Skype Connect troubles

anzigo

Joined
Oct 11, 2010
Messages
8
Likes
0
Points
0
#1
Hi, I'm new to the forums, so please be gentle.

I am trying to get Skype Connect (previously known as Skype for SIP) to work with my Elastix 2.0 install. I primarily would like to utilise the Skype Call Me button on my website, so that customers may initiate a Skype call from my website. I have purchased a Skype Channel and I have my Elastix box successfully registering the SIP profile for the Skype Connect account. However, I am unable to receive incoming calls.

At the Asterisk CLI (asterisk -vvvr), nothing shows up when an incoming call should be coming in. At the command line, the registration for the Skype SIP account does show. All of the recommended ports are being forwarded to my Elastix box. I am using a dd-wrt firmware based Linksys router.

I have also tried the Skype Connect Test Tool released by Voxygen (http://voxygen.co.uk/) and it reports that the NAT Traversal is failing. I then hooked my computer directly to my cable modem (public IP, no routing/NAT) and re-ran the test tool, but it failed again.

Has anyone successfully setup Skype Connect on their Elastix installation (or any other FreePBX based system)? Any help/suggestions would be appreciated.
 

milauria

Joined
Mar 27, 2010
Messages
27
Likes
0
Points
0
#2
I have installed Skype for Asterisk from Digium and installed according to the wiki described in Elastix support section and works perfectly, no experience with Skype Connect.

hope it helps...
 

anzigo

Joined
Oct 11, 2010
Messages
8
Likes
0
Points
0
#3
Ok, I have Skype Connect working now. It was a silly oversight on my part. I did not set up Incoming Calls in the Skype Manager. I created a Skype business account within the Skype Manager and configured incoming calls to be sent there. Incoming calls work now.

As for the Voxygen Skype Connect Test tool, I would have to say that it is of very, very limited use.

Thanks milauria... I have not tried Digium's Skype for Asterisk, though it does look promising with it's flat rate per channel/license of US$66. Skype Connect is a pay per month affair at US$6.95 per month per channel. Perhaps it is worth it to have a Skype official SIP trunk and avoid messing with channel drivers. On the flip side, Asterisk (and Asterisk based distros such as Elastix) are not yet officially supported by Skype.

If anyone would like some assistance with getting Skype Connect up and running, feel free to hit me up.

Anton
 

pranasoftvoip

Joined
Jul 16, 2010
Messages
7
Likes
0
Points
0
#4
Hi Anzigo!

I'm new at both Elastix and Skype, and I need to receive call from Skype to my Elastix base PBX.

I've made some research throught the web and found some pages that refered to skype but with digium cards.

Can you give me some advise on how to set up my sip trunk in Elastix with skype connect?

Thank you!
 

milauria

Joined
Mar 27, 2010
Messages
27
Likes
0
Points
0
#5

anzigo

Joined
Oct 11, 2010
Messages
8
Likes
0
Points
0
#6
pranasoftvoip, as you can see that are (at least) two commercial solutions for integrating Skype into your Elastix/Asterisk pbx.

Skype for Asterisk, which milauria is referring to, has been available for a longer period of time and as such you're likely to find more how-to information about that online right now.

Skype Connect (formerly called Skype for SIP during it's beta testing stage) is a product offered directly by Skype. It is not specific to Asterisk, and will work with most SIP based PBX solutions. At the moment, no flavour of Asterisk is officially supported by Skype for Skype Connect use. My guess as for the reason for this is because of the fragmented nature of the Asterisk ecosystem and the lack of a dominant Asterisk distribution in the marketplace; meanwhile sipXecs is officially supported by Skype Connect - one source, one set of testing, once certification.

