unable to make outbound calls via VoipVoip

[JW Ronken]

Hi all
trying for hours now to get outbound calling to work with VoipVoip but no luck. Inbound works fine, other sip trunks work fine. Softphones (non 3cx) work fine with outbound calling but 3CX refuses to make the call. Trunk is registered and outbound rule correct (in logs)
Below an excerpt from eventlog, hopefully anyone has an idea?

03/02/2019 12:37:51 PM - [CM503003]: Call(C:152): Call to <sip:[email protected]:5060> has failed; Cause: 408 Request Timeout/INVITE from local
03/02/2019 12:37:51 PM - Session 35332 has failed in leg L:152.2[Line:10001>>3146yyyyyyy] ; Cause: 408 Request Timeout/INVITE from local
03/02/2019 12:37:51 PM - Failure from <sip:[email protected]:5060>;tag=9edca047 to "33xxxxxxxx"<sip:33xxxxxxxxsip3.voipvoip.com:5060>;tag=fee82571
03/02/2019 12:37:19 PM - Route to L:152.2[Line:10001>>3146yyyyyyy] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=70562e77839cb756 Call-ID=sDPN0Sdy2nUxP-m6LJxfqg..:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---70562e77839cb756;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "33xxxxxxxx"<sip:[email protected]:5060>;tag=fee82571
Call-ID: sDPN0Sdy2nUxP-m6LJxfqg..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE
Content-Type: application/sdp
Supported: replaces, timer
P-Preferred-Identity: <sip:[email protected]:5060>
Remote-Party-ID: "33xxxxxxxx"<sip:[email protected]:5060>;party=calling
Content-Length: 292
v=0
o=3cxPS 17866391486988288 21967003627552769 IN IP4 213.193.234.40
s=3cxPS Audio call
c=IN IP4 212.190.x.x
t=0 0
m=audio 9380 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=sendrecv

Many Thanks
Janwillem

thasss

 

[leejor]

What type of registration are you using with the provider? You might try using Wireshark to confirm an Invite is sent to the correct IP, is there an Invite at all? Is there something preventing the URL being resolved ? DNS issue?

 

[JW Ronken]

Using regiser/account based, the invite is sent to the correct IP (problem is same with hostname and IP and upon suggestion from VoipVoip tested with port 5060 and 443.
3CX in on public IP so no NAT etc,

03/02/2019 8:00:15 PM - [CM503003]: Call(C:210): Call to <sip:[email protected]:443> has failed; Cause: 408 Request Timeout/INVITE from local
03/02/2019 7:59:43 PM - [CM503025]: Call(C:210): Calling
T:Line:10001>>314xxxxxxxxx@[Dev:sip:[email protected]:443] for L:210.1[Extn:161]
03/02/2019 7:59:43 PM - Route to L:210.2[Line:10001>>31xxxxxxx] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=e53f2c45f9be2920 Call-ID=W7C99L7JNUJtBGcTmb10mg..: INVITE sip:[email protected]:443 SIP/2.0

trunk works with pbxes.org (which we used before) and on any softphone. pretty sure it is one of the outbound settings for the trunk (all are default) but can't see where...
thanks!

 

[nub]

http://www.voipvoip.com/3cx/
looks straightforward to me, odd that incoming calls are working but outbound are not.

Get a test key and see if it works in the cloud proof of concept test, takes about 20 min to setup.

Try the ip of sip3.voipvoip.com maybe a dns issue? sip3.voipvoip.com [66.220.10.66]

Best of luck.

 

[JW Ronken]

Tried already with IP. Same issue...
curious if you find something...

 

[YiannisH_3CX]

Using regiser/account based, the invite is sent to the correct IP (problem is same with hostname and IP and upon suggestion from VoipVoip tested with port 5060 and 443.
3CX in on public IP so no NAT etc,

So you have confirmed that the Invite is sent to the correct IP address. Do you get a reply back from the provider?

