Displaying 20 results from an estimated 200 matches similar to: "Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?"
2016 Jan 18
2
Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".
Would greatly appreciate any input into this currently-unanswered
question on the forum:
http://forums.asterisk.org/viewtopic.php?f=1&t=96496
I posted it on Jan 6th, have tried so many things, so much forum/list
searching and late nights since, but have had to admit defeat.
Rather than duplicate it all here, I've posted my logs and conf files
on that thread, too.
Problem is that while
2018 Jul 28
2
SRV with pjsip on Asterisk 15.5: yes or no?
I'm trying to configure sip2sip, which says:
http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk
"Asterisk, is currently unable to handle more that one result for a
DNS SRV lookup, and the Asterisk configuration needed for getting it
work with the SIP2SIP service is not trivial"
It then gives a complex multi-section workaround in SIP. I remember
reading there'd be
2013 Jan 24
1
How configure asterisk server extension.conf.
Hi,
I have to create scenario like following,
I have 2 sip soft phone.I configured Asterisk server on local network, on Linux.With two soft-phone , local asterisk sever, i able to communicate.Now i have communicate with other network SIP client.For that i have opened account at @sip2sip.info, they provided me credentials.Then i registered one SIP phone to local Asterisk sever and another to
2016 Apr 25
2
Is set_var allowed with pjsip_wizard.conf ?
Hello,
I've just discovered PJSIP 's support of set_var setting in pjsip.conf.
Is this setting also supported in pjsip_wizard.conf ?
On a fresh 13.8.2, it doesn't seem but I may have missed somthing.
Regards
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2016 Apr 25
2
Is set_var allowed with pjsip_wizard.conf ?
On Mon, Apr 25, 2016 at 10:00 AM, George Joseph <gjoseph at digium.com> wrote:
>
>
> On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4h07 at gmail.com> wrote:
>
>> Hello,
>>
>> I've just discovered PJSIP 's support of set_var setting in pjsip.conf.
>> Is this setting also supported in pjsip_wizard.conf ?
>> On a fresh 13.8.2, it
2016 Apr 25
2
Is set_var allowed with pjsip_wizard.conf ?
On Mon, Apr 25, 2016 at 11:11 AM, Olivier <oza.4h07 at gmail.com> wrote:
>
>
>
> 2016-04-25 18:14 GMT+02:00 George Joseph <gjoseph at digium.com>:
>
>>
>>
>> On Mon, Apr 25, 2016 at 10:00 AM, George Joseph <gjoseph at digium.com>
>> wrote:
>>
>>>
>>>
>>> On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4h07
2011 Apr 22
2
Cannot call to my server with SIP
Hello,
I cannot call my server over the internet with SIP anymore.
Even when I do a maximum logging on my firewall, I don't see packets
coming from outside. I've tried it from an ekiga.net account and an
sip2sip.info account. What could be wrong? I would expect incoming
traffic on port 5060 UDP...
The account is "paul at vandervlis.nl". This should connect trought DNS to
the
2010 Sep 16
4
one way audio for xlite clients behind NAT
I am having a one way audio issue with xlite clients behind NAT. They can
connect to the server and make calls but no audio is heard on the other
end.
my sip conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" <XXXX>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
2011 Apr 20
1
dtmf payload type problem during faxing..
Hello,
We have a sip trunk between our voip operator and our asterisk 1.6.2.9
We have no problem during voice communications.
But we can not send any t38 fax via this gateway.
We tried to trace the error made some tests..
There are 2 main tests we tried to do.
As i learned their voip path is like .. we connect to session border
controller..then it routes the call to a cisco media gateway if the
2014 Oct 13
1
asterisk stun setup , not using public ip returned by stun server
Dear all,
I have enabled stun module and configured it in asterisk , but
asterisk not using stun returned public ip address for any of the sip
requests going out of my network.
i have done settings as below
res_stun_monitor.conf settings:
[general]
stunaddr = stun.ideasip.com
stunrefresh = 30
stun show status
Hostname Port Period Retries Status ExternAddr
2016 Feb 17
2
Asterisk 13.6.0/The simplest TCP configuration does not work
On Wed, Feb 17, 2016 at 8:56 AM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:
> I made some progress. The first thing I have realized is that it is my
> Twilio configuration in pjsip_wizard.conf that was killing me. I have since
> removed that entire file from /etc/asterisk and I am able to make
> "from-internal" context calls (i.e., calls that do not
2016 Feb 17
2
Asterisk 13.6.0/The simplest TCP configuration does not work
Yes, it is enabled on port 5060. I do receive a TCP ACK back from the
server, so I know the TCP segment is received at the server hosting the
Asterisk build.
