Skype Connect SIP trunking to Yeastar U100

I am trying to get this service running for the first time. I signed up for a SIP profile through Skype Manager and followed Yeastar's setup instructions. Both ends indicate the regestration was successful. I can receive incoming Skype calls. I followed Yeastar's example to set up and Outgoing Route and I am attempting to call the Skype Echo Test at 001760-660-4690 as they recommend. All I ever hear is a recording saying that "All Circuiits are Busy". I have tried dialing other PSTN number and I get the same failure. Yeastar Support has look at the outgoing call msgs between me and Skype and they say what I'm sending looks right but Skype Connect is saying that the number I'm calling is invalid.
I have been in 3 separate Chats with Skype Business support and they have not be able to give me any help as to what is going on.

Each channel carries one phone call. So if you have 4 channels, you can carry 4 simultaneous phone calls, in any combination of incoming or outgoing.

Similar Messages

Skype Connect - SIP Channels

Hi everybody,
Sorry for being a noob, and trying not to be a troll... look at me using all this web lingo, don't even know if I got it right?
Anyway never used a message board before because I've always managed to figure it out myself, and there seems to be a lot of issues here that I just can't wade through to find what I am looking for, nevertheless, you can ask and see what happens.
With Skype Connect, I read the instructions, and purchased 4-channels to receive inbound calls, from the payment area instructions it appears that 4-channels will be able to handle at least 10-inbound calls, but when you hover over the help question mark on the features page once you've assigned the channels to a SIP Profile it says you can only have 4 inbound/outbound calls.
I have trawled the internet, looking for guidance on this? Which is it? Can someone please help? I am setting up a project where I will be expecting a high volume of inbound calls, there will be very few, if any outbound calls, and 10 was a good testing point.
Can someone please advise me how many channels I would need to get at least 10 simultaneous inbound calls? Even if someone could point me in the right direction I would be really appreciative of your assistance.
Thanks,
Russell Hunt

Each channel carries one phone call. So if you have 4 channels, you can carry 4 simultaneous phone calls, in any combination of incoming or outgoing.

Unity Connection SIP trunk integration vs alerting name

Hello,
I have been implementing a Unity Connecion 8.6 with a CUCM cluster 8.6 via a SIP trunk.
The previous implemention of the Unity server was set up using voicemail ports;
When users used to call the voicemail pilot number, we were able to configure an alerting name on the vm ports, saying (to VoiceMail) for example.
I am looking for a solution for the new implementation via the SIP trunk. I just want users to see 'to VoiceMail' on their phone when they call the Unity Connection system.
Thanks for the help!
Best regards,
Antoine

I have found the solution:
If you want to dispaly a name such as Voicemail, etc you can change that in Unity Connection under Port Group --> Advanced Settings --> "Remote-Party-ID"
cheers!

Belgacom BBox 2 and Skype Connect

Hello to all,
I need some help
1st my configuration
1 Belgacom BBox2 (with 2 pst connextion linked to sip account)
1 Belgacom Forum 3000 (with 1 pstn incomming line)
2nd what i want to do
I want to configure a skype connect account into my bbox2 and connecte the bbox 2 pstn port to the incoming port of the Forum3000 so i can recive incomming calls by skype and callers can skype us for free.
3rd my question
Can somone help me to configure the skype connect sip account into the bbox?
Tks to all for help

Nevermind. I found a solution on a Forum. I tried downloading Skype for Android on my computer. I was asked for my phone number and a link to download Skype was sent on my phone. I accessed it and Skype start installing (have no idea what was installed, anyway). I start Skype and it is working fine.
The most interesting thing is that I still have the same Skype version (5.1.0.57240) but it is localized in Romanian.
Best regards,
Sebastian

Skype connect & VoIP Gateway

Hi,
Could you please any give me more information about VoIP gateway, and how to connect it with PBX and how to buy it and what is the price?
Thanks,
Mohamed ElDieb

Hello ToshibaJoe,
If your PBX doesn’t support SIP check with your manufacturer to see whether a SIP gateway, module or upgrade is available for your existing PBX, that would enable you to use Skype Connect/SIP.
Also, it is highly recommended that a Skype Connect certified device is used, to gaurantee all features and functions work. Following is a link to that lists all Skype Connect certified products:
http://www.tekvizionlabs.com/3rdpartyprograms/skype/skype_verified_products.php
Please let us know if you have further questions.
Thank you,
MariaA
Skype Enterprise Support

Skype Connect outside US

Hi, I'm interested on implementing Skype Connect in my company, that is located in Colombia.
So I have some questions:
Is Skype connect avaiable for Colombia?
Can I have and use a US Skype number for incomming calls?
Thank you.

