I’m reading some information that recommends using SER / OpenSER for
large installation to offload SIP traffic from the Asterisk server.

http://www.voip-info.org/wiki/view/Asterisk+at+large

However, it looks like the information might be dated.

I’m looking at a potential 750 SIP phone and 150 Analog installation,
all internal network, PRI trunks, and am trying to nail down an
architecture.

Opinions? You think I skip the SER box if I’m using 1.8?

Thanks!

Hi,

I got a problem with asterisk 1.8.9.2. The same scenario is working fine
in 1.8.8.2.

Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on
the same LAN, no NAT.

Asterisk sends the INVITE to the proxy, the proxy sends INVITE to the
phone. The phone sends 180 RINGING back to the proxy. The proxy sends
180 RINGING to asterisk. So far so good. If the calling side decides to
cancel the call, asterisk sends the CANCEL directly to the phone. The
phone doesn’t find the call and answers 404. In asterisk 1.8.8.2
asterisk sends the CANCEL to the proxy, which sends the CANCEL to the
phone and all ist fine.

I think, the new behavior comes from the lines
parse_ok_contact(p, req);
if (!reinvite) {
build_route(p, req, 1);
}
which are inserted in the handling of provisional SIP response.

Am I doing something wrong or is this a bug?

Thanks,

Karsten

Hi all !

We currently have an asterisk box that is rather old (runs Asterisk
1.4.21.2), and it’s connected to the PSTN with a sangoma A104d card.

Now we have a new PRI at another location, and I use that occasion to
build 2 new servers, one to replace our aging one and a new one for this new
pri.

So I downloaded the lastest libpri / asterisk / wanpipe driver, but the
previous version of dahdi (2.5), since the latest wanpipe isn’t compatible
with dahdi 2.6. All is built from source

Now, all seems to be working OK. I can connect a SIP phone to my new box,
make calls to the outside, receive calls etc.

But, I can’t seem to bridge a call. So on my new server, with the new PRI, I
got a Sangoma a104 card (no echo-canceler on this one).

In my extensions.ael, I got this :

418nxxxxx1 => {
Answer();
Wait (2);
Playback(demo-thanks);
Dial(${TRUNK}/418nxxxxx2);
};

TRUNK is DAHDI/G1

Where 418nxxxxx1 is a DID on my new PRI and 418nxxxxx2 is my cellphone
number.

When I do a call from my home phone or cell phone to my new PRI to
418nxxxxx1, I hear the demo-thanks file, and then it dials out. My cellphone
rings, but as soon as I pick up the call, the calls hangs up :

Hi All,

I want to play a file in agi but dont want to answer the call

I am dialing through sip phone and running asterisk 1.8.6,

I tried following with no luck

$agi->exec(“Progress”);
$agi->exec(“Playback $filetoplay,noanswer”);
$agi->hangup();

When I dial I can’t hear the audio but if I answer the call or remove
noanswer argument I can hear the audio.

phpAGI’s stream_file didn’t help either.

I ended up with ResetCDR() before hangup to reset billsec, duration and
disposition but don’t want to do it this way.

What could be the problem?

From Voip-info.org :
*noanswer*: Play the sound file, but don’t answer the channel first (if
hasn’t been answered already). Not all channels support playing messages
while still on hook.

Is it because the channel is not supported?

Regards,
Zohair Raza

On 01/31/2012 12:17 AM, Ira wrote:
> Tonight I tried 4 versions of Asterisk; 10.0.0, 10.0.1, 10.1.0 and trunk.
>
> On 10.1.0 and trunk, I can’t successfully enter the password for any
> mailbox in voicemailmain on my Aastra 480i phones. All four version work
> with a Snom cordless SIP phone. In 10.0.0 and 10.0.1 the Aastra works
> perfectly. So needless to say I’m back to running 10.0.1. The WAF is
> very low for stuff like that.

It is quite unlikely that there were any changes between 10.0.1 and
10.1.0 that would affect DTMF detection or app_voicemail itself, but
it’s certainly possible. That’s why we have an issue reporting system,
and it’s also why we produce release candidates to get testing prior to
making official releases.

> I notice that comedian mail has <> instead of [] brackets. Does that
> mean it’s on its way to being deprecated?

I assume you are referring to how app_voicemail (not ‘comedian mail’) is
listed the menuselect tool. Umm… no, those are completely unrelated.
How did you reach that assumption?