* You are viewing Posts Tagged ‘call’

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.

IAX Trunking Stopped Working

I administer a group of Asterisk servers running a mix of 10.3, 10.4, and 1.8.8.1 (mostly 10.4). One of those servers is a call concentrator/relay for E911 service. All of the other servers make an IAX connection to the relay server, which then hands off to a SIP trunk to my E911 provider. It all worked as recently as 2 weeks ago, but I discovered that sometime between then and now it stopped working without any explanation. Last modified time on the config files is over 2 months ago.

The setup is as follows:

On the call relay (IAX “receiver”)
[my-remote-server]
type=user host=dynamic username=my-remote-username encryption=yes secret=my-remote-secret context=my-call-context deny=0.0.0.0/0.0.0.0
permit=remote.server.ip.address/255.255.255.255

On the VoIP servers (IAX “sender”)

- One of the servers is set to register: register => my-remote-username:my-remote-secret@call.relay.server.ip

- Another is set to just use the peer definition as below without trying to register
[my-remote-server]
type=peer host=call.relay.server.ip username=my-remote-username secret=my-remote-secret qualify=no

Dialplan on the VoIP servers:
exten => 911,1,Verbose()
same => n,Dial(IAX2/my-remote-server/911)

Dialplan on the relay server:
[my-call-context]
exten => 911,1,Verbose()
same => n,Dial(Relay to E911)

The issue I’m seeing is this:

- On the servers that are set to register, the relay server is rejecting the registration (I’ve confirmed the username/peername/secrets are an exact match on both sides, and nothing has changed from when they were working). IAX debug on the relay server shows the auths come in and the relay server send REGREJ – Registration Refused, Cause Code 29. IAX debug on the server attempting to register shows sending the REGAUTH packets and receiving the REGREJ packets. The IP address shown in the IAX debug packets matches the IP address in the permit rule for each peer that’s supposed to register.

- On the server that is set to just send the calls, an attempt to dial 911 just hangs for 60 seconds and eventually times out without sending the call. IAX debug on the relay server shows the call start frame get RX’d, shows the relay server try to TX a CTOKEN frame, and nothing further (other than retransmissions of the call start frame). IAX debug on the server trying to send the call to the relay server shows the TX for the call start frame, but no RX for the CTOKEN frames.

Ultimately, this has gone from working to totally broken without any apparent change to my configuration. I need help to try to troubleshoot it further, I’ve tried everything I can think of (including transferring the backed-up working config files to a brand new clean-load server, upgrading Asterisk, and recreating the configurations by hand), and nothing seems to be helping.

Thank you,

How To Play Different Different Hold Music.

Dear All,

I have two server ‘A‘ and ‘B‘ . In Server ‘A‘, five different ivr (Sevices) is playing and call is *forwarding *into Server ‘B‘. Server ‘B‘ basically use for agent login(Extension). I want to play different hold music(Server ‘B‘) bases on the corresponding services which is running into server ‘A‘.

A single agent takes the call from different different services but hold music is play astrisk own by default. Is there any way to play different hold music bases on services which run into server A.

I have some changes into musiconhold.conf (server B) but problem is no solve.

please help me.

Regards

port 5060 is blocked by ISP

dear
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)

PORT STATE SERVICE VERSION
5060/tcp closed sip

telnet localhost 5060 (could not connect)

Park function and billsec

When a call comes in and is parked using the Park() command it appears that
the billing seconds do not include the time while the caller was parked. Is
there a simple way to correct this with an option or setting in asterisk.
This is not acceptable as we are loosing min when callers park calls.

I am using 1.8.x release currently.

Thanks
zktech

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?