Some time ago, I needed to configure an SIP trunk between a Trixbox/FreePBX (Asterisk on Linux) PBX and a Cisco Call Manager PBX. It was pretty hard to find any relevant information on the internet, however eventually I figured out how to do it. Originally this article was written for Trixbox, however the same configuration applies to FreePBX (with minor differences in steps due to the UI differences).
Please note that the following configuration reflects a Trixbox/FreePBX PBX configured with phones with extensions of 1XX and the Cisco Unified Call Manager configured with extensions of 3XX.
If you are simply using CUCM for Cisco IP Phone handset connectivity, you don’t even need CUCM anymore, you can simply use the commercial “EndPoint Manager” on FreePBX to handle Cisco IP Phone connectivity to FreePBX (includes the Cisco 7961 phone’s I use).
Trixbox/FreePBX Configuration
Create an SIP Trunk (Leave settings default unless otherwise specified below)
Outgoing Settings
Trunk Name: CallManager
Peer Details:
type=friend
qualify=yes
nat=no
insecure=very
host=ip.address.of.CUCM
fromdomain=ip.address.of.CUCM
dtmf=rfc2833
disallow=all
context=from-internal
canreinvite=no
allow=ulaw
Incoming Settings
USER Context: ip.address.of.CUCM
USER Details:
type=friend
qualify=yes
nat=no
insecure=very
host= ip.address.of.CUCM
fromdomain= ip.address.of.CUCM
dtmf=rfc2833
disallow=all
context=from-internal
canreinvite=no
allow=ulaw
Create an Outbound Route to route calls made to 3XX to the Cisco Call Manager
Create outbound route “Cisco”. Check the “Intra Company Route”, and inside of the Dial Patterns type in 3XX. Under Trunk Sequence select “CallManager”.
This pretty much sums up the amount of configuration required on the Trixbox/FreePBX side of things. Now onto the Cisco stuff.
Cisco Unified Call Manager Configuration
Create an SIP Trunk
Device -> Trunk -> Add New
Trunk Type: SIP Trunk
Device Protocol: SIP
Device Name: TrixboxPBX
Call Classification: OnNet
Check the “Media Termination Point Required” checkbox (this is to handle transfers, hold music, etc…)
Check “Remote-Party-Id”
Check “Asserted-Identity”
SIP Information
Destination Address: IP.address.of.trixboxfreepbx
Uncheck “Destination Address is an SRV”
Destination Port: 5060
MTP Preferred Originating Code: 711ulaw
SIP Trunk Security Profile: Non-Secure SIP Trunk Profile
Change the “Non-Secure SIP Trunk Profile” security profile from TCP to UDP
System -> Security Profile -> SIP Trunk Security Profile
Hit the “Find” button
Select “Non Secure SIP Trunk Profile”
Incoming Transport Type: TCP+UDP
Outgoing Transport Type: UDP
Uncheck “Enable Digest Authentication”
Incoming Port: 5060
Out of the last 6 checkboxes, all should be checked except the First and Last.
Create a Route Pattern to route calls from the Cisco Call Manager to Trixbox
Call Routing -> Route/Hunt -> Route Pattern
Create New
Route Pattern: 1XX
Gateway/Route List: TrixboxPBX
Route Option: Route this pattern
Call Classification: OnNet
Enable Required Services on CUCM
I’m not too sure which ones are actually required, however the below configuration works great. To get to the CUCM services go to the “Cisco Unified Serviceability” section (Top right of web interface).
Enable Services
Tools -> Serviceability
Enable the following:
CM Services
Cisco CallManager
Cisco Tftp
Cisco Messaging Interface
Cisco Unified Mobile Voice Access Service
Cisco IP Voice Media Streaming App
CTI Services
Cisco CallManager Attendant Console Server
Cisco IP Manager Assistant
Cisco WebDialer Web Service
Select “Save”, afterwards select “Set to Default”. Please note that it may take some time to bring the services up.
It’s always a good idea to restart both the Trixbox/FreePBX PBX and the CUCM PBX.
After you have configured the above, configure phones in the 1XX range for the trixbox, configure phones on the CUCM for the 3XX range and they should be able to call each other. Please remember that if you have a PSTN line on your Trixbox or FreePBX you will need to create another route pattern for how to transfer 9XXXXXXXXXX from your CUCM -> Trixbox, then configure the applicable route in Trixbox -> PSTN.
Feedback is welcome, leave a comment!
