Greetings fellow listers,

I have an application where I have
approximately 300 files that I playback individually or in blocks to
simulate “text-to-speech” in a “less mechanical” voice than normal Allison
files provide. These files are presently in GSM format and sound pretty
good when I play them on my computer speakers or on my in-house Polycom
501′s over SIP connections. The “problem” I have is that the intended use
of the application is going to be over SIP/DAHDI trunks that will connect to
VM’s over IAX trunks. What is your best suggestion for maintaining the
quality of the audio as much as possible?

Best Case presently – SIP phone in-house to IAX

Worst Case presently – Cell phone calls Asterisk 1 on TDM400P which connects
to VM Asterisk 2 via IAX.

Asterisk version is 1.4.30

Thanks in Advance

Danny Nicholas

Hi Everyone,

I have setup an OpenVPN tunnel between Server A (running Asterisk) and
Server B suppling it’s SIP Phones with DHCP pool of IPs.

So, the tunnel is established nicely and everyone can ping others. “sip show
peers” shows the local subnet of the SIP Phones registered (192.168.100.0/24
).

But there is the old bad one-way audio. Calls also drop after few seconds.
In the SIP debug I can see that asterisk uses it’s external public IP
address to communicate to endpoints that are known to it as the
192.168.100.0/24 endpoints and the endpoints identify themselves with the
OpenVPN tunnel IP address scheme in one part of the sip handshake. How can
this be fixed? After all, with the OpenVPN this should all look like an
internal network to Asterisk.

I have added my comments followed by # to lines below that are problematic.

<--- SIP read from UDP:192.168.100.5:5060 ---> #This line is good as it
uses the local DHCP supplied network address scheme
INVITE sip:203@172.16.0.1:5060 SIP/2.0 #This line is BAD. Why are we
inviting Ext. 203 with it’s OpenVPN IP while it’s on the same network of
192.168.50.0/24 as 202?
Via: SIP/2.0/UDP
192.168.100.5:5060;branch=z9hG4bK695f8c1cfc7cdee96.1c65dc2eb25a46fc6
Max-Forwards:
70
From: “SIP Phone – Ext. 202″ ;tag=6d6f8c4226
#BAD line again. Should be SIP:202@192.168.100.6
To: “203″ #Bad again….
Call-ID: 43af67a634e06e75
CSeq: 32058 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: “SIP Phone – Ext. 202″ ;transport=udp>;+sip.instance=”“
Supported: gruu, path, timer, 100rel, replaces
User-Agent: Aastra 55i/2.5.2.1500
Content-Type: application/sdp
Content-Length: 594

Basically the phones should only send with FROM their local
192.168.100.0/24address and Asterisk should only send ANSWER and ACK
back to
192.168.100.0/24 rather than sending it to 172.16.0.0/24 (which is the
openvpn client ip).

Once above is fixed, I think all the audio and call cut will go away. I hate
to use a sip proxy in this situation since I already have an openvpn
connection.

Any feed back is appreciated.

Thanks,

Hi,

It seems ive broken my settings and now, asterisk isnt detecting my DTMF tones.

What kind of diagnostics can I do to work this out?

I’ve set the extension in sip.conf to everything listed on this page but no result. I’ve also played around with the settings on the phone with no help either. Someone once said on here that Asterisk and the SIP phone have to match, but that doesnt seem to work either.

Any ideas?

Thanks
Dan

I’m running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a Linksys
SPA942 SIP phone and outgoing SIP and IAX routes.

When I dial local PSTN numbers from the SPA942 using the FXO channels I
observe a 5-7 second delay between when the PSTN number answers the call
and when Asterisk connects the call at my end. There’s enough delay
time that I hear an additional ring after the PSTN number has answered
the call. I’ve had people hang-up since they don’t hear anything when
answering.

If I try the exact same call using an IAX route, the call is connected
at my end just as soon as the PSTN number answers.

I don’t have any connection delays for incomming FXO calls. They are
connected as soon as I answer the calls.

Can anyone give me some pointers on where to start looking?

Frank

Hi,

I’m currently programming an interface for my Asterisk service.

I’ve noticed an issue if someone sets up call forwarding on their sip phone.
Asterisk receives a 302 “Moved Temporarily” message, and forwards the call successfully.

However, the CDR is not correct.

If I set up call forwarding to a mobile on extension 201, and then place a call from extension 202, the call gets diverted.
I answer the call and talk for 30 seconds, then I hang up.

The CDR shows two calls:-

2010-08-27 13:38:24 – 202 -> 201 – Answered – Billsec is 30
2010-08-27 13:38:24 – 202 -> 5551234 – Answered – Billsec is 0

5551234 is the mobile number.
The second CDR entry should read 30 seconds, and the first should read 0 (or 30)

Since it isn’t behaving like I want, is there any way to disable the feature that allows a SIP phone to perform call forwarding?

Thanks
Dan