Re: [asterisk-users] Problems Solved, two left
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. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Solved, two left
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. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Solved, two left
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Solved, two left
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. -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Solved, two left
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
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'. -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Solved, two left
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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