Hello, I am a new user of FreeSwitch and I try to switch from Asterisk to FreeSwitch. For now, I just registered a SIP line and I made an extension in the dialplan to playback music on hold when I call : <context name="default"> <extension name="test"> <condition> <!--<condition field="destination_number" expression="^9000$">--> <action application="answer"/> <action application="playback" data="$${hold_music}"/> </condition> </extension> </context> When I call I have a problem with the sound, it is very bad, I hear very fast crackle. On the console I can see a warning during my call and that's all: 2010-11-12 13:02:45.119517 [NOTICE] switch_channel.c:784 New Channel sofia/external/anonymous@anonymous.invalid [c27034e2-ee54-11df-ad91-7fc19b304698] 2010-11-12 13:02:45.122548 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <anonymous>->test in context default 2010-11-12 13:02:45.125555 [NOTICE] mod_dptools.c:920 Channel [sofia/external/anonymous@anonymous.invalid] has been answered 2010-11-12 13:02:45.508790 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20 If I add <X-PRE-PROCESS cmd="set" data="timer_name=soft"/> in vars.xml, when I call the sound is good but I still get the warning : 2010-11-12 13:05:42.789161 [NOTICE] switch_channel.c:784 New Channel sofia/external/anonymous@anonymous.invalid [2c565760-ee55-11df-a0d9-0f91e325cd76] 2010-11-12 13:05:42.791179 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <anonymous>->test in context default 2010-11-12 13:05:42.794179 [NOTICE] mod_dptools.c:920 Channel [sofia/external/anonymous@anonymous.invalid] has been answered 2010-11-12 13:05:43.174443 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20 2010-11-12 13:05:43.175461 [INFO] sofia.c:709 sofia/external/anonymous@anonymous.invalid Update Callee ID to "anonymous" <anonymous> Is this normal? Why should I add this line in vars.xml for it to work ? What does this warning? How to solve it? Thank you in advance for your help. Loïc
What is your device? is it a cisco/linksys/sipura ? you can change the packet size from 0.30 to 0.20 in the UI of your phone. On Fri, Nov 12, 2010 at 6:09 AM, Loïc <loic.latreille@ovh.net> wrote:
Hello,
I am a new user of FreeSwitch and I try to switch from Asterisk to FreeSwitch.
For now, I just registered a SIP line and I made an extension in the dialplan to playback music on hold when I call :
<context name="default">
<extension name="test"> <condition> <!--<condition field="destination_number" expression="^9000$">--> <action application="answer"/> <action application="playback" data="$${hold_music}"/> </condition> </extension>
</context>
When I call I have a problem with the sound, it is very bad, I hear very fast crackle. On the console I can see a warning during my call and that's all:
2010-11-12 13:02:45.119517 [NOTICE] switch_channel.c:784 New Channel sofia/external/anonymous@anonymous.invalid [c27034e2-ee54-11df-ad91-7fc19b304698] 2010-11-12 13:02:45.122548 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <anonymous>->test in context default 2010-11-12 13:02:45.125555 [NOTICE] mod_dptools.c:920 Channel [sofia/external/anonymous@anonymous.invalid] has been answered 2010-11-12 13:02:45.508790 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20
If I add <X-PRE-PROCESS cmd="set" data="timer_name=soft"/> in vars.xml, when I call the sound is good but I still get the warning :
2010-11-12 13:05:42.789161 [NOTICE] switch_channel.c:784 New Channel sofia/external/anonymous@anonymous.invalid [2c565760-ee55-11df-a0d9-0f91e325cd76] 2010-11-12 13:05:42.791179 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <anonymous>->test in context default 2010-11-12 13:05:42.794179 [NOTICE] mod_dptools.c:920 Channel [sofia/external/anonymous@anonymous.invalid] has been answered 2010-11-12 13:05:43.174443 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20 2010-11-12 13:05:43.175461 [INFO] sofia.c:709 sofia/external/anonymous@anonymous.invalid Update Callee ID to "anonymous" <anonymous>
Is this normal? Why should I add this line in vars.xml for it to work ? What does this warning? How to solve it?
Thank you in advance for your help.
Loïc
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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
Hi Anthony, I have this warning as well with my Siemens C470IP as my Linksys SPA942. And my Linksys has been configured with a RTP Packet Size to 0.020. If I change to 0.20 I have the same warning :-/ Loïc On Fri, 12 Nov 2010 11:23:46 -0600, Anthony Minessale <anthony.minessale@gmail.com> wrote:
What is your device?
is it a cisco/linksys/sipura ?
you can change the packet size from 0.30 to 0.20 in the UI of your phone.
