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[Freeswitch-users] SIP provider with extern rtp server


 
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ivan at myrvold.org
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PostPosted: Tue Oct 27, 2009 1:56 am    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

I have used a SIP provider for more than a year. A few days ago, he
said he was moving to a new server, and asked me to reconfigure. I
did, and everything seemed to work fine, until I did an outgoing call
to an external telephone. I found out I had no audio, in neither
direction. Incoming calls was working fine.

My provider said that the rtp is not going through the sip server, as
it did earlier, but now through several other IP's.

Do I have to do some special configuration to handle that?

Ivan

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PostPosted: Tue Oct 27, 2009 9:15 am    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

Make sure you let their media IPs through your firewall. Also, if you
are behind a NAT, check you have things passing to the correct
internal address.

On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold <ivan@myrvold.org> wrote:
Quote:
I have used a SIP provider for more than a year. A few days ago, he
said he was moving to a new server, and asked me to reconfigure. I
did, and everything seemed to work fine, until I did an outgoing call
to an external telephone. I found out I had no audio, in neither
direction. Incoming calls was working fine.

My provider said that the rtp is not going through the sip server, as
it did earlier, but now through several other IP's.

Do I have to do some special configuration to handle that?

Ivan

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--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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ivan at myrvold.org
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PostPosted: Tue Oct 27, 2009 11:12 am    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

The server is on a public IP, so there is no nat issue here.

I can also see the rtp messages on wireshark starting just after the
183 Session Progress message on the server, but just in one direction,
coming in to the server.
So it looks like Freeswitch is stopping the rtp.
Is this because the rtp originates from another ip than the sip
provider ip?

Ivan

Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable:

Quote:
Make sure you let their media IPs through your firewall. Also, if you
are behind a NAT, check you have things passing to the correct
internal address.

On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold <ivan@myrvold.org>
wrote:
Quote:
I have used a SIP provider for more than a year. A few days ago, he
said he was moving to a new server, and asked me to reconfigure. I
did, and everything seemed to work fine, until I did an outgoing call
to an external telephone. I found out I had no audio, in neither
direction. Incoming calls was working fine.

My provider said that the rtp is not going through the sip server, as
it did earlier, but now through several other IP's.

Do I have to do some special configuration to handle that?

Ivan

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
users
http://www.freeswitch.org




--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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PostPosted: Tue Oct 27, 2009 12:51 pm    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

No, the IP address the media originates from does not need to be tied
to the SIP IP address. Can you send a Wireshark capture taken on the
FreeSWITCH server of both call legs? Or, if you can, pastebin a debug
log from FreeSWITCH console with sofia loglevel set to 9 and siptrace
on for any Sofia SIP profiles involved.

On Tue, Oct 27, 2009 at 11:52 AM, Ivan C Myrvold <ivan@myrvold.org> wrote:
Quote:
The server is on a public IP, so there is no nat issue here.

I can also see the rtp messages on wireshark starting just after the
183 Session Progress message on the server, but just in one direction,
coming in to the server.
So it looks like Freeswitch is stopping the rtp.
Is this because the rtp originates from another ip than the  sip
provider ip?

Ivan

Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable:

Quote:
Make sure you let their media IPs through your firewall. Also, if you
are behind a NAT, check you have things passing to the correct
internal address.

On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold <ivan@myrvold.org>
wrote:
Quote:
I have used a SIP provider for more than a year. A few days ago, he
said he was moving to a new server, and asked me to reconfigure. I
did, and everything seemed to work fine, until I did an outgoing call
to an external telephone. I found out I had no audio, in neither
direction. Incoming calls was working fine.

My provider said that the rtp is not going through the sip server, as
it did earlier, but now through several other IP's.

Do I have to do some special configuration to handle that?

Ivan

_______________________________________________
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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




--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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ivan at myrvold.org
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PostPosted: Wed Oct 28, 2009 9:53 am    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

Here is a debug log from a call from an internal phone out to an
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578

Ivan

Den 27. okt. 2009 kl. 18:34 skrev Eliot Gable:

Quote:
No, the IP address the media originates from does not need to be tied
to the SIP IP address. Can you send a Wireshark capture taken on the
FreeSWITCH server of both call legs? Or, if you can, pastebin a debug
log from FreeSWITCH console with sofia loglevel set to 9 and siptrace
on for any Sofia SIP profiles involved.

On Tue, Oct 27, 2009 at 11:52 AM, Ivan C Myrvold <ivan@myrvold.org>
wrote:
Quote:
The server is on a public IP, so there is no nat issue here.

