Has anyone ever trunked Cisco Call Manager to Shoretel? PRI? H323? SIP?
I am looking for some interoperability docs
We have a potential customer running shoretel at several sites, we would like to add new sites on Call Manager 4.2 or possible 7.1 and have the sites communicate to the shoretel sites (4 digit dialing)
I havent found any doc in our website, I will go for SIP/H.323 so u dont add extra costs.
Also open a TAC case so Interop team gets engaged or contact your Account team to engage them as well.
We have a shoretel in our lab, and I was just asked this same thing. Im going to try to trunk the 2 together and post the results/findings.
Curious to know this as well. Do not have a shoretel to test an integration, but feel h323 is the best choice for the integration.
Has anyone gotten an answer on this yet? I've tried, but due to limited knowledge the shoretel side its hasn't been successful. The call is going to the shoregear switch and SIP is enabled but get error messages stating objectid not found or SIP error 488. I know this can be done with a PRI, but would like to do it with SIP. Just curious what others have done in this situation...
Hello, I was wondering how we can get this to work between ShoreTel Shoregear switch and Cisco 2811 VGW with cross-over cable between the two serving as tie-line T1 PRI. We have the 2811 acting as the Network/CO to the Shoregear switch as User. The Cisco VGW has two T1 ports configured; Subunit0 is connected to the CO as true PRI and Subunit2 is connected to the ShoreTel switch.
- Cisco Call Mgr ver is 6.1.4
I have route patterns configured in UCCM to allow successful dialing from cisco ip phones to ShoreTel DNs and vice-versa. However, we're unable to make outside calls from ShoreTel phones; we get fast busy tone. The cisco phones dial out fine. The T1/PRI port on the cisco side is currently configured with MGCP and we'd like to see if it's possible to make that work without having to reconfigure the port as H.323.
Has anyone had any success with tandem trunking ShoreTel with Cisco MGCP? If so, please give insight as to how I can configure route pattern (or present current route patterns) to work with calls coming from ShoreTel side and hit the VGW. Currently, debug is showing the calls as "Unrecognized" on the VGW.
Admittedly, I haven't worked with Shoretel but we do tandem routing with other PBX's all the time in migrations. I haven't heard it discussed or asked so I will broach the following topic. There was mention of 4-digit dialing but other folks have chimed in on what they're doing as well. Anyway, my assumption is that you are saying if a Shoretel call is destined to PSTN and goes over the tandem to the Cisco, it doesn't work. So, my first question would be is how many digits are you passing with the call? You would want to make sure you are passing the leading 9 all the way thru to the CUCM and let it be evaulated there. The same would be true vice versa. As for integration type, I almost always MGCP so unless there is a requirement to do H323 - I don't see why it would cause an issue.
I got it working with no problem. One thing I had to do is enable external routing on a per station DN on the shortel side.
Without this setting it will not work at all.
Where do I find the control in the Shoretel Director console to enable external routing? We are permitting our users access to the PRI interface on the Cisco VGW via a User Group within Shoretel.
We got this working.
Here's what we did:
1) Switched the PRI config on the Cisco VGW from Network to User (PRI NI2)
2) Switched the PRI config on the ShoreTel switch from User to Network
3) Changed the Call Search Space in the "Call Routing Information - Inbound Calls" under Gateway Configuration in Cisco Call Mgr; the Cisco Dialed Number Analyzer was showing that the original Call Search Space was somehow blocking all external calls from reaching the outside from ShoreTel
It's now working flawlessly. Thanks for your help on this.
I'm wondering if anyone has successfully used a SIP trunk between Shoretel and CUCM (I'm on CUCM version 6.1(5)). I have a SIP trunk configured and I can successfully call from a Shoretel phone to a Cisco phone using the SIP trunk and I'm receiving the Caller ID and Name Display from the Shoretel phone. However, when I try to call in the reverse direction, from the Cisco Phone to the Shoretel phone over the SIP trunk I get a fast busy. I can successfully ping between CUCM and the Shoregear 220 T1A. I've used the DN Analyzer and it shows the call routing out the SIP trunk. I did a CUCM trace of the call and sent it to TAC and here is what they say:
I have checked the CUCM traces and I found that you have a connectivity issue or firewall blocking the sip traffic (TCP, port 5060). That is the reason why we can not see any INVITE message in the CUCM traces.