Hello ToshibaJoe,
If your PBX doesn’t support SIP check with your manufacturer to see whether a SIP gateway, module or upgrade is available for your existing PBX, that would enable you to use Skype Connect/SIP.
Also, it is highly recommended that a Skype Connect certified device is used, to gaurantee all features and functions work. Following is a link to that lists all Skype Connect certified products:
http://www.tekvizionlabs.com/3rdpartyprograms/skype/skype_verified_products.php
Please let us know if you have further questions.
Thank you,
MariaA
Skype Enterprise Support

Skype Connect & Unlimited Call Plans

We are currently using Elastix (Asterisk based) to receive and make calls in our office. Since we are already using Skype on our laptops, it would make sense to use Skype in the office as well. However, as a startup company with minimum needs, we need to keep an eye on cost.
I already browsed through the forum and it appears that people are using Skype with Elastix successfully, but what about the calling plans? Can we just sign up for Skype Connect and link to an unlimited plan to make outgoing calls? I am guessing that we would require one plan per channel?
I saw the call limitations for the unlimited calling plans, but we would never exceed the listed totals.
Thanks!

Synaesthesia wrote:
With a Skype Connect SIP channel subscription and Cisco SPA504g, am I able to use my Unlimited World calling plan for outgoing calls, or am I forced to use the standard Skype calling rates for outgoing calls, or is another entirely different option possible?
If I am forced to use the standard Skype calling rates, does that mean 4.5¢1/min for Johannesburg, South Africa?
Thank you.
Unlimited world or other calling subscription available for personal accounts won't work for SIP account(Skype Connect). Calls using Skype Connect will use Skype Credits or the 5000 calling minutes to US (The only calling subscription for SIP).
For more information, please visit this link: https://support.skype.com/en/faq/FA10253/
Our greatest weakness lies in giving up. The most certain way to succeed is always to try just one more time.
If you found my post useful, please give "Kudos".
If it helped to answer your problem/question, please mark it as "Accepted Solution" so other people can find it easily.

Skype Connect and AudioCodes E-SBC

Hi
We want to use Skype Connect in our company with AudioCodes gateway. What I see from skype-connect-requirements-guide the requirements for codec is G.711 or G.729. What I know is SKype supports iLBC codec too. Do I need to make transcoding on AudioCodes gateway?

Dear scivici,
Skype Connect (SIP) only supports the following codecs:
G.729 and G.711 ulaw/ALaw
The question you are asking, in regards to iLBC, may be referring to the Skype Client which is on another network.
Please let us know if you have further questions.
Thank you,
MariaA
Skype Enterprise Support

Skype Connect and Toshiba CIX 670 PBX

Does Skype Connect work with a Toshiba CIX 670 PBX? Do I need a Media Gateway?

Hello ToshibaJoe,
If your PBX doesn’t support SIP check with your manufacturer to see whether a SIP gateway, module or upgrade is available for your existing PBX, that would enable you to use Skype Connect/SIP.
Also, it is highly recommended that a Skype Connect certified device is used, to gaurantee all features and functions work. Following is a link to that lists all Skype Connect certified products:
http://www.tekvizionlabs.com/3rdpartyprograms/skype/skype_verified_products.php
Please let us know if you have further questions.
Thank you,
MariaA
Skype Enterprise Support

Unable to perform call transfer or call park for an outbound call via SIP Trunk (SKYPE)

We have configured the SIP Trunk & SIP profile and successfull make outbound call through SIP Trunk (SKYPE). However, we are not able to perform call transfer or call park when the call is connected.
The scenario is:
A call to an phone number via SIP trunk, when call established, A perform call-transfer to B. After the call-transfer, the call Drop and Phone B show error code "Temp Fail"
When i select "enable MTP" in SIP trunk, we are able to call transfer and call park. But it limit the number of call session to 1.

You are probably running into some sort of Codec issue. IE, your phone is G.711 and the trunk is G.729. You will need to transcode the call at somepoint.

