Hello Asterisk users
:-)
Server: Asterisk 11.7.0~dfsg-1ubuntu1 on Raspberry
Client: Zoiper on Android device
If "Transport=tcp" everything works fine, without any trouble.
If "Transport=tls" registration is fine. But after a few minutes, the
peer continuously is "reachable...unreachable".
Maybe shut off qualify for the peer? I think I tried twinkle a few
years ago and it didna (yes didna) like the qualify packet. the sip
options qualify packet is only needed to keep the UDP state tables in a
firewall if the peer is remote
--
Zitat von Adrian Serafini adrian-li...@wombit.com:
Maybe shut off qualify for the peer? I think I tried twinkle a few
years ago and it didna (yes didna) like the qualify packet. the sip
options qualify packet is only needed to keep the UDP state tables
in a firewall if the peer is remote
Hi list!
I have a problem and I hope someone can help me...
I configured an Asterisk on a VM to serve more accounts and act as a proxy to
other SIP-providers.
The first account running on my phone works without any problem.
A second account, running on the phone of my wife, is always
What kind of phone are we talking about, both yours that works and
your
wife's that does not?
Right!
Can you ping the unreachable phone and does it respond to a ping?
I can ping both phones from the VM
Many phones will have a network test function built in to them to help
you
Kevin Larsen kevin.lar...@pioneerballoon.com schrieb:
I am still curious why you have both an Asterisk setup and an AsteriskNow
setup? Is that just to play around with? At the end of the day you should
just need one or the other.
Just why I need a second SIP-provider to check if all works,
Kevin Larsen kevin.lar...@pioneerballoon.com schrieb:
What kind of phone are we talking about, both yours that works and your
wife's that does not?
Right!
Can you ping the unreachable phone and does it respond to a ping?
I can ping both phones from the VM
Many phones will have a network
Ahh. Seen that before! That suggests to me that you don't have your
sip.conf records setup right.
What's your sip.conf look like?
On 15-05-28 04:51 PM, Luca Bertoncello wrote:
Kevin Larsen kevin.lar...@pioneerballoon.com schrieb:
The phone you gave your wife is really old. Are you sure it
Darryl Moore dar...@moores.ca schrieb:
I think your phone may be trying to register with the username '1234',
while your sip configuration is expecting 'luca'. Can you try changing
your phone registration credentials to use 'luca'? Can you give us a sip
transcript when you try to place a
I have a problem and I hope someone can help me...
I configured an Asterisk on a VM to serve more accounts and act as a
proxy to
other SIP-providers.
The first account running on my phone works without any problem.
A second account, running on the phone of my wife, is always
UNREACHABLE.
Darryl Moore dar...@moores.ca schrieb:
I'd start by turning on sip debugging in asterisk
sip set debug ip [your_phone_ip]
Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d@172.
16.34.133' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.200.11:5060:
OPTIONS
I think your phone may be trying to register with the username '1234',
while your sip configuration is expecting 'luca'. Can you try changing
your phone registration credentials to use 'luca'? Can you give us a sip
transcript when you try to place a call from it?
On 15-05-28 05:09 PM, Luca
Darryl Moore dar...@moores.ca schrieb:
Ahh. Seen that before! That suggests to me that you don't have your
sip.conf records setup right.
What's your sip.conf look like?
Well, here what I wrote in my sip.conf:
register = 004935:MYSECRET@pbxluca/004935
register =
I'd start by turning on sip debugging in asterisk
sip set debug ip [your_phone_ip]
and use tcpdump or wireshark to see what the OS sees
tcpdump host [your_phone_ip] and udp port 5060
On 15-05-28 03:58 PM, Luca Bertoncello wrote:
Hi list!
I have a problem and I hope someone can help me...
Darryl Moore dar...@moores.ca schrieb:
I think your phone may be trying to register with the username '1234',
while your sip configuration is expecting 'luca'. Can you try changing
your phone registration credentials to use 'luca'? Can you give us a sip
transcript when you try to place a
No, I'm not sure.
And no, I can't make any call, right now... At least, not connected to
my
Asterisk...
If I connect it to the other VM with AsteriskNOW I can call my Twinkle,
but
NOT my phone connected on my Asterisk, using the proxy.
I can see that in the log:
[May 28 22:49:51]
Kevin Larsen kevin.lar...@pioneerballoon.com schrieb:
Can you post the Manufacturer and Model of your phones (both of them if
they are different)? That will help us look up what diagnostics/log files
there might be on the phones.
Of course!
My phone is a Thomson ST2022 and my wife has a
Darryl Moore dar...@moores.ca schrieb:
I'd start by turning on sip debugging in asterisk
sip set debug ip [your_phone_ip]
Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d@172.16.34.133'
Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.200.11:5060:
OPTIONS
Kevin Larsen kevin.lar...@pioneerballoon.com schrieb:
The phone you gave your wife is really old. Are you sure it supports SIP
OPTIONS? Can you make a call in or out to it? If you can, it is more
likely that it just doesn't support that and you can't use a qualify
statement.
No, I'm not
Dear Asterisker,
I am using java-asterisk-IAX, java based application, that implemented the IAX2
protocol.
I am registred with this applet to my asterisk; when i try to show iax2 peers
in my asterisk, i get my status UNREACHABLE.
I have no firewall in my network; i tried qualify=3000 and still
Hi,
I have 3 friends trying to connect to my Asterisk using x-lite, all of them
are using 3 dif. adsl-provider.
For each of them I got this in sip.conf:
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=g729
allow=g723.1
[seholm]
type=friend
secret=**
auth=md5
nat=yes
host=dynamic
Hans-Henrik Andresen wrote:
Hi,
I have 3 friends trying to connect to my Asterisk using x-lite, all
of them are using 3 dif. adsl-provider.
For each of them I got this in sip.conf:
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=g729
allow=g723.1
[seholm]
type=friend
Senad Jordanovic wrote:
Hans-Henrik Andresen wrote:
Hi,
I have 3 friends trying to connect to my Asterisk using x-lite, all
of them are using 3 dif. adsl-provider.
For each of them I got this in sip.conf:
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=g729
allow=g723.1
[seholm]
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