Skype outgoing calls?

[Furry]

Hi

Just wondering what the method is for making an outgoing Skype call from an extension, when using 3CX Gateway for Skype?

When using Uplink with Asterisk, I was able to give each Skype ID I wanted to call a shortcode but the latest version of Skype, that the 3CX Gateway for Skype requires to be used, doesn't seem to allow this.

I have tried dialling using the full alphanumeric name (from my Nokia E51), and changing the Skype name to be digits, but with no luck. I get 'address is busy'.

Or was this 3CX gateway for Skype only conceived to allow handling of incoming calls?

Thanks,
Dave.

 

[Vali_3CX]

Hi, Dave

Suppose you have something like this

PBX--->(outboundrule prefix "ak")---->PSTN gateway01---->SkypeGatePort01--->SkypeAccount01
PBX--->(outboundrule prefix "bk")---->PSTN gateway02---->SkypeGatePort02--->SkypeAccount02

Assume SkypeGatePort01 is gateway's master.
In such case, from your phone dial akecho123 - it means you call SkypeAccount01's echo service. It will get busy, since SkypeGatePort01 is master port. But, dialing bkecho123, calling SkypeGatePort02's echo service, should work if this port has no other call in progress.

If none of them are master, then both ak- and bk- calls should have the same behavior as described bk- above. But, even here, if SkypeGatePort02 has the same Skype account as SkypeGatePort01, say SkypeAccount01, if SkypeGatePort01 has already an in-progress call, the bk-call will get busy, since the Skype account is obviously busy, taken by SkypeGatePort01.

Hope it clarified.

Regards
vali

 

[Furry]

Vali,

Thanks.

I am using 2 Skype accounts. Dialling the echo123 service is successful, but dialling a Skype contact is not - I get 'user busy'.

I have set up the outbound rule for my 'master' and 'slave' skype accounts with different prefixes, and outbound rules accordingly. If I attempt to dial the echo123 service on the master account, I get 'user busy'. If I do this on the 'slave' account it works fine.

But, if I try to call one of my skype contacts on the 'slave' account (which has all my contacts defined), it seems to attempt to make the call (makes the 'tom tom drums' skype sound) but then also goes 'user busy'.

Dave.

 

[Furry]

Vali,

Another query, possibly related...

Neither my 'master' nor 'slave' accounts are shown as being online, as contacts from a third Skype account - yet the Gateway shows that they're 'Online - On Hook' under 'Skype status'.

Is this 'normal'?

Dave.

 

[Vali_3CX]

Hi, Dave
For me, all the "simptoms" you describe leads to following reasons:
- port's Skype account is not "fully" logged in (I mean, it appears as online, but not all Skype clients around, especially the one you want to call, were notified yet, and you will have to wait a little)
- the Skype account you want to call is not online or, as in the case above, your account is not "fully" logged in, and it has not been yet "informed" about other's statuses.
- the Skype account you want to call is indeed busy (less probable, but possible)

No matter what, the best test you can do is, first, to open gateway's port Skype:
- stop the gateway service, then select one of its ports, and click Skype button, to open its Skype client.
- wait until it gets logged in (splash' "Account" field displays "Configured as...")

Ok, now:
- if the contact you want to call is in Skype's contact list, check its status (online, offline, DND).
- when it becames online, try to call it from your Skype.
- if you have access to the Skype you want to call (for instance , is running on another machine next to you) check to see when your Skype becomes online.

This test performs a direct Skype-to-Skype call, excluding gateway's involvement. If it works, basically on the running gateway you will have to wait aproximately the same time you wait in this test for online "appearance".

Hope it helps and tell me how's going.

Regards
vali

 

[Furry]

Vali,

From a third Skype account ('C'), I can place a call to either of the accounts ('A' and 'B') that I'm using for 3CX.

C can see that both A and B are online.
Oddly, one of the accounts that I'm using for 3CX cannot see that the other is online (i.e. A cannot see B as being online, but B can see A).

PBX can accept incoming calls via either A or B, when placed from C.

This is very puzzling!

Dave.

