Outbound SIP Issue

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Outbound SIP Issue

TelcomPBX

Hi, Folks we are setting up our First Issabel instance but we are running Elastix 4 already for this client. But we were migrating.

We use Vitelity.net SIP trunks which are IP authenticated no registration string. etc. It should be very straight forward. Inbound works fine. Is there anything different from Elastix 4 to Issabel that would effect this. Otherwise we assume we are just missing a step etc. Is there any known Outbound SIP issue.

supa

TelcomPBX Nothing is different. Outbound SIP works fine.

Fail2ban was added. So maybe check iptables to make sure fail2ban is not banning anything.

venturinog

TelcomPBX Hi. If your PBX has not the public IP, did you configured the exterip parameter?

TelcomPBX

We are still struggling with outbound SIP yet it worked the exact same way on Elastix 4 (the real 4) not 3CX. It is an IP authentication style so very simple. Any ideas what we could look at it, what we may have over look at. We are getting close to just installing a saved copy of Elastix 4 we have. We know at some point this will be an issue.

supa

It works perfectly for me 100% of the time. You did not post any information about your setup or any log files so it could be anything. We don't have anything to go on.

Did you at least watch asterisk CLI while making a call to see what happens? I am assuming you created an outbound route? You didn't post any information other than to say it doesn't work.

TelcomPBX

SIP log for outbound call:

<--- SIP read from UDP:24.112.1.61:5063 --->
INVITE sip:12027621401@SERVERHOSTNAME.com SIP/2.0
Via: SIP/2.0/UDP 24.112.1.61:5063;branch=z9hG4bK-22107674
From: "Main" <sip:9001@SERVERHOSTNAME.com>;tag=afe115ae4f310064o3
To: <sip:14432439618@SERVRHOSTNAME.com>
Call-ID: 1b2eef6-bba3b7bc@192.168.0.10
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Main" <sip:9001@24.112.1.61:5063>
Expires: 240
User-Agent: Linksys/SPA942-6.1.5(a)
Content-Length: 395
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0^M
o=- 38843 38843 IN IP4 24.112.1.61^M
s=-^M
c=IN IP4 24.112.1.61^M
t=0 0^M
m=audio 16458 RTP/AVP 0 2 4 8 18 96 97 98 101^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:2 G726-32/8000^M
a=rtpmap:4 G723/8000^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:18 G729a/8000^M
a=rtpmap:96 G726-40/8000^M
a=rtpmap:97 G726-24/8000^M
a=rtpmap:98 G726-16/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-15^M
a=ptime:30^M
a=sendrecv^M
<------------->
[2017-10-29 20:51:38] VERBOSE[16019] chan_sip.c: --- (14 headers 18 lines) ---
[2017-10-29 20:51:38] VERBOSE[16019] chan_sip.c: Sending to 24.112.1.61:5063 (no NAT)
[2017-10-29 20:51:38] VERBOSE[16019][C-00000011] chan_sip.c: Sending to 24.112.1.61:5063 (no NAT)
[2017-10-29 20:51:38] VERBOSE[16019][C-00000011] chan_sip.c: Using INVITE request as basis request - 1b2eef6-bba3b7bc@192.168.0.10
[2017-10-29 20:51:38] VERBOSE[16019][C-00000011] chan_sip.c: Found peer '9001' for '9001' from 24.112.1.61:5063
[2017-10-29 20:51:38] VERBOSE[16019][C-00000011] chan_sip.c:
<--- Reliably Transmitting (NAT) to 24.112.1.61:5063 --->
SIP/2.0 401 Unauthorized^M
Via: SIP/2.0/UDP 24.112.1.61:5063;branch=z9hG4bK-22107674;received=24.112.1.61;rport=5063^M
From: "Main" <sip:9001@SERVERHOSTNAME.com>;tag=afe115ae4f310064o3^M
To: <sip:12027621401@SERVERHOSTNAME.com>;tag=as1ca005f3^M
Call-ID: 1b2eef6-bba3b7bc@192.168.0.10^M
CSeq: 101 INVITE^M
Server: FPBX-2.11.0(11.25.0)^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE^M
Supported: replaces, timer^M
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="41506cca"^M
Content-Length: 0^M
^M

Using IP authentication , outbound trunk settings:

type=friend
dtmfmode=auto
host=64.2.142.93
context=from-trunk
allow=all
insecure=port,invite
canreinvite=no
allow sip guests= yes
qualify=yes

Enable anonymous SIP calls: is turned on... as we typically get this up and running and then limit IPs in the firewall. But still not working. If there is a better way to do this then let me know. By the log, it looks like the hard phone connected to the hosted PBX server and does not see the number as a local call, so then it sends to the phonenumber@pbxserverhostname . But that says it is unauthorized.

supa

You should not need to allow sip guests which does not matter for outbound anyways.

I am not sure if allow=all is valid or recommended.

Try

disallow=all
allow=ulaw

Are you sure you have your public IP set correctly on the provider end?