Because Skype Connect is SIP based, configuring your Skype Connect trunks to work in Elastix is fairly trivial once you have sample SIP Trunk configs to work with. I'll post my config here (minus my number/password of course):

Outgoing Settings

PEER Details:
username=xxxxxxxxxxx <--- SIP Username from the SIP Profile area of Skype Manager
type=peer
secret=yyyyyyyyyyy <--- password generated by the SIP Profile area of Skype Manager
qualify=yes
insecure=invite
host=sip.skype.com
fromuser=xxxxxxxxxxx <--- SIP Username from the SIP Profile area of Skype Manager
fromdomain=sip.skype.com
canreinvite=no
disallow=all
allow=g729 <--- g729 or g711 - both are supported by Skype Connect
context=from-trunk

For the Incoming Settings, both USER Context and USER Details are left blank.

Apart from SIP Trunk set up, you may need to make a few more changes. Skype Connect uses from port 8000 and up to RTP connections (two consecutively numbered ports per SIP channel). As such you may need to modify your rtp.conf to read like this (the default rtpstart value is 10000):

rtpstart=8000
rtpend=20000

At this point you would be able to make outgoing calls once you have set up an Outbound Route to use this SIP Trunk and assigned some Skype Credit to the account.

Incoming calls will also work, assuming you have configured an Inbound Route which applies to this trunk and have configured Incoming Calls in the Skype Manager interface/website.

Also, do not forget to modify your firewall rules to forward the appropriate additional ports to you Elastix box.

Of course, before you do any of these steps, you would need to create a Skype Manager account, assign Skype Credit to the account and pay for as many SIP trunks as you would need. Pricing and other details are of course available at the Skype website.

If you need any more assistance, let me know.

UPDATE Oct 25 2010:
Previously omitted:

Register String under Registration. It should be:
MY_SKYPE_USERNAME:MY_SKYPE_PASSWORD@sip.skype.com
 

pranasoftvoip

Joined
Jul 16, 2010
Messages
7
Likes
0
Points
0
#7
Re: Re:Skype Connect troubles

Anzigo, thank you very much for all of your help.

I followed your intructions and set up my sip trunk as follows:
Peer Details:
username=MY_SKYPE_USERNAME
type=peer
secret=MY_SKYPE_PASSWORD
qualify=yes
insecure=invite
host=sip.skype.com
fromuser=MY_SKYPE_USERNAME
fromdomain=sip.skype.com
canreinvite=no
disallow=all
allow=g711
context=from-trunk
Incoming Settings, both USER Context and USER Details are left blank.
Then with sip show peers I've check the new trunk I've created.
Name/username Host Dyn Nat ACL Port Status
Skype/99051000117232 63.209.144.201 5060 OK (198 ms)
102 (Unspecified) D N 0 UNKNOWN
101/101 (Unspecified) D N 0 UNKNOWN
3 sip peers [Monitored: 1 online, 2 offline Unmonitored: 0 online, 0 offline]
Then I made the changes in rtp.conf as you suggested.

I've also created an outbound route to my new SIP trunk:


At this stage I'm only testing the outbound calls.

I've follow the intruction in the user manual for skype connect in order to test the oubound calls, but it's not working.
http://www.skype.com/go/skype.connect.quick.start.guide
In my sip profile in skype manager, it tells me my SIP user is not yet registered in sip.skype.com.
Do you have any idea what it may be happening?
 

pranasoftvoip

Joined
Jul 16, 2010
Messages
7
Likes
0
Points
0
#8
Re: Re:Skype Connect troubles

Anzigo, thank you very much for all of your help.

I followed your intructions and set up my sip trunk as follows:
Peer Details:
username=MY_SKYPE_USERNAME
type=peer
secret=MY_SKYPE_PASSWORD
qualify=yes
insecure=invite
host=sip.skype.com
fromuser=MY_SKYPE_USERNAME
fromdomain=sip.skype.com
canreinvite=no
disallow=all
allow=g711
context=from-trunk
Incoming Settings, both USER Context and USER Details are left blank.
Then with sip show peers I've check the new trunk I've created.
Name/username Host Dyn Nat ACL Port Status
Skype/99051000117232 63.209.144.201 5060 OK (198 ms)
102 (Unspecified) D N 0 UNKNOWN
101/101 (Unspecified) D N 0 UNKNOWN
3 sip peers [Monitored: 1 online, 2 offline Unmonitored: 0 online, 0 offline]
Then I made the changes in rtp.conf as you suggested.