 

[JW Ronken]

Hi YiannisH
Not 100% sure where to check that, but here is the log from a test call.
VoiPVoip account is 55zzzzzzzz, my number is 33yyyyyyyyy calling to 003519xxxxxxxxx
Spent a couple of hours mailing back and forth with VoipVoip yesterday but they have no idea...
DNS is OK, IP and lookup are OK, direct connection to outside
Anonimized IP's and numbers in log
many thanks
jan

03/04/2019 10:53:46 AM - Leg L:65.2[Line:10001>>003519xxxxxxx] is terminated: Cause: 408 Request Timeout/INVITE from local
03/04/2019 10:53:46 AM - L:65.2[Line:10001>>003519xxxxxxx] got Terminated Recv 408/INVITE from 0.0.0.0:0 tid=83e71115d475e84f Call-ID=eSI_4zY7YhzAMv75dtR5LA..:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 213.193.y.y:5060;branch=z9hG4bK-524287-1---83e71115d475e84f;rport
To: <sip:[email protected]:5060>;tag=72034906
From: "339yyyyyyyy"<sip:[email protected]:5060>;tag=382c0f3d
Call-ID: eSI_4zY7YhzAMv75dtR5LA..
CSeq: 1 INVITE
Content-Length: 0
03/04/2019 10:53:46 AM - Terminated from <sip:[email protected]:5060>;tag=72034906 to "339yyyyyyyy"<sip:[email protected]:5060>;tag=382c0f3d; reason: Error
03/04/2019 10:53:46 AM - ~Route=Dev:sip:[email protected]:5060
03/04/2019 10:53:46 AM - Call to T:Line:10001>>003519xxxxxxx@[Dev:sip:[email protected]:5060] from L:65.1[Extn:161] failed, cause: Cause: 408 Request Timeout/INVITE from local
03/04/2019 10:53:46 AM - L:65.2[Line:10001>>003519xxxxxxx] got Failure: Failure Recv 408/INVITE from 0.0.0.0:0 tid=83e71115d475e84f Call-ID=eSI_4zY7YhzAMv75dtR5LA..:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 213.193.y.y:5060;branch=z9hG4bK-524287-1---83e71115d475e84f;rport
To: <sip:[email protected]:5060>;tag=72034906
From: "339yyyyyyyy"<sip:[email protected]:5060>;tag=382c0f3d
Call-ID: eSI_4zY7YhzAMv75dtR5LA..
CSeq: 1 INVITE
Content-Length: 0
03/04/2019 10:53:46 AM - [CM503003]: Call(C:65): Call to <sip:[email protected]:5060> has failed; Cause: 408 Request Timeout/INVITE from local
03/04/2019 10:53:46 AM - Session 32650 has failed in leg L:65.2[Line:10001>>003519xxxxxxx] ; Cause: 408 Request Timeout/INVITE from local
03/04/2019 10:53:46 AM - Failure from <sip:[email protected]:5060>;tag=72034906 to "339yyyyyyyy"<sip:[email protected]:5060>;tag=382c0f3d
03/04/2019 10:53:14 AM - Outbound CID reformat rule for DN:10001 is disabled (<Rules />)
03/04/2019 10:53:14 AM - Dev(147438667):[sip:[email protected]:5060 / 55zzzzzzz]: PBX contact is public IP: <sip:[email protected]:5060>
03/04/2019 10:53:14 AM - Outbound URI is used: sip:66.220.10.66;lr;ftag=9ae22b06-92dd-422c-8fda-0e5ddd6d748f;did=08b.1621d077
03/04/2019 10:53:14 AM - [CM503027]: Call(C:65): From: Extn:161 ("JW Rxxxxx" <sip:[email protected]:5060>)  to  T:Line:10001>>003519xxxxxxx@[Dev:sip:[email protected]:5060]
03/04/2019 10:53:14 AM - [CM503004]: Call(C:65): Route 1: from L:65.1[Extn:161] to T:Line:10001>>003519xxxxxxx@[Dev:sip:[email protected]:5060]
03/04/2019 10:53:14 AM - Line limit check: Current # of calls for line Lc:10001(@GDC_FR[<sip:[email protected]:5060>]) is 0; limit is 10
03/04/2019 10:53:14 AM - Call(C:65): Call from Extn:161 to 003519xxxxxxx matches outbound rule 'VOIPVOIP'
03/04/2019 10:53:14 AM - [Flow] Call(C:65): has built target endpoint: Out#:>>Rule{VOIPVOIP}>>003519xxxxxxx for call from L:65.1[Extn:161]
03/04/2019 10:53:14 AM - [Flow] Target endpoint for 003519xxxxxxx is Out#:>>Rule{VOIPVOIP}>>003519xxxxxxx
03/04/2019 10:53:14 AM - Caller '161' is found in the list of allowed extensions: 161-161
03/04/2019 10:53:14 AM - Looking for outbound rule: dialed = [003519xxxxxxx], processed: [003519xxxxxxx]; from-ext:
03/04/2019 10:53:14 AM - [Flow] Building target endpoint to 003519xxxxxxx from "JW Rxxxxx" <sip:[email protected]:5060>
03/04/2019 10:53:14 AM - [CM503010]: Call(C:65): Making route(s) from Extn:161 to <sip:[email protected]:5060>
03/04/2019 10:53:14 AM - Remote SDP is set for leg L:65.1[Extn:161]
03/04/2019 10:53:14 AM - OnOffer from "JW Rxxxxx"<sip:[email protected]:5060>;tag=0hniAVB5NNtl6t0UeTQLWtt9QMxAcK-e
03/04/2019 10:53:14 AM - [CM503001]: Call(C:65): Incoming call from Extn:161 to <sip:[email protected]:5060>
03/04/2019 10:53:14 AM - Outbound URI is used: sip:[email protected]:5080;user=proxy;uri=clnt.0-ctmcstyrjujpzl4xwy0sg2dadymf8yc-
03/04/2019 10:53:14 AM - IncomingCall: C:65 from <sip:[email protected]:5060> to <sip:[email protected]:5060>
03/04/2019 10:53:14 AM - Added leg L:C:65.1[No endpoint yet]
03/04/2019 10:53:14 AM - Call from "JW Rxxxxx"<sip:[email protected]:5060>;tag=0hniAVB5NNtl6t0UeTQLWtt9QMxAcK-e to <sip:[email protected]:5060;tag3cx=b0vp8jan7qavxoqoe.cyyunrxvx7grig>;tag=fab84516

