I have a cloud based pbx install with various remote extensions at two sites. I’ve opened up SIP on the freepbx firewall for just the external IPs at those two sites.
All extensions register fine but at site B i get one way audio. This is usually down to NAT or a firewall issue. yet at both sites im allowing all traffic from internally to the pbx. I’ve looked at the firewall settings on both and they are the same.
I have NAT set to yes on all extensions too.
Handsets at site A are Sangoma and at site B are Yealink. I’m a little puzzled on where to look next?
EDIT: after further investigation it appears only external calls are not working. When i try to call out externally i get a “all circuits are busy message”. This can’t be the case as it’s the only external call going on.
Internal calls work fine, which in my mind would rule out the firewall or nat issues.
I’ve been looking into this some further, i’ve spoken to our SIP supplier and they mentioned our contact IP was showing as a 127.0.0.1 IP which would obviously fail. I’ve since changed this and now i believe it’s showing the external IP address of our PBX box.
I still can’t make external calls though. This is the output with sip debug turned on.
<------------>
[2018-07-04 10:24:24] WARNING[5703][C-00000082] channel.c: Prodding channel ‘SIP/1051-00000085’ failed
[2018-07-04 10:24:24] VERBOSE[5703][C-00000082] app_macro.c: Spawn extension (macro-outisbusy, s, 5) exited non-zero on ‘SIP/1051-00000085’ in macro ‘outisbusy’
[2018-07-04 10:24:24] VERBOSE[5703][C-00000082] pbx.c: Spawn extension (restrictedroute-83f1807ba6c14eef109c67d41a45063c, 07793204033, 8) exited non-zero on ‘SIP/1051-00000085’
[2018-07-04 10:24:24] VERBOSE[5703][C-00000082] pbx.c: Executing [h@restrictedroute-83f1807ba6c14eef109c67d41a45063c:1] Hangup(“SIP/1051-00000085”, “”) in new stack
[2018-07-04 10:24:24] VERBOSE[5703][C-00000082] pbx.c: Spawn extension (restrictedroute-83f1807ba6c14eef109c67d41a45063c, h, 1) exited non-zero on ‘SIP/1051-00000085’
[2018-07-04 10:24:24] VERBOSE[5703][C-00000082] app_stack.c: SIP/1051-00000085 Internal Gosub(crm-hangup,s,1) start
[2018-07-04 10:24:24] VERBOSE[5703][C-00000082] pbx.c: Executing [s@crm-hangup:1] NoOp(“SIP/1051-00000085”, “Sending Hangup to CRM”) in new stack
[2018-07-04 10:24:24] VERBOSE[5703][C-00000082] pbx.c: Executing [s@crm-hangup:2] NoOp(“SIP/1051-00000085”, “HANGUP CAUSE: 34”) in new stack
[2018-07-04 10:24:24] VERBOSE[5703][C-00000082] pbx.c: Executing [s@crm-hangup:3] ExecIf(“SIP/1051-00000085”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2018-07-04 10:24:24] VERBOSE[5703][C-00000082] pbx.c: Executing [s@crm-hangup:4] NoOp(“SIP/1051-00000085”, “MASTER CHANNEL: 1530696261.389 = 1530696261.389”) in new stack
[2018-07-04 10:24:24] VERBOSE[5703][C-00000082] pbx.c: Executing [s@crm-hangup:5] GotoIf(“SIP/1051-00000085”, “0?return”) in new stack
[2018-07-04 10:24:24] VERBOSE[5703][C-00000082] pbx.c: Executing [s@crm-hangup:6] Set(“SIP/1051-00000085”, “__CRM_HANGUP=1”) in new stack
[2018-07-04 10:24:24] VERBOSE[5703][C-00000082] pbx.c: Executing [s@crm-hangup:7] AGI(“SIP/1051-00000085”, “sangomacrm.agi”) in new stack
[2018-07-04 10:24:24] VERBOSE[5703][C-00000082] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2018-07-04 10:24:24] VERBOSE[2206] chan_sip.c:
<— SIP read from UDP:SIPTRUNKIP:65476 —>
ACK sip:00000000000@PBXEXTIP:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.40.152:5060;branch=z9hG4bK4264235943
From: “Handset 1” sip:1051@PBXEXTIP:5160;tag=1024722605
To: sip:00000000000@PBXEXTIP:5160;tag=as2b28a287
Call-ID: [email protected]
CSeq: 2 ACK
Content-Length: 0
To me it reads like the call is being blocked, restrictedroute but i haven’t restricted it anywhere (i’m using Class of Service too and i’ve checked that too)
I’ve now had this back from our SIP Supplier (Gamma)
As discussed, these calls are failing as we use IP Authentication on our SIP Trunks and the IP you present out in your Contact and Via Header: 127.0.0.1 does not match your Endpoint IP: EXTIP. You also send the wrong Connection Info IP in the SDP portion of the invite which is why you get one way audio.
As discussed, these calls are failing as we use IP Authentication on our SIP Trunks and the IP you present out in your Contact and Via Header: 127.0.0.1 does not match your Endpoint IP: EXTIP. You also send the wrong Connection Info IP in the SDP portion of the invite which is why you get one way audio.
I’ve recently disabled the VPN so this would be the first time we are using NAT, so i suspect it’s something to do with the NAT settings.
On the general SIP Settings page, under NAT settings i have my external IP configured and only under local networks i have just 128.0.0.1 configured.
Under Chan_SIP settings i have;
NAT=Yes
IP Config: Yes, external IP