I've also created an outbound route to my new SIP trunk:


At this stage I'm only testing the outbound calls.

I've follow the intruction in the user manual for skype connect in order to test the oubound calls, but it's not working.
http://www.skype.com/go/skype.connect.quick.start.guide
In my sip profile in skype manager, it tells me my SIP user is not yet registered in sip.skype.com.
Do you have any idea what it may be happening?
 

anzigo

Joined
Oct 11, 2010
Messages
8
Likes
0
Points
0
#9
MY BAD! I'm sorry for the oversight.

I forgot the Register String under Registration. It should be:

MY_SKYPE_USERNAME:MY_SKYPE_PASSWORD@sip.skype.com

Lemme know how it goes.
 

pranasoftvoip

Joined
Jul 16, 2010
Messages
7
Likes
0
Points
0
#10
Re: Re:Skype Connect troubles

Great! Now it's registered!
But still does not work.
I'm trying to make and outbound call but all lines appear to be busy!

I set the verbose in 4 and this is the log:
[hide]
Code:
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [726000325@from-internal:1] Macro("SIP/101-090d0528", "user-callerid|SKIPTTL|") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-user-callerid:1] Set("SIP/101-090d0528", "AMPUSER=101") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-user-callerid:2] GotoIf("SIP/101-090d0528", "0?report") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-user-callerid:3] ExecIf("SIP/101-090d0528", "1|Set|REALCALLERIDNUM=101") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: ExecIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-user-callerid:4] Set("SIP/101-090d0528", "AMPUSER=101") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-user-callerid:5] Set("SIP/101-090d0528", "AMPUSERCIDNAME=101") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-user-callerid:6] GotoIf("SIP/101-090d0528", "0?report") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-user-callerid:7] Set("SIP/101-090d0528", "AMPUSERCID=101") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-user-callerid:8] Set("SIP/101-090d0528", "CALLERID(all)="101" <101>") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-user-callerid:9] Set("SIP/101-090d0528", "REALCALLERIDNUM=101") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-user-callerid:10] ExecIf("SIP/101-090d0528", "0|Set|CHANNEL(language)=") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: ExecIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-user-callerid:11] GotoIf("SIP/101-090d0528", "1?continue") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Goto (macro-user-callerid,s,20)
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-user-callerid:20] NoOp("SIP/101-090d0528", "Using CallerID "101" <101>") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Noop
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [726000325@from-internal:2] Set("SIP/101-090d0528", "_NODEST=") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [726000325@from-internal:3] Macro("SIP/101-090d0528", "record-enable|101|OUT|") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-record-enable:1] GotoIf("SIP/101-090d0528", "1?check") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Goto (macro-record-enable,s,4)
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-record-enable:4] AGI("SIP/101-090d0528", "recordingcheck|20101025-155527|1288029327.12") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Oct 25 15:55:27] VERBOSE[15109] logger.c:   recordingcheck|20101025-155527|1288029327.12: Outbound recording not enabled
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- AGI Script recordingcheck completed, returning 0
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: AGI
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-record-enable:5] MacroExit("SIP/101-090d0528", "") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [726000325@from-internal:4] Macro("SIP/101-090d0528", "dialout-trunk|3|26000325||") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:1] Set("SIP/101-090d0528", "DIAL_TRUNK=3") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] DEBUG[15109] func_db.c: DB: AMPUSER/101/pinless not found in database.
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/101-090d0528", "0?sub-pincheck|s|1") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: GosubIf
[Oct 25 15:55:27] DEBUG[15109] func_db.c: DB: AMPUSER/101/pinless not found in database.
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/101-090d0528", "0?disabletrunk|1") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:4] Set("SIP/101-090d0528", "DIAL_NUMBER=26000325") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:5] Set("SIP/101-090d0528", "DIAL_TRUNK_OPTIONS=tr") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:6] Set("SIP/101-090d0528", "OUTBOUND_GROUP=OUT_3") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/101-090d0528", "1?nomax") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Goto (macro-dialout-trunk,s,9)
