* You are viewing Posts Tagged ‘phone’

Asterisk With OpenBTS And Mobile Phone

Hello people,

I want to connect Asterisk with OpenBTS and make a call with a mobile phone.

I use:
Ubuntu 11.10 + Kernel 3.0.22
GnuRadio 3.3.0
Asterisk 1.8.13
OpenBTS 2.8
Nokia Mobile Phone

OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also OpenBSC is working with Asterisk successfully (OpenBSC is another project). Perhaps you can help me because I think it is an issue with Asterisk.

I have tried both contexts, [open-bts] and [sip_external] and both don’t work. If I want to call the mobile phone (6201) with a Twinkle soft phone (6000) I get following message in the CLI-window from Asterisk:

*CLI> sip show peers
*CLI> sip show peer 123456789101112

Asterisk log file (path: /var/log/asterisk/cdr-csv/Master.csv):

If you need more informations write me and I will give you. It would be very appreciated if some of you can help me or has an idea how I can fix this erorr. Best regards and thanks for helping.

port 5060 is blocked by ISP

i have configured properly asterisk. At the one end i am using x-lite soft
ph and another end twinkle. call is going properly from both end but after
picking the phone not able to listen other one.
when i checked the port 5060 on the asterisk server it is always showing
closed while i have flushed all the rules from iptables (iptables -F)

5060/tcp closed sip

telnet localhost 5060 (could not connect)

Dahdi Dropping Calls

Hi Guys

Has anyone seen Dahdi dropping incoming calls with Hangup cause 27?
It only drops whilst we are on the phone?
Its not every single call
Any ideas?

One-way audio when calling multiple SIP


On one of our locations, I am having issues with one-way audio when I call
several phones with SIP/Phone_A&SIP/Phone_B&SIP/Phone_C. When I call the
phones individually, they work fine, so it’s not a volume setting on the
phone. Also this setup has worked at other locations.

Any idea’s what to look for?

Thanks in advance.

Proactive problem monitoring on SIP on Asterisk


1) I am wondering what is the best practice to monitor if there are or were
problems with SIP calls on my Asterisk box. E.g. how about a software that
extracts all calls from the /var/log/asterisk/full (I have permanently
enabled verbose 10 and sip debug) log and tells me on which of them were
problems? Checking the logs manually is very hard, but as SIP is a
standardized protocoll, there should be tools doing that for you? As an
example, a person calling me recently got a 488 Not acceptable error as
reply from my Asterisk box. Nothing came through to my SIP phone, so I
didn’t know anything about the call or the problems (which were on his
phone btw). I would like to be informed about such cases, know that there
was a call to my Asterisk box that made problems.

2) How about monitoring speech quality? E.g. sometimes it seems like a
packet is missing (I then have a short pause during the call), how to
monitor such things and create statistics out of this data?

So basically I want to monitor my Asterisk installation proactively for
reliability/problems and (speech) quality.

Thanks for any hints!

Best regards

Missing voicemail prompt beginning


I am using the voicemail module of asterisk. When I did some test calls
from my mobile phone, sometimes the beginning of the prompt was missing,
e.g. instead of something like “number 12345 not available” I was only
hearing “345 not available”. Verbose level 5 on the asterisk console didn’t
give me any hint on this, it only shows that playback of the prompt started
correctly in every test case. Any hints on how I can debug this? I think
it’s some problem on my local configuration, I doubt it’s a problem with my
SIP provider or mobile phone provider, they are both very reliable (Sipgate
and T-Mobile).

Thanks for any hint!

Best regards