Best SIP trunk providers for Lync in UK

Hi,
Does anyone have any experience with SIP trunk providers in UK? can you suggest one that does not require me to purchase a MPLS line (like BT) or one that requires me to have a SBC on site. I have tried Gamma too but they are extremely slow in any kind of
response, any request can take over a month for them to complete.
P.S. please do not send me the link for approved SIP providers, i know about that, I am trying to see if anyone from UK has some good experience with a provider that I can utilize.
Thanks,

This is most likely because the analogue phones are sending inband message in the progress indicator. We can see this from the ISDN logs, which is not present in the digital phones.
Oct 7 21:36:38.991: ISDN Se0/0/0:15 Q931: RX <- ALERTING pd = 8 callref = 0x8213
Progress Ind i = 0x8582 - Destination address is non-ISDN
This is because usually analogue ports/devices always send 183 Session Progress.
Where are these analogue phones connected? On a cisco router? if they are then you can use the command below to get them to send alerting (180 ringing) to the sip provider.
voice call send-alert

Similar Messages

SIP trunk source ports

I have a customer who has anti fraud settings on their SBC that checks to make sure the SOURCE port of incoming SIP connections is 5060. Mediation servers assign an ephemeral port as the SOURCE port. Anyone know of a way to change this behavior?
Thanks.

Agree with Anthony, For more info., You can check below link
http://blog.insidelync.com/2013/04/sip-trunking-101-with-lync-server-2013/
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Mai Ali

SIP trunk INCOMING/OUTGOING calling issue

Dears,
i am facing issue i.e inbound / outbound calls are not wokring after configuring SIP trunk . the call flow is
SERVICEPROVIDER--------sip trunk------VG --------------sip trunk ----------CUCM
For Outbound calls ---error-----tot tot tot
For Inbound Calls -----error--- Silence
Below are the configuration and debugs:

Appreciate if some provide me Sip configuration and troubleshooting guide...Thankyou

Cca xml export for SIP Trunk service providers, request or schema.

Hi
Please can I have the complete xml schema for SIP trunks so I can create a proper template?
Exporting from CCA 3.2.2 gives me an incomplete and sometimes inaccurate xml file
example: toll fraud protection - I would like to add this in to the template - how
SIP registration: If I set this to 120 on CCA and export - the exported template remains as 3600
thanks
Adam

Hi Adam,
Are you saying when you try to export your SIP trunk settings via CCA that your settings are incorrect in the xml file? If so, I recommend opening a case to troubleshoot the issue as it sounds like a bug to me.

Configuring Level3 SIP trunk with Lync 2013

Hi, I ran into some issues trying to configure SIP trunk from Level 3 and I was hoping someone here can help. We have our mediation server collocated with FE and SIP traffic goes from public IP, port 5060 via NAT, to local IP on FE, port 5060.
Level 3 provided us with one signaling IP and two RTP IPs.
I tried multiple trunk configuration settings and I can see that when I'm placing a call from Lync to an outside number I'm getting INVITE from Level 3 signaling IP, the session is established, phone rings, but there is no audio on either side. There's also
a METHOD NOT ALLOWED message coming from them, which doesn't tell me much about what's happening.
If I call to a Level 3 DID (assigned to my Lync user account) there's also INVITE from their side, but later I receive a CANCEL from them due to idle session. The phone never rings.
Questions:
1) Does anyone have Level 3 SIP trunks configured and can share their Get-TrunkConfiguration settings? What settings should I have for encryption, refer, sessionTimer / RTCP, and others? Level 3 refuses to provide any additional information besides IPs.
2) Do I understand this correctly that when configuring PSTN gateways in topology, one of the RTP IPs should be entered in the "alternate media IP" field? We have SIP trunks from another provider (which work fine), and they only use one IP
for everything, so I don't have any experience configuring separate SIP and media IPs with Lync.
Thanks, and let me know if I should provide additional info.

