Hi, i installed FS and configured 2 IP Phones 1000 and 1001, when i call from 1000 to 1001 or backwards, i get too high Post Dial Delay, about 2-3 seconds, any help/sugestion on this? PS: with openser and Asterisk there is not appreciable PDD. Luis Jimenez
What do you mean? can you be a bit more descriptive about this? /b On Apr 20, 2008, at 8:11 AM, Luis Jimenez wrote:
Hi, i installed FS and configured 2 IP Phones 1000 and 1001, when i call from 1000 to 1001 or backwards, i get too high Post Dial Delay, about 2-3 seconds, any help/sugestion on this?
PS: with openser and Asterisk there is not appreciable PDD.
Luis Jimenez
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Brian West sip:brian@freeswitch.org
Ok, when you dial from say 1000 to 1001 you wait 3 seconds before the phone start ringing nad you listen ringback, i'll send some debugs from the FS console later. On Sun, Apr 20, 2008 at 8:14 PM, Brian West <brian@freeswitch.org> wrote:
What do you mean? can you be a bit more descriptive about this?
/b
On Apr 20, 2008, at 8:11 AM, Luis Jimenez wrote:
Hi, i installed FS and configured 2 IP Phones 1000 and 1001, when i call from 1000 to 1001 or backwards, i get too high Post Dial Delay, about 2-3 seconds, any help/sugestion on this?
PS: with openser and Asterisk there is not appreciable PDD.
Luis Jimenez
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Brian West sip:brian@freeswitch.org <sip%3Abrian@freeswitch.org>
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I haven't seen this issue can you describe your network topology? On Apr 21, 2008, at 5:23 AM, Luis Jimenez wrote:
Ok, when you dial from say 1000 to 1001 you wait 3 seconds before the phone start ringing nad you listen ringback, i'll send some debugs from the FS console later.
Brian West sip:brian@freeswitch.org
Ok, my network topology is: 1 server HP ML-110 FS installed. 2 Snom 360 1 Switch Linksys SRW224P using default dialplan installed by make samples this is the debug of de FS console when you dial from 1000 to 1001: freeswitch@pbx> 2008-04-22 08:33:43 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1000@10.0.0.100[eddcc31a-3dcb-4644-8258-fcb5f49e6900] 2008-04-22 08:33:43 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing Juan Perez->1001@default 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 switch_ivr_bind_dtmf_meta_session() Bound: 1 execute_extension::dx XML features 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 switch_ivr_bind_dtmf_meta_session() Bound: 2 record_session::/opt/freeswitch/recordings/1000.2008-04-22-08-33-44.wav 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 switch_ivr_bind_dtmf_meta_session() Bound: 3 execute_extension::cf XML features 2008-04-22 08:33:45 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1001@10.0.0.16:2051;line=1nepvhw6 [dfaf0d12-a892-46f1-bfb1-1b0c385e97a5] 2008-04-22 08:33:45 [NOTICE] sofia.c:1713 sofia_handle_sip_i_state() Ring-Ready sofia/default/1001@10.0.0.16:2051;line=1nepvhw6! 2008-04-22 08:33:45 [NOTICE] mod_sofia.c:1018 sofia_receive_message() Ring-Ready sofia/default/1000@10.0.0.100! 2008-04-22 08:33:45 [NOTICE] switch_ivr_originate.c:1036 switch_ivr_originate() Ring Ready sofia/default/1000@10.0.0.100! Ring starts after last line Any help appreciated. Luis jimenez On Mon, Apr 21, 2008 at 6:28 AM, Brian West <brian@freeswitch.org> wrote:
I haven't seen this issue can you describe your network topology?
On Apr 21, 2008, at 5:23 AM, Luis Jimenez wrote:
Ok, when you dial from say 1000 to 1001 you wait 3 seconds before the phone start ringing nad you listen ringback, i'll send some debugs from the FS console later.