 

[YiannisH_3CX]

From the provided logs i can see that the PBX is sending the call but receives no answer so the call times out after 32 seconds which is the correct behaviour.
This means that either the call is not reaching the provider or the reply from the provider is not reaching you. The first step is to run the firewall checker of the PBX and make sure that you pass all tests to ensure that this is not a firewall issue.
If everything checks out you need to call your provider and ask if they are seeing an Invite from you. If so do they see anything wrong?
Alternatively i would recommend using a supported provider to avoid any compatibility issues.
https://www.3cx.com/partners/sip-trunks/

 

[JW Ronken]

Hi
unfortunately, these numbers are well published so I can't move them to another provider easily.
I will contact them again and see if they have a solution. It's too bad that 3CX doesn't work well with providers which whom we had no problems on our Asterisk or on a hosted pbxes.org system. But as all other features are much better than any competitive product I think I have to live with that.

 

[cobaltit]

Porting to another provider keeps the number so that's not an issue. 3CX works well with many providers, it's just that supported providers work out of the box. When you use an unsupported provider and the provider doesn't have a guide for 3CX you need to figure out the settings. Many providers help with this.

 

[JW Ronken]

I am a bit further.. as VoipVoip uses a different authentication id (i.e 55xxxxxxx) than the callerid (in this case 339yyyy) I replaced the outbound caller id with the authentication id and it worked.
Major drawback and issue: customer who call back dial now to 0055xxxx (seems Latvia) instead of to 00339xxxxx which is our number in france and calls don't work of course. I know there are many fields in the Trunk -->Outbound parameters which likely allow for changing this, but which one is the correct one? I need to show callerid 339xxxxxx while being authenticated to VoipVoip as [email protected] (taken from Trunk -> CallerID.
many thanks to all for helping me out so far figuring things out.

 

[YiannisH_3CX]

If VoipVoip uses the From sip field for authentication then perhaps they are using a different sip field to read the number from. It could be something like Remote Party Id or P-Asserted-Identity but that is a question only VoipVoip can answer.