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Mango
What router are you using?
Premium
join:2008-12-25
www.toao.net
kudos:13
Reviews:
·AcroVoice
·Callcentric
·Anveo
reply to neftv

Re: Asus WL520GU, Asterisk, VOIP

To clarify, incoming calls are routed directly to my phone. Using my phone's dial plan, I can selectively route outgoing calls via the Asterisk server if it needs to perform some special routing that is not possible to do with my phone itself.
--
Who is the best VoIP provider? | Which ATA should I buy? | Dial Plan Tips and Tricks

neftv

join:2000-10-01
Broomall, PA

1 edit
So like in my instance in the SP2000 would I try to make proxy be QV and outbound be my Asterisk/router IP?
Is that what you mean?

That would probably solve the caller ID problem too right?

Mango
What router are you using?
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join:2008-12-25
www.toao.net
kudos:13
That sounds like what I did. Keep in mind though, just because I do something doesn't mean it's the "right" way to do it. The right way is the way it works best for you.

Dan_voip

join:2007-01-03
Saint-Hubert, QC
kudos:4
reply to neftv
said by neftv:

1) Caller id pass through from QuantumVoice to Asterisk to phones
2) Message waiting indicator from QuantumVoice to Asterisk to phones

3) In figuring out how things work I discovered that if I am on my one extension I can place an outbound call on my other extension too. But inbound is just one call at a time, if a second call comes in it goes to my QuantumVoice voice mail but I don't get the busy greeting I get the unavailable greeting.
Even though I'm registered to 8 providers with my Asterisk box I have just 1 DID for now and that is an European phone number, so I didn't do much tests for incoming.
1) I can see the caller ID on my phone and I think we have pretty much the same settings.
2) Sometimes the voicemail indicator is getting lit on my phone but I'm thinking that could be a false alert, I need to do more tests
3) I'm thinking QV is ringing your Asterisk box who is trying to transfer the call to your busy extensions and Asterisk doesn't send back to QV a busy or QV doesn't understand. I wonder if you can catch what really happens from log.
My provider has just 1 channel but I can have 2 calls in the same time, 1 outgoing and 1 incoming, I guess because of call waiting option. That number is ringing on 2 extensions and that's how I can take the call on the 2nd extension.

For me Asterisk was an old project but I didn't start it until I saw Mango's thread about Asus router so it's something new for me also. What I did more I setup call transfer(or it's by default, I don't remember) and the option to pick up from another extension no matter which one is ringing.
I have in plan to do more like dial URI in/out, to route an incoming call to another phone (cell) and so on. I just need time

I'm building a PC in my A+ class which won't be done till end of April and I am thinking of making this a model for what I want to do with that new PC and perhaps more. It will have Windows 7 Professional 64bit.
Depend what you want to do with that PC but you might want to consider installing Linux

neftv

join:2000-10-01
Broomall, PA

1 edit
Which logs should I try to see what is happening with Caller ID, and if Asterisk is sending the busy TONE? How do I get to those logs?

I'll have to see if the instructor will let me install Linux because the school is giving us one copy of Win 7 Pro. I have a Ubuntu 7.10 Book with the OS setting under my home desk here.

Dan_voip

join:2007-01-03
Saint-Hubert, QC
kudos:4
said by neftv:

Which logs should I try to see what is happening with Caller ID, and if Asterisk is sending the busy TONE? How do I get to those logs?
/opt/sbin/asterisk -rvvvvv
After you run this you'll see logs on the console for incoming and outgoing calls and when the box will register again with a provider and so on.

neftv

join:2000-10-01
Broomall, PA
Just seeing how a incoming call is handled I get this
*CLI> -- Remote UNIX connection
-- Saved useragent "Sipura/SPA2000-3.1.5" for peer neftv
-- Executing [5555555555@voip-inbound:1] Dial("SIP/5555555555-0059d028", "SIP/neftv&SIP/dph541|30|tR") in new stack
-- Called neftv
-- Called dph541
-- SIP/neftv-005a23e0 is ringing
-- SIP/dph541-005a6c48 is ringing
== Spawn extension (voip-inbound, 5555555555, 1) exited non-zero on 'SIP/5555555555-0059d028'

Where 5555555555 is my where my home number appeared.
I placed the inbound call on my cell phone and that does not appear.