I can also see the rtp messages on wireshark starting just after the
183 Session Progress message on the server, but just in one
direction,
coming in to the server.
So it looks like Freeswitch is stopping the rtp.
Is this because the rtp originates from another ip than the sip
provider ip?

Ivan

Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable:

Quote:
Make sure you let their media IPs through your firewall. Also, if
you
are behind a NAT, check you have things passing to the correct
internal address.

On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold <ivan@myrvold.org>
wrote:
Quote:
I have used a SIP provider for more than a year. A few days ago, he
said he was moving to a new server, and asked me to reconfigure. I
did, and everything seemed to work fine, until I did an outgoing
call
to an external telephone. I found out I had no audio, in neither
direction. Incoming calls was working fine.

My provider said that the rtp is not going through the sip
server, as
it did earlier, but now through several other IP's.

Do I have to do some special configuration to handle that?

Ivan

_______________________________________________
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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




--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from
our
children." ~David Brower

"I decided the words were too conservative for me. We're not
borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
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users
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--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
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PostPosted: Wed Oct 28, 2009 1:35 pm    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan@myrvold.org (ivan@myrvold.org)> wrote:
Quote:
Here is a debug log from a call from an internal phone out to an
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578

Ivan

Uh... you wanna try that PB number again?
-MC
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PostPosted: Wed Oct 28, 2009 4:02 pm    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

Oh, what happened to it?

Anyway, here is a new pb:


http://pastebin.freeswitch.org/10867


Ivan
Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
Quote:


On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan@myrvold.org (ivan@myrvold.org)> wrote:
Quote:
Here is a debug log from a call from an internal phone out to an
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578

Ivan

Uh... you wanna try that PB number again?
-MC



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PostPosted: Wed Oct 28, 2009 9:30 pm    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

See that 200 OK that keeps coming in over and over and over and over
again? That's because they never received your ACK. If you can turn on
sofia loglevel to 9 and then watch where you send the ACK, you will
probably have your answer to why the other system did not receive it.
If you're still not sure what's going on, post another pastebin with
sofia loglevel set to 9.


On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan@myrvold.org> wrote:
Quote:
Oh, what happened to it?
Anyway, here is a new pb:
http://pastebin.freeswitch.org/10867
Ivan
Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:


On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan@myrvold.org> wrote:
Quote:

Here is a debug log from a call from an internal phone out to an
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578

Ivan

Uh... you wanna try that PB number again?
-MC

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--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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ivan at myrvold.org
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PostPosted: Fri Oct 30, 2009 11:25 am    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

I have already set debug to 9, on both profiles.

Ivan


Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:

Quote:
See that 200 OK that keeps coming in over and over and over and over
again? That's because they never received your ACK. If you can turn on
sofia loglevel to 9 and then watch where you send the ACK, you will
probably have your answer to why the other system did not receive it.
If you're still not sure what's going on, post another pastebin with
sofia loglevel set to 9.


On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan@myrvold.org>
wrote:
Quote:
Oh, what happened to it?
Anyway, here is a new pb:
http://pastebin.freeswitch.org/10867
Ivan
Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:


On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan@myrvold.org>
wrote:
Quote:

Here is a debug log from a call from an internal phone out to an
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578

Ivan

Uh... you wanna try that PB number again?
-MC

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--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
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PostPosted: Fri Oct 30, 2009 12:42 pm    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

fsctl loglevel debug
console loglevel debug
sofia profile internal siptrace on
sofia profile external siptrace on
sofia loglevel all 9
^^^^^^^^^^^^^^^^^^^^^

Then run your call, then do this:

sofia loglevel all 0
sofia profile external siptrace off
sofia profile internal siptrace off
fsctl loglevel warning
console loglevel warning

On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold <ivan@myrvold.org> wrote:
Quote:
I have already set debug to 9, on both profiles.

Ivan


Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:

Quote:
See that 200 OK that keeps coming in over and over and over and over
again? That's because they never received your ACK. If you can turn on
sofia loglevel to 9 and then watch where you send the ACK, you will
probably have your answer to why the other system did not receive it.
If you're still not sure what's going on, post another pastebin with
sofia loglevel set to 9.


On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan@myrvold.org>
wrote:
Quote:
Oh, what happened to it?
Anyway, here is a new pb:
http://pastebin.freeswitch.org/10867
Ivan
Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:


On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan@myrvold.org>
wrote:
Quote:

Here is a debug log from a call from an internal phone out to an
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578

Ivan

Uh... you wanna try that PB number again?
-MC

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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
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--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
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http://www.freeswitch.org




--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

_______________________________________________
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PostPosted: Fri Oct 30, 2009 3:35 pm    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

Yes, now I got a more detailed trace. Thank you for helping me with
this.