On Fri, Nov 12, 2010 at 6:09 AM, Loïc <loic.latreille@ovh.net> wrote:
Hello,
I am a new user of FreeSwitch and I try to switch from Asterisk to FreeSwitch.
For now, I just registered a SIP line and I made an extension in the dialplan to playback music on hold when I call :
<context name="default">
<extension name="test"> <condition> <!--<condition field="destination_number" expression="^9000$">--> <action application="answer"/> <action application="playback" data="$${hold_music}"/> </condition> </extension>
</context>
When I call I have a problem with the sound, it is very bad, I hear very fast crackle. On the console I can see a warning during my call and that's all:
2010-11-12 13:02:45.119517 [NOTICE] switch_channel.c:784 New Channel sofia/external/anonymous@anonymous.invalid [c27034e2-ee54-11df-ad91-7fc19b304698] 2010-11-12 13:02:45.122548 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <anonymous>->test in context default 2010-11-12 13:02:45.125555 [NOTICE] mod_dptools.c:920 Channel [sofia/external/anonymous@anonymous.invalid] has been answered 2010-11-12 13:02:45.508790 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20
If I add <X-PRE-PROCESS cmd="set" data="timer_name=soft"/> in vars.xml, when I call the sound is good but I still get the warning :
2010-11-12 13:05:42.789161 [NOTICE] switch_channel.c:784 New Channel sofia/external/anonymous@anonymous.invalid [2c565760-ee55-11df-a0d9-0f91e325cd76] 2010-11-12 13:05:42.791179 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <anonymous>->test in context default 2010-11-12 13:05:42.794179 [NOTICE] mod_dptools.c:920 Channel [sofia/external/anonymous@anonymous.invalid] has been answered 2010-11-12 13:05:43.174443 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20 2010-11-12 13:05:43.175461 [INFO] sofia.c:709 sofia/external/anonymous@anonymous.invalid Update Callee ID to "anonymous" <anonymous>
Is this normal? Why should I add this line in vars.xml for it to work ? What does this warning? How to solve it?
Thank you in advance for your help.
Loïc
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Try changing your RTP packet time to 30ms and see what happens.
Hi Anthony,
I have this warning as well with my Siemens C470IP as my Linksys SPA942. And my Linksys has been configured with a RTP Packet Size to 0.020. If I change to 0.20 I have the same warning :-/
Loïc
On Fri, 12 Nov 2010 11:23:46 -0600, Anthony Minessale <anthony.minessale@gmail.com> wrote:
What is your device?
is it a cisco/linksys/sipura ?
you can change the packet size from 0.30 to 0.20 in the UI of your phone.
On Fri, Nov 12, 2010 at 6:09 AM, Loïc <loic.latreille@ovh.net> wrote:
Hello,
I am a new user of FreeSwitch and I try to switch from Asterisk to FreeSwitch.
For now, I just registered a SIP line and I made an extension in the dialplan to playback music on hold when I call :
<context name="default">
<extension name="test"> <condition> <!--<condition field="destination_number" expression="^9000$">--> <action application="answer"/> <action application="playback" data="$${hold_music}"/> </condition> </extension>
</context>
When I call I have a problem with the sound, it is very bad, I hear very fast crackle. On the console I can see a warning during my call and that's all:
2010-11-12 13:02:45.119517 [NOTICE] switch_channel.c:784 New Channel sofia/external/anonymous@anonymous.invalid [c27034e2-ee54-11df-ad91-7fc19b304698] 2010-11-12 13:02:45.122548 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <anonymous>->test in context default 2010-11-12 13:02:45.125555 [NOTICE] mod_dptools.c:920 Channel [sofia/external/anonymous@anonymous.invalid] has been answered 2010-11-12 13:02:45.508790 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20
If I add <X-PRE-PROCESS cmd="set" data="timer_name=soft"/> in vars.xml, when I call the sound is good but I still get the warning :
2010-11-12 13:05:42.789161 [NOTICE] switch_channel.c:784 New Channel sofia/external/anonymous@anonymous.invalid [2c565760-ee55-11df-a0d9-0f91e325cd76] 2010-11-12 13:05:42.791179 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <anonymous>->test in context default 2010-11-12 13:05:42.794179 [NOTICE] mod_dptools.c:920 Channel [sofia/external/anonymous@anonymous.invalid] has been answered 2010-11-12 13:05:43.174443 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20 2010-11-12 13:05:43.175461 [INFO] sofia.c:709 sofia/external/anonymous@anonymous.invalid Update Callee ID to "anonymous" <anonymous>
Is this normal? Why should I add this line in vars.xml for it to work ? What does this warning? How to solve it?