I tried again and let it go to voicemail. Here the phones stopped ringing but was ringing on my cell phone for a few rings before QV voicemail picked up
*CLI> -- Executing [5555555555@voip-inbound:1] Dial("SIP/5555555555-0059e9e0", "SIP/neftv&SIP/dph541|30|tR") in new stack
-- Called neftv
-- Called dph541
-- SIP/neftv-005a3978 is ringing
-- SIP/dph541-005a80a8 is ringing
-- Got SIP response 486 "Busy Here" back from 192.168.123.191
-- SIP/dph541-005a80a8 is busy
-- ast_get_srv: SRV lookup for '_sip._udp.sipdr.quantumvoice-sip.com' mapped to host sipdr.quantumvoice-sip.com, port 5060
== Spawn extension (voip-inbound, 5555555555, 1) exited non-zero on 'SIP/5555555555-0059e9e0'

Where is it getting '_sip._udp.sipdr.quantumvoice-sip.com' mapped from? That is not in the spec of QV web site.
So what is happening from all this?

mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA
kudos:4
said by neftv:

Where is it getting '_sip._udp.sipdr.quantumvoice-sip.com' mapped from?
From Quantum DNS server.

neftv

join:2000-10-01
Broomall, PA
reply to Dan_voip
I found this page about Caller ID »www.the-asterisk-book.com/unstab···rid.html don't know if that helps with Caller ID pass-through.

Also, sorry if this is such a basic question but how do you close out of the CLI prompt of Asterisk without actually shutting Asterisk down. It seems like clicking X at the top of the window is not the way to do it.

mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA
kudos:4

1 recommendation

said by neftv:

Also, sorry if this is such a basic question but how do you close out of the CLI prompt of Asterisk without actually shutting Asterisk down.
exit.

It seems like clicking X at the top of the window is not the way to do it.
If you think asterisk CLI (Command Line Interface) is a GUI, then you are not smarter than a 5th grader.
--
Mazilo always prays for FREEBIES!
UK Phone: +44-703-194-2574

neftv

join:2000-10-01
Broomall, PA
haha fifth grade huh lol

Interesting the exit works on the busybox to the router (Which I knew) but not in the Asterisk window. it says no such command.

mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA
kudos:4
At least on my FON2100 unit, exit command terminates the CLI to the busybox shell as shown below:

BusyBox v1.14.4 (2009-11-29 16:35:45 EST) built-in shell (ash)
Enter 'help' for a list of built-in commands.
 
  _______                     ________        __
 |       |.-----.-----.-----.|  |  |  |.----.|  |_
 |   -   ||  _  |  -__|     ||  |  |  ||   _||   _|
 |_______||   __|_____|__|__||________||__|  |____|
          |__| W I R E L E S S   F R E E D O M
 KAMIKAZE (bleeding edge, r18615) ------------------
  * 10 oz Vodka       Shake well with ice and strain
  * 10 oz Triple sec  mixture into 10 shot glasses.
  * 10 oz lime juice  Salute!
 ---------------------------------------------------
root@Fonerisk:~# rasterisk
Asterisk 1.6.1.11, Copyright (C) 1999 - 2009 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.6.1.11 currently running on Fonerisk (pid = 1639)
Verbosity is at least 4
    -- Remote UNIX connection
Fonerisk*CLI> exit
root@Fonerisk:~#
 

--
Mazilo always prays for FREEBIES!
UK Phone: +44-703-194-2574

Dan_voip

join:2007-01-03
Saint-Hubert, QC
kudos:4
reply to neftv
said by neftv:

Interesting the exit works on the busybox to the router (Which I knew) but not in the Asterisk window. it says no such command.
exit or quit, both are working on my box

neftv

join:2000-10-01
Broomall, PA
Click for full size
This is at the prompt after I start asterisk with asterisk -vvvc I get the CLI prompt. As you can see.

mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA
kudos:4
said by neftv:

This is at the prompt after I start asterisk with asterisk -vvvc I get the CLI prompt.
You should have a init scripts file on /etc/init.d/asterisk to start the asterisk. If you have configured your asterisk and/or system properly, the asterisk should be started during its booting process. Then, you can either issue rasterisk or asterisk -r to remotely connect to asterisk server to get the asterisk CLI. In your case, your started asterisk as a process. Perhaps, hitting a cntrl-C will cause the process to terminate and get you out to the shell prompt.
--
Mazilo always prays for FREEBIES!
UK Phone: +44-703-194-2574

neftv

join:2000-10-01
Broomall, PA
if I do "control c" that backs me to main prompt but it also shuts down Asterisk cleanly.
I have not put the command in the "run after booting" USB section in Tomato yet till I can see if can get Caller ID, MWI working. I did it once just to see and even after I connect as Asterisk -r then I try to back out it will shut down Asterisk.
I don't see /init.d when looking in that directory.

mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA
kudos:4
said by neftv:

if I do "control c" that backs me to main prompt but it also shuts down Asterisk cleanly.
In this case, you will need to type core stop now on the asterisk CLI to shutdown the asterisk. For futher information, type help core on the asterisk CLI.

I don't see /init.d when looking in that directory.
I don;t now how you install your asterisk, on my system, the asterisk init scripts file is on /etc/init.d/asterisk. If you had installed asterisk on /opt directory, then chances are the init scripts file is on /opt/etc/init.d/asterisk. This is a simple example why I emphasized (as shown on this post) that it will make life easier to make a router, with a USB2 port, boots off of an external partition so that all the programs can normally be installed in the default paths.
--
Mazilo always prays for FREEBIES!
UK Phone: +44-703-194-2574

neftv

join:2000-10-01
Broomall, PA
core stop now is not listed in help core or work when I try to use it.
I just wanted to close that window without having to shutdown Asterisk.
The only thing I did was followed Mango's initial instructions on his post. Almost word for word.

mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA
kudos:4
Here is the excerpt of help core stop on my asterisk-1.6.11:
Fonerisk*CLI> help core stop
          core stop gracefully Gracefully shut down Asterisk
                 core stop now Shut down Asterisk immediately
     core stop when convenient Shut down Asterisk at empty call volume
Fonerisk*CLI>
 
If you are using asterisk-1.4.x, I have no idea how do to this. Perhaps, you will need to use help to find out which instruction you will need to stop your asterisk. If you just want to close the window without shutting down the asterisk, then I reckon you will need to click the [-] icon on the right-top corner of the window to stow it away. Other than that, you perhaps you will need to start asterisk through the init scripts file or start asterisk as usual except in a background using &. If you already start the asterisk, then perhaps pressing ctrl-z will stop the asterisk and give you the busybox shell prompt. Then, just execute bg to put the stopped process into background. Whenever you feel like to get back to the background process (of asterisk), all you need is to execute fg. Whilte the asterisk process has been put in the background, you can click the [-] icon on the right-top corner of the window to stow it away.

--
Mazilo always prays for FREEBIES!
UK Phone: +44-703-194-2574

neftv

join:2000-10-01
Broomall, PA
I decided to finish out the step in putting /opt/sbin/asterisk in the "run after booting" section in Tomato. It will now behave better that is I can get into Asterisk using the -r and then in the CLI prompt I can type exit to go back to the main prompt.

I got email support at QV to tell me that MWI does not pass through Asterisk at the present time. He recommended using Asterisk as the voicemail and timing it so that QV voicemail kicks in with a later duration so that if my Asterisk goes down my QV picks up the call to Voicemail.
I was told that Caller ID basically should just work because I NOT overriding it with a command like
exten => s,n,Set(CALLERID(all)="Quantumvoice" )
so I guess my asterisk is broken in that regard no caller ID. It just shows unknown caller, Unknown Caller.

After my class tonight I might try to run a virtual environment over my older windows pc and install AsteriskNOW which has OS and Asterisk and see if I can get that going to test it. Would have been nice to seen Asterisk on my router work with Caller ID but I guess it's not my day.

