Re: [asterisk-users] chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
> On Jan 2, 2024, at 23:17, the...@sys-concept.com wrote:
>
> On 1/2/24 15:13, aster...@phreaknet.org wrote:
>>> On 1/2/2024 4:03 PM, the...@sys-concept.com wrote:
>>> I'm using asterisk-16.30.1
>>>
>>> When I try to call another asterisk server over IAX I get a busy signal,
>>>
>>> chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
>>> -- IAX2/192.168.143.1:4569-656 is circuit-busy
>>>
>>> Asterisk-16.16 is working normally, no congestion error.
>> There is not enough information for anyone to really help or comment on this.
>> Dialplan and IAX2 configuration on both sides of the trunk?
>> CLI output on both sides with iax2 debug enabled?
>
> It is very simple:
>
> Local Asterisk, iax.conf:
>
> [clinic_server]
> type=friend
> host=dynamic
> context=internal
> disallow=all
> allow=ulaw
> allow=alaw
> requirecalltoken=no
> callgroup=1
> pickupgroup=1
>
> extension.conf:
>
> exten => 4,1,Dial(IAX2/home_server:@${clinic_server}/${EXTEN},60,rw)
> exten => 4,n,Hangup() Remote Asterisk iax.conf:
>
> [home_server]
> type=friend
> host=dynamic
> secret=
> context=extensions
> disallow=all
> allow=ulaw
> allow=alaw
> callgroup=1
> pickupgroup=1
>
> Remote extension.conf:
>
> exten => 4,1,Dial(SIP/4,15,trw)
> exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2)
> exten => 4,n(line2),Dial(SIP/54,20,rw)
> exten => 4,n(vmail),Voicemail(4)
> exten => 4,n,Voicemail(4)
> exten => 4,n,Hangup()
>
>
You have no internal context in your dialplan. But in your iax.conf you specify
internal as your context.
-H
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Re: [asterisk-users] PJSIP not performing outbound authentication
On Wed, Jun 21, 2023 at 05:19:11PM +, 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 > logging I notice that the outgoing invite does not have an authentication > line. Why is Asterisk not sending credentials to the ISP? SIP transactions > are: > > INVITE > < 100 TRYING > < 403 FORBIDDEN > > Or is this normal? Must Twilio respond with a 407 which will cause Asterisk > to authenticate? > > Twilio has a nice technical document to setup a trunk with PJSIP. It includes an example for a pjsip_wizard.conf https://assets.cdn.prod.twilio.com/documents/TwilioElasticSIPTrunking-AsteriskPBX-Configuration-Guide-Version2-1-FINAL-09012018.pdf Maybe that helps. And make sure for your outgoing calls to set the callerid to a valid caller Id which ist authorized with your twilio account. It will not allow outgoing calls if the number is not recognized by twilio -H -- Henning Follmann | hfollm...@itcfollmann.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with inbound connection and registering phone
On Tue, May 23, 2023 at 02:24:02PM -0400, Steve Matzura wrote: > 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 thought I'd try > Zoiper, but none of the download buttons at > https://www.zoiper.com/en/voip-softphone/download/current did anything when > clicked, so I set up a spare line on a Yealink TA33 phone that is connected > to another much much older Asterisk implementation running a piece of > amateur radio gear called Allstar. The version of Asterisk used in the > Allstar project is ancient--like 1.4 or 1.6--and the configuration syntax > and options are quite different, so I didn't get lulled into thinking I'd > just clone that configuration on my newer Asterisk implementation. > > > The "Definitive Guide" shows everything about adding phones as SQL > statements, so I made some educated guesses as to what to put into > pjsip.conf. Something's obviously wrong because the phone won't authenticate > (see below). > > > Here's how I set it up in pjsip. > > > [yealink] > transport=udp > type=auth > auth_type=userpass > username=Steve > password=Steve > [yealink] type=aor > [yealink] > type = endpoint > transport = transport-udp > context = phones > disallow = all > allow = ulaw > ; allow=g729 ; uncomment if you support g729 > auth = yealink > aors = yealink > > > Here's how I set it up in extensions: > > > [phones] > exten => 101,1,Dial(PJSIP/yealink) > > Here's the error I get on the Asterisk console: > > > [May 23 13:42:56] NOTICE[45189]: res_pjsip/pjsip_distributor.c:676 > log_failed_request: Request 'REGISTER' from '"Steve" > ' failed for '192.168.1.228:5060' (callid: > 0_1554187534@192.168.1.228) - Failed to authenticate > > > What did I omit? > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Henning Follmann | hfollm...@itcfollmann.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP dial via proxy
On Thu, Jul 21, 2022 at 02:46:07PM +1200, David Cunningham wrote:
> Hello,
>
> We have an Asterisk dial which sends the call via a proxy using //, for
> example:
>
> Dial(SIP/${EXTEN}@peer_address//proxy_address)
>
> Does anyone know how we can make the SIP to the proxy use TCP? We tried
> making proxy_address match a peer in sip.conf with "transport = tcp" but
> that didn't seem to work. We are using chan_sip.
>
> Thanks very much for any advice.
>
Have you tried to define
outboundproxy=proxy_address
in your sip.conf?
-H
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Re: [asterisk-users] Listen on 2 of 3 IP addresses
On Fri, Jul 15, 2022 at 08:57:46AM -0300, Joshua C. Colp wrote: > On Fri, Jul 15, 2022 at 1:37 AM David Cunningham > wrote: > > > Hello, > > > > We have an Asterisk server with 3 IP addresses, and need to listen on only > > 2 of those. This is with chan_sip. Does anyone know if it's possible? > > > > If Asterisk listens on the third address then it seems to cause problems > > with the media address put in the SDP for our use case. > > > > It's not. The chan_sip module allows you to bind to one thing, either a > specific address or an any address. > Well... maybe chan_sip cannot, but your OS can restrict traffic on any port/iface. -H -- Henning Follmann | hfollm...@itcfollmann.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] stir/shaken
Hello,
I have some trouble reading the headers.
Asterisk 16
in my dial plan I have these:
...
exten => _X,n,NoOp(Number of STIR/SHAKEN identities: ${STIR_SHAKEN(count)})
exten => _X,n,NoOp(First STIR/SHAKEN identity: ${STIR_SHAKEN(0,identity)})
exten => _X,n,NoOp(First STIR/SHAKEN attestation: ${STIR_SHAKEN(0,attestation)})
...
and I do get this:
-- Executing [@incoming:2] NoOp("PJSIP/flowroute-003c", "Number of
STIR/SHAKEN identities: 1") in new stack
-- Executing [**@incoming:3] NoOp("PJSIP/flowroute-003c",
"First STIR/SHAKEN identity: +1") in new stack
-- Executing [**@incoming:4] NoOp("PJSIP/flowroute-003c",
"First STIR/SHAKEN attestation: ") in new stack
Why do I not see the attestation?
Also I do not see any validation. What am I missing here?
-H
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