Brian West sip:brian@freeswitch.org <sip%3Abrian@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
The ringing is not passed across until the other phone (the one you are calling) sends a 180 Ringing. As soon as it sends it, we pass that indication to the calling phone (your phone). Asterisk just assumes you should hear ringing and sends it instantly on it's own before anyone knows that the call is going to work. If you want this same behaviour, add this to the dialplan: go to line 145 of default.xml and put the following line as the first anti-action <anti-action application="ring_ready"/> This will make FreeSWITCH send your calling phone a 180 ringing before it knows for sure if it should. On Tue, Apr 22, 2008 at 8:53 AM, Luis Jimenez <ljjimenez@gmail.com> wrote:
Ok, my network topology is:
1 server HP ML-110 FS installed. 2 Snom 360 1 Switch Linksys SRW224P using default dialplan installed by make samples
this is the debug of de FS console when you dial from 1000 to 1001:
freeswitch@pbx> 2008-04-22 08:33:43 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1000@10.0.0.100[eddcc31a-3dcb-4644-8258-fcb5f49e6900] 2008-04-22 08:33:43 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing Juan Perez->1001@default 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 switch_ivr_bind_dtmf_meta_session() Bound: 1 execute_extension::dx XML features 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 switch_ivr_bind_dtmf_meta_session() Bound: 2 record_session::/opt/freeswitch/recordings/1000.2008-04-22-08-33-44.wav 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 switch_ivr_bind_dtmf_meta_session() Bound: 3 execute_extension::cf XML features 2008-04-22 08:33:45 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1001@10.0.0.16:2051;line=1nepvhw6 [dfaf0d12-a892-46f1-bfb1-1b0c385e97a5] 2008-04-22 08:33:45 [NOTICE] sofia.c:1713 sofia_handle_sip_i_state() Ring-Ready sofia/default/1001@10.0.0.16:2051;line=1nepvhw6! 2008-04-22 08:33:45 [NOTICE] mod_sofia.c:1018 sofia_receive_message() Ring-Ready sofia/default/1000@10.0.0.100! 2008-04-22 08:33:45 [NOTICE] switch_ivr_originate.c:1036 switch_ivr_originate() Ring Ready sofia/default/1000@10.0.0.100!
Ring starts after last line
Any help appreciated. Luis jimenez
On Mon, Apr 21, 2008 at 6:28 AM, Brian West <brian@freeswitch.org> wrote:
I haven't seen this issue can you describe your network topology?
On Apr 21, 2008, at 5:23 AM, Luis Jimenez wrote:
Ok, when you dial from say 1000 to 1001 you wait 3 seconds before the phone start ringing nad you listen ringback, i'll send some debugs from the FS console later.
Brian West sip:brian@freeswitch.org <sip%3Abrian@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
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
What about OpenSer, i get the same results as Asterisk. On Tue, Apr 22, 2008 at 10:11 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
The ringing is not passed across until the other phone (the one you are calling) sends a 180 Ringing.
As soon as it sends it, we pass that indication to the calling phone (your phone).
Asterisk just assumes you should hear ringing and sends it instantly on it's own before anyone knows that the call is going to work. If you want this same behaviour, add this to the dialplan:
go to line 145 of default.xml and put the following line as the first anti-action <anti-action application="ring_ready"/>
This will make FreeSWITCH send your calling phone a 180 ringing before it knows for sure if it should.
On Tue, Apr 22, 2008 at 8:53 AM, Luis Jimenez <ljjimenez@gmail.com> wrote:
Ok, my network topology is:
1 server HP ML-110 FS installed. 2 Snom 360 1 Switch Linksys SRW224P using default dialplan installed by make samples
this is the debug of de FS console when you dial from 1000 to 1001:
freeswitch@pbx> 2008-04-22 08:33:43 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1000@10.0.0.100[eddcc31a-3dcb-4644-8258-fcb5f49e6900] 2008-04-22 08:33:43 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing Juan Perez->1001@default 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 switch_ivr_bind_dtmf_meta_session() Bound: 1 execute_extension::dx XML features 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 switch_ivr_bind_dtmf_meta_session() Bound: 2 record_session::/opt/freeswitch/recordings/1000.2008-04-22-08-33-44.wav 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 switch_ivr_bind_dtmf_meta_session() Bound: 3 execute_extension::cf XML features 2008-04-22 08:33:45 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1001@10.0.0.16:2051;line=1nepvhw6 [dfaf0d12-a892-46f1-bfb1-1b0c385e97a5] 2008-04-22 08:33:45 [NOTICE] sofia.c:1713 sofia_handle_sip_i_state() Ring-Ready sofia/default/1001@10.0.0.16:2051;line=1nepvhw6! 2008-04-22 08:33:45 [NOTICE] mod_sofia.c:1018 sofia_receive_message() Ring-Ready sofia/default/1000@10.0.0.100! 2008-04-22 08:33:45 [NOTICE] switch_ivr_originate.c:1036 switch_ivr_originate() Ring Ready sofia/default/1000@10.0.0.100!