On Wed, Feb 17, 2016 at 8:36 AM, A J Stiles <asterisk_list at earthshod.co.uk>
wrote:
> On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote:
> > OK. Let me ask this. Is anything else necessary, except choosing TCP as
> the
2020 Apr 06
2
Outgoing PJSIP using Kamailio
Hello,
We have a provider which is using Kamailio as front end. Our asterisk
13/chan_sip server has no problem to register and pass/receive calls
form this provider.
Now we want to move to asterisk 16/pjsip and face problem. Registration
is OK but when we pass a call our INVITE never receive answer from the
provider. We opened a ticket to their support but in the mean time we
want to know
2016 Feb 19
2
No matching endpoint found for incoming call from SIP trunk
Thanks George, for your mighty quick response.
I made the changes (re: server_uri_pattern etc.) and still, no luck--it
fails for the same error.
BTW, there is nothing for transport (but this is the same config from my
SIP/UDP + Twilio days, which worked):
*CLI> pjsip show transport twilio-siptrunk
Unable to find object twilio-siptrunk.
*CLI> pjsip show identifies
No objects found.
I did
2016 Feb 19
2
No matching endpoint found for incoming call from SIP trunk
Hello,
I have an Asterisk 13.6.0 PBX using PJSIP connected to the Twilio gateway.
I am able to make calls outbound through the gateway, but I am not able to
make calls into the PBX from external PSTN.
Specifically, an incoming call is _received_ by Asterisk, but it is not
able to route the call internally owing to the following error:
[Feb 18 21:08:47] NOTICE[4606]:
2016 Feb 17
2
Asterisk 13.6.0/The simplest TCP configuration does not work
On Wed, Feb 17, 2016 at 12:13 PM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:
> Wow. Incredible. That worked. The backslash is important there; I kept
> trying with no backslash and followed the instructions in
> pjsip_wizard.conf.sample (in configs/samples) and it says we have to say
>
> transport=tcp ; the only example however talks about ipv4.
>
> Is
2020 Feb 27
2
pjsip: how to survive rejected registrations?
Hi,
sometimes 'pjsip show registrations' shows registrations to the VOIP provider
as Rejected. I have already added
max_retries = 0
auth_rejection_permanent = no
in pjsip_wizard.conf and still asterisk does not recover.
I need asterisk to keep trying to register and to renew the registration
without requiring manual intervention. How can I make asterisk do that?
2020 Feb 29
2
pjsip: how to survive rejected registrations?
Le 28/02/2020 à 23:43, hw a écrit :
> On Thursday, February 27, 2020 3:03:47 PM CET hw wrote:
>> Hi,
>>
>> sometimes 'pjsip show registrations' shows registrations to the VOIP
>> provider as Rejected. I have already added
>>
>>
>> max_retries = 0
>> auth_rejection_permanent = no
>>
>>
>> in pjsip_wizard.conf and still
2023 Jun 21
1
PJSIP not performing outbound authentication
On Wed, Jun 21, 2023 at 05:19:11PM +0000, TTT wrote:
> I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP
> (Twilio) who requires outbound authentication. My pjsip.auth.conf contains:
>
> [Twilio]
> type=auth
> auth_type=userpass
> password=mysecret
> username=myun
>
> However, my calls using the trunk are rejected with a 403. Using pjsip
>
2023 Jun 21
1
PJSIP not performing outbound authentication
I didn't use pjsip_wizard, I'm kind of crafting this by hand as I learn. I actually have a plain asterisk, and a FreePBX, system to help me learn. I sometimes create something in FreePBX to see what it does to the config files. So that's how I modelled my pjsip.X.conf files
If I issue the command "pjsip show endpoint Twilio" it does show that outbound_auth=Twilio
Does