On 09/30/2013 12:09 PM, Sean Darcy wrote:
On 09/28/2013 11:11 AM, Asghar Mohammad wrote:
Hi,
If you post your configuration someone may help you.


On Sat, Sep 28, 2013 at 5:03 PM, Sean Darcy <seandar...@gmail.com
<mailto:seandar...@gmail.com>> wrote:

    On 09/27/2013 09:08 PM, Sean Darcy wrote:

        We have zoiper connected over iax to asterisk in Sydney. The
        call is to
        asterisk in New York. The caller in NZ can hear clearly. Nothing
        in NY.

        Here's the sydney server:

        -- Accepting AUTHENTICATED call from <zoiperipaddr>:
                 > requested format = speex,
                 > requested prefs = (),
                 > actual format = ulaw,
                 > host prefs = (silk16|ulaw|gsm|g722),
                 > priority = mine
              -- Executing [8447@nz-in:1] Dial("IAX2/n4-270",
        "IAX2/sydney") in
        new stack
              -- Called IAX2/sydney
              -- Call accepted by <nyipaddr> (format ulaw)
              -- Format for call is (ulaw)
              -- IAX2/sydney-8819 is ringing
              -- IAX2/sydney-8819 answered IAX2/n4-270
              -- Channel 'IAX2/n4-270' unable to transfer
              -- Channel 'IAX2/sydney-8819' unable to transfer
              -- Channel 'IAX2/sydney-8819' unable to transfer
              -- Channel 'IAX2/sydney-8819' unable to transfer

        The NY server:

             -- Accepting AUTHENTICATED call from <sydneyipaddr>:
              --        > requested format = ulaw,
              --        > requested prefs = (ulaw|silk16|gsm|g722),
              --        > actual format = ulaw,
              --        > host prefs = (ulaw|gsm|g722),
              --        > priority = mine
              -- Executing [s@incoming-nz:1] Goto("IAX2/home-2152",
        "incoming,s,nz-in") in new stack
              -- Goto (incoming,s,5)
              -- Executing [s@incoming:5] Dial("IAX2/home-2152",
        "DAHDI/g0&SIP/250&SIP/251,60,__tT") in new stack
            == Using SIP RTP TOS bits 184
            == Using SIP RTP CoS mark 5
            == Using SIP RTP TOS bits 184
            == Using SIP RTP CoS mark 5
              -- Called DAHDI/g0
              -- Called SIP/250
              -- Called SIP/251
              -- DAHDI/1-1 is ringing
              -- SIP/251-0000001d is ringing
              -- SIP/250-0000001c is ringing
              -- DAHDI/1-1 is ringing
              -- DAHDI/1-1 answered IAX2/home-2152
              -- Channel 'IAX2/home-2152' unable to transfer
              -- Hanging up on 'DAHDI/1-1'

        Any help appreciated.

        sean



    FWIW, sydney server is 11.5.1, ny server 11.6.0-rc1.

    sean



Thanks for the reply.

Here's sydney iax.conf:

[general]
bandwidth=medium

trunkmtu=1240
disallow=all
allow=silk16
allow=ulaw
allow=gsm
allow=g722
jitterbuffer=yes
forcejitterbuffer=no
trunktimestamps=yes

authdebug=yes

tos=ef
cos=5
autokill=yes
codecpriority=caller

[default](!)
type=friend
auth=md5
host=dynamic
context=nz-in
qualify=1000
setvar=Protocol=IAX2

[n4](default)
secret=<n4pw>
callerid="<callerid>"

[sydney](default)
secret=<pwsydney>
username=home-sydney


home iax.conf:

[general]
bandwidth=medium
disallow=all
allow=ulaw
allow=gsm
allow=g722
jitterbuffer=yes
forcejitterbuffer=no

tos=0x10
autokill=yes

register => sydney:<pwsydney>@<sydneyipaddr>

[nz](!)
type=friend
secret=<pwhome>
context=incoming-nz

[home-sydney](nz)
host=<sydneyipaddr>
username=sydney
callerid="House"

sean




Any thoughts? Anybody?

sean


--
_____________________________________________________________________
-- 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

Reply via email to