Displaying 20 results from an estimated 700 matches similar to: "Outgoing PJSIP using Kamailio"
2020 Apr 08
0
Outgoing PJSIP using Kamailio
On Mon, Apr 6, 2020 at 2:06 PM Administrator <admin at tootai.net> wrote:
> 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
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]
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 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 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
2018 Apr 06
2
PJSip CallerID Question
I have multiple Asterisk instances set up in different locations and
would like to modify the callerID of inbound calls to identify which
instance the call is coming from.? I knew how to do that with the old
sip format, but can't seem to figure it out with PJSip.
For example:
Currently Location A, extension 10 calls Location B, extension 20.?
CallerID on Extension 20 displays
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 Jan 31
3
how to make asterisk set cos values
Hi,
examining the network traffic with wireshark shows that asterisk does not set
any QoS values at all.
What do I need to do to make asterisk set QoS values (on Centos 7)?
The wiki says to use vconfig to set QoS values[1]. What does the skb-priority
need to be set to? How do you use vconfig on interfaces that are not VLAN
interfaces?
Is it generally impossible to set QoS values on
2020 Apr 19
1
how to make a bug report
On Saturday, April 18, 2020 5:42:11 PM CEST Joshua C. Colp wrote:
> On Sat, Apr 18, 2020 at 8:47 AM hw <hw at gc-24.de> wrote:
> > Hi,
> >
> > how do I make a bug report? I filled in the form to make a report and
> > https://issues.asterisk.org/jira/issues/?filter=-2 still shows no issues
> > reported by me.
>
> If successful then JIRA will redirect
2020 Apr 18
2
how to make a bug report
Hi,
how do I make a bug report? I filled in the form to make a report and
https://issues.asterisk.org/jira/issues/?filter=-2 still shows no issues
reported by me.
If someone knows how to get asterisk to re-register when using pjsip after the
registration shows as Rejected, like after the internet connection to the VOIP
provider goes away (and comes back), please let me know. This bug makes
2015 Oct 19
2
Modify Contact in PJsip
Hi Guys
We are using the wizard to configure our pjsip trunk(see below)
How do we get this setting to work
contact_user=username
We want to change the contact field in the sip invite to display the
username of the trunk
[trunk_defaults](!)
type = wizard
transport = transport-udp
endpoint/allow_subscribe = no
endpoint/allow = !all,g729
aor/qualify_frequency = 30
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 Apr 16
2
PJSIP error No auth credentials for realm(s) 'asterisk' in challenge
Hi all,
we are trying to move our servers from chan_sip to chan_pjsip. At this
time no problems with phones, they all register fine and can place
calls. But for a trunk we face problem and can't place calls despite the
fact that registration is OK. What we get is:
[2018-04-16 16:08:33] WARNING[18665]:
res_pjsip_outbound_authenticator_digest.c:178
2020 Jun 01
1
Asterisk 16 Certified 16.8 and MagicJack Incoming Calls
I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and
converted form SIP to PJSIP using the python script as a start and then
mofiying from there. I ran into an issue when testing that incoming calls
from MagicJack would go silent after about 10 seconds. This happened while in
the automated attendant area. This problem did not occur with Asterisk 13
LTS. I reverted PJSIP
2017 Sep 26
2
asterisk pjsip as voip client with multiple registrations
hi,
i want use asterisk+pjsip as voip client with multiple registrations
(perf testing)
i'm using this example configuration for one account
[308]
type=registration
outbound_auth=308
server_uri=sip:308 at example.com:5060
client_uri=sip:308 at example.com:5060
[308](auth-userpass)
username=308
password=pass
[308](aor-single-reg)
contact=sip:example.com:5060
[308](endpoint-basic)
2017 Jun 05
3
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 06/05/2017 at 11:30 AM, Joshua Colp wrote:
> On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote:
>> On 06/04/2017 at 01:41 PM Telium Technical Support wrote:
>>> Just a guess (without knowing about your network), but are the two ends
>>> points on public networks and visible to one another? If not the reinvite
>>> may be passing an internal (nat'ed)
2016 Jul 01
2
CALLERID on pjsip doesn't work?
Asterisk 13.8
Is CALLERID(all) supposed to wok for pjsip? When I do this:
exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>)
same => n,Dial(PJSIP/phone123, 30)
I expect the callerid to be as set, but is always seems to be "phone123",
the name of the endpoint.
Andrew
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2015 Mar 06
2
res_pjsip endpoint config object's 'identify_by' option needs new value "uri".
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 requesting to add new value for endpoint option identify_by. The value
'uri'.
Simple config (cutted):
[siptrunk]
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
2015 Mar 15
2
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic
configuration works, and I am connected to a SIP trunk using SIP.US, and
have set up my inbound calling which works correctly (when I call my PBX
DID, the call does come into my PBX network).
The issue is that I am not able to make outbound calls, because the call
fails with the error: