Can anyone shed any light on this? I've set up 3 trunks - Voiptalk, Voipfone
and Sipgate. With Voipfone and Sipgate I get 2 way audio but with Voiptalk
the other party can hear me but I can't hear them.
Voiptalk CS won't support trixbox at the moment, although they helpfully
sent me a config that another customer had used successfully. If the problem
was across all Carriers I'd suspect a NAT or port forwarding issue but with
just one I'm scratching my head a bit.
Telephoneman wrote:
> Can anyone shed any light on this? I've set up 3 trunks - Voiptalk, Voipfone
> and Sipgate. With Voipfone and Sipgate I get 2 way audio but with Voiptalk
> the other party can hear me but I can't hear them.
>
> Voiptalk CS won't support trixbox at the moment, although they helpfully
> sent me a config that another customer had used successfully. If the problem
> was across all Carriers I'd suspect a NAT or port forwarding issue but with
> just one I'm scratching my head a bit.
Are you using SIP on Sipgate & Voipfone but IAX with Voiptalk?
If so, IAX requires a different (single) port to operate.
Have you forwarded TCP / UDP or both?
Looking at their site it appears they support Asterisk, tell them your
using Asterisk and get them to tell you what should be in the *.conf
files and what ports should be set. Make a note, then add it to the
freepbx interface later (will overight *.conf files).
Telephoneman wrote:
> Can anyone shed any light on this? I've set up 3 trunks - Voiptalk, Voipfone
> and Sipgate. With Voipfone and Sipgate I get 2 way audio but with Voiptalk
> the other party can hear me but I can't hear them.
>
> Voiptalk CS won't support trixbox at the moment, although they helpfully
> sent me a config that another customer had used successfully. If the problem
> was across all Carriers I'd suspect a NAT or port forwarding issue but with
> just one I'm scratching my head a bit.
>
Calling Jono, What did you do to get Voiptalk working?
Liam's tried my settings which work fine here, but gets this one-way
audio issue.
Funnily, I tried my settings from my normal Trixbox in my experimental
Vmware version, and I'm getting this same issue......
paul123 wrote:
> Telephoneman wrote:
> > Can anyone shed any light on this? I've set up 3 trunks - Voiptalk, Voipfone
> > and Sipgate. With Voipfone and Sipgate I get 2 way audio but with Voiptalk
> > the other party can hear me but I can't hear them.
> >
> > Voiptalk CS won't support trixbox at the moment, although they helpfully
> > sent me a config that another customer had used successfully. If the problem
> > was across all Carriers I'd suspect a NAT or port forwarding issue but with
> > just one I'm scratching my head a bit.
> >
>
> Calling Jono, What did you do to get Voiptalk working?
>
> Liam's tried my settings which work fine here, but gets this one-way
> audio issue.
>
> Funnily, I tried my settings from my normal Trixbox in my experimental
> Vmware version, and I'm getting this same issue......
Sorry to answer my own post, but it now works on the vmware version
too.
In config edit->sip_nat.conf I had to add:
externip=myfixedIP (not my dyndnsname)
localnet=192.168.1.0/255.255.255.0 (this would be the same on your
716g)
"paul123" <paul@redy.net> wrote in message
news:1157107929.164849.36800@b28g2000cwb.googlegro ups.com...
>
> paul123 wrote:
>> Telephoneman wrote:
>> > Can anyone shed any light on this? I've set up 3 trunks - Voiptalk,
>> > Voipfone
>> > and Sipgate. With Voipfone and Sipgate I get 2 way audio but with
>> > Voiptalk
>> > the other party can hear me but I can't hear them.
>> >
>> > Voiptalk CS won't support trixbox at the moment, although they
>> > helpfully
>> > sent me a config that another customer had used successfully. If the
>> > problem
>> > was across all Carriers I'd suspect a NAT or port forwarding issue but
>> > with
>> > just one I'm scratching my head a bit.
>> >
>>
>> Calling Jono, What did you do to get Voiptalk working?
>>
>> Liam's tried my settings which work fine here, but gets this one-way
>> audio issue.
>>
>> Funnily, I tried my settings from my normal Trixbox in my experimental
>> Vmware version, and I'm getting this same issue......
>
> Sorry to answer my own post, but it now works on the vmware version
> too.
