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[asterisk-users] Connecting two Asterisk

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lucabert at lucabert.de
Guest

Posted: Sun Jun 07, 2015 3:08 pm Post subject: [asterisk-users] Connecting two Asterisk

Hi again!

I always try to get my mobile phone work with my Asterisk.
I tried to install Asterisk on my PC (with public IP), but it has problems,
too...
I think, my UMTS-Provider doesn't want to connect to dynamic IP or my DSL-Provider
does not want it, too, since I have no problem to connect and get a very good
audio quality if I connect to other SIP-Provider or to an Asterisk (SAME
configuration!!) installed on my Server...

Well, I will try to configure the Asterisk on my Server to act as "proxy" so
that all phones at home talk with my Asterisk at home (now called "wrt", and
my mobile phone talk with my Asterisk on my server (now called "lucabert").

I followed this HowTo http://sysmagazine.com/posts/125303/ and I got both
Servers talking together.

I can call my mobile phone (logged in at "lucabert") from a phone logged in
on "wrt" and a phone at "wrt" from my mobile phone at "lucabert".
Wonderful!

Now the problem: on my phones at "wrt" I can hear what the mobile phone at
"lucabert" sends (with a very good audio-quality), but on this mobile phone
I cannot hear a single word spoken with the phone at "wrt", not even the music
on hold I configured...

When I call my mobile phone from a phone logged on at "wrt" I see on the
Asterisk at "wrt":

== Using SIP RTP CoS mark 5
-- Executing [4@default:1] Verbose("SIP/00493511111111-0000001e", "2,Internal call for Mobile - [00493511111111]") in new stack
== Internal call for Mobile - [00493511111111]
-- Executing [4@default:2] Dial("SIP/00493511111111-0000001e", "IAX2/lucabert:MYVERYSECRET@lucabert/00491773333333,,R") in new stack
-- Called IAX2/lucabert:MYVERYSECRET@lucabert/00491773333333
-- Call accepted by X.Y.Z.K (format gsm)
-- Format for call is gsm
-- IAX2/lucabert-1298 is ringing
-- IAX2/lucabert-1298 answered SIP/00493511111111-0000001e
-- Started music on hold, class 'default', on IAX2/lucabert-1298
-- Stopped music on hold on IAX2/lucabert-1298
-- Hungup 'IAX2/lucabert-1298'
== Spawn extension (default, 4, 2) exited non-zero on 'SIP/00493511111111-0000001e'

On the Asterisk at "lucabert" I see:

-- Accepting AUTHENTICATED call from A.B.C.D:

Quote:

requested format = ulaw,
requested prefs = (ulaw|gsm|g729|alaw),
actual format = gsm,
host prefs = (gsm|g729|alaw|ulaw),
priority = mine

-- Executing [00491773333333@default:1] Macro("IAX2/lucabert-94", "stdexten,00491773333333,SIP/00491773333333&DAHDI/1") in new stack
[Jun 7 21:59:09] WARNING[19888]: app_macro.c:302 _macro_exec: No such context 'macro-stdexten' for macro 'stdexten'
-- Executing [00491773333333@default:2] Set("IAX2/lucabert-94", "CHANNEL(musicclass)=default") in new stack
-- Executing [00491773333333@default:3] Dial("IAX2/lucabert-94", "SIP/00491773333333,,R") in new stack
== Using SIP RTP CoS mark 5
-- Called 00491773333333
-- SIP/00491773333333-00000008 is ringing
-- SIP/00491773333333-00000008 is ringing
-- SIP/00491773333333-00000008 is ringing
-- SIP/00491773333333-00000008 answered IAX2/lucabert-94
== Spawn extension (default, 00491773333333, 3) exited non-zero on 'IAX2/lucabert-94'
-- Hungup 'IAX2/lucabert-94'

Well, I'm very puzzled...
Can someone help me?

Thank you very much!
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

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lucabert at lucabert.de
Guest

Posted: Sun Jun 07, 2015 3:51 pm Post subject: [asterisk-users] Connecting two Asterisk

Some other data...

I changed both iax.conf and wrote:

bandwidth=high
allow=all

Now I see in the log:

Quote:

requested format = ulaw,
requested prefs = (),
actual format = ulaw,
host prefs = (),
priority = mine

and I can hear somewhat, but with a VERY poor quality on my mobile phone...
On the other phone however, the quality is very good...

I'm very very puzzled...

Thanks for any help!
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

asterisk.org at sedwar...
Guest

Posted: Sun Jun 07, 2015 5:01 pm Post subject: [asterisk-users] Connecting two Asterisk

On Sun, 7 Jun 2015, Luca Bertoncello wrote:

Quote:

I think, my UMTS-Provider doesn't want to connect to dynamic IP or my DSL-Provider
does not want it, too, since I have no problem to connect and get a very good
audio quality if I connect to other SIP-Provider or to an Asterisk (SAME
configuration!!) installed on my Server...

Maybe fiddling with the SIP and RTP ports would help.

--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

asterisk.org at sedwar...
Guest

Posted: Sun Jun 07, 2015 5:03 pm Post subject: [asterisk-users] Connecting two Asterisk

On Sun, 7 Jun 2015, Luca Bertoncello wrote:

Quote:

Now the problem: on my phones at "wrt" I can hear what the mobile phone at
"lucabert" sends (with a very good audio-quality), but on this mobile phone
I cannot hear a single word spoken with the phone at "wrt", not even the music
on hold I configured...

Quote:

-- Call accepted by X.Y.Z.K (format gsm)
-- Format for call is gsm

I thought GSM regurgitated by cell had issues. Can you try alaw/ulaw?

--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

lucabert at lucabert.de
Guest

Posted: Mon Jun 08, 2015 12:19 am Post subject: [asterisk-users] Connecting two Asterisk

Steve Edwards <asterisk.org@sedwards.com> schrieb:

Quote:

On Sun, 7 Jun 2015, Luca Bertoncello wrote:

Quote:

Now the problem: on my phones at "wrt" I can hear what the mobile phone at
"lucabert" sends (with a very good audio-quality), but on this mobile
phone I cannot hear a single word spoken with the phone at "wrt", not
even the music on hold I configured...

Quote:

-- Call accepted by X.Y.Z.K (format gsm)
-- Format for call is gsm

I thought GSM regurgitated by cell had issues. Can you try alaw/ulaw?

Yes, I do, but the quality is always very poor...

Thanks
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

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