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/101-090d0528", "0?skipoutcid") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:10] Set("SIP/101-090d0528", "DIAL_TRUNK_OPTIONS=") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:11] Macro("SIP/101-090d0528", "outbound-callerid|3") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/101-090d0528", "0|SetCallerPres|") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: ExecIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/101-090d0528", "0|Set|REALCALLERIDNUM=101") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: ExecIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/101-090d0528", "1?normcid") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Goto (macro-outbound-callerid,s,6)
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-outbound-callerid:6] Set("SIP/101-090d0528", "USEROUTCID=") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] DEBUG[15109] func_db.c: DB: DEVICE/101/emergency_cid not found in database.
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-outbound-callerid:7] Set("SIP/101-090d0528", "EMERGENCYCID=") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-outbound-callerid:8] Set("SIP/101-090d0528", "TRUNKOUTCID=") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/101-090d0528", "1?trunkcid") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Goto (macro-outbound-callerid,s,12)
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/101-090d0528", "0|Set|CALLERID(all)=") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: ExecIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-outbound-callerid:13] GotoIf("SIP/101-090d0528", "1?exit") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Goto (macro-outbound-callerid,s,11)
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-outbound-callerid:11] MacroExit("SIP/101-090d0528", "") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Macro
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/101-090d0528", "0|AGI|fixlocalprefix") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: ExecIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:13] Set("SIP/101-090d0528", "OUTNUM=26000325") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:14] Set("SIP/101-090d0528", "custom=SIP/Skype") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Set
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/101-090d0528", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: ExecIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:16] Macro("SIP/101-090d0528", "dialout-trunk-predial-hook|") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/101-090d0528", "") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Macro
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/101-090d0528", "0?bypass|1") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/101-090d0528", "0?customtrunk") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:19] Dial("SIP/101-090d0528", "SIP/Skype/26000325|300|") in new stack
[Oct 25 15:55:27] NOTICE[15109] app_dial.c: Hey! chan SIP/101-090d0528's context='macro-dialout-trunk', and exten='s'
[Oct 25 15:55:27] WARNING[15109] chan_sip.c: No audio format found to offer. Cancelling call to 26000325
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Couldn't call Skype/26000325
[Oct 25 15:55:27] VERBOSE[15109] logger.c:   == Everyone is busy/congested at this time (0:0/0/0)
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Dial
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-dialout-trunk:20] Goto("SIP/101-090d0528", "s-CHANUNAVAIL|1") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Goto
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/101-090d0528", "1?noreport") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/101-090d0528", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks") in new stack
[Oct 25 15:55:27] DEBUG[15109] app_macro.c: Executed application: Noop
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [726000325@from-internal:5] Macro("SIP/101-090d0528", "outisbusy|") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- Executing [s@macro-outisbusy:1] Playback("SIP/101-090d0528", "all-circuits-busy-now|noanswer") in new stack
[Oct 25 15:55:27] VERBOSE[15109] logger.c:     -- <SIP/101-090d0528> Playing 'all-circuits-busy-now' (language 'en')
[Oct 25 15:55:31] DEBUG[15109] app_macro.c: Executed application: Playback