>
> In config edit->sip_nat.conf I had to add:
>
> externip=myfixedIP (not my dyndnsname)
> localnet=192.168.1.0/255.255.255.0 (this would be the same on your
> 716g)
>
> Calls in and out are working fine
Hi Paul,
I had the above in sip.conf under [general] but with the dyndns name. I
removed it and put it in sip_nat.conf with the IP address. Still no joy.
Re-booted linux/asterisk and made 1 call on Voiptalk - still one-way audio
but subsequent attempts to call on this trunk get the asterisk lady telling
me it's busy...
Jono wrote:
> on 01/09/2006, paul123 supposed :
> > Calling Jono, What did you do to get Voiptalk working?
>
> Alas, nothing. It's not a provider I use.
Aha, never mind, it's just that I had assumed you'd got it working -
voiptalk had come up in an earlier thread (about the free signup for a
geo number), where you'd had some probs.
paul123 laid this down on his screen :
> Jono wrote:
>> on 01/09/2006, paul123 supposed :
>>> Calling Jono, What did you do to get Voiptalk working?
>>
>> Alas, nothing. It's not a provider I use.
>
> Aha, never mind, it's just that I had assumed you'd got it working -
> voiptalk had come up in an earlier thread (about the free signup for a
> geo number), where you'd had some probs.
>
> Not to worry, we'll get there in the end.
Bugger. Yes, I was thinking "voipfone". You're right. Duh!
I had to play around with getting the inbound DID working, I can't say
there was much wrong with the audio, however, here're my trunk
settings:
I've also got this in extensions_custom.conf:
[custom-voiptalk]
exten => 12345678,1,Goto(ext-did,12345678,1)
I have also created an inbound route for 12345678.
I think I have a bit more in sip.conf:
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
context = from-trunk
;defaultexpirey = 600 ; include this only if necessary
;maxexpirey = 3600 ; include this only if necessary
progressinband = yes
dtmfmode=auto
Jono wrote:
> paul123 laid this down on his screen :
> > Jono wrote:
> >> on 01/09/2006, paul123 supposed :
> >>> Calling Jono, What did you do to get Voiptalk working?
> >>
> >> Alas, nothing. It's not a provider I use.
> >
> > Aha, never mind, it's just that I had assumed you'd got it working -
> > voiptalk had come up in an earlier thread (about the free signup for a
> > geo number), where you'd had some probs.
> >
> > Not to worry, we'll get there in the end.
>
> Bugger. Yes, I was thinking "voipfone". You're right. Duh!
<snip>
You're not the only one to mix those two up. Easily done! I was
recently in email contact with someone and I sent them to voipfone
instead of voiptalk. Not surprisingly, they couldn't find what I was
talking about.
I didn't write my configs myself! I got them here from.....paul123
<http://groups.google.co.uk/group/uk.telecom.voip/tree/browse_frm/thread/6f602f2a3e0468a1/e90748215e800e83?rnum=21&hl=en&q=unbeatable+voipta lk+offer%3F&_done=%2Fgroup%2Fuk.telecom.voip%2Fbro wse_frm%2Fthread%2F6f602f2a3e0468a1%2F676376528958 64be%3Ftvc%3D1%26q%3Dunbeatable+voiptalk+offer%3F% 26hl%3Den%26#doc_3fae756fba793790>
.......paul123, why are you asking me how I got voiptalk working, when
you told me?
Jono wrote:
> linker3000 pretended :
> > Jono wrote:
> >
> > defaultexpirey=160
> >
> >
> > Eh?
>
> Hmm.
>
> I didn't write my configs myself! I got them here from.....paul123
> <http://groups.google.co.uk/group/uk.telecom.voip/tree/browse_frm/thread/6f602f2a3e0468a1/e90748215e800e83?rnum=21&hl=en&q=unbeatable+voipta lk+offer%3F&_done=%2Fgroup%2Fuk.telecom.voip%2Fbro wse_frm%2Fthread%2F6f602f2a3e0468a1%2F676376528958 64be%3Ftvc%3D1%26q%3Dunbeatable+voiptalk+offer%3F% 26hl%3Den%26#doc_3fae756fba793790>
>
> ......paul123, why are you asking me how I got voiptalk working, when
> you told me?
lol, :) yes, I know, but I thought that you had initial audio problems
too and that maybe you'd had to adjust something to get it going OK.
But now, looking back at that thread, I see that it was dialling format
that was the problem.