I forget now why I needed it on, >> > but >> > there >> > was a reason I added it.) >> > >> > Yes, the bridge doesn't start until after the A-leg has answered. >> > >> > Thanks for the suggestion about nat/auto-changing port, I'll have >> > a >> > look >> > into that - would that be in the cli output or in a sip trace? >> > I've >> > already >> > looked and it's not appearing in the CLI output (with >> > loglevel=debug), >> > haven't looked in the sip trace yet. >> > >> > Cheers, >> > Fraser >> > >> > >> > >> > >> > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale >> > <anthony.minessale@gmail.com> wrote: >> >> >> >> are you setting ignore_early_media=true in the first vars=values >> >> area? >> >> >> >> This looks like you could be calling one leg who is still not >> >> answered >> >> and then bridging it to another dest. The bridge app will wait >> >> for >> >> the first leg to answer before bridging. >> >> >> >> Also if you have any NAT anywhere, look for an "auto-changing >> >> port" >> >> type message which can also be attributed to this due to a >> >> detection >> >> period for incorrect ports. >> >> >> >> >> >> >> >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond >> >> <fraserredmond@gmail.com> wrote: >> >> > event_socket: >> >> > api originate {vars=values}user/$fromExtn@Domain >> >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline >> >> > >> >> > then >> >> > >> >> > dialplan: >> >> > <action application="set" >> >> > data="effective_caller_id_number=+1800number"/> >> >> > <action application="set" data="effective_caller_id_name="/> >> >> > (set and/or export a bunch of other vars too) >> >> > <action application="set" >> >> > >> >> > >> >> > >> >> > >> >> > data="dial_string=sofia/gateway/ gatewayname.com/00${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D> "/> >> >> > <action application="bridge" data="${dial_string}"/> >> >> > >> >> > Cheers, >> >> > Fraser >> >> > >> >> > >> >> > >> >> > >> >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale >> >> > <anthony.minessale@gmail.com> wrote: >> >> >> >> >> >> how are you accomplishing that? by which technique? >> >> >> >> >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond >> >> >> <fraserredmond@gmail.com> wrote: >> >> >> > The call is originated from Freeswitch (via CLI) to a >> >> >> > softphone, >> >> >> > then >> >> >> > when >> >> >> > that is connected it bridges out to the gateway. >> >> >> > >> >> >> > Cheers, >> >> >> > Fraser >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale >> >> >> > <anthony.minessale@gmail.com> wrote: >> >> >> >> >> >> >> >> Where is the other side of this call coming from? >> >> >> >> >> >> >> >> [ ( ) ] -> FS -> (PSTN via SIP) >> >> >> >> >> >> >> >> What goes in the empty space above? >> >> > >> >> > >> >> > _______________________________________________ >> >> > 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 >> >> >> >> _______________________________________________ >> >> 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 >> > >> > >> >> >> >> -- >> 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 >> >> _______________________________________________ >> 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 > >
-- 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
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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
-- 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
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_______________________________________________ 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
_______________________________________________ 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
Ok, so I've finally found the problem. I'm connecting to the softphone first then transferring to the gateway, but in the a-leg connection to the softphone I had set ignore_early_media=true. I tried removing that and it works now. Presumably it's correct that SDP wouldn't be sent at 180 Ringing with that flag on - but that's above my pay-grade, I'm just happy to have it working. Thanks for your help Anthony, Regards, Fraser On Fri, Oct 29, 2010 at 11:38 PM, Fraser Redmond <fraserredmond@gmail.com>wrote:
Ah, ok, thanks, I'll do some more tests with the trace running, and see whats happening with different softphones and gateways.
Thanks, Fraser
On Fri, Oct 29, 2010 at 10:53 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I'm not sure what you mean it doesn't fit.
Did you look at your trace you sent me in wireshark ? (see bad_call.png) You can see the bria phone answer 180 ringing with no sdp in packet 7 Then it starts sending RTP .. ??? That is completely wrong.
It never told us it's sending RTP how can we receive it? We don't have a SDP.
Then in bad_call2.png it sends a 200+sdp (yes that's what we wanted) Now you have 2 way audio.
On Fri, Oct 29, 2010 at 4:23 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks, but that doesn't fit - I've tried it with Bria, and with the recent version of XLite 4, and its happened on both. They're both made by Counterpath, so I've also tried 3CXphone, and it's happening on that too.
More suspicious though is that it only happens with some gateways, and not others. Or did you mean that you think it's a bug with the gateway's systems? The two I've tried that have this problem are both big outfits, not some guy operating out of his garage :-) Also, both were already listed in the Gateways on the wiki, so I'm guessing someone else is already using them with Freeswitch.
My server is running on Amazon AWS - could it be a timing/virtualization type problem, or something like that?
Any other ideas?
Cheers, Fraser
On Fri, Oct 29, 2010 at 9:39 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
OK so, The phone sends a 180 ringing with NO SDP then it starts sending RTP That's is not right. It's a bug in the phone.