Hi,
On Lync topology PSTN gateways interface, please check if you enter gateway listening port 5060 and enable TCP option.
Please also check if you enable refer support on Lync Server Control Panel, if you enable it please uncheck it.
You can compare the trunk configuration for Level 3 in the part “Sample Trunk Configuration for Level 3” in the link below with yours’, it is for Lync server 2010 but similar for Lync server 2013:
http://blogs.technet.com/b/nexthop/archive/2013/04/10/configuring-lync-2010-server-to-work-with-level-3-sip-trunking-services.aspx
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

Best way to implement SIP Options Pings on a SIP Trunk

I wanted to see if anyone had suggestions on the best way to configure SIP Options Pings.
Typically I would configure them per dial-peer. However, I really want to do it per destination IP address. I do not want a SIP Options ping for every single dial-peer being sent out every X seconds.
Example:
In my case I have 4 SIP trunks in the same CUBE. Each pointing to a different destination IP. There are 6 dial-peers per SIP trunk. I really do not want 24 option pings going out every X seconds. I guess I've never actually did a debug to see how many pings are going out at a time but I am assuming it sends one for each dial-peer or does it?
If I am correct in my assumption, is there way to only send one ping per destination IP and if that single IP goes unresponsive then all 6 dial-peers go down?

Hello,
The OOD option ping is sent per dial-peer to the destination.
Restrictions
•The Cisco Unified Border Element OOD Options ping feature can only be configured at the VoIP Dial-peer level.
•All dial peers start in an active (not busied out) state on a router boot or reboot.
•If a dial-peer has both an outbound proxy and a session target configured, the OOD options ping is sent to the outbound proxy address first.
•Though multiple dial-peers may point to the same SIP server IP address, an independent OOD options ping is sent for each dial-peer.
•If a SIP server is configured as a DNS hostname, OOD Options pings are sent to all the returned addresses until a response is received.
•Configuration for Cisco Unified Border Element OOD and TDM Gateway OOD are different, but can co-exist.
//Suresh
Please rate all the useful posts.

Best Practice to Integrate CER with RedSky E911 Anywhere via SIP Trunk

We are trying to integrate CER 9 with RedSky for V911 using a SIP trunk and need assistance with best practice and configuration. There is very little documentation regarding "best practice" for routing these calls to RedSky. This trunk will be handling the majority of our geographically dispersed company's 911 calls.
My question is: should we use an IPsec tunnel for this? The only reference I found was this: http://www.cisco.com/c/en/us/solutions/collateral/enterprise-networks/virtual-office/deployment_guide_c07-636876.htmlm which recommends an IPsec tunnel for the SIP trunk to Intrado. I would think there are issues with an unsecure SIP trunk for 911 calls. Looking for advice or specifics on how to configure this. Does the SIP trunk require a CUBE or is a CUBE only required for the IPsec tunnel?
Any insight is appreciated.
Thank you.

you can use Session Trace in RTMT to check who is disconnecting the call and why.

Lync Enterprise Voice Deployment - Can't get SIP trunks to work

Hey guys,
So i'm doing a proof-of-concept for Lync 2013 Enterprise voice. I've just purchased two sip trunks with 3 DIDs. I've followed their directions but have had no luck making outgoing PSTN numbers. I always get the error message "Disconnected service unavailable".
I did some troubleshooting with their tech support, did some logging with Lync & gave them our SIP Logs.
Here is what they told me back:
I have reviewed the log and found the following:
Your logs state the below:
Direction: outgoing;source="local"
Peer: lync2013.network.caedm.ca:5070
sip:[email protected]:5070
ß this particular line states to send the call of 780-719-8033 to 208.82.88.120 at port 5070. Please change this to port 5060.
In your outbound To field you have the following:
To: <sip:7807198033;phone-context=[email protected];user=phone>;epid=EC06E70A33;tag=b91b46715a
Please change this to reflect: To:
sip:[email protected];user=phone>;epid=EC06E70A33;tag=b91b46715a
Where can I set the outgoing port to 5060? In the topology builder I have already configured the listening port under "PSTN gateway" to be TCP 5060. I don't see anything in the topology builder relating to outgoing ports, only "listening ports".
Thanks.