Dan_voip

join:2007-01-03
Saint-Hubert, QC
kudos:4
said by neftv:

I decided to finish out the step in putting /opt/sbin/asterisk in the "run after booting" section in Tomato. It will now behave better that is I can get into Asterisk using the -r and then in the CLI prompt I can type exit to go back to the main prompt.
Me I have in Tomato under USB and NAS->USB support in "Run after mounting" box:
[ -d /mnt/Asterisk ] && mount /mnt/Asterisk /opt
sleep 15
/opt/sbin/asterisk
 
and of course I have Automount checked
What helps me a lot to learn how to setup Asterisk, I've created free accounts with Callcentric, Voip.ms and Future9 and did many tests.
By example Callcentric has on the web site a way to call your phone number alocated (1777xxxxxxx) and you can test the caller id and so on
»my.callcentric.com/click2dial.php
You don't need any amount in your account to do that.

Dan_voip

join:2007-01-03
Saint-Hubert, QC
kudos:4
To see what incoming caller ID is getting Asterisk when somebody is calling you, use this dial plan:

exten => s,1,NoOp,${CALLERID(num)}
exten => s,2,Dial(SIP/neftv,30,tR)
 

You need to be in Asterisk console when the call is coming to see the logs. If you see it there Asterisk should pass it to your extension.

neftv

join:2000-10-01
Broomall, PA
Thanks for those suggestion I will try them when I get a chance. Something interesting having Tomato loaded on this router there is some anomaly going on when it comes to Streaming media. I frequently have Windows Media Player up looking at TV stations from Greece and I leave it on all day. Now with the Verizon Fios Supplied router "Actiontec rev 1" or my previous router LevelOne 6001 I did not have this symptom. When i loaded the ASUS with DD-WRT I didn't have the symptom either. With Tomato on this router I seem to get perhaps 2 or 3 times an hour where the Windows Media Feed will Video will freeze and audio might pause then sometimes I get this stutter of Video and Audio sort of like what a broken record sounds like and it will occur for a few seconds sometimes. Its funny to hear and see it it does remind me of the old LPs. Then I did some googling on tomato and found that Tomato has some sort of issue once in a while of doing something to the TCP packets destroying them or something potentially causing issues. Now I noticed this anomaly with the feeds before I loaded Asterisk.
Not sure what that is about but I might go back to making Asus just a router with DD-WRT or the open-WRT if someone figured out how to load that version 2.6 on this particular Asus model in simple 1 2 3 steps that are spelled out easy. I read that 2.4 is suppose to be simple but the newer 2.6 is out now.

I not done my AsteriskNow install on my other PC that I mentioned in a previous post because one of the files is missing in the iso or is corrupt so I gave up on it. It gets like to the point when its tries to copy the image to the hard drive and just stops with some error on a file corruption or missing. I didn't want to drive into it deeper. Gotta love how they state AsteriskNow is a complete solution though.

Anyway I have an older Dell pc sitting on the shelf that I might try to load my Ubuntu 7.1 since I have the unleashed book with the disk. My only concern it the 7.5gig drive might be two small. The books say with Linux you really want 20gig. And they seem to favor intel cpus and that Dell is the only Intel based I have. I have AMD on my other two systems.

mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA
kudos:4
said by neftv:

Not sure what that is about but I might go back to making Asus just a router with DD-WRT or the open-WRT if someone figured out how to load that version 2.6 on this particular Asus model in simple 1 2 3 steps that are spelled out easy. I read that 2.4 is suppose to be simple but the newer 2.6 is out now.
If you want to flash your WL-520GU router with an OpenWRT SVN firmware, you are better off to download the OpenWRT SVN source and compile it yourself. Otherwise, use pre-built OpenWRT Kamikazi firmware.
--
Mazilo always prays for FREEBIES!
UK Phone: +44-703-194-2574

neftv

join:2000-10-01
Broomall, PA
reply to Dan_voip
I did what you said and yes the number appear there but...

I sent you a pm of the log if you want to look at it.
I took out my phone numbers and placed a label for the number to identify what was there.
Please let me know what you think.

Thanks.