[Oct 25 15:55:31] VERBOSE[15109] logger.c:     -- Executing [s@macro-outisbusy:2] Playback("SIP/101-090d0528", "pls-try-call-later|noanswer") in new stack
[Oct 25 15:55:31] VERBOSE[15109] logger.c:     -- <SIP/101-090d0528> Playing 'pls-try-call-later' (language 'en')
[Oct 25 15:55:36] DEBUG[15109] app_macro.c: Executed application: Playback
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Executing [s@macro-outisbusy:3] Macro("SIP/101-090d0528", "hangupcall") in new stack
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/101-090d0528", "w") in new stack
[Oct 25 15:55:36] DEBUG[15109] app_macro.c: Executed application: ResetCDR
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Executing [s@macro-hangupcall:2] NoCDR("SIP/101-090d0528", "") in new stack
[Oct 25 15:55:36] DEBUG[15109] app_macro.c: Executed application: NoCDR
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Executing [s@macro-hangupcall:3] GotoIf("SIP/101-090d0528", "1?skiprg") in new stack
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Goto (macro-hangupcall,s,6)
[Oct 25 15:55:36] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Executing [s@macro-hangupcall:6] GotoIf("SIP/101-090d0528", "1?skipblkvm") in new stack
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Goto (macro-hangupcall,s,9)
[Oct 25 15:55:36] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Executing [s@macro-hangupcall:9] GotoIf("SIP/101-090d0528", "1?theend") in new stack
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Goto (macro-hangupcall,s,11)
[Oct 25 15:55:36] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Executing [s@macro-hangupcall:11] Hangup("SIP/101-090d0528", "") in new stack
[Oct 25 15:55:36] VERBOSE[15109] logger.c:   == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/101-090d0528' in macro 'hangupcall'
[Oct 25 15:55:36] VERBOSE[15109] logger.c:   == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/101-090d0528' in macro 'outisbusy'
[Oct 25 15:55:36] VERBOSE[15109] logger.c:   == Spawn extension (from-internal, 726000325, 5) exited non-zero on 'SIP/101-090d0528'
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Executing [h@from-internal:1] Macro("SIP/101-090d0528", "hangupcall") in new stack
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/101-090d0528", "w") in new stack
[Oct 25 15:55:36] DEBUG[15109] app_macro.c: Executed application: ResetCDR
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Executing [s@macro-hangupcall:2] NoCDR("SIP/101-090d0528", "") in new stack
[Oct 25 15:55:36] DEBUG[15109] app_macro.c: Executed application: NoCDR
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Executing [s@macro-hangupcall:3] GotoIf("SIP/101-090d0528", "1?skiprg") in new stack
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Goto (macro-hangupcall,s,6)
[Oct 25 15:55:36] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Executing [s@macro-hangupcall:6] GotoIf("SIP/101-090d0528", "1?skipblkvm") in new stack
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Goto (macro-hangupcall,s,9)
[Oct 25 15:55:36] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Executing [s@macro-hangupcall:9] GotoIf("SIP/101-090d0528", "1?theend") in new stack
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Goto (macro-hangupcall,s,11)
[Oct 25 15:55:36] DEBUG[15109] app_macro.c: Executed application: GotoIf
[Oct 25 15:55:36] VERBOSE[15109] logger.c:     -- Executing [s@macro-hangupcall:11] Hangup("SIP/101-090d0528", "") in new stack
[Oct 25 15:55:36] VERBOSE[15109] logger.c:   == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/101-090d0528' in macro 'hangupcall'
[Oct 25 15:55:36] VERBOSE[15109] logger.c:   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-090d0528'
Here is the trunk info:
Code:
* Name       : Skype
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : from-trunk
  Subscr.Cont. : <Not set>
  Language     : 
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  FromUser     : MY_SKYPE_USERNAME
  FromDomain   : sip.skype.com
  Callgroup    : 
  Pickupgroup  : 
  Mailbox      : 
  VM Extension : *97
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : No
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : invite
  Nat          : RFC3581
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       : sip.skype.com
  Addr->IP     : 63.209.144.201 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 0
  Def. Username: MY_SKYPE_USERNAME
  SIP Options  : (none)
  Codecs       : 0x0 (nothing)
  Codec Order  : (none)
  Auto-Framing:  No 
  Status       : OK (195 ms)
  Useragent    : 
  Reg. Contact :
[/hide]
Any ideas?
 

