Re: [asterisk-users] Problem using Android SIP-Client

[Luca Bertoncello]

Harel Cohen  schrieb:

> Is the Sophos a home router or professional one? In many cases what home

Of course the professional firewalls (we have two Sophos in Cluster, to
manage our two SDSLs)

> router does by default needs to be configured manually on professional one.
> E.G. a home router will allow outgoing sessions and create a return path by
> default where professional one won't.
> Two things I would look for:
> 1. Look for, and disable, ALG for SIP. The idea of ALG is nice but I
> haven't encountered a device that implements this properly (I'm not a
> network expert so it doesn't mean that there isn't such a router out there).
> 2. On the Sophos try to statically open the UDP port range of your RTP to
> outgoing traffic from your phone to your SIP server. Note that outgoing
> port range is what your SIP server defines as its port range, not your
> phone. If you get one way voice (remote hears phone) then you are on the
> right direction. You'll then need to open the incoming ports too for the
> ports that your phone is expecting to get its RTP from.

OK, tomorrow I'll check it...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Problem using Android SIP-Client

[Harel Cohen]

Hi,
Is the Sophos a home router or professional one? In many cases what home
router does by default needs to be configured manually on professional one.
E.G. a home router will allow outgoing sessions and create a return path by
default where professional one won't.
Two things I would look for:
1. Look for, and disable, ALG for SIP. The idea of ALG is nice but I
haven't encountered a device that implements this properly (I'm not a
network expert so it doesn't mean that there isn't such a router out there).
2. On the Sophos try to statically open the UDP port range of your RTP to
outgoing traffic from your phone to your SIP server. Note that outgoing
port range is what your SIP server defines as its port range, not your
phone. If you get one way voice (remote hears phone) then you are on the
right direction. You'll then need to open the incoming ports too for the
ports that your phone is expecting to get its RTP from.
KR
Harel
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Re: [asterisk-users] Problem using Android SIP-Client

[Andre Gronwald]

the issue is quiet sure codec based:

[Oct 24 15:32:41] NOTICE[3731]: channel.c:4280 __ast_read: Dropping
incompatible voice frame on SIP/messagenet-028e of format gsm since our
native format has changed to 0x8 (alaw)

shorter:
Dropping incompatible voice frame on SIP/messagenet-028e of format 
gsm since our

native format has changed to 0x8 (alaw)

looks like your android phone uses gsm, but only alaw is supported. just 
try to put gsm as allowed codec.


the sip invite and ok-message would be interesting as well (especially sdp)-

regards,
andre


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Re: [asterisk-users] Problem using Android SIP-Client

[Luca Bertoncello]

Luca Bertoncello  schrieb:

Hallo again

> I configured an user for my mobile phone and I can call, but as soon  
> as the other party answer, I get this error in Log:
> 
> [Oct 24 15:32:41] NOTICE[3731]: channel.c:4280 __ast_read: Dropping  
> incompatible voice frame on SIP/messagenet-028e of format gsm  
> since our native format has changed to 0x8 (alaw)
> 
> and I can't hear anything...

I tried to call the same number I called before using LTE instead of WLAN,
and it worked...
Then I tried to call the same number again using my WLAN at home, and it
worked again.

So, I must conclude that the problem is somewhere in the WLAN at office...
Very curiously I can initiate the SIP-communication, but as soon as the other
party answer the connection will be closed...

Since I'm one of the admins at office, I'd like to solve this problem.
Can someone give me some advice what can be wrong in our firewall (Sophos)?

Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Problem using Android SIP-Client

[Marcelo Terres]

You should try another SIP client, just to check it. (Zoiper or
cSipSimple, for example).

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 24 October 2017 at 14:42, Luca Bertoncello  wrote:
> Hi list!
>
> I use Asterisk 1.8.30.0 on an OpenWRT-Router (I know, not the last version,
> but I can't upgrade).
> It always runned very well, and it runs very well with our home phones, too,
> but now I have problems using the native Android SIP-Client...
>
> I configured an user for my mobile phone and I can call, but as soon as the
> other party answer, I get this error in Log:
>
> [Oct 24 15:32:41] NOTICE[3731]: channel.c:4280 __ast_read: Dropping
> incompatible voice frame on SIP/messagenet-028e of format gsm since our
> native format has changed to 0x8 (alaw)
>
> and I can't hear anything...
>
> This is the configuration of the user:
>
> [00491771234567]
> fullname = 00491771234567
> secret = MYVERYSECRET
> dahdichan = 1
> hassip = yes
> hasiax = no
> hash323 = no
> hasmanager = no
> callwaiting = no
> context = default
> host = dynamic
> dtmfmode=rfc2833
> canreinvite=no
> sendrpid=pai
> type=friend
> ;nat=force_rport,comedia
> nat=yes
> qualify=yes
> qualifyfreq=60
> ;transport=Auto
> avpf=no
> force_avp=no
> icesupport=no
> encryption=no
> callgroup=1
> pickupgroup=1
> dial=SIP/00491771234567
> allow = all
>
> Any idea?
> The user worked very well with my old mobile phone (Android 4), I __THINK__
> the problem happens since I use my new phone with Android 7...
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
>
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>
> New to Asterisk? Start here:
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[asterisk-users] Problem using Android SIP-Client

[Luca Bertoncello]

Hi list!

I use Asterisk 1.8.30.0 on an OpenWRT-Router (I know, not the last  
version, but I can't upgrade).
It always runned very well, and it runs very well with our home  
phones, too, but now I have problems using the native Android  
SIP-Client...


I configured an user for my mobile phone and I can call, but as soon  
as the other party answer, I get this error in Log:


[Oct 24 15:32:41] NOTICE[3731]: channel.c:4280 __ast_read: Dropping  
incompatible voice frame on SIP/messagenet-028e of format gsm  
since our native format has changed to 0x8 (alaw)


and I can't hear anything...

This is the configuration of the user:

[00491771234567]
fullname = 00491771234567
secret = MYVERYSECRET
dahdichan = 1
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = default
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
;nat=force_rport,comedia
nat=yes
qualify=yes
qualifyfreq=60
;transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=1
pickupgroup=1
dial=SIP/00491771234567
allow = all

Any idea?
The user worked very well with my old mobile phone (Android 4), I  
__THINK__ the problem happens since I use my new phone with Android 7...


Thanks
Luca Bertoncello
(lucab...@lucabert.de)


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