Thank you in advance for your help.
Loïc
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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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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Loïc, I would suspect the issue to be on the leg B, because I use a Siemens C470IP on FS regularly for testing, and I never had a such warning. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 12/11/2010 à 18:45, Loïc a écrit :
Hi Anthony,
I have this warning as well with my Siemens C470IP as my Linksys SPA942. And my Linksys has been configured with a RTP Packet Size to 0.020. If I change to 0.20 I have the same warning :-/
Loïc
On Fri, 12 Nov 2010 11:23:46 -0600, Anthony Minessale <anthony.minessale@gmail.com> wrote:
What is your device?
is it a cisco/linksys/sipura ?
you can change the packet size from 0.30 to 0.20 in the UI of your phone.
On Fri, Nov 12, 2010 at 6:09 AM, Loïc <loic.latreille@ovh.net> wrote:
Hello,
I am a new user of FreeSwitch and I try to switch from Asterisk to FreeSwitch.
For now, I just registered a SIP line and I made an extension in the dialplan to playback music on hold when I call :
<context name="default">
<extension name="test"> <condition> <!--<condition field="destination_number" expression="^9000$">--> <action application="answer"/> <action application="playback" data="$${hold_music}"/> </condition> </extension>
</context>
When I call I have a problem with the sound, it is very bad, I hear very fast crackle. On the console I can see a warning during my call and that's all:
2010-11-12 13:02:45.119517 [NOTICE] switch_channel.c:784 New Channel sofia/external/anonymous@anonymous.invalid [c27034e2-ee54-11df-ad91-7fc19b304698] 2010-11-12 13:02:45.122548 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <anonymous>->test in context default 2010-11-12 13:02:45.125555 [NOTICE] mod_dptools.c:920 Channel [sofia/external/anonymous@anonymous.invalid] has been answered 2010-11-12 13:02:45.508790 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20
If I add <X-PRE-PROCESS cmd="set" data="timer_name=soft"/> in vars.xml, when I call the sound is good but I still get the warning :
2010-11-12 13:05:42.789161 [NOTICE] switch_channel.c:784 New Channel sofia/external/anonymous@anonymous.invalid [2c565760-ee55-11df-a0d9-0f91e325cd76] 2010-11-12 13:05:42.791179 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <anonymous>->test in context default 2010-11-12 13:05:42.794179 [NOTICE] mod_dptools.c:920 Channel [sofia/external/anonymous@anonymous.invalid] has been answered 2010-11-12 13:05:43.174443 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20 2010-11-12 13:05:43.175461 [INFO] sofia.c:709 sofia/external/anonymous@anonymous.invalid Update Callee ID to "anonymous" <anonymous>
Is this normal? Why should I add this line in vars.xml for it to work ? What does this warning? How to solve it?
Thank you in advance for your help.
Loïc
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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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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0.020 is correct but what provider are you going out and what do the logs say... I think tony mis typed the bottom line is you want 20ms packets if at all possible. /b On Nov 12, 2010, at 11:45 AM, Loïc wrote:
Hi Anthony,
I have this warning as well with my Siemens C470IP as my Linksys SPA942. And my Linksys has been configured with a RTP Packet Size to 0.020. If I change to 0.20 I have the same warning :-/
Loïc
I have no phone connected directly to FS. I go through a SIP provider. My 2 SIP phones (Siemens and Linksys SPA942 C470IP) are connected to my SIP provider. I opened a new line with the SIP provider to configure a gateway on FS. When I call with my Phone, I call the line configured on SIP gateway. Loïc On Sat, 13 Nov 2010 12:00:11 -0600, Brian West <brian@freeswitch.org> wrote:
0.020 is correct but what provider are you going out and what do the logs say... I think tony mis typed the bottom line is you want 20ms packets if at all possible.
/b
On Nov 12, 2010, at 11:45 AM, Loïc wrote:
Hi Anthony,
I have this warning as well with my Siemens C470IP as my Linksys SPA942. And my Linksys has been configured with a RTP Packet Size to 0.020. If I change to 0.20 I have the same warning :-/
Loïc
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participants (5)
-
Anthony Minessale
-
Brian West
-
David Ponzone
-
Jeffrey Leung
-
Loïc