Looks exactly the same as mine. (I also use a SIP trunk.)
I assume you can ping the SIP provider from your mediation server? (and telnet to TCP 5060) without any issue.
Sorry, try netstat -ano
You should have some entries that look this: (where 10.0.0.3 is your mediation servers local IP)
TCP 10.0.0.3:5060 208.82.88.120:58869 ESTABLISHED 1780
TCP 10.0.0.3:5060 208.82.88.120:58924 ESTABLISHED 1780
TCP 10.0.0.3:5060 208.82.88.120:59094 ESTABLISHED 1780
TCP 10.0.0.3:5060 208.82.88.120:59216 ESTABLISHED 1780
TCP 10.0.0.3:5060 208.82.88.120:59658 ESTABLISHED 1780
TCP 10.0.0.3:5060 208.82.88.120:59734 ESTABLISHED 1780
TCP 10.0.0.3:5060 208.82.88.120:59953 ESTABLISHED 1780
TCP 10.0.0.3:5060 208.82.88.120:60255 ESTABLISHED 1780
TCP 10.0.0.3:5060 208.82.88.120:60318 ESTABLISHED 1780
TCP 10.0.0.3:5060 208.82.88.120:60631 ESTABLISHED 1780
TCP 10.0.0.3:5060 208.82.88.120:60765 ESTABLISHED 1780
TCP 10.0.0.3:5060 208.82.88.120:60834 ESTABLISHED 1780
TCP 10.0.0.3:5060 208.82.88.120:60885 ESTABLISHED 1780
TCP 10.0.0.3:5060 208.82.88.120:60934 ESTABLISHED 1780
TCP 10.0.0.3:5060 208.82.88.120:61141 ESTABLISHED 1780
If this helped you please click "Vote As Helpful" if it answered your question please click "Mark As Answer"
Georg Thomas | Lync MVP
Blog www.lynced.com.au | Twitter
@georgathomas
Lync Edge Port Check (Beta)
This forum post is my own opinion and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs.

Lync server sip trunk , IM and presence forwarded in trunk, is possible?

tank you in advance for all the help i can obtain.
I have managed to install, configure a lync server 2013 evaluation.
One domain controller and one front end server (without Edge Server)
My first goal was to configure a SIp trunk with an Open Source SIP proxy/server. (OpenSIPS)
All work properly, it was many days of suffering and hard work. The call are correcly routed in trunk in both direction.There is some possibility to forward sip message about presence (subscribe,publish, notify) inside the trunk?
Lync manage it internally but when i want to send the message cross domain, a SUBSCRIBE as example(my lync domain is lynclab.local, my OpenSIPS domain is opensips.local) i obtain this error in lync:
"Non-trusted source sent an FQDN/IP that doesn't match a .... The SIP 404 Not Found" (referring my esternal domain Opensips.local.
Probably i have deeply misudestood someting, reading for many days documents and manuals i cannot find a solution.
I need to configure an XMPP Gateway for "exchange" status message between sip domain?
thank you for all, and excuse me if the questions is tryvial or absolutely out of context.

You'll need a trusted application for this, you won't be able to forward the subscribe events through a standard sip trunk sent through the mediation role unfortunately. You'll have to find something specifically written to help the two systems interoperate,
but out of the box this would not be native functionality from the Lync side.
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications
This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs.

Lync 2013 with SIP trunk with panasonic kx-tde200

Hi
My company has installed a panasonic ip-pbx kx-tde for multiline with 100 number range for telephone service.
Now my company is going to replace multiline by sip trunk . It will still work with Panasonic pbx box just need to reprogramme to be able to connected to the sip proxy which is managed by internet service provider.
For this scenario , would Lync 2013 voice work if I just add PSTN gateway which is the ip of panasonic pbx address to the frontend in topology ? Or I may need a mediation server as a must requirement to make lync voice work?
Thanks
WenFei

Media bypass allow a call to basically skip the mediation server once it's established and go directly from gateway (in this case the PBX) and the endpoint (the telephone handset or Lync client) More information here: http://technet.microsoft.com/en-us/library/gg398719.aspx
By having this (if your PBX supports it) you reduce the load on the mediation servers. Before you go too far down the road also make sure that your PBX supports SIP trunks that are SIP over TCP (as Lync doesn't work with SIP over UDP)
Sort of, the easiest way is to add the .com as an additional SIP domain in Topology builder, you will need to create DNS records for it (both internal and external) and you will need to reissue the certs with additional SANs to support the second domain.
YOu will also need to update all the users to use the new suffix of xxx.com. So it's not a small task.
If this helped you please click "Vote As Helpful" if it answered your question please click "Mark As Answer" | Blog
www.lynced.com.au | Twitter
@imlynced

What is best recommendstion for DNS LB for lync 2013 Edge servers

What is best recommendation for DNS LB for lync 2013 Edge servers ?. We have F5 LB for edge and want to decide if we can go with DNS base LB for Edge servers.
Anil MCC 2011,ITIL V3,MCSA 2003,MCTS 2010, My Blog : http://messagingschool.wordpress.com

It will be better to Use Hardware Load balancing (F5).
If you choose to use DNS load balancing for a pool but still need to implement hardware load balancers for traffic such as HTTP traffic, the administration of the hardware load balancers is greatly simplified. For example, configuring the hardware load balancer
will be simpler as it will only manage the HTTP and HTTPS traffic, while all other protocols will be managed by DNS load balancing
Also for more info., you can check below links
http://technet.microsoft.com/en-us/library/gg615011.aspx
http://technet.microsoft.com/en-us/library/gg398634.aspx
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question, please click "Mark As Answer"
Mai Ali | My blog: Technical | Twitter:
Mai Ali

Why do we need MTP in the SIP trunk for CVP warm transfers

Hi All,
Why do we need to enable MTP in SIP trunk between CUCM and CVP for CVP based trasnfers???
Thanks in advance!!
Regards,
Thammaya Gupta K.

