Hi, If you post your configuration someone may help you.
On Sat, Sep 28, 2013 at 5:03 PM, Sean Darcy <[email protected]> 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 > > > -- > ______________________________**______________________________**_________ > -- 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<http://lists.digium.com/mailman/listinfo/asterisk-users> >
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