 

[Vali_3CX]

Hi, Dave
Dave, I wasn't an intensive Skype user :mrgreen: so perhaps is better for you to ask somebody who used seriously Skype and know better how it works.
However, all I can tell you (from the tests I've done) is following:

the only "normal" case when I've got the same simptoms as you describe was when an "empty" A/B skype account (that mean, having an empty contact list) has been started by the gateway, and from "external" C I was trying to call one of them. I was warned that A/B is not in my contact list, so it started to search Skype directory, it found A/B, I was asked to send that "presentation" and, after that, A/B were in my contact list as grayed (not online). And, in my opinion, they will remain obviously forever grayed, since neither A or B accepted my request to be added in my contact list - matter of privacy, I guess. If I stop the gateway and from A I allow C, then restart the gateway, on C i will seee (after several seconds) A is online while B - not (because it doesn't granted it access).

Hope it helps. If you find some more precise details about this behavior, please tell me.

Regards
vali

 

[Furry]

Vali,

Well, I did have an issue with my three skype accounts seeing other as online, which I've now fixed. Skype seems a bit confusing in that regard.

But, I still can't make outgoing calls, still get 'busy'. It does, however, ring for a while before going to 'busy', with an orange indication and 'ringing' in the ports/trunks status.

Dave.

 

[Furry]

Vali,

I've worked out what the problem was, apart from the possible issues with Skype accounts (above):

I was using my name for the Skype contact, rather than the actual 'Skype Name' (doh!). I think this was a 'hangover' from using shortcode dialling with uplink (I still think it's a shame that Skype seem to have removed that facility - wonder why they did?).

Also, I wasn't dialling SkypeOut numbers correctly (doh! again); I wasn't putting '00' at the front, followed by the country code (44) - I somehow forgot to do that.

An outstanding slight issue is that while my softphone client (zoiper) can dial out via Skype using either '+' (as in +44...) or '00' (as in 0044...), after my prefix, when I use '+' on my Nokia E51 it doesn't work - I have to use '00'. I haven't yet worked out how Setttings/Advanced/e164 number processing relates to this.

Dave.

 

[heinbense]

Hi,

So how do you dial the 'Skype name' from an office phone?

Lets say you use prefix '5' and skype name = heinbense How would you dial it?

Thanks,
Hein

 

[sipero123]

Hi,

The way I've done mine is 8+skypeid and then set the rules for this to strip the 8 and dial the skypeid using the relevant trunk.



Jonathan Hamon

 

[Vali_3CX]

So how do you dial the 'Skype name' from an office phone?
Lets say you use prefix '5' and skype name = heinbense How would you dial it?

Hi, Hein

Simply, prefix immediately followed by account name: 5heinbense

P.S Not all phones - mainly hardphones - allows letters, but only digits to be typed. In such a case, use their Speeddial feature (I've used this to be able to call Skype accounts from my Grandstream deskphone)

Regards
vali

 

[Furry]

Hi vali,

Re. the 'speedial feature' you mention: as I said in my first post on this thread (I described it as 'shortcode' or something), I used that with an earlier version of Skype, but I can't see how to use it with the latest version - what am I missing? Edit: just been to the Skype site, and it states that latest version doesn't support speed dial - but it doesn't say why.

TIA,
Dave.

 

[Vali_3CX]

Re. the 'speedial feature' you mention: as I said in my first post on this thread (I described it as 'shortcode' or something), I used that with an earlier version of Skype, but I can't see how to use it with the latest version - what am I missing?

Hi, Dave

I guess it's a confusion here: I referred PHONE (hardphone or softphone) speeddial feature, where available.
My Grandstream doesn't allow me to call, for instance, "skecho123", therefore, from my computer, I had to login in its configuration program and there I've set one of its speeddials to this "number", skecho123 - after that, everytime I press its "1" speeddial button, it dial and connects that Skype account (of course, "sk" is the outbound rule).