SKYPE CONNECT Trunk to CUCM.

im interested in creating a sip trunk to skype connect. im new to voice but i think i can get it done.
is a cube absolutely necessary or the trunk can be created directly to cucm.
also can a regular cisco router with access to the internet work just as well as a cube ?
my current system has a voice gateway E1 to the PSTN.. is it possible to give this voicerouter access to the internet and use to connect to skype instead of the cube?

hey anas.. i have set up the skype trunk.. and calls seem to be going out my problem is incoming calls i have a skype id .. this is the matching incoming dial peer.. dial-peer voice 6 voip
description incoming skype
translation-profile incoming IncomingSkype
preference 1
destination-pattern 13478093543
session target ipv4:192.168.xxx.x (cucm)
incoming called-number 13478093543
voice-class codec 1
dtmf-relay h245-alphanumeric.
the translation profile changes the calling number to a 4 digit number
a number which matches this dial peer which is already on the working system
dial-peer voice 10 voip
description Codec Match
destination-pattern 2...
session target ipv4:192.168.xxx.x (cucm)
voice-class codec 1
dtmf-relay h245-alphanumeric
no vad

Problem connecting two trunks to sip provider using same CUBE

We need to connect two SIP trunks from service provider to Cisco CUCM 7.1 using CUBE “Cisco 2821”, SP using the following configuration:
First SIP PSTN Link Configuration(In-Out DID/DOD 218 7700 – 218 7799)
Customer IP Address =   10.196.191.158/30
SP IP Address = 10.196.191.157/30
Protocol= SIP
SIP Port = 5060
Transport Protocol=UDP
Voice Codec= G711 A-Law
DTMF = IN-Band DTMF without RFC2833
Signaling IP address = 10.201.20.49
IP Address 10.201.20.10 (Media IP) must be visible from IP PABX
Second SIP PSTN Link Configuration( Inbound Only 920009999)
Customer IP Address =   10.196.192.94/30
SP IP Address = 10.196.192.93/30
Protocol= SIP
SIP Port = 5060
Transport Protocol=UDP
Voice Codec= G711 A-Law
DTMF = IN-Band DTMF without RFC2833
SIP server IP address = 10.201.20.49
IP Address 10.201.20.10 (Media IP) must be visible from IP PABX
When we tried to configure both links on the same CUBE we faced two problems:
-          Routing issue, as we can’t route traffic using single CUBE through two different interfaces to the same destination “ i.e we have to configure static route commend (ip route 10.201.20.49 255.255.255.255 10.196.191.157 & ip route 10.201.20.49 255.255.255.255 10.196.192.93), sip traffic coming from one link can’t be sure to send it back to the same link.
-          SIP media & signaling control binding issue, as CUBE support sip binding using one interface only “one IP Address”, if we not using binding commands on the CUBE we can’t receive any calls though any link.
We have two options:
SP to send both traffic on the same trunk link
Or
Have another CUBE for the second link.
Attached network diagram.
Any solution?????
Regards,
Ahmed Rizk

I didn't mean NAT CUCM, I meant the interface towards it. But since you're using a single interface then yes that is what you NAT. You have a lot going on in that config. Probably a lot more than you need. Like I said you should work on this in two legs. CUBE to ITSP, and then CUCM to CUBE. You're trying to make the whole thing work in one shot which is going to cause you some headaches.
Install XLite free version. In the account settings set your UserID to a generic 10 digit phone number, domain to something generic, then at the bottom set the Proxy Address to the IP of your CUBE. The media ports will be negotiated dynamically between the CUBE and the ITSP. Since you said you're not registering you will also need to give the ITSP YOUR peer IP (this is how they secure the trunk) which is whatever IP you're sourcing from when you leave your network (what you're NAT'ing the CUBE to).
For testing, reduce your config to something like this:
voice service voip
allow-connections sip to sip
allow-connections h323 to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
signaling forward none
sip
dial-peer voice 10 voip
description CUBE_TO_ITSP
session protocol sipv2
session target ipv4:SIGNALING IP PROVIDED BY ITSP
destination-pattern [2-9].........
codec g711ulaw
dtmf-relay rtp-nte sip-notify
no vad
dial-peer voice 20 voip
description ITSP_TO_CUBE
destination-pattern .
session protocol sipv2
session target ipv4:Eventually your CUCM IP...for now set it to your computers IP.
codec g711ulaw
dtmf-relay rtp-nte sip-notify
no vad
Use XLite to place a phone call from your PC (if you have a mic and speakers you can have audio if the call connects). This should come pretty close to getting your outward leg established. Once you get this part working you can add in more codecs and translation profiles if you want. Let me know what happens. Include any debug or packet cap results if you can.

