stuck with asterisk

[akogan82]

I am running asterisk 1.4 and keep having issues with error 404 when trying to dial out. Can somebody please provide me with a basic sip.conf and extensions.conf that they have working with viatalk. this would help eliminate a lot of issue. I have been working with tech support but not able to get this resolved yet.

I basically have 1 softphone for now connecting to asterisk that i want to connect to viatalk.

Thanks

[KLH]

Have you tried TrixBox? www.trixbox.org
It makes setting up asterisk alot easier.

[rvtango]

Look at the previous post. I run A*1.2 and it works fine.

[KLH]

http://support.viatalk.com/index.php?_a=knowledgebase&_j=questiondetails&_i=123&nav=+%26gt%3B+%3Ca+href%3D%27index.php%3F_a%3Dknowledgebase%26_j%3Dsubcat%26_i%3D42%27%3EInstallatio n+Guides%2FTech%3C%2Fa%3E+%26gt%3B+%3Ca+href%3D%27index.php%3F_a%3Dknowledgebase%26_j%3Dsubcat%26_i% 3D60%27%3EAsterisk+Setup+And+Configuration%3C%2Fa%3E

What are the working settings for Asterisk
Due to a larger need for configuration of Asterisk PBX's, we've worked out a set of default settings that work for several users. Below you will find snippets of the configuration files that have the various settings you will need.

This is a known working config, all you have to do is insert the right data where we have all caps words placed.

SIP.CONF

[general]
register => YOURNUMBER:YOURPASS@INSERT PROXY HERE/YOURNUMBER; registration of your pbx to viatalk

The number after the slash (/) is required, otherwise you won't receive calls from ViaTalk because you tell our server that you want to answer calls for "s".
[viatalk]
context=global ;context to use
type=peer ; user type
fromuser=YOUR NUMBER ; callerid number
username=YOUR NUMBER
authuser=YOUR NUMBER
secret=YOUR PASSWORD
host= INSERT PROXY HERE ;host to contact
fromdomain= INSERT PROXY HERE ; domain you're on
nat=no ; network address translation (set to yes if you have problems)
canreinvite=yes
insecure=very ; security level
qualify=yes ; require password (we do require this)
dtmfmode=inband (needed on BOTH ends to allow keytone passing)
dtmf=inband ; same as previous item

EXTENSIONS.CONF

[outgoing]
exten => _1NXXNXXXXXX,1,dial(SIP/${EXTEN:0}@viatalk,60,Tt)
exten => _1NXXNXXXXXX,2,congestion
exten => _NXXNXXXXXX,1,dial,(SIP/1${EXTEN:0}@viatalk,60,Tt)
exten => _NXXNXXXXXX,2,Hangup

[incoming]
The incoming section MUST include as it's start YOUR NUMBER as the pattern match. Example for number 15181231234
exten =>_15181231234,1,START PROCESSING CALL FROM VIATALK TRUNK
OR
exten =>_1NXXNXXXXXX,1,START PROCESSING ANY VALID NUMBER

If you do not have this pattern in your incoming, you will not receive calls from viatalk, because you have not instructed your system to do so.

Additionally, be aware that your service codes may conflict with ViaTalk service codes. One primary example is *123, which is the service code to grab voicemail from the viatalk server. If you wish to use voicemail internall as well as through ViaTalk, you must use a code different than *123, or setup your outgoing plan to intercept the number you wish to use, but dial outbound as *123.

Example to use *86 (*VM) as your external voicemail config. Put the following in your outgoing section.

exten => *86,1,Dial(SIP/*123$@viatalk,120,t)


*4378 (*HELP) is our quick support dial, please make sure you're not using this code if you wish the feature to work.

Also, if you want per extension callback last caller, you'll want to define it in your extensions.conf. Otherwise, the last number to call your office will be used.

The extensions.conf file edit does not need to be in there, but is posted to allow understanding of a working extension setup for a viatalk "trunk" (outgoing/incoming line)

Thank you,
ViaTalk Engineering Dept

[rvtango]

To KLH: I've tried that and had problems with DTMF recognition. Had to talk with VT support guys and we came up with slightly different configuration that was posted in a previous thread

[akogan82]

I did get it working thanks for the help.

I am still having some issues with inbound DTMF when I called in from my cell phone. Still experimenting hopefully i'll find your old posts rvtango

[rvtango]

Have you set it as I posted?
I have no problem calling in from landline or cellie

[akogan82]

I've been messing with it and still have issues. I tested the authenticate command it worked once then kept saying incorrect the next call. What version of asterisk are you using?

[rvtango]

1.2.13
my old post is right before this thread ... just make sure you set

exten => s,n,SIPDtmfMode(inband)

in a context where your call comes in

[cjunevicus]

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they have flash movie walkthroughs.