A new pastebin at http://pastebin.freeswitch.org/10905

Ivan

Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable:

Quote:
fsctl loglevel debug
console loglevel debug
sofia profile internal siptrace on
sofia profile external siptrace on
sofia loglevel all 9
^^^^^^^^^^^^^^^^^^^^^

Then run your call, then do this:

sofia loglevel all 0
sofia profile external siptrace off
sofia profile internal siptrace off
fsctl loglevel warning
console loglevel warning

On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold <ivan@myrvold.org>
wrote:
Quote:
I have already set debug to 9, on both profiles.

Ivan


Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:

Quote:
See that 200 OK that keeps coming in over and over and over and over
again? That's because they never received your ACK. If you can
turn on
sofia loglevel to 9 and then watch where you send the ACK, you will
probably have your answer to why the other system did not receive
it.
If you're still not sure what's going on, post another pastebin with
sofia loglevel set to 9.


On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan@myrvold.org>
wrote:
Quote:
Oh, what happened to it?
Anyway, here is a new pb:
http://pastebin.freeswitch.org/10867
Ivan
Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:


On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan@myrvold.org>
wrote:
Quote:

Here is a debug log from a call from an internal phone out to an
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578

Ivan

Uh... you wanna try that PB number again?
-MC

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--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from
our
children." ~David Brower

"I decided the words were too conservative for me. We're not
borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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ivan at myrvold.org
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PostPosted: Sun Nov 01, 2009 5:52 am    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

No one have any idea why this is not working? I have combed through
the log, but couldn't find any clue there.
Incoming calls from my sip provider is working perfect, but for
outgoing calls it looks like Freeswitch is not letting the incoming
rtp to the local sip phone.

Ivan

On 30. okt. 2009, at 21:26, Ivan C Myrvold wrote:

Quote:
Yes, now I got a more detailed trace. Thank you for helping me with
this.

A new pastebin at http://pastebin.freeswitch.org/10905

Ivan

Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable:

Quote:
fsctl loglevel debug
console loglevel debug
sofia profile internal siptrace on
sofia profile external siptrace on
sofia loglevel all 9
^^^^^^^^^^^^^^^^^^^^^

Then run your call, then do this:

sofia loglevel all 0
sofia profile external siptrace off
sofia profile internal siptrace off
fsctl loglevel warning
console loglevel warning

On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold <ivan@myrvold.org>
wrote:
Quote:
I have already set debug to 9, on both profiles.

Ivan


Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:

Quote:
See that 200 OK that keeps coming in over and over and over and
over
again? That's because they never received your ACK. If you can
turn on
sofia loglevel to 9 and then watch where you send the ACK, you will
probably have your answer to why the other system did not receive
it.
If you're still not sure what's going on, post another pastebin
with
sofia loglevel set to 9.


On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan@myrvold.org>
wrote:
Quote:
Oh, what happened to it?
Anyway, here is a new pb:
http://pastebin.freeswitch.org/10867
Ivan
Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:


On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan@myrvold.org>
wrote:
Quote:

Here is a debug log from a call from an internal phone out to an
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578

Ivan

Uh... you wanna try that PB number again?
-MC

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--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from
our
children." ~David Brower

"I decided the words were too conservative for me. We're not
borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

_______________________________________________
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--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from
our
children." ~David Brower

"I decided the words were too conservative for me. We're not
borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

_______________________________________________
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FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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PostPosted: Sun Nov 01, 2009 3:29 pm    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

Session-Expires: -1;refresher=uas

nta: 200 OK has fatal syntax errors

This is a know-bug in asterisk.

see: https://issues.asterisk.org/view.php?id=15621



On Sun, Nov 1, 2009 at 4:40 AM, Ivan C Myrvold <ivan@myrvold.org (ivan@myrvold.org)> wrote:
Quote:
No one have any idea why this is not working? I have combed through
the log, but couldn't find any clue there.
Incoming calls from my sip provider is working perfect, but for
outgoing calls it looks like Freeswitch is not letting the incoming
rtp to the local sip phone.

Ivan


On 30. okt. 2009, at 21:26, Ivan C Myrvold wrote:

Quote:
Yes, now I got a more detailed trace. Thank you for helping me with
this.

A new pastebin at http://pastebin.freeswitch.org/10905

Ivan

Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable:

Quote:
fsctl loglevel debug
console loglevel debug
sofia profile internal siptrace on
sofia profile external siptrace on
sofia loglevel all 9
^^^^^^^^^^^^^^^^^^^^^

Then run your call, then do this:

sofia loglevel all 0
sofia profile external siptrace off
sofia profile internal siptrace off
fsctl loglevel warning
console loglevel warning

On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold <ivan@myrvold.org (ivan@myrvold.org)>
wrote:
Quote:
I have already set debug to 9, on both profiles.