Unable to perform call transfer & call park through SIP Trunk (SKYPE)

The Scenario is:
I have set up a SIP trunk to SKYPE and we are able to make outbound call to a number via SIP Trunk.
After the call is established, when we tried to make call transfer, the call DROP and the phone at the other end shows error "Temp Fail".
I tried to "enable MTP" in SIP Trunk and We are able to perform call-transfer but it limits the call session to 1.
Anyone has facing the same issue?

MTP is needed to invoke supplementary functions like hold, transfer etc. Make sure that the MTP is checked on SIP trunk, MTP is assigned to the MRGL of the device pool on SIP trunk and has sufficient resources.
HTH
Manish

Unity Connection not passing CallerID to CUCM over SIP Trunk

I'm trying to get CallerID working for Unity Connection Device Notification (and it seems everything else), however, when I run UC Remote Port Status Monitor and the Call-Out goes to CUCM for the Device Notification, no caller ID is presented to the CUCM SIP trunk.
06:06:02, New Call, CalledId=, RedirectingId=, Origin=16, Reason=1024, CallGuid=,
CallerName=, LastRedirectingId=, LastRedirectingReason=1024, PortDisplayName=LFC_CUCM-1-134,
[Origin=Unknown],[Reason=Unknown]
06:06:02,
Dialing '99254753'
06:06:32, Idle
06:06:33, Idle
Therefore, the out-going call to the PRI PSTN is:
10:59:01.005: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x5B03
        Sending Complete
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98397
                Exclusive, Channel 23
        Calling Party Number i = 0x0081, N/A
                Plan:Unknown, Type:Unknown
        Called Party Number i = 0xC1, '9254753'
                Plan:ISDN, Type:Subscriber(local)
*Dec 6 10:59:01.513: ISDN Se0/0/0:23 Q931: RX <- CALL_PR
I've looked through my SIP trunk on the CUCM side and for Inbound Calls, Connected Line ID and Presentation Name are set to "allowed" or "default" doesn't make a difference. RTMT Port Status also shows no "caller", so I'm thinking there is some way to set or allow the calling number on the Unity Connection (8.5) side.
Oddly enough, I also noticed that in Unity Connection> Telephony Integrations > Port Group, if I change the Contact Line Name from nothing to "Unity" (or whatever), the Q931 debug outbound doesn't show ANY "Calling Party Numer - = XXXXX" and the carrier throws out the BTN as the ANI.
10:46:00.837: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x5AFF
        Sending Complete
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98397
                Exclusive, Channel 23
        Called Party Number i = 0xC1, '9254753'
                Plan:ISDN, Type:Subscriber(local)
Any ideas on where/how CallerID comes from, on Unity Connection with a SIP integration?
THANKS!!
Mike.

I did not- my work around has been to put in a name for Contact Line Name under Port Group Basics Switch configuration in Unity Connection- this for some reason keeps CUCM from sending ANI TYPE/PLAN information in the Q931 message, and my carrier then sends a default ANI of the circuit's BTN. When I have time, I'll open up a TAC ticket.
Mike.

A SIP trunking connection that scales with growth

Since delivering its first inflight Internet experience in 2008, Gogo has grown to over 800 employees in five offices around the country. With 1000s of customers thirsty for in air data connections, its crucial that Gogo is highly available to support its staff and its airline partners globally. Capacity was near its max, but the […]
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This topic first appeared in the Spiceworks Community

Hi,
No as you said you have already created the Trunk between the Sonus and Lync you can use the same trunk. but in the Sonus you have to configure the outbound routes from which Trunk you have to send the calls for the new Trunk provider or the Level3.
check this
https://support.sonus.net/display/UXDOC41/SIP+Trunking+Between+SIP+Border+Elements
Whenever you see a helpful reply, click on Vote As Helpful & click on Mark As Answer if a post answers your question.

Skype Connect SIP trunking to Yeastar U100

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