No ring back tone on external calls

[DrNemo]

Hi

I have created a nice little setup with a 3CX (ver. 8 ) and a Linksys SPA-2102 which have 2 analog phones connected to it, and my external line is a Danish VoIP provider (Telsome). The SPA connects from a remote location via the 3CX proxy tunnel to the PBX.

Everything works great, except from getting a ring back tone from the SPA when doing external calls via my VoIP provider. When I call from one analog phone to the other (internal extension to extension call), I do get the Ring back tone.

Apart from the tone, the call connects when the remote user picks up, and it works great.

Any ideas?

PS. it seems like the PBX takes a long time (10 sec) to disconnect the external connection, after put the receiver on hook again after a call. Is this normal.

Thanks
Anders

 

[leejor]

What does the 3CX log look like for a call duration (start to finish)? Have you used this VoIP provider before, with the 2102 directly? What happens if you set up one of the lines directly? Do you hear ringback? Some of the messaging may not be getting through from the Provider to 3CX or it is being misinterpreted.

 

[DrNemo]

I can't set up the provider directly, since SIP is blocked where i live. Hence the tunneling proxy.

The 3CX proxy SW runs on a pc on the local network here in my apartment, which connects to the 3CX PBX, which runs on a PC i Denmark.
The 2102 routes all traffic through this tunnel.

here is the log.... a call to an external no that doesn't answer, aborted from my end... no ring back tone:
(numbers replaced with x'es by me)

17:29:33.308 [CM505003]: Provider:[Telsome] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Asterisk PBX] PBX contact: [sip:[email protected]:5060]
17:29:33.308 [CM503002]: Call(70): Alerting sip:[email protected]:5060
17:29:32.308 [CM503025]: Call(70): Calling VoIPline:xxxxxxxx@(Ln.10000@Telsome)@[Dev:sip:[email protected]:5060]
17:29:32.261 [CM503004]: Call(70): Route 1: VoIPline:xxxxxxxx@(Ln.10000@Telsome)@[Dev:sip:[email protected]:5060]
17:29:32.261 [CM503010]: Making route(s) to <sip:[email protected]>
17:29:32.261 [CM505001]: Ext.11: Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA2102-5.2.10] PBX contact: [sip:[email protected]:5060]
17:29:32.245 [CM503001]: Call(70): Incoming call from Ext.11 to <sip:[email protected]>

 

[leejor]

I'm wondering, do calls placed from an extension in Denmark (same LAN as the 3CX) have the same problem? Have you tried the 3CX softphone, using the tunnel option, running on your laptop at home? I don't see the line and trunk call being "joined", which may explain the lack of audio until the call is answered. If a call on the LAN has the same issue then it is most likely something NOT being sent back from the VoIP provider until the far end actually answers. A log showing an answered call may confirm that. If the PC running the 3CX soft phone also has no ringback then it eliminates the remote server and your set as being part the problem.

 

[DrNemo]

I'm using the softphone with the tunnel option from the remote site, and it produces the ring back tone.

same no, UNanswered call... WITH ring back tones

her is the log when using the softphone:

20:42:38.683 [MS105000] C:89.1: No RTP packets were received:remoteAddr=127.0.0.1:10340,extAddr=0.0.0.0:0,localAddr=127.0.0.1:7332
20:42:37.558 [CM503008]: Call(89): Call is terminated
20:42:29.339 [CM505003]: Provider:[Telsome] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Asterisk PBX] PBX contact: [sip:[email protected]:5060]
20:42:29.339 [CM503002]: Call(89): Alerting sip:[email protected]:5060
20:42:29.073 [CM503025]: Call(89): Calling VoIPline:xxxxxxxx@(Ln.10000@Telsome)@[Dev:sip:[email protected]:5060]
20:42:29.027 [CM503004]: Call(89): Route 1: VoIPline:xxxxxxxx@(Ln.10000@Telsome)@[Dev:sip:[email protected]:5060]
20:42:29.027 [CM503010]: Making route(s) to <sip:[email protected]:5060>
20:42:29.027 [CM505001]: Ext.10: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 4.0.9878.0] PBX contact: [sip:[email protected]:5060]
20:42:29.011 [CM503001]: Call(89): Incoming call from Ext.10 to <sip:[email protected]:5060>

 

[leejor]

Perhaps the softphone produces "early audio", if set to do so, so you may be getting ringback after sending the digits as a matter of course. What about an extension on the PBX LAN itself?

 

[DrNemo]

I cant test pysically on the PBX network, since it is a long way away. I do of course have full control over it, just not physical access.
I can install a softphone on the PBX itself, but i wouldn't be able to hear if it produced the right tones.

 

[leejor]

I suspect it is the message sent back (or not being sent) after you send the digits to them. Before you go pointing a finger at them...try this. Set up a trunk to Sipbroker.com, it doesn't cost anything and you don't even have to have the trunk register to be able to make calls through them. Their website gives you all the info you need. They allow you to call through to other providers, using *XXX codes and access to toll free numbers. This way you can make some test calls to a provider other than yours in Denmark.

Install a program such as Wireshark. It can capture all of the messaging between 3CX and you provider. If you place a call to a number using SipBroker and you get ringback/busy tone before the party answers then compare the 3CX logs and have a look at a capture of the packets, compare the messages coming back from both calls. Make sure that you do a test to a line that answers as well. 3Cx users have had problems with some providers NOT sending standard messages, perhaps your provider in Denmark is set up to work with ATA's and not PBX's, maybe their customers using ATA's don't have an issue.

You may want to look this over...www.delmarnorth.com/namaste/.../SIP_PSTN_Call_Flow.pdf

So..... you (3CX) may not be getting the SIP 183 but does receive the SIP 200 when the call is answered.