P.S On the other hand, I cannot figure out how SKYPE's speeddial would be useful when used in the gateway :roll:

Regards
vali

 

[Furry]

Hi, Dave

I guess it's a confusion here: I referred PHONE (hardphone or softphone) speeddial feature, where available.
My Grandstream doesn't allow me to call, for instance, "skecho123", therefore, from my computer, I had to login in its configuration program and there I've set one of its speeddials to this "number", skecho123 - after that, everytime I press its "1" speeddial button, it dial and connects that Skype account (of course, "sk" is the outbound rule).

P.S On the other hand, I cannot figure out how SKYPE's speeddial would be useful when used in the gateway :roll:

Ah, yes, I see. I found the speed dial facility useful when using Uplink Skype-to-Sip with Asterisk (as I said above), because it allowed me to dial my chosen prefix plus a 2-digit speed dial number, e.g. '511'. I guess the Skype API (or whatever) previously allowed the speed dial to be used via that interface.

Dave.

 

[dpoynter01]

Looking for a step to step guide for outgoing callls. I used this guide: http://www.3cx.com/voip-gateways/skype.html ( for setting this up with skype )

I can call in fine, I can dial ext. fine with inbound calls, and ext to ext works fine. However when I dial out I setup the "9" to get outside line and go nowhere and then I setup "0" still getting nowhere.. According to the 3CX log it looks like its telling me the line is busy when its not. I setup an additional outbound gateway in 3cx and in the skype gateway. not sure if that is what i was suppose to do. Any help would be great. Here is my log.


00:03:07.937 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

23:58:07.718 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

23:58:06.609 [CM306003]: SIP IPort mapping (24.210.189.72:5060) resolved by STUN server 75.101.138.128:3478 differs from the one (24.210.189.72:6767 resolved by STUN server 69.0.208.27

23:58:06.500 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.101.138.128:3478 over Transport 192.168.0.91:5060

23:53:08.500 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

23:48:08.281 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

23:43:08.062 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

23:40:28.265 [CM504002]: Ext.100: a contact is unregistered. Contact(s): []

23:40:27.765 [CM504002]: Ext.100: a contact is unregistered. Contact(s): []

23:40:13.468 [CM503008]: Call(21): Call is terminated

23:40:13.468 [CM503015]: Call(21): Attempt to reach <sip:[email protected]:5060> failed. Reason: Not Found

23:40:13.468 [CM503014]: Call(21): No known route to target: <sip:[email protected]:5060>

23:40:13.453 [CM503010]: Making route(s) to <sip:[email protected]:5060>

23:40:13.453 [CM505001]: Ext.100: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 3.1.6288.0] Transport: [sip:192.168.0.91:5060]

23:40:13.437 [CM503001]: Call(21): Incoming call from Ext.100 to <sip:[email protected]:5060>

23:39:48.328 [CM503008]: Call(20): Call is terminated

23:39:48.312 [CM503015]: Call(20): Attempt to reach <sip:[email protected]:5060> failed. Reason: Busy

23:39:48.312 [CM503003]: Call(20): Call to sip:[email protected]:6062 has failed; Cause: 486 Busy Here; from IP:127.0.0.1:6062

23:39:48.312 [CM505002]: Gateway:[outbound] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXSkypeGateway 2.0.7919.0] Transport: [sip:127.0.0.1:5060]

23:39:48.312 [CM503002]: Call(20): Alerting sip:[email protected]:6062;rinstance=eae3f4e91f680d94

23:39:48.187 [CM503024]: Call(20): Calling PSTNline:19375201242@(Ln.10001@outbound)@[Dev:sip:[email protected]:6062;rinstance=eae3f4e91f680d94]

23:39:48.187 [CM503004]: Call(20): Route 1: PSTNline:19375201242@(Ln.10001@outbound)@[Dev:sip:[email protected]:6062;rinstance=eae3f4e91f680d94]

23:39:48.171 [CM503010]: Making route(s) to <sip:[email protected]:5060>

23:39:48.171 [CM505001]: Ext.100: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 3.1.6288.0] Transport: [sip:192.168.0.91:5060]

23:39:48.171 [CM503001]: Call(20): Incoming call from Ext.100 to <sip:[email protected]:5060>