On Fri, Oct 29, 2010 at 1:35 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you try another one with just udp and not "port 5060" so I can see the rtp too
On Fri, Oct 29, 2010 at 12:28 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Thanks Anthony, it's here: http://pastebin.freeswitch.org/14350
And pcap is attached.
The call connects around (or just before) the 16:58:35 mark (line
558
is what I see in the terminal while waiting for it to connect - both early-media and the missing start of the media)
Cheers, Fraser
On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: > > Can you do this trace with debug level logging in addition to the sip > trace > console loglevel debug > > you also may want to get a pcap of it > > tshark udp and port 5060 -w test.pcap > > > > On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond > <fraserredmond@gmail.com> wrote: > > Thanks Anthony, > > > > Finally managed to get a sip trace - could you do me a favor and > > take a > > look > > and/or give me some ideas of what to look for? > > > > http://pastebin.freeswitch.org/14300 > > > > I've highlighted lines 168 and 193. In between these lines is where > > the > > number is dialed and rings once, then picks up, then theres silence > > for > > a > > second or two, and that second SIP message is when I start hearing > > audio. > > > > Thanks, > > Fraser > > > > > > > > > > On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale > > <anthony.minessale@gmail.com> wrote: > >> > >> its a blue message on cli > >> > >> It could also be the other side expecting us to send media first or > >> something silly. > >> try getting a sip trace and figure out when the rtp starts > >> arriving. > >> > >> > >> On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond > >> <fraserredmond@gmail.com> wrote: > >> > Sorry, yes, I am setting ignore_early_media=true in the first > >> > area. > >> > (Or > >> > are > >> > you saying that should be off? I forget now why I needed it on, > >> > but > >> > there > >> > was a reason I added it.) > >> > > >> > Yes, the bridge doesn't start until after the A-leg has answered. > >> > > >> > Thanks for the suggestion about nat/auto-changing port, I'll have > >> > a > >> > look > >> > into that - would that be in the cli output or in a sip trace? > >> > I've > >> > already > >> > looked and it's not appearing in the CLI output (with > >> > loglevel=debug), > >> > haven't looked in the sip trace yet. > >> > > >> > Cheers, > >> > Fraser > >> > > >> > > >> > > >> > > >> > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale > >> > <anthony.minessale@gmail.com> wrote: > >> >> > >> >> are you setting ignore_early_media=true in the first vars=values > >> >> area? > >> >> > >> >> This looks like you could be calling one leg who is still not > >> >> answered > >> >> and then bridging it to another dest. The bridge app will wait > >> >> for > >> >> the first leg to answer before bridging. > >> >> > >> >> Also if you have any NAT anywhere, look for an "auto-changing > >> >> port" > >> >> type message which can also be attributed to this due to a > >> >> detection > >> >> period for incorrect ports. > >> >> > >> >> > >> >> > >> >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond > >> >> <fraserredmond@gmail.com> wrote: > >> >> > event_socket: > >> >> > api originate {vars=values}user/$fromExtn@Domain > >> >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline > >> >> > > >> >> > then > >> >> > > >> >> > dialplan: > >> >> > <action application="set" > >> >> > data="effective_caller_id_number=+1800number"/> > >> >> > <action application="set" data="effective_caller_id_name="/> > >> >> > (set and/or export a bunch of other vars too) > >> >> > <action application="set" > >> >> > > >> >> > > >> >> > > >> >> > > >> >> > data="dial_string=sofia/gateway/ gatewayname.com/00${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D> "/> > >> >> > <action application="bridge" data="${dial_string}"/> > >> >> > > >> >> > Cheers, > >> >> > Fraser > >> >> > > >> >> > > >> >> > > >> >> > > >> >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale > >> >> > <anthony.minessale@gmail.com> wrote: > >> >> >> > >> >> >> how are you accomplishing that? by which technique? > >> >> >> > >> >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond > >> >> >> <fraserredmond@gmail.com> wrote: > >> >> >> > The call is originated from Freeswitch (via CLI) to a > >> >> >> > softphone, > >> >> >> > then > >> >> >> > when > >> >> >> > that is connected it bridges out to the gateway. > >> >> >> > > >> >> >> > Cheers, > >> >> >> > Fraser > >> >> >> > > >> >> >> > > >> >> >> > > >> >> >> > > >> >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale > >> >> >> > <anthony.minessale@gmail.com> wrote: > >> >> >> >> > >> >> >> >> Where is the other side of this call coming from? > >> >> >> >> > >> >> >> >> [ ( ) ] -> FS -> (PSTN via SIP) > >> >> >> >> > >> >> >> >> What goes in the empty space above? > >> >> > > >> >> > > >> >> > _______________________________________________ > >> >> > 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 > >> >> > >> >> _______________________________________________ > >> >> 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 > >> > > >> > > >> > >> > >> > >> -- > >> 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 > >> > >> _______________________________________________ > >> 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 > > > > > > > > -- > 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 > > _______________________________________________ > 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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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
-- 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
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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
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participants (2)
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Anthony Minessale
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Fraser Redmond