TIPS TRUNK registered, No call can be made, why?

[JimmyKhine]

Hi all,

Is there expert in this forum, plz help?

I have an issue as the following:
TRUNK has registered with status " OK " and outbound call has created with prefix as 3000. However, call can't be made and I have received the log as below

== Using SIP RTP CoS mark 5
-- Called SIP/PQ/254207600000
[2016-07-01 18:06:35] WARNING[18785][C-00000008]: chan_sip.c:23220 handle_response_invite: Received response: "Forbidden" from '<sip:[email protected]>;tag=as0d753417'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:23] NoOp("SIP/132-00000000", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/132-00000000", "0?continue,1:s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/132-00000000", "RC=21") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/132-00000000", "21,1") in new stack
-- Goto (macro-dialout-trunk,21,1)
-- Executing [21@macro-dialout-trunk:1] Goto("SIP/132-00000000", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/132-00000000", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] Set("SIP/132-00000000", "CALLERID(number)=132") in new stack
-- Executing [9000254207600000@from-internal:8] Macro("SIP/132-00000000", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/132-00000000", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/132-00000000", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/132-00000000", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/132-00000000", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/132-00000000> Playing 'all-circuits-busy-now.gsm' (language 'en')
-- <SIP/132-00000000> Playing 'pls-try-call-later.gsm' (language 'en')
-- Executing [s@macro-outisbusy:5] Congestion("SIP/132-00000000", "20") in new stack
[2016-07-01 18:06:39] WARNING[19416][C-00000008]: channel.c:4861 ast_prod: Prodding channel 'SIP/132-00000000' failed
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/132-00000000' in macro 'outisbusy'

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With soft phone (X-Lite) I can registered and make the call. The setting on the soft phone, I have set with Proxy adress as 11.22.33.44:5061

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Here are the details of the TRUNK that I have done

type=peer
username=123456789
secret=xxxxxxxx
host=IP host(11.22.33.44)
fromuser=123456789
fromdomain=host server (sip.domain.com)
context=default
canreinvite=no
insecure=very
qualify=yes
nat=yes
port=5061
dtmfmode=rfc2833
disallow=all
allow=g729

This TRUNK has different port as 5061 and my asterisk is 5060

 

[geopeterwc]

@JimmyKhine ... You haven't provided enough information to be able to give you much help. Please provide the "status" information presented from login to the PBX console.

This will tell us what version of PBX (Incredible or PIAF) are you using. It would be helpful to also know: Is this installed on hardware (computer or Pi) or in the cloud? and ... Who is the VoIP Telephony Service provider?

/Pete./

 

[Sushant Parasher]

I will suggest you to try 'connection-reuse' under sip-ua and see the result.
or To make a outgoing call, your extensions should have context from-internal, not from-trunk.

 

[Ilan Stern]

Make sure you are using the correct port. some providers are very specific as to which port you can use. 5060 tends to be the standard and by default it is open, however some sip registrations will register to a variety of ports other than 5060. You might need to configure the firewall to specifically have the SIP registration talk to one port alone. Also be sure to check your NAT settings.

 

[JimmyKhine]

@ llan Stern - I have changed the port as the same with provider ans let the PBX port to be like that as well, also IP phone too. However, it doesn't work the Outbound still has a message as " Forbidden ". Why?

I have turned of the Firewall and and enable the NAT on router, unfortunately the situation doesn't change. Why?

 

[Ilan Stern]

Did you enable nat in the sip_nat.conf

externip=his nated IP
localnet=his local subnet

 

[Jake]

I'd ask your provider for help. The "forbidden" reply is from your provider usually an incorrect username.

 

[JimmyKhine]

Well, - llan Stern. How do I go to NAT as you explained?
The PBX server I am using, currently I have 4 providers of VOIP and they are working well with all the same setting from TRUNK, outbound calls and the rest. Don't know why this provider has strange things which I can't send Outbound calls. Since they are a telecom provider so I believe that the error would start from my end. Also, the PBX server from this provider which is PBX server Teles

 

[mainenotarynet]

Jimmy - you keep asking why things don't work, but you do not give us enough information, we can't help without certain information.

