Displaying 20 results from an estimated 2000 matches similar to: "Problems Solved, Two Remaining"
2023 May 24
0
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
2023 May 23
3
Problems with inbound connection and registering phone
I have two problems. The first is that when I dial my number from a
phone on the Internet or any phone outside my LAN, Asterisk does not
respond in any way, which means somehow my system is not picking up the
fact that there's an incoming call to it.
The second problem is that I thought I'd try an internal phone to see if
I could get the hello-world stuff working at the least. I
2019 Apr 05
2
pjsip endoint woes
I'm trying to set up pjsip to work with an obi202 and google voice. But
I can't configure the endpoint.
pjsip:
[obi202-auth](!)
type = auth
auth_type = userpass
password = <mypass>
[obi202-aor](!)
type = aor
max_contacts = 2
; ===== endpoints ========
[gv-voice](obi202-endpoint)
auth = gv-voice
aors = gv-voice
identify_by=auth_username
;identify_by=username ; I also tried
2023 May 23
3
Problems Solved, two left
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
2023 May 24
0
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
> _NXXNXXXXXX handler will match the incoming call and create a loop
> include => voipms-inbound
> include => voipms-outbound
>
> [voipms-outbound]
> exten =>
2019 Apr 08
2
pjsip endoint woes
On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote:
> On 4/5/19 10:36 AM, sean darcy wrote:
> > I'm trying to set up pjsip to work with an obi202 and google voice. But
> > I can't configure the endpoint.
> >
> > pjsip:
> >
> > [obi202-auth](!)
> > type = auth
> > auth_type = userpass
> > password = <mypass>
> >
>
2019 Jan 26
3
INVITE from DID: No matching endpoint found but completes the call anyway
I have a trunk set up for the DID from my provider:
[my_provider]
type=registration
outbound_auth=my_provider
server_uri=sip:sip.example.com
client_uri=sip:my_username at sip.example.com
retry_interval=60
[my_provider]
type=auth
auth_type=userpass
password=123456
username=my_username
[my_provider]
type=aor
contact=sip:sip.example.com:5060
[my_provider]
type=endpoint
context=from-my_provider
2017 Dec 03
2
PJSIP OPTIONS
Right now it reply 401 Unauthorized with message in log "No matching
endpoint ..."
on Content 0 should reply 200 OK I guess
<--- Received SIP request (376 bytes) from UDP:10.30.100.41:5060 --->
OPTIONS sip:10.30.100.27:5080 SIP/2.0
Via: SIP/2.0/UDP
10.30.100.41;branch=z9hG4bKf5eb.1ac76487000000000000000000000000.0
To: <sip:10.30.100.27:5080>
From:
<sip:vprx00 at
2018 Jul 28
2
Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?
Using pjsip 2.7.2 on Asterisk 15.5
Really struggling to make sense of translating these old 1.8 SIP
instructions into a neat pjsip_wizard conf suitable for 2018
http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18
In pjsip_wizard.conf, I have the following, which seems to get me
registered, and it responds to an incoming call, but I always get
this:
[Jul 28 18:32:29]
2017 Dec 03
2
PJSIP OPTIONS
If understand correctly type=identify is more for sip trunk
configuration ?
;[mytrunk]
;type=identify
;endpoint=mytrunk
;match=198.51.100.1
;match=198.51.100.2
In chan_sip it was just reply 200 OK on keepalive packet without need
define trunks.
volga629
On Sun, 3 Dec, 2017 at 10:45 AM, Joshua Colp <jcolp at digium.com> wrote:
> On Sun, Dec 3, 2017, at 10:42 AM, volga629 at
2010 Oct 28
0
Intermittent failure when placing calls - unable to create channel of type SIP
Hello community,
I've been running Asterisk on an embedded device for about six months, and
my operation has been largely trouble-free. I'm hoping I could get some help
with a minor problem:
Every week or three, my PBX gets stuck in a state where it can receive
calls, but it becomes completely unable to originate outgoing calls until I
do a "sip reload". After doing the SIP
2016 Oct 25
0
Asterisk 13.12.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 13.12.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.12.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New
2019 Jun 06
2
Fail2ban for asterisk 16 PJSIP
Hello
Anyone have a working copy of Fail2ban asterisk filter asterisk.conf
for Asterisk 16 running PJSIP.
I have tried 10 different filters but none of them show any matches when testing with
fail2ban-regex
I see date template hits but no matches....
My log
[2019-06-06 15:37:20] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" <sip:2405 at
2016 Oct 25
0
Asterisk 14.1.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 14.1.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.1.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New
2015 Mar 13
2
PJSIP/Asterisk 13.1.0 incoming call via DID: No matching endpoint found
I have a working Asterisk 13.1.0 running, and I am trying to configure a
SIP trunk for outbound and inbound calling, and a DID for the Asterisk
server, which is used for incoming calls from PSTN.
I configured my SIP.US trunks (showing one gateway, gw1, here for brevity,
have two: gw1 & gw2, which are both configured on my end):
[sonnyGW1]
type=registration
transport=transport-udp
2015 Mar 06
0
res_pjsip endpoint config object's 'identify_by' option needs new value "uri".
On Fri, Mar 6, 2015 at 3:46 PM, Dmitriy Serov <serov.d.p at gmail.com> wrote:
> 07.03.2015 0:24, Kevin Harwell ?????:
>
> On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com>
> wrote:
>
>> Hello.
>>
>> Asterisk 13.2.
>> I transfer configs from chan_sip to res_pjsip.
>> In chan_sip i have
2017 Jun 15
2
asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'
On 06/14/2017 at 10:17 PM, Joshua Colp wrote:
> On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote:
>
> <snip>
>
>>
>> I can now say, that asterisk / pjsip seams to work *mostly* as expected.
>> Just one exception - and that's the package in question, which can't be
>> seen in tcpdump.
>>
>> I extended the above patch by adding
2015 Mar 06
0
res_pjsip endpoint config object's 'identify_by' option needs new value "uri".
On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com> wrote:
> Hello.
>
> Asterisk 13.2.
> I transfer configs from chan_sip to res_pjsip.
> In chan_sip i have "match_auth_username=yes" and have nothing in pjsip.
>
> I have a lot of endpoints and registrations on same SIP server. And it's
> problem in pjsip now. Is not it?
>
> I
2016 Sep 09
2
13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
Hello!
Upgraded 13.10 to 13.11.1 today and now I see messages in log:
[Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request
'REGISTER' from '"3563" <sip:3563 at 192.168.32.254>' failed for
'192.168.32.116:5060' (callid: 0_1409534529 at 192.168.32.116) - No
matching endpoint found
or
[Sep 9 12:56:14] NOTICE[10163]
2013 May 21
1
Failed to authenticate device "Ext 110"
I'm having a strange problem recently with a Yealink SIP-T28P phone
connected to Asterisk 11.4.0 via openvpn. It was working fine for months,
and now when I dial anything from the phone, it shows "Forbidden", and the
Asterisk console shows:
[May 21 10:47:49] NOTICE[28518][C-00000004]: chan_sip.c:25189
handle_request_invite: Failed to authenticate device "Ext 110" <