Trouble with outbound calls

[edwards2020]

I am trying to place outbound calls from a SIP client on my IPhone to external numbers. We are trying the free version with the possiblity of purchasing it.

I am able to dial into the SIP client from a land/cell line; however, I am not able to dial out. I've configured outbound rules and believe that I've gone throught all of the setup steps properly. I temporarily turned off the firewall to see if this was the issue to no avail.

I am using Bandwidth.com as the PTSN. Here is what I'm seeing in the logs. (The x's aren't the log - just blocking out the cell number.

09:43:51.203 [CM503008]: Call(32): Call is terminated
09:43:51.156 [CM503020]: Normal call termination. Reason: Not found
09:43:51.156 [CM503016]: Call(32): Attempt to reach <sip:[email protected]> failed. Reason: Not Found
09:43:51.156 [CM503014]: Call(32): No known route to target: <sip:[email protected]>
09:43:51.156 [CM503010]: Making route(s) to <sip:[email protected]>
09:43:51.156 [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [SipPhone v3.0/iPhoneOS] PBX contact: [sip:[email protected]:5060]
09:43:51.156 [CM503001]: Call(32): Incoming call from Ext.100 to <sip:[email protected]>

Any help would be greatly appreciated!!

 

[KerryG]

Looks like a DNS issue, it cant find the host you are trying to reach.

 

[edwards2020]

OK, I don't understand.... I can ping the server from on external source using complete different DNS servers. Here's what I get...

Pinging istphones.gotdns.com [70.90.194.195] with 32 bytes of data:

Reply from 70.90.194.195: bytes=32 time=20ms TTL=124
Reply from 70.90.194.195: bytes=32 time=18ms TTL=124
Reply from 70.90.194.195: bytes=32 time=18ms TTL=124
Reply from 70.90.194.195: bytes=32 time=28ms TTL=124

Ping statistics for 70.90.194.195:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 18ms, Maximum = 28ms, Average = 21ms

It seems that the server to trying to connect to external cell phone number @istphones.gotdns.com. Is that correct? Should it be doing that? Inbound calls come from a different IP address.

 

[KerryG]

I dont know, is that the server information that Bandwidth.com gave you? I wouldn't think that Bandwidth.com would use a dynamic DNS server for their SIP trunks.

 

[edwards2020]

OK, remember. I'm new at this.

Istphones.gotdns.com is a dynamic DNS entry for my local 3CX box. I have Bandwidth.com setup as a VOIP provider. I don't understand how the sip client is supposed to get to the outside world. When I am dialing an outside line the log says sip:[email protected] which which makes me think it is trying to find this number on my server. It should be routing out to bandwidth.coms server. Right? How do I make that happen?

 

[KerryG]

First off, use a softphone on the local network as the PBX instead of a SIP client on the iPhone. This will help eliminate variables. However, since I have no idea what your outbound trunk or outbound rules look like, I can't begin to guess what is wrong.

 

[MichaelB_3CX]

Specify the number format you are trying to dial (replace the last 4 digits with X not all)

How did you configure the outbound rule to go out. Not found means that bandwidth.com cannot route this number.

Either you are missing a Plus, or you are stripping the incorrect number of digits - put a screenshot of the outbound rule.

You can do a very quick test - get a 3CXphone, register it directly with bandwidth.com (bypassing the pbx) and dial this number. If you get a not found in this case as well, the answer is pretty straight forward.

 

[leejor]

I am able to dial into the SIP client from a land/cell line; however, I am not able to dial out.

So, let me get this straight, you are truing to dial into the 3CX from any land/cell phone (not a 3Cx extension) and then get dial tone and dial back out again? Or.... are you trying to dial in and be forwarded to a specific number? Or ...are you trying to register a SIP client as a 3Cx extension and then dial outbound calls from the PBX to various numbers?

 

[robhazes]

I am having a similar problem.

I am using Linksys PAP2 as an external extension, logged in on my 3cx-system.
I can call to my PAP2-phone, for instance from another external softphone connected to the same 3cx-system, all ok.

Making a call the other way does not work. I can not call from my PAP2, neither to another internal phone nor to an external phone.

The IP-address of the 3cx system is dynamic, so I use dynamic DNS to be able to find my 3cx-system. This works fine, my PAP2 always registers with the 3cx-system.

The error message at the 3cx-side is:
Unidentified incoming call. Review INVITE and adjust source identification. etc

When I in "Line 1" section in the PAP2 change the Proxy-field from mydynamicdns.XXX (my dynamic DNS placeholder) to the actual IP address of the 3cx system it all works.

Anyone who can tell me what to change to make this work with mydynamicdns.XXX in the Proxy-field of the PAP2?

 

[leejor]

Ok, first make sure you have STUN enabled on the remote PAP2, this is a given for any remote extension. Since putting in the actual IP of the 3CX works, then I'm going to assume, that the PAP2 isn't able to resolve the DynDNS address of the 3Cx location. In the settings for the PAP2 there is a place (or two) to put in DNS server info. This can normally just be the IP of the local router that will then pass the request on to the DNS server(s) that it "knows" about via the ISP. You might try getting the IPs of a couple of DNS servers and "manually" put them into the PAP2.

 

[robhazes]

Thanks for the answer.