anzigo

Joined
Oct 11, 2010
Messages
8
Likes
0
Points
0
#11
Looks like you may have some codec issues. From your asterisk CLI log, we can see this:
Code:
WARNING[15109] chan_sip.c: No audio format found to offer. Cancelling call to 26000325
Are you certain that the disallow=all comes BEFORE your allow=g711? The order does matter.
 

pranasoftvoip

Joined
Jul 16, 2010
Messages
7
Likes
0
Points
0
#12
Re: Re:Skype Connect troubles

Yes, I'm sure :p
Code:
username=MY_SKYPE_USERNAME
type=peer
secret=MY_SKYPE_PASSWORD
qualify=yes
insecure=invite
host=sip.skype.com
fromuser=MY_SKYPE_USERNAME
fromdomain=sip.skype.com
canreinvite=no
disallow=all
allow=g711
context=from-trunk
I've tried to allow all the codecs to work, but stil can't make an outbound call.
 

anzigo

Joined
Oct 11, 2010
Messages
8
Likes
0
Points
0
#13
pranasoftvoip, I had seen the config before, but just had to ask as I'm out of ideas right now. Have you had any luck since your last post?
 

pranasoftvoip

Joined
Jul 16, 2010
Messages
7
Likes
0
Points
0
#14
Re: Re:Skype Connect troubles

Anzigo,

I'm sorry for not anwsering sooner :blush:

After almost going crazy about this issue, I think I have a clue of what might be going on. :blink:

I think is an issue with the Firewall in the office!!! :(

Next week I'll install ZeroShell and make a new battery of tests, I'll let you know how it works! ;)
 

alang

Joined
Mar 19, 2008
Messages
47
Likes
0
Points
0
#15
Re: Re:Skype Connect troubles

I saw the configuration you posted, there is a incorrect parameter within that.

This one:
allow=g711

Sould be:
allow=ulaw&alaw

Try it before you change firewall.
 

hijasahmed

Joined
Feb 8, 2011
Messages
6
Likes
0
Points
0
#16
Re: Re:Skype Connect troubles

Hi this configuration is working for me for outbound. This will surely work if the firewall configuration and your outbound route is correct


username=99051xxxxxxxxx
type=peer
secret=xxxx
qualify=yes
insecure=invite
host=sip.skype.com
fromuser=99051xxxxxxxxx
fromdomain=sip.skype.com
disallow=all
context=from-pstn
canreinvite=no
allow=g729&alaw&ulaw
 

kenneth.menken

Joined
Nov 9, 2011
Messages
2
Likes
0
Points
0
#17
Sorry for bumping an old post. I'm actually having an issue with the inbound calls. I set up a business account for incoming calls and an inbound route to a specific extension. Calls will not connect.

For example, I created a business account called skype1 and added that under the incoming call section of the skype connect sip profile.

I created an inbound route where the DID number is skype1 and pointed it to an extension.

Is there anything I'm missing? All my firewall settings are correct.
 

hijasahmed

Joined
Feb 8, 2011
Messages
6
Likes
0
Points
0
#18
1.do you have other trunks working fine in the same elastix server???

2. Have you tried the outbound with this skype route???

3. which version of elastix you are using?
 

kenneth.menken

Joined
Nov 9, 2011
Messages
2
Likes
0
Points
0
#19
1.do you have other trunks working fine in the same elastix server???

Yup, all my other trunks are working correctly on the same server.

2. Have you tried the outbound with this skype route???

Outbound works with the trunk settings from this thread.

3. which version of elastix you are using?

Elastix 2.2.0
 

hijasahmed

Joined
Feb 8, 2011
Messages
6
Likes
0
Points
0
#20
Please try this config

Peer details

username=99051000XXXXXX
type=peer
secret=XXXX
qualify=yes
insecure=invite
host=sip.skype.com
fromuser=99051000XXXXXX
fromdomain=sip.skype.com
disallow=all
directmedia=no
allow=g729&alaw&ulaw


uSER DETAILS

username=99051000XXXXXX
type=user
secret=XXXX
qualify=yes
insecure=invite
host=sip.skype.com
fromuser=99051000XXXXXX
fromdomain=sip.skype.com
dtmfmode=rfc2833
disallow=all
directmedia=no
allow=g729&alaw&ulaw


REGISTRATIN STRING

99051000XXXXXX:YOURPASSWORD@sip.skype.com/99051000XXXXXX
 

Members online

No members online now.

Latest posts

Forum statistics

Threads
30,902
Messages
130,887
Members
17,566
Latest member
Fpino
Top