Ivan


Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:

Quote:
See that 200 OK that keeps coming in over and over and over and
over
again? That's because they never received your ACK. If you can
turn on
sofia loglevel to 9 and then watch where you send the ACK, you will
probably have your answer to why the other system did not receive
it.
If you're still not sure what's going on, post another pastebin
with
sofia loglevel set to 9.


On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan@myrvold.org (ivan@myrvold.org)>
wrote:
Quote:
Oh, what happened to it?
Anyway, here is a new pb:
http://pastebin.freeswitch.org/10867
Ivan
Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:


On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan@myrvold.org (ivan@myrvold.org)>
wrote:
Quote:

Here is a debug log from a call from an internal phone out to an
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578

Ivan

Uh... you wanna try that PB number again?
-MC

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--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from
our
children." ~David Brower

"I decided the words were too conservative for me. We're not
borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

_______________________________________________
FreeSWITCH-users mailing list
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--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from
our
children." ~David Brower

"I decided the words were too conservative for me. We're not
borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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 ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
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ivan at myrvold.org
Guest





PostPosted: Mon Nov 02, 2009 4:17 pm    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

That was it. My sip provider applied the patch to his Asterisk server that was referenced in the link you was so kind to provide, and again everything worked as it should.

Thank you very much!


Ivan

Den 1. nov. 2009 kl. 21:20 skrev Anthony Minessale:
Quote:
Session-Expires: -1;refresher=uas

nta: 200 OK has fatal syntax errors

This is a know-bug in asterisk.

see: https://issues.asterisk.org/view.php?id=15621



On Sun, Nov 1, 2009 at 4:40 AM, Ivan C Myrvold <ivan@myrvold.org (ivan@myrvold.org)> wrote:
Quote:
No one have any idea why this is not working? I have combed through
the log, but couldn't find any clue there.
Incoming calls from my sip provider is working perfect, but for
outgoing calls it looks like Freeswitch is not letting the incoming
rtp to the local sip phone.

Ivan


On 30. okt. 2009, at 21:26, Ivan C Myrvold wrote:

Quote:
Yes, now I got a more detailed trace. Thank you for helping me with
this.

A new pastebin at http://pastebin.freeswitch.org/10905

Ivan

Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable:

Quote:
fsctl loglevel debug
console loglevel debug
sofia profile internal siptrace on
sofia profile external siptrace on
sofia loglevel all 9
^^^^^^^^^^^^^^^^^^^^^

Then run your call, then do this:

sofia loglevel all 0
sofia profile external siptrace off
sofia profile internal siptrace off
fsctl loglevel warning
console loglevel warning

On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold <ivan@myrvold.org (ivan@myrvold.org)>
wrote:
Quote:
I have already set debug to 9, on both profiles.

Ivan


Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:

Quote:
See that 200 OK that keeps coming in over and over and over and
over
again? That's because they never received your ACK. If you can
turn on
sofia loglevel to 9 and then watch where you send the ACK, you will
probably have your answer to why the other system did not receive
it.
If you're still not sure what's going on, post another pastebin
with
sofia loglevel set to 9.


On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold <ivan@myrvold.org (ivan@myrvold.org)>
wrote:
Quote:
Oh, what happened to it?
Anyway, here is a new pb:
http://pastebin.freeswitch.org/10867
Ivan
Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:


On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold <ivan@myrvold.org (ivan@myrvold.org)>
wrote:
Quote:

Here is a debug log from a call from an internal phone out to an
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578

Ivan

Uh... you wanna try that PB number again?
-MC

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--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from
our
children." ~David Brower

"I decided the words were too conservative for me. We're not
borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

_______________________________________________
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--
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from
our
children." ~David Brower

"I decided the words were too conservative for me. We're not
borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

_______________________________________________
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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 ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
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msc at freeswitch.org
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PostPosted: Mon Nov 02, 2009 4:26 pm    Post subject: [Freeswitch-users] SIP provider with extern rtp server Reply with quote

On Mon, Nov 2, 2009 at 12:58 PM, Ivan C Myrvold <ivan@myrvold.org (ivan@myrvold.org)> wrote:
Quote:
That was it. My sip provider applied the patch to his Asterisk server that was referenced in the link you was so kind to provide, and again everything worked as it should.

Thank you very much!




This is why Tony's Asterisk karma is still so high!
-MC
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