I do not think this is the problem. The PAP2 does find the 3cx system, also with the dynamic DNS placeholder. My 3cx-system registers the call I try to make, but rejects it, see log below.

Below som log-data from my 3cx system, first the call in the working direction, and then a call in opposite direction which fails.

Global IP-address 66.166.166.55 and dynamic DNS placeholder mydynamicdns.XXX have been modified for safety-reasons.


This call from remote internal 14 (3cx softphone) to remote internal 16 (analog phone on PAP2) works:

23:58:24.375 [CM503008]: Call(64): Call is terminated

23:58:24.359 [CM503008]: Call(64): Call is terminated

23:58:22.625 [CM503007]: Call(64): Device joined: sip:[email protected]:51242

23:58:22.609 [CM503007]: Call(64): Device joined: sip:[email protected]:56196;rinstance=e988eaedb8ffe0df

23:58:14.390 [CM503024]: Call(64): Calling Ext:Ext.16@[Dev:sip:[email protected]:51242]

23:58:14.171 [CM503004]: Call(64): Route 1: Ext:Ext.16@[Dev:sip:[email protected]:51242]

23:58:14.140 [CM503001]: Call(64): Incoming call from Ext.14 to <sip:[email protected]:5060>


This call in opposite direction fails:

00:00:21.468 [CM503008]: Call(65): Call is terminated

00:00:21.468 [CM502001]: Source info: From: "16"<sip:[email protected]>;tag=2dfbf6c3f613605co0<sip:[email protected]>

00:00:21.468 [CM503013]: Call(65): Incoming call rejected, caller is unknown; msg=SipReq: INVITE [email protected] tid=-cb78e2b9 cseq=INVITE [email protected]:51242 / 102 from(wire)

00:00:21.250 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 66.166.166.55:51242;branch=z9hG4bK-2e1c7245;rport=51242

Max-Forwards: 70

Contact: "16"<sip:[email protected]:51242>

To: <sip:[email protected]>

From: "16"<sip:[email protected]>;tag=2dfbf6c3f613605co0

Call-ID: [email protected]

CSeq: 101 INVITE

Expires: 240

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

Supported: x-sipura, replaces

User-Agent: Linksys/PAP2T-5.1.6(LS)

Content-Length: 0

00:00:21.250 [CM302001]: Authorization system can not identify source of: SipReq: INVITE [email protected] tid=-2e1c7245 cseq=INVITE [email protected]:51242 / 101 from(wire)

Hope this might help to find out where the problem lies.

 

[leejor]

When I in "Line 1" section in the PAP2 change the Proxy-field from mydynamicdns.XXX (my dynamic DNS placeholder) to the actual IP address of the 3cx system it all works.

Anyone who can tell me what to change to make this work with mydynamicdns.XXX in the Proxy-field of the PAP2?

Ok, I'm a bit confused. This led me to believe that the extension wasn't working unless you put the actual IP (of the 3CX location) in the setting. But then you said...

The PAP2 does find the 3cx system, also with the dynamic DNS placeholder.

So which is it, did you fix that problem??

As far as the outgoing calls failing. They should not be trying to go out to anything at Istphones.gotdns.com if your VoIP provider is at Bandwidth.com. I'm assuming that the trunk that you have created to Bandwidth.com is showing as registered?
It sounds like you haven't created a valid outbound rule to route the call (based on your dialled digits) to the trunk at Bandwidth.com. Since you've X'd out all the numbers I have no idea what digits you are sending 3CX or what you want to go out on the trunk. If you post another log, just X out the last couple of digits of the phone number.

 

[robhazes]

leejor:

Please check member-name of the posts. There are 2 different problems mentioned in this thread, it seems you are mixing the problems...

My problem is not fixed. Please check error log in previous post. Hope you have some advice...

 

[SY]

leejor:

Please check member-name of the posts. There are 2 different problems mentioned in this thread, it seems you are mixing the problems...

My problem is not fixed. Please check error log in previous post. Hope you have some advice...

Two log messages in your post:
first
00:00:21.250 [CM302001]: Authorization system can not identify source of: SipReq: INVITE [email protected] tid=-2e1c7245 cseq=INVITE [email protected]:51242 / 101 from(wire)
and second
00:00:21.250 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:

provide hints for you.

Thanks

 

[leejor]

There are 2 different problems mentioned in this thread, it seems you are mixing the problems...

Sorry, my mistake.

 

[robhazes]

Two log messages in your post:
first
00:00:21.250 [CM302001]: Authorization system can not identify source of: SipReq: INVITE [email protected] tid=-2e1c7245 cseq=INVITE [email protected]:51242 / 101 from(wire)
and second
00:00:21.250 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:

provide hints for you.

Thanks Stepan. I have been trying to understand at which side to make changes, but have not succeeded. My best guess is I have to make a change somewhere in the 3cx setup. Any suggestions?

 

[mfm]

HI Rob,

Try and avoid posting problems in threads which are being used to diagnose another users issue, please open a new thread people will surely assist you that way.

So in regard to you rob please see this video, the last minute and a half is the part that you need to look at.
http://www.3cx.com/docs/manual/sip-trunks/ - Updated link

 

[robhazes]

No problem, I start a new thread next time. This one seemed to be dead, and the problem was similar, thats why...

I had a look at the video, but it does not really help. I checked the 3CX doc as well without results. Hope someone can suggest the solution for this problem.