I saw also in the CDR logs that the IP Phone media transport going to CUBE is in G711.And as well in the wireshark capture of the IP communicator that the CUCM invoke to use the g711 codec but as per ITSP logs they are now in the g729.
@ Jamie If I un-tick the MTP point required in SIP trunk will make the call leg from IP Phone to CUBE g729 (w/o hw resource), I have also tried to use g729 preferred originating codec, but still the IP Phone is using g711.
I have seen a documentation states:
" To configure G.729 codecs for use with a SIP trunk, you must use a hardware MTP or transcoder that supports the G.729 codec." - I read this on the CUCM help page under configuring SIP trunk setting.
Our ultimate goal is to use g729 without using HW MTP/ transcoder.
IP Phone ->CUCM SIP Trunk ->CUBE-> ITSP

CUCM SIP Trunk Unable to find a device handler for the request received

I have a lab running CUCM 10.0 that has a SIP trunk to a VCS. Previously I was able to place and receive calls across the trunk, but now I can only place calls, not receive. I did recently upgrade to 10.5, but I really can't recall if it ever worked after the upgrade. The error I'm getting in CUCM is
Unable to route message, Cannot find the SIP Device with Name=192.1.1.61, Source Port=5060, IpAddress Type=0
This confuses me, because the destination IP/port of the VCS SIP trunk is 192.1.1.61:5060. The trunk on CUCM show as up and the Zone in VCS shows as up. What is going on?
Thanks, Mike

I figured it out. Long story short, it was a port mismatch. When you configure Mobile Remote Access on the Expressway Core it creates a SIP trunk from Expressway to CUCM using port 5060. If you then create another SIP trunk from the same Expressway to CUCM for firewall traversal or external calling, you must use a different port. The port used for the zone in Expressway must match the SIP security profile used on the CUCM side SIP trunk. So I had created another SIP Security profile in CUCM with port 5062 for this purpose. However, the VCS in question that I was working with was only being used for FW traversal, so I used 5060 as the port on the VCS side, but I applied the SIP security profile that used 5062 on the CUCM side. That's why CUCM was throwing that "device not found" error. Once I changed the Security Profile to match the same port used on VCS (5060) all was well. Sorry for the long winded answer. Hope it helps someone else down the road.

Testing new sip trunk for an upgraded system....

Hey all,
Hoping I can get a quick direction on this.....
We are upgrading our call center servers and need to point a few test toll free numbers.
I have already created a test sip trunk....here's where it all gets muddy for me. What do I need to do to route 4 specific toll free numbers to hit this specific sip trunk instead of going to our current one?
Any docs and verbal guidance would be appreciated.
Thanks in advance all!

Ok, so here's what i did....
I created the following >>> RG with new server as member>RL with the new RG as member>RP
We already have a RP of 5[1-9]XXX which matches the DNIS of the 4 toll free numbers we are using for this test, so I just went ahead and created an exact match RP of 59796. (I'll create the other 3 if I can get this to work)
Before I created all this, the call went to our sales dept like designed. After creating it, now I get a "this number has been disconnected" message. What did I miss?

CME required CUBE license for ISP SIP Trunk

Hi,
We have IP telephony setup with CME 9.1 with 2921 ISRG2 router.
Client would like to take ISP SIP trunk.
Do we require CUBE license for the same.Because I tried in the 2851 router without the CUBE license and its working fine.
How I can check CUBE license installed my CME.
When I check license in CME I can see that UC K9 and IPBasek9 license are permanent.
Do this enough for the SIP trunk configuration.
Thanks & Regards
Nithin Louis.

Just to add to George answer - CUBE licenses are somekind "honor" based right now which means you can configure everything and it will work but you need license paper for proving that you have adequate number of CUBE licenses for your system.
Maybe in future Cisco will improve this with option that you will need to install license and only then it will work - like many other licensing right now...
BR,
Dragan

Best SIP trunk providers for Lync in UK

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