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 and packets now arrive and go where they should. Two problems remain. 1. Still can't register my phone The username and password are correct. I don't know what else to try. 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'. I changed the name of the context in pjsip's to 'voipms-inbound' and removed reference to '[mycontext]' from pjsip.conf and extensions.conf as they were superfluous. The endpoint section of pjsip.conf now reads: [voipms] type = endpoint transport = transport-udp context = voipms-inbound ... The bottom part of extensions.conf (with the phone number obfuscated) is now: [voipms-inbound] exten => 3115552368,1,Goto(hello,200,1) [phones] exten => 101,1,Dial(PJSIP/yealink) [hello] exten => 200,1,Answer() same => n,Playback(hello-world) same => n,Hangup() The idea was for any inbound call to the public network number to immediately go to extension 200, play the message and hang up, and you could still call extension 200 to here it from inside.
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-inboundSteve, 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230524/31b767dd/attachment.html>
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 )
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.