23:39:38.437 [CM504001]: Ext.100: new contact is registered. Contact(s): [sip:[email protected]:63608;rinstance=0d7efe1a5c1237c8/100]

23:39:28.609 [CM503008]: Call(19): Call is terminated

23:39:28.609 [CM503008]: Call(19): Call is terminated

23:39:17.953 [CM503007]: Call(19): Device joined: sip:[email protected]:40600;rinstance=a1493e7827d4e539

23:39:17.953 [CM503007]: Call(19): Device joined: sip:[email protected]:6060

23:39:17.937 [CM505001]: Ext.800: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX MakeCall helper;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX MakeCall helper] Transport: [sip:127.0.0.1:5060]

23:39:17.937 [CM503002]: Call(19): Alerting sip:[email protected]:40600;rinstance=a1493e7827d4e539

23:39:17.515 [CM503024]: Call(19): Calling Ext:Ext.800@[Dev:sip:[email protected]:40600;rinstance=a1493e7827d4e539]

23:39:17.484 [CM503004]: Call(19): Route 1: Ext:Ext.800@[Dev:sip:[email protected]:40600;rinstance=a1493e7827d4e539]

23:39:17.484 [CM503010]: Making route(s) to <sip:[email protected]:5060>

23:39:17.484 [CM505002]: Gateway:[skype] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXSkypeGateway 2.0.7919.0] Transport: [sip:127.0.0.1:5060]

23:39:17.468 [CM503001]: Call(19): Incoming call from +19375201242@(Ln.10000@skype) to <sip:[email protected]:5060>

23:39:17.468 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10000 forwards to DN:800

23:39:17.406 [CM503008]: Call(18): Call is terminated

23:39:17.390 [CM503015]: Call(18): Attempt to reach <sip:[email protected]:5060> failed. Reason: Not Registered

23:39:17.390 [CM503015]: Call(18): Attempt to reach <sip:[email protected]:5060> failed. Reason: Not Registered

23:39:17.390 [CM503016]: Call(18): Target is not registered: Ext:Ext.100

23:39:17.390 [CM503010]: Making route(s) to <sip:[email protected]:5060>

23:39:17.390 [CM505002]: Gateway:[outbound] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXSkypeGateway 2.0.7919.0] Transport: [sip:127.0.0.1:5060]

23:39:17.375 [CM503001]: Call(18): Incoming call from +19375201242@(Ln.10001@outbound) to <sip:[email protected]:5060>

23:39:17.359 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10001 forwards to DN:100

23:38:16.250 [CM504002]: Ext.100: a contact is unregistered. Contact(s): []

23:38:15.718 [CM504002]: Ext.100: a contact is unregistered. Contact(s): []

23:38:12.093 [CM503008]: Call(17): Call is terminated

23:38:12.093 [CM503008]: Call(17): Call is terminated

23:38:07.843 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

23:38:06.156 [CM306003]: SIP IPort mapping (24.210.189.72:5060) resolved by STUN server 75.101.138.128:3478 differs from the one (24.210.189.72:6649 resolved by STUN server 69.0.208.27

23:38:06.046 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.101.138.128:3478 over Transport 192.168.0.91:5060

23:38:05.250 [CM503015]: Call(17): Attempt to reach <sip:[email protected]:5060> failed. Reason: Busy

23:38:05.250 [CM503015]: Call(17): Attempt to reach <sip:[email protected]:5060> failed. Reason: Busy

23:38:05.250 [CM503014]: Call(17): No known route to target: <sip:[email protected]:5060>

23:38:05.250 [CM503010]: Making route(s) to <sip:[email protected]:5060>

23:37:51.921 [CM503007]: Call(17): Device joined: sip:[email protected]:40600;rinstance=a1493e7827d4e539

23:37:51.921 [CM503007]: Call(17): Device joined: sip:[email protected]:63607;rinstance=f1f5b95cd402daaa

23:37:51.906 [CM505001]: Ext.800: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX MakeCall helper;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX MakeCall helper] Transport: [sip:127.0.0.1:5060]

