Asterisk With OpenBTS And Mobile Phone

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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]…

Asterisk Users 3.1 years ago 3 Answers

IAX Trunking Stopped Working

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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")…

Asterisk Users 3.1 years ago 2 Answers

How To Play Different Different Hold Music.

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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…

Asterisk Users 3.1 years ago 4 Answers

port 5060 is blocked by ISP

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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)

Asterisk Users 3.1 years ago 8 Answers

Park function and billsec

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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

Asterisk Users 3.1 years ago 0 Answers

Forcing SIP trunk matching order?

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Hi,
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
connection.
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
authentication.
I can guess that this…

Asterisk Users 3.1 years ago 3 Answers

.lock file issue

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I'm currently running Asterisk 10.5.1, compiled from source, and just had someone call saying they couldn't get their voice mail. Looking into the user's voice mail folder, I saw a .lock file. Removing this file, enabled them to get voice mail. Is anybody else seeing this? The system is a new install and has only been running for a week with very little traffic (8 person office). Doug

Asterisk Users 3.1 years ago 5 Answers

FastAGI script and DIAL execution

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Hi all, I am trying to control the whole call using a FastAGI script. To that
effect I launch a FastAGI script (written with asterisk-java). Basically, I want to DIAL from within the FastAGI script. When the call
ends I want to control the hangup (if executed at the remote end), and
depending on the cause, dial again, play a message, or hang up. This is a
pretty standard telephony scenario. I did it before by executing the AGI,
setting variables, calling the DIAL command from the dialplan, and then
executing a second AGI…

Asterisk Users 3.1 years ago 1 Answer

Timeout for Huntgroup

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I have not found a solution so I am checking with the Masses here. I have a client who has a old 5 line key system without voicemail. Currently, I can set up a huntgroup and ring each line for 15 seconds and after the 5th line has reached its limit, the call goes to
voicemail. The problem with this is that the caller will hear roughly 1m15s worth of ringing. I trying to give them the same ability to always ring the huntgroup, starting with line 1, and hunting only if its busy or the
timeout is…

Asterisk Users 3.1 years ago 0 Answers