Re: [asterisk-users] Problems Solved, two left

2023-05-27 Thread Steve Matzura
Thanks, Daryl. I fixed this before I saw this message by changing my 
connectivity from SIP to IVR/IAX on voip.ms's Manage DID Numbers page. 
I'll keep this one in my notes, though, should I ever do this again with 
SIP.



On 5/26/2023 7:42 PM, Daryl Richards wrote:

On 2023-05-23 7:22 p.m., Steve Matzura wrote:

And I think they're both small.


[May 23 18:34:12] NOTICE[46582]: res_pjsip_session.c:3968 new_invite: 
voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected 
because extension not found in context 'voipms-inbound'.


Steve,

In your voip.ms console, go to Account Settings -> Inbound Settings, 
and set Device Type to "IP PBX Server..." instead of "ATA device..."


This will fix the 's' instead of the number.




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Re: [asterisk-users] Problems Solved, two left

2023-05-26 Thread Daryl Richards

On 2023-05-23 7:22 p.m., Steve Matzura wrote:

And I think they're both small.


[May 23 18:34:12] NOTICE[46582]: res_pjsip_session.c:3968 new_invite: 
voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected because 
extension not found in context 'voipms-inbound'.


Steve,

In your voip.ms console, go to Account Settings -> Inbound Settings, and 
set Device Type to "IP PBX Server..." instead of "ATA device..."


This will fix the 's' instead of the number.



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Re: [asterisk-users] Problems Solved, two left

2023-05-24 Thread Doug Lytle

On 5/24/23 09:56, Steve Matzura wrote:
I don't understand your explanation because in the two files whose 
contents I posted, there's nothing routed to anything called just 's'. 
However, I've seen that in the error messages and it stumped me, too. 
No 'start' either.


Steve,

Please make sure you reply back to the list, so others can help also.

As for why it's sending to the start extension, I cannot say since I am 
using IAX trunking with voip.ms and I get a DID for inbound matching.


Doug
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Re: [asterisk-users] Problems Solved, two left

2023-05-24 Thread Stefan Tichy
Am Wed, May 24, 2023 at 09:40:18AM -0400 schrieb Steve Matzura:
> 
> On 5/24/2023 7:49 AM, Stefan Tichy wrote:
> > Am Tue, May 23, 2023 at 07:22:22PM -0400 schrieb Steve Matzura:
> > 
> > > 1. Still can't register my phone
> > > The username and password are correct. I don't know what else to try.
> > You can start a sip trace from the asterisk console.

> > REGISTER sip:192.168.1.185:5060 SIP/2.0

> > Authorization: Digest username="Steve", realm="asterisk",

> > [May 24 09:26:13] NOTICE[47903]: res_pjsip/pjsip_distributor.c:676 
> > log_failed_request:
> > Request 'REGISTER' from .'  - No matching endpoint found

In the endpoint section there is a parameter identify_by (default:
"username,ip"). "username" means, the the from-user is used.


Use "yealink" or "Steve" for both user names. Using different names
for a phone make it just more complicated. You might have to change
the phone configuration.




> [yealink]
> type = aor
> contact = sip:Steve@192.168.1.185

There should be no "contact" parameter for a phone. The phone sends
the required information with the register request.



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Stefan Tichy  ( asterisk3 at pi4tel dot de )

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Re: [asterisk-users] Problems Solved, two left

2023-05-24 Thread Doug Lytle

On 5/24/23 08:03, Steve Matzura wrote:


***  extensions.conf  ***


[general]

[globals]

; Make sure to include inbound prior to outbound because the 
_NXXNXX handler will match the incoming call and create a loop

include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXX,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _1NXXNXX,n,Hangup()
exten => _NXXNXX,1,Dial(PJSIP/1${EXTEN}@voipms)
exten => _NXXNXX,n,Hangup()
exten => _011.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 
1 in front


[voipms-inbound]
exten => {redacted},1,Goto(hello,200,1) ; My  DID

[phones]
exten => 101,1,Dial(PJSIP/yealink)

[hello]
exten => 200,1,Answer()
    same => n,Playback(hello-world)
    same => n,Hangup()




Your inbound is being sent to s (start extension) instead of your DID, 
so it's not matching.  So, you'll need to find out where in your 
dialplan it's being mapped to s.


Did you know that voip.ms supports IAX2 natively?  Working much better, 
in my opinion, that SIP.


Doug

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Re: [asterisk-users] Problems Solved, two left

2023-05-24 Thread Stefan Tichy
Am Tue, May 23, 2023 at 07:22:22PM -0400 schrieb Steve Matzura:

> 1. Still can't register my phone
> The username and password are correct. I don't know what else to try.

You can start a sip trace from the asterisk console.

pjsip set logger on

There should be a REGISTER from the phone, a Response 401 and an ACK
from the phone. Then asterisk should receive another REGISTER with
an additional "WWW-Authenticate" header. The response could be 401
again or 403 or something else.


> 2. Asterisk can't find the extension in my inbound context.


> [May 23 18:34:12] NOTICE[46582]: res_pjsip_session.c:3968 new_invite: 
> voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected because
> extension not found in context 'voipms-inbound'.

This can happen if there is no contact_user parameter.

"contact_user=" sets the SIP contact header's user portion of the SIP URI
this will affect the extension reached in dialplan when the far end calls
you at this ; registration. The default is 's'.


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Re: [asterisk-users] Problems Solved, two left

2023-05-24 Thread Doug Lytle

On 5/23/23 19:22, Steve Matzura wrote:
voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected 
because extension not found in context 'voipms-inbound


Steve,

Could we see your dialplan for voipms-inbound?


I'm using voip.ms as well, but have not converted from chan_sip yet.  My 
voip-ms inbound extensions.conf below (Phone number changed to protect 
the innocent)


[voipms]

include => voicemail

exten => 5175551212,1,Answer()
   same => n,Gosub(check_blacklist,s,1)
   same => n,Gosub(get_callerid,s,1)
   same => n,Gosub(check_for_direct,s,1)
   same => n,Set(_ARG1=4259)
   same => n,Gosub(extension_timeouts,s,1(${ARG1}))
   same => n,Queue(home,WwtTkKr,,,23)
   same => n,NoOP(Dial Status: ${QUEUESTATUS})
   same => n,NoOP(Hangup Cause: ${HANGUPCAUSE})
   same => n,Gosub(s-${QUEUESTATUS},s,1(${ARG1}))

Doug

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