23:37:51.906 [CM503002]: Call(17): Alerting sip:[email protected]:40600;rinstance=a1493e7827d4e539

23:37:51.390 [CM503024]: Call(17): Calling Ext:Ext.800@[Dev:sip:[email protected]:40600;rinstance=a1493e7827d4e539]

23:37:51.375 [CM503004]: Call(17): Route 1: Ext:Ext.800@[Dev:sip:[email protected]:40600;rinstance=a1493e7827d4e539]

23:37:51.375 [CM503010]: Making route(s) to <sip:[email protected]:5060>

23:37:51.375 [CM505001]: Ext.100: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 3.1.6288.0] Transport: [sip:192.168.0.91:5060]

23:37:51.375 [CM503001]: Call(17): Incoming call from Ext.100 to <sip:[email protected]:5060>

23:37:48.046 [CM503008]: Call(16): Call is terminated

23:37:48.031 [CM503015]: Call(16): Attempt to reach <sip:[email protected]:5060> failed. Reason: Busy

23:37:48.031 [CM503003]: Call(16): Call to sip:[email protected]:6062 has failed; Cause: 486 Busy Here; from IP:127.0.0.1:6062

23:37:48.031 [CM505002]: Gateway:[outbound] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXSkypeGateway 2.0.7919.0] Transport: [sip:127.0.0.1:5060]

23:37:48.031 [CM503002]: Call(16): Alerting sip:[email protected]:6062;rinstance=eae3f4e91f680d94

23:37:47.875 [CM503024]: Call(16): Calling PSTNline:52012542@(Ln.10001@outbound)@[Dev:sip:[email protected]:6062;rinstance=eae3f4e91f680d94]

23:37:47.875 [CM503004]: Call(16): Route 1: PSTNline:52012542@(Ln.10001@outbound)@[Dev:sip:[email protected]:6062;rinstance=eae3f4e91f680d94]

23:37:47.859 [CM503010]: Making route(s) to <sip:[email protected]:5060>

23:37:47.859 [CM505001]: Ext.100: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 3.1.6288.0] Transport: [sip:192.168.0.91:5060]

23:37:47.859 [CM503001]: Call(16): Incoming call from Ext.100 to <sip:[email protected]:5060>

23:37:39.750 [CM503008]: Call(15): Call is terminated

23:37:39.750 [CM503015]: Call(15): Attempt to reach <sip:[email protected]:5060> failed. Reason: Not Found

23:37:39.750 [CM503014]: Call(15): No known route to target: <sip:[email protected]:5060>

23:37:39.734 [CM503010]: Making route(s) to <sip:[email protected]:5060>

23:37:39.734 [CM505001]: Ext.100: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 3.1.6288.0] Transport: [sip:192.168.0.91:5060]

23:37:39.734 [CM503001]: Call(15): Incoming call from Ext.100 to <sip:[email protected]:5060>

23:37:33.937 [CM503008]: Call(14): Call is terminated

23:37:33.937 [CM503015]: Call(14): Attempt to reach <sip:[email protected]:5060> failed. Reason: Not Found

23:37:33.937 [CM503014]: Call(14): No known route to target: <sip:[email protected]:5060>

23:37:33.921 [CM503010]: Making route(s) to <sip:[email protected]:5060>

23:37:33.921 [CM505001]: Ext.100: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 3.1.6288.0] Transport: [sip:192.168.0.91:5060]

23:37:33.906 [CM503001]: Call(14): Incoming call from Ext.100 to <sip:[email protected]:5060>

23:36:41.609 [CM503008]: Call(13): Call is terminated

23:36:41.593 [CM503015]: Call(13): Attempt to reach <sip:[email protected]:5060> failed. Reason: Busy

23:36:41.593 [CM503003]: Call(13): Call to sip:[email protected]:6062 has failed; Cause: 486 Busy Here; from IP:127.0.0.1:6062

23:36:41.593 [CM505002]: Gateway:[outbound] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXSkypeGateway 2.0.7919.0] Transport: [sip:127.0.0.1:5060]

23:36:41.593 [CM503002]: Call(13): Alerting sip:[email protected]:6062;rinstance=eae3f4e91f680d94

