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 3.2 years ago 3 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 3.2 years ago 8 Answers

One-way audio when calling multiple SIP

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Hi, 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.

Asterisk Users 3.2 years ago 0 Answers

Proactive problem monitoring on SIP on Asterisk

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Hello, 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

Asterisk Users 3.2 years ago 12 Answers

Missing voicemail prompt beginning

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Hello, 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…

Asterisk Users 3.3 years ago 9 Answers

Clipping issue with SIP over satellite

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I'm having a wierd clipping issue with one employee who's using a phone
over a satellite Internet. He was sold that system specifically for use
with VoIP. Ping times show average round-trip time as around 700 ms with a
range of 560 to 841, so considerable jitter. Things work fine when he's talking to another Asterisk phone or to a SIP
trunk provider, but when connecting to a T1, there's clipping where about
1/3 of his voice (in intervals of maybe 200ms) are removed. This sounds
like an echo canceller conflict, but I've set…

Asterisk Users 3.3 years ago 5 Answers

voicemail password with phone instrument

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Hello, voicemail password is not getting changed through phone handset while IVR indicates that password has been changed. During google I found that uniqueid column must not be changed so it is not changed. Please guide on this. During debug log I found below but in mysql db new password is not getting updated, [Jun 15 13:54:07] VERBOSE[6418] file.c: -- Playing 'vm-newpassword.gsm' (language 'en')
[Jun 15 13:54:10] DEBUG[6418] channel.c: Set channel SIP/123-00000005 to write format ulaw[Jun 15 13:54:15] VERBOSE[6418] file.c: --
Playing 'vm-reenterpassword.gsm' (language 'en')[Jun 15 13:54:22] DEBUG[6418] app_voicemail.c: User 123 set password to 3333…

Asterisk Users 3.3 years ago 1 Answer

Polycom, Dial Specific Number on Handset Pickup

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Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards
David.

Asterisk Users 3.3 years ago 2 Answers

Problems installing DPMA

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Hi, I'm just trying to install the DPMA on my Asterisk. I already made the
upgrade from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did: *mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185
*
*compiling Asterisk-Cert2 1.8.11*
*./configure
make
make install
make config
*
Afther that i register the DPMA license, and finally copied the *
res_digium_phone.so* to */usr/lib/asterisk/modules * When i try to load the module on asterisk console this is what i get> **CLI> module load res_digium_phone.so
Unable to load module res_digium_phone.so
Command 'module load res_digium_phone.so' failed. * With…

Asterisk Users 3.3 years ago 5 Answers