First check that you can ping the sip server from the CUCM, then verify if you have any ACL, firewall in your network blocking the SIP traffic.
I'm finding that hard to believe in that I have voice traffic working in one direction and that i can ping from the CUCM server to the Shoregear device. Any ideas?
Lots of stuff to check here... CCM accepts both TCP and UDP for SIP, which is nice, but might mask the fact that Shoretel is using UDP. Your TCP connection may be rejected if it only accepts UDP (guesswork, I know nothing of Shoretel). What TAC will see is an attemped TCP connection that doesn't complete the handshake, and therefore no SIP packet is sent...
A quick test for TCP is to telnet to the shoretel IP like so:
Telnet x.x.x.x 5060
If you get a black screen, and can CTRL+] out of it (from a windows box, this is)... then it's responding on that ip/port. If not, it doesn't use TCP - set up another SIP Security Profile, set it to UDP outbound, and apply to the trunk.
It might help to carry out a packet cap from your CCM, covering call from Shoretel-Cisco.
Utils network capture size all count 100000 file SIP-1
Then CTRL+C when you have done the test call... download the packet cap using RTMT 'collect logs' in trace/log central, and then look at the capture to determine whether shoretel is using UDP, and verify which IP it comes from (that IP will be a likely place to send the trunk).
Thanks for all the info. I will start checking. One thing I would like to add is that I have tried changing the SIP Security Profile to UDP. With UDP selected, saved and rest, then when I try a test call from the Cisco phone, instead of getting a fast busy I get a recorded message that says "Your call cannot be completed as dialed, please consult your directory and call again or ask your operator for assistance, this is a recording..........then fast busy". The curious this is that calling in the reverse direction still works (Shoretel to Cisco).
Yeah, it's really not best to pay too much attention to what the Cisco Lady says... it's really not very descriptive, certainly when you get to the point of these interop issues.
I would suggest you take and post a few packet captures as I suggested before - one running TCP, one UDP, and one calling happily from Shoretel to CCM.
We should be able to help you further then...
One other thing if you've not done it already - check that the server to which the Shoretel is pointed is the highest priority in the CCM group assigned to the Dev Pool of the SIP trunk. This will concentrate all your logs on one server, and if the Shoretel has some fancy restriction on source IP should ensure you can get past that...
I am also having a very similar issue. I have a SIP trunk to the PSTN on an asterisk box and calls come in fine through the PSTN --> asterisk --> CUCM DN. If I try to dial out from an IP phone or softphone registered in CUCM through the SIP trunk, I get the message "Your call can not be completed as dialed, check your directory and dial again." I tried creating a softphone extension in asterisk and I can dial the CUCM DN perfectly, yet even when I created a specific route pattern to dial the one extension in asterisk, I get the same message from CUCM. I tried the DNA and it shows the call as routing out through the SIP trunk, no issue. I have tested network connectivity between the systems and it is fine also. SIP trunk is set to use UDP. Any suggestions?
Also, I ran wireshark on the connection. When I made a call from the xlite softphone from asterisk --> CCM, i showed the normal SIP traffic. When I make a call from IP Communicator --> asterisk, NO SIP traffic appears.
The same suggestions apply to you. 'Cisco Lady' tells users it isn't working; she doesn't tell you what the problem is.
Also just because you can't send a call out of a trunk doesn't mean you have the same issue - you have a different system you are integrating to, and the resolution is likely to be different. You should open a new thread.
OLD post, but I recently ran across this and found _no_ useful information on SIP trunking online.
Symptom was that we could receive calls over a SIP trunk to a Shoretel switch but we could not send calls across the SIP trunk (from an 8.6 CUCM).
We determined today that the Shortel switch didn't like an INVITE without the codec SDP, so enabling Early Offer or checking the MTP Required on the SIP trunk fixed the problem.
Same old post & same old question.
We have a new customer who is trying to migrate from Shoretel to Cisco. At present they have Shoretel PBX in US and India. Now as per the 1st phase of the migration they are going to remove the shoretel from india and place a cucm. They will remove shoretel from US after one year only in the 2nd phase. So now we have to some how route the calls between CUCM & Shoretel PBX. Any information on the CUCM & Shoretel PBX integration steps would be a great help!!!