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Sip Trunk Failing To Register Causes Sip Phones To Become Unreachable


I have only seen this problem when using SIPgate SIP trunks which actually “register”. If the ADSL connection goes down that the sip trunk uses, the sip phones registered locally become unreachable. This happens on any 1.6.x or 1.8 version of asterisk I’ve tried. Is there a work around that doesn’t involve putting an opensips server between the asterisk server and the sip trunk?


Regards, John

PRI trunk between Asterisk servers does not work.

In article <4FECCD0C.1020000@fivecats.org>,
James Sharp wrote:
> On 6/28/2012 3:53 PM, Ernie Dunbar wrote:
> > We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
> > Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen),
> > and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our
> > PRI to the PSTN and we hope will allow us to failover to other Asterisk
> > servers (ie, Voip2 and Voip3). Voip2 is our current production server,
> > and Voip3 is being turned into our next production server.
> >
> > We’re trying to build a PRI trunk between Voip1 and Voip3. Curiously
> > enough, we’ve already done this between Voip1 and Voip2, so one would
> > think that the same configuration would work between Voip1 and Voip3 as
> > well. However, it hasn’t gone so smoothly. If you’re wondering why we
> > don’t just use SIP trunking between these servers, it’s because faxes
> > are not reliable over SIP trunks. I am open to suggestions however.
> >
> > At any rate, the PRI trunk between Voip1 and Voip3 isn’t working, and
> > that’s my current problem.
> >
> > – I have built a T1 crossover cable, and it’s plugged in between Span 3
> > on Voip1, and Span 1 on Voip3.
> > – I have a green light on both PRI cards for the appropriate spans.
> > – Both servers detect their cards on boot.
> > – DAHDI is installed on both servers, and all diagnostics are good, ie.
> > dahdi_test returns good results, dahdi_tool shows that the alarms are
> > OK, and executing ‘dahdi show status’ on the Asterisk console shows the
> > same.
> >
> > The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like
> > this:
> >
> > ; Span 3: TE4/0/4 “T4XXP (PCI) Card 0 Span 4″
> > group=3
> > context=default
> > switchtype = national
> > signalling = pri_net
> > channel => 49-71
> > group = 63
> >
> > ; Span 4: TE4/0/4 “T4XXP (PCI) Card 0 Span 4″
> > group=4
> > context=default
> > switchtype = national
> > signalling = pri_net
> > channel => 73-95
> > context = default
> > group = 63
> >
> > Span 4 goes to Voip2, which has a working PRI trunk.
> >
> > The chan_dahdi configuration for Voip3 looks like this:
> >
> > group=1
> > signalling=pri_cpe
> > switchtype=national
> > context=local
> > channel=>1-23
> > dchannel=>24
> > ;channel=25-47,49-71,73-95
> > rxgain=0
> > txgain=0
> > busydetect=yes
> > busycount=5
> >
> > resetinterval=1800
> >
> > I have a test DID, the dialplan for which on Voip1 looks like this:
> >
> > exten => 604484XXXX,1,Dial(DAHDI/g3/604482YYYY)
> >
> > But when I call 604484XXXX from my cell phone, I get no output on the
> > Asterisk console on Voip3, and this output on Voip1:
> >
> >
> > — Executing [604484XXXX@local:1] Dial(“DAHDI/5-1″,
> > “DAHDI/g3/604482XXXX”) in new stack
> > [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable
> > to create channel of type ‘DAHDI’ (cause 34 – Circuit/channel congestion)
> > == Everyone is busy/congested at this time (1:0/1/0)
> > == Auto fallthrough, channel ‘DAHDI/5-1′ status is ‘CONGESTION’
> > — Accepting call from ‘778839ZZZZ’ to ‘604484XXXX’ on channel 0/5,
> > span 1
> >
> > I’ve also tried connecting span 3 to one of the other ports on Voip2
> > with the same configuration, and I get the same results. I’ve run
> > loopback tests on the TE110P and tested the cable thoroughly.
> >
> > Any input on this problem is greatly appreciated.
> You’ve got the spans configured as “group = 63″ but you’re trying to
> dial out on group 3 (DAHDI/g3 rather than DAHDI/g63).

No, the group=63 lines are actually redundant. It is the settings *above*
each channel=> line that get applied to the channels when they are created.

To the OP: what does “pri show span 3″ give you on Voip1?

It might be useful to see the complete chan_dahdi.conf from Voip1.
To save space, you can list it without comments like this:

# grep -v ‘^;’ /etc/asterisk/chan_dahdi.conf


Forcing SIP trunk matching order?

I have a bunch of different customers on an Asterisk Box (the PBX).
This Asterisk Box is behind another Asterisk box that provides a PSTN
Up to this point I’ve been using IAX between the 2 Asterisk boxes, but
I would like to use SIP instead.
After doing some testing I have the following issue.

If customer A calls customer B, but the call goes out through the PSTN
and comes back in, the call is rejected at the PBX because it wants
I can guess that this must be because it matches the “To” address to
the friend SIP trunk that provides registration for the phone.
(All phones are setup as type=friend, host=dynamic). Is there any way
to force matching against the peer SIP trunk to PSTN so as to not
require authentication for calls
coming in from the PSTN server?


Asterisk and the media path

I am attempting to get an asterisk server to step out of the media
path, but am running into a brick wall. Can someone assist? Here’s my

Ultimate SIP Provider —> LCR Trunk (Asterisk 1.6) —-> PBX (Asterisk 1.8).

I am attempting to get the trunk to step out of the media stream.
There is no NAT involved, all machines have a public IP.

In the trunk’s sip.conf I have:


And on the connection to the pbx I have canreinvite=yes

On the pbx I have the trunk connection set to canreinvite=yes.

In the CLI on the LCR trunk I see:

SPA3102 asterisk signaling

Hy all,

Recently a have a little problem with a Cisco device, SPA3102. I use
this device with asterisk to dial out with outbound trunk. (SPA3102
has 1 FXO port)
It working ok , but the device SPA3102 do this : when a call is placed
for outgoing in asterisk and send to SPA3102 , this device “answer
and dial the number in the same time” , in my CLI I see the channel
is open , but on the phone I hear the ringing from the provider PTSN ,
then the answer…. ,

So , in the end, asterisk don’t know when the real answer was made on
PTSN line.
It like SPA3102 don’t notify asterisk the ringing , then open the channel.

There is a problem for signaling ?

Sincerely ,
Alexandru Achim

better timing source for an asterisk gateway


I have to make an asterisk gateway in front of several other asterisk.
This gateway will essentialy be used for outbound call.
This gateway will be connected to other asterisk by IAX trunk, outbound
call will use SIP trunk (voip provider or patton isdn).
I have a TE220BF available than i can use for dahdi timing source. Is a
good idea, or this will give me zero benefit for timerfd timing source
(will host this gateway on debian squeeze or centos 6.2) ?