23:36:41.406 [CM503024]: Call(13): Calling PSTNline:9375201242@(Ln.10001@outbound)@[Dev:sip:[email protected]:6062;rinstance=eae3f4e91f680d94]

23:36:41.406 [CM503004]: Call(13): Route 1: PSTNline:9375201242@(Ln.10001@outbound)@[Dev:sip:[email protected]:6062;rinstance=eae3f4e91f680d94]

23:36:41.390 [CM503010]: Making route(s) to <sip:[email protected]:5060>

23:36:41.390 [CM505001]: Ext.100: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 3.1.6288.0] Transport: [sip:192.168.0.91:5060]

23:36:41.375 [CM503001]: Call(13): Incoming call from Ext.100 to <sip:[email protected]:5060>

23:34:57.640 [CM503008]: Call(12): Call is terminated

 

[Vali_3CX]

when I dial out I setup the "9" to get outside line and go nowhere and then I setup "0" still getting nowhere.. According to the 3CX log it looks like its telling me the line is busy when its not.


23:39:48.328 [CM503008]: Call(20): Call is terminated
23:39:48.312 [CM503015]: Call(20): Attempt to reach <sip:[email protected]:5060> failed. Reason: Busy
23:39:48.312 [CM503003]: Call(20): Call to sip:19375201242@127.0.0.1:6062 has failed; Cause: 486 Busy Here; from IP:127.0.0.1:6062
23:39:48.312 [CM505002]: Gateway:[outbound] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXSkypeGateway 2.0.7919.0] Transport: [sip:127.0.0.1:5060]
23:39:48.312 [CM503002]: Call(20): Alerting sip:[email protected]:6062;rinstance=eae3f4e91f680d94
23:39:48.187 [CM503024]: Call(20): Calling PSTNline:19375201242@(Ln.10001@outbound)@[Dev:sip:[email protected]:6062;rinstance=eae3f4e91f680d94]
23:39:48.187 [CM503004]: Call(20): Route 1: PSTNline:19375201242@(Ln.10001@outbound)@[Dev:sip:[email protected]:6062;rinstance=eae3f4e91f680d94]
23:39:48.171 [CM503010]: Making route(s) to <sip:[email protected]:5060>
23:39:48.171 [CM505001]: Ext.100: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 3.1.6288.0] Transport: [sip:192.168.0.91:5060]
23:39:48.171 [CM503001]: Call(20): Incoming call from Ext.100 to <sip:[email protected]:5060>

Hi
At this moment - I mean in this Beta version - gateway responds with "busy" everytime it's unable to make the Skype call. That's quite confusing, we will have to improve this.

back to your issue - the configuration you have seems to be ok, the reason of the "busy" answer you get being, almost for sure, the missing of the country code in front of the number you want to call ("Call to sip:19375201242", not "Call to sip:00xx19375201242")
Also, IMPORTANT, be sure you have credit on the Skype account you want to make phone calls, otherwise Skype will drop the call and you will get another "busy".

Hope it helps

Regards
vali

 

[sipero123]

Hi,

I'm sure Vali is right. You just need to update your outgoing rule to add 00 by using prepend or what ever is appropriate for the numbers you are dialling. I'm in the UK so I have a rule like this

number starts 01,02,07,08 strip 1 digit prepend 0044



Jonathan Hamon

 

[dpoynter01]

Thanks for the information...

I have purchased an incoming phone number for Skype... I have almost purchased the monthly unlimited plan for usa and canada. I'm really only interested in calling usa to usa... So it should be like so?

Route: "skypeoutbound"
Strip Digits: "1"
Prepend: "011"

will i be billed for using the country code or is it just for the setup of 3cx?

What should I be putting into the " calls to numbers starting with " and "calls from extension " and " calls to a number with a length of"

 

[dpoynter01]

I have accomplished getting this to work being able to dial out via client... Now the problem is if I setup an extention to forward to external number it disconnects because it of the busy error again. Then states no outbound rule for external number 0000000000 etc.. Where do I set the outboud rule up for that?