Dan_voip

join:2007-01-03
Saint-Hubert, QC
kudos:4
Your log:
-- Executing [actual_home_number@voip-inbound:1] NoOp("SIP/Actual_HOME_NUMBER-0059a190", "Actual_CELL_NUMBER") in new stack
-- Executing [actual_home_number@voip-inbound:2] Dial("SIP/Actual_HOME_NUmBER-0059a190", "SIP/neftv|30|tR") in new stack
Audio is at 192.168.123.254 port 10060
....
From: "UNKNOWN CALLER" ;tag=as050f4711
To: 
Contact:
 

My log:
Asus-WL-520GU*CLI> 
    -- Executing [1777xxxxxxx@voip-inbound:1] NoOp("SIP/1777xxxxxxx-0059bc28", "17771234567") in new stack
Asus-WL-520GU*CLI> 
    -- Executing [1777xxxxxxx@voip-inbound:2] NoOp("SIP/1777xxxxxxx-0059bc28", "17771234567") in new stack
Asus-WL-520GU*CLI> 
    -- Executing [1777xxxxxxx@voip-inbound:3] Dial("SIP/1777xxxxxxx-0059bc28", "SIP/dan|30|tR") in new stack
...
From: "17771234567" <sip:17771234567@callcentric.com>;tag=ycw1wjvcu2
 
To: <sip:1777xxxxxxx@ss.callcentric.com>;tag=as33285ae8
 
Call-ID: 2e142b4bf045-iu014pxkqd02@snom.com
 
CSeq: 1 INVITE
 
User-Agent: asterisk
 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
 
Supported: replaces
 
Contact: <sip:1777xxxxxxx@192. 168.2.10>
 

From: "UNKNOWN CALLER" this is the problem but I don't know why.

neftv

join:2000-10-01
Broomall, PA
Hey Dan,
In the hibernation from the snow yesterday here in PA I decided to get VMware on my windows PC and run Ubuntu on it virtually. (BTW ubuntu makes it easy to like Linux and having VMware is really a nice software as long as you have hard drive real estate to accommodate the virtual environment.) Ubuntu had Asterisk 1.62 in the repository so I installed it and used the same extension.conf and sip.conf files as I have for my Asus Asterisk. I had to change the pipe symbols to commas and I have to change the insecure=very to invite because that is how they doing things in version 1.62. Anyway I still have the same symptom of unknown Caller ID for incoming calls on my phones. Must be something in the Conf files that is not liking the caller ID or something ELSE that is overlooked in all the other conf files. I can't imagine that I am the only one in the planet to get this symptom even though I couldn't find anything in my google searches.

Dan_voip

join:2007-01-03
Saint-Hubert, QC
kudos:4
said by neftv:

Must be something in the Conf files that is not liking the caller ID or something ELSE that is overlooked in all the other conf files. I can't imagine that I am the only one in the planet to get this symptom even though I couldn't find anything in my google searches.
Your incoming calls are only from QV. I'm thinking you might want to try to take a free account with Callcentric by example and do some tests. I'll extract my settings for CC and send them to you, in that way we gonna be using the same device, version of Asterisk, settings and that should work. Since you see the incoming phone number in Asterisk's logs shouldn't be a problem with your lan/wan configuration. In other conf files I did just a couple of changing for call transfer and parked calls, nothing related with CID. On CC it's an option Click 2 Dial and using the number 17771234567 and your CC number you can do tests.

neftv

join:2000-10-01
Broomall, PA
Anyway to extend the time a call rings the extensions? What happens is when I call into Asterisk via my cell phone I hear the extension ring 3 times then it stops. In the earpiece of my phone it rings many more times before my QV voicemail picks up. Waitexten or digittimeout don't work the way I want it to work.

I tried Callcentric as you recommended I did see either number or name stored in phone book show up in Caller ID when I called from my Gizmo5 to Callcentric Asterisk set up. I could not for the life of me make a call out to the test number 17771234567 with the setup you sent. I decided to concentrate on the QV set-up.


Motofreak
Premium
join:2009-08-03
Oshawa, ON
Reviews:
·voip.ms
·Start Communicat..

1 recommendation

reply to mazilo
I just wanted to say thanks for everyone that added info to this thread. I went out and got a WL500GPV2 and installed asterisk 1.4 loaded all the modules (needed for gui) installed the asterisk gui (just to configure it) and a few minutes later, I have setup a IVR,vm routing etc.. and 6 extensions for my house and started to make calls.

This router rocks and so far (past 24 hours) its solid unit. With everything that is on by default with QOS enabled in the Tomato firmware.

thanks