Ring starts after last line
Any help appreciated. Luis jimenez
On Mon, Apr 21, 2008 at 6:28 AM, Brian West <brian@freeswitch.org> wrote:
I haven't seen this issue can you describe your network topology?
On Apr 21, 2008, at 5:23 AM, Luis Jimenez wrote:
Ok, when you dial from say 1000 to 1001 you wait 3 seconds before the phone start ringing nad you listen ringback, i'll send some debugs from the FS console later.
Brian West sip:brian@freeswitch.org <sip%3Abrian@freeswitch.org>
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
What about it? Did you try my suggestion? I am not sure I could have explained it any better? Watch your console trace and maybe take a pcap of it. you can export TPORT_LOG=1 in your shell to see the sip messages in the console. The instant the phone you are calling sends us 180, we send you 180. So if you add the line i told you to your dialplan, we will send 180 instantly instead which is what you want right? On Tue, Apr 22, 2008 at 9:18 AM, Luis Jimenez <ljjimenez@gmail.com> wrote:
What about OpenSer, i get the same results as Asterisk.
On Tue, Apr 22, 2008 at 10:11 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
The ringing is not passed across until the other phone (the one you are calling) sends a 180 Ringing.
As soon as it sends it, we pass that indication to the calling phone (your phone).
Asterisk just assumes you should hear ringing and sends it instantly on it's own before anyone knows that the call is going to work. If you want this same behaviour, add this to the dialplan:
go to line 145 of default.xml and put the following line as the first anti-action <anti-action application="ring_ready"/>
This will make FreeSWITCH send your calling phone a 180 ringing before it knows for sure if it should.
On Tue, Apr 22, 2008 at 8:53 AM, Luis Jimenez <ljjimenez@gmail.com> wrote:
Ok, my network topology is:
1 server HP ML-110 FS installed. 2 Snom 360 1 Switch Linksys SRW224P using default dialplan installed by make samples
this is the debug of de FS console when you dial from 1000 to 1001:
freeswitch@pbx> 2008-04-22 08:33:43 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1000@10.0.0.100[eddcc31a-3dcb-4644-8258-fcb5f49e6900] 2008-04-22 08:33:43 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing Juan Perez->1001@default 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 switch_ivr_bind_dtmf_meta_session() Bound: 1 execute_extension::dx XML features 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 switch_ivr_bind_dtmf_meta_session() Bound: 2 record_session::/opt/freeswitch/recordings/1000.2008-04-22-08-33-44.wav 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 switch_ivr_bind_dtmf_meta_session() Bound: 3 execute_extension::cf XML features 2008-04-22 08:33:45 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1001@10.0.0.16:2051;line=1nepvhw6 [dfaf0d12-a892-46f1-bfb1-1b0c385e97a5] 2008-04-22 08:33:45 [NOTICE] sofia.c:1713 sofia_handle_sip_i_state() Ring-Ready sofia/default/1001@10.0.0.16:2051;line=1nepvhw6! 2008-04-22 08:33:45 [NOTICE] mod_sofia.c:1018 sofia_receive_message() Ring-Ready sofia/default/1000@10.0.0.100! 2008-04-22 08:33:45 [NOTICE] switch_ivr_originate.c:1036 switch_ivr_originate() Ring Ready sofia/default/1000@10.0.0.100!
Ring starts after last line
Any help appreciated. Luis jimenez
On Mon, Apr 21, 2008 at 6:28 AM, Brian West <brian@freeswitch.org> wrote:
I haven't seen this issue can you describe your network topology?
On Apr 21, 2008, at 5:23 AM, Luis Jimenez wrote:
Ok, when you dial from say 1000 to 1001 you wait 3 seconds before the phone start ringing nad you listen ringback, i'll send some debugs from the FS console later.
Brian West sip:brian@freeswitch.org <sip%3Abrian@freeswitch.org>
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ 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/ 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> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (3)
-
Anthony Minessale
-
Brian West
-
Luis Jimenez