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.,

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

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

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.

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]

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

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

[asterisk-users] Problems Solved, two left

2023-05-23 Thread Steve Matzura
And I think they're both small. Solved: tcpdump showed no packets coming in, so I went to my DID provider's Website to discover to my intense embarrassment that the DID number had been set up forwarded to their voicemail. I got egg on my face for this one. I changed that setting to SIP/IAX