What version Asterisk
Is it on a Pi, a Virtual Machine, in the cloud, in your closet on an old chromebook, what.
Go into your PBX as root and copy the contents of

piafprompt />status

This status command will give us what we need (just change your IP that it shows (bad idea to post the real one)

We also nee the providers name that is hosed -- is it did4sale, callcentric, SIPStation - who is it giving you the problem

Providers usually give a sample config for FreePBX on their websites (I know voip.ms dos and it is 'usually' correct.

Give US mor information and we can give YOU more help

 

[Ilan Stern]

Well, - llan Stern. How do I go to NAT as you explained?
The PBX server I am using, currently I have 4 providers of VOIP and they are working well with all the same setting from TRUNK, outbound calls and the rest. Don't know why this provider has strange things which I can't send Outbound calls. Since they are a telecom provider so I believe that the error would start from my end. Also, the PBX server from this provider which is PBX server Teles

@JimmyKhine you can use Putty to SSH into the PBX server and check your config file. If you are hosting your own PBX then you or your IT person who setup the PBX should be able to SSH into the server and check the config file. If it is being hosted in the cloud then you should contact the company to make sure that they have all the correct settings.

 

[JimmyKhine]

#mainenotorynet.
I have installed FreePBX in a flash with the latest version

What I have seen from the TRUNK here is when I changed it the port " 5060 ". The TRUNK is UNREACHABLE, however, when I changed it to the port "50xx" which is given from provider. The TRUNK is registered with status " OK ", however, I can't make any outbound calls

Any solution, Please help?

 

[mainenotarynet]

You still won't tell us the provider - we cant help without it and if the provider isn't using 5060 that could be the problem - you need to contact them then. Also you MAY need to add the provider and the ports they use to the IPtables whitelist (unless of course you are not using that and you don't mind security holes you can drive a truck through and $1M phone bills.

I am no Guru by any stretch of tyeh imagination, but you need to provide as much info on your system (FreePBXin a flash latest version means squat as there is asterisk 11 & 13 running on CentOS, Raspian, BBB(I think) Debian, you get the idea) what hardware, VM, (in cloud or at home. As I said before, post the output of status on your pbx - that gives us everything, and we need that provider.

I don't say this to be mean or sound like an @$$H0l3 but your question is akin to asking a mechanic 'My car doesn't start -- Why). The more information you can give to us, the more we can TRY to help you. However a provider that uses special ports instead of 5060 sound like more trouble than it's worth.

 

[JimmyKhine]

You still won't tell us the provider - we cant help without it and if the provider isn't using 5060 that could be the problem - you need to contact them then. Also you MAY need to add the provider and the ports they use to the IPtables whitelist (unless of course you are not using that and you don't mind security holes you can drive a truck through and $1M phone bills.

I am no Guru by any stretch of tyeh imagination, but you need to provide as much info on your system (FreePBXin a flash latest version means squat as there is asterisk 11 & 13 running on CentOS, Raspian, BBB(I think) Debian, you get the idea) what hardware, VM, (in cloud or at home. As I said before, post the output of status on your pbx - that gives us everything, and we need that provider.

I don't say this to be mean or sound like an @$$H0l3 but your question is akin to asking a mechanic 'My car doesn't start -- Why). The more information you can give to us, the more we can TRY to help you. However a provider that uses special ports instead of 5060 sound like more trouble than it's worth.

#maineno. That is MTN and the latest version I am talking about here is asterisk 11, running on Centos. Also, I am self-hosting PBX server

 

[mainenotarynet]

I tried looking up MTN as a voip provider and got that they are in South AMerica.

Can you make calls on other trunks? Have you checked your outbound routes and Trunk Dialplan to one that MTN will allow? there are many possibilities as to why the calls are not working?

As I said I am no Guru and have done whatI know. Other Gurus may take it from here.

Good Luck

 

[JimmyKhine]

I tried looking up MTN as a voip provider and got that they are in South AMerica.

Can you make calls on other trunks? Have you checked your outbound routes and Trunk Dialplan to one that MTN will allow? there are many possibilities as to why the calls are not working?

As I said I am no Guru and have done whatI know. Other Gurus may take it from here.

Good Luck

100% that the other TRUNKs are working well. Only this TRUNK is not. I have done with what exactly they told me however, it can't be resolved