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Give it a try... The hard work's been done for you:D

[baba2s]

I did get it working thanks for the help.

I am still having some issues with inbound DTMF when I called in from my cell phone. Still experimenting hopefully i'll find your old posts rvtango

Hello Akogan82,

I too like to setup ViaTalk with Asterisk 1.4.2. Can you please tell me the working sip.conf and extensions.conf settings ? After registration the incomming calls work for few minutes and later all go to Answering...

I read all the posts related to Asterisk and looks like the working settings for Asterisk 1.2 is not working properly in 1.4.2.

I see the OK Registration and out bound calls are working good
Name/username Host Dyn Nat ACL Port Status

ViaTalk/1xxxxxxxxxx 216.246.88.130 N 5060 OK (37 ms)

I see many 404
<--- SIP read from 216.246.88.130:5060 --->
SIP/2.0 404 Not Found



[May 11 14:13:30] NOTICE[2956]: chan_sip.c:15120 sip_poke_noanswer: Peer 'ViaTalk' is now UNREACHABLE! Last qualify: 34
[May 11 14:13:40] NOTICE[2956]: chan_sip.c:12140 handle_response_peerpoke: Peer 'ViaTalk' is now Reachable. (32ms / 2000ms)

Name/username Host Dyn Nat ACL Port Status
ViaTalk/1xxxxxxxxxx 216.246.88.130 N 5060 UNREACHABLE

Name/username Host Dyn Nat ACL Port Status
ViaTalk/1xxxxxxxxxx 216.246.88.130 N 5060 LAGGED (2052 ms)

[baba2s]

Lastly i am able to do setting to get incomming call too... Thanks to all who posted such important information.

Now i have one issue left. Out going calls with 7 or 10 digits are not working. Only 11 digits call r going through...Here is my extensions.conf

Asterisk 1.4.2
Not working..
exten => _NXXNXXXXXX,1,dial,(SIP/1${EXTEN:0}@viatalk,60,Tt)
exten => _NXXNXXXXXX,2,Hangup
Error is
-- Executing [xxxyyyzzzz@from-internal:1] Dial("SIP/7001-1007eef8", "(SIP/1xxxyyyzzzz@viatalk,60,rTt)") in new stack
[May 11 17:59:07] WARNING[3125]: channel.c:3024 ast_request: No channel type registered for '(SIP'
[May 11 17:59:07] WARNING[3125]: app_dial.c:1090 dial_exec_full: Unable to create channel of type '(SIP' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)

working...
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@ViaTalk,60,Tt)
exten => _1NXXNXXXXXX,2,congestion


Thanks

[IronHelix]

first, i generally recommend using the format tech/channel/exten, ie SIP/viatalk/${EXTEN}. If you aren't stripping any digits use ${EXTEN} not ${EXTEN:0}

also you don't want 'r' because it will override any progress tones (ringing) vt sends. You may not want tT because it means the callee can trasnfer the call... t and T mean caller and callee dont recall which one is which, at the asterisk prompt do 'show application dial' and itwill tell you.

Try something like this:

exten => _1NXXNXXXXXX,1,Dial(SIP/viatalk/${EXTEN},tT)
exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1)
exten => _NXXXXXX,1,Goto(1xyz${EXTEN},1) ; replace xyz with your area code

the result is that the first one is the only one that actually dials. If you dial 10 or 7 digit, it will reformat the number as 11 digit and Goto that number in the current context, which matches the first line.
ie if you dial 1234567, then 1234567 is the ${EXTEN}, so we goto 1 xyz 1234567. That matches the first line's pattern (1NXXNXXXXXX) so we start there, and now the full number is the ${EXTEN}.

hope that helps

[baba2s]

first, i generally recommend using the format tech/channel/exten, ie SIP/viatalk/${EXTEN}. If you aren't stripping any digits use ${EXTEN} not ${EXTEN:0}

also you don't want 'r' because it will override any progress tones (ringing) vt sends. You may not want tT because it means the callee can trasnfer the call... t and T mean caller and callee dont recall which one is which, at the asterisk prompt do 'show application dial' and itwill tell you.

Try something like this:

exten => _1NXXNXXXXXX,1,Dial(SIP/viatalk/${EXTEN},tT)
exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1)
exten => _NXXXXXX,1,Goto(1xyz${EXTEN},1) ; replace xyz with your area code

the result is that the first one is the only one that actually dials. If you dial 10 or 7 digit, it will reformat the number as 11 digit and Goto that number in the current context, which matches the first line.
ie if you dial 1234567, then 1234567 is the ${EXTEN}, so we goto 1 xyz 1234567. That matches the first line's pattern (1NXXNXXXXXX) so we start there, and now the full number is the ${EXTEN}.

hope that helps


Thanks IronHelix

The Dialplan worked for me
Thanks