Excellent explanation, as you said that there is almost none information about this configuration, i´ve been testing different confs and the only one that works is your conf, but i have a problem because it works only in the way CCM-> Tribox and in the opossite way it doesnt and i hear the record “All ciruits are busy now please try your call again later”.
Well i´ll try to find the problem and i´ll send my feedback.
Thanks a lot for share your info
I have to apologize for the late feedback! For some reason I didn’t receive an e-mail notification on your comment…
I’m not too sure if you have this checked/enabled, but inside of the web interface, under “PBX Settings”, then “General Settings” look for the “Allow anonymous inbound connections” and set it to yes.
Let me know if this works, if not let me know and I’ll do some more research in to it!
Thanks for the comment by the way!
(P.S. Make sure you check you inbound routes, etc… so the calls get come into the Trixbox and get terminated on an extension!)
My friend, these are the best set of instructions so far.. I am so close, i can taste it..
I am a Noob.. and am coming at this blind…
The trunk between the CCM and Trixbox
I can’t however get the calls to go between the Trixbox and the CCM
I already have the CCM setup with extentiosn using the 65xx format
On Trixbox i’d like to mirror the extentions OR have 165xx as the extention number..
Basically our CCM goes out of an ISDN30, and what i’d like users to be able to do, is if the user is at home, have them connect to the Trixbox, dial 7 to divert outgoing calls out via the isdn30, or dial 8 to divert outgoing calls via a sipgateway (lets say for the sake of example sipgate.co.uk) the users should also be able to dial the 65XX extentions as well and get to people in the office..
The main reason for the use of Trixbox, is we use a product called Trusted Client for home users, which is Ubuntu based, and i can use X-Lite simply make calls over our VPN.. It also provides me with a backup phone system if/when the ISDN30 goes down, which it has 4 times this year..
Any assistance, help, direction will be taken on board.. I’m still earning,but nearly there.
Well I’m glad to hear I can help!
Let me check out my config and play around with a few things and get back to you.
So I just want to confirm, can you make internal calls from CCM -> Trixbox, and Trixbox -> CCM? Or is it just Trixbox -> CCM -> PSTN that isn’t working?
You could probably get all extensions (Trixbox and CCM) going on 65XX, however one would have to be 6500-6550 and the other would have to be 6551 to 6599.
Also, tell me if you have “Allow anonymous calls” setup on your Trixbox general config enabled. This could be it. If not, try it out, it might work.
If you do have one way calling from either the CCM or Trixbox to the other, let me know.
Stephen
Hi there, since posting yesterday, i setup the dial plan correctly, and am able to make calls out of the Trixbox to the big wide world VIA the Cisco CCUM..
I’d like to make CCM -> Trixbox, and Trixbox -> CCM?
Tried “Allow anonymous calls” both settings, no change..
Oh, well I’m glad to hear.
I guess if you had the CUCM handling the PSTN the dialplan would be different. In my scenario the Trixbox handled PSTN termination.
Glad to hear you got it going though 🙂
Stephen
I followed your instructions and it’s working CCM -> Trixbox but not the other way around. I enabled “Allow anonymous calls” but it’s not working yet. Do you have any further suggestions?
Thanks in advanced….
BTW, great tutorial!!!!!!
Did you configure your outbound dial plans on Trixbox?
Stephen
be sure that you select Calling Search Space that match to your phone in Inbound Calls part of
Trunk Configuration in CUCM
This solves “all circuits are busy” problem for me.
btw great tutorial
Hi Jepes,
Thanks!
jepes could you help me with the Calling Search Space, what is the right config at this?
Thanks
Ronald,
I’ve noticed no one’s replied. I checked my “Calling Search Space” to see what my config is, however I have nothing configured inside of there.
Mind explaining your exact issue? I’ll try to help out if I can.
Stephen
I managed to make a call from a my mobile phone through a PRI connect to cisco5300 to an extension on the trixbox. Thanks very much for the info. It was not easy to find this .
Hi
Do you have the command line cli instruction for cme to connect to asterisk using sip. For example what command enables Media Termination Point Required using ios?
Can you forward me your copy of your config? Thanks again.
Hi Mike,
I’m sorry but I did all the above configuration in the web interface. I would have absolutely no idea how to do this using the cli.
Sorry about that. Can anyone else help Mike out?
Hi Stephen,
What is extensions.conf setting? I have followed every steps but it doesn’t work for me 🙁
BTW you did a great work 🙂
Thanks ShUaib,
Are you referring to the extensions.conf on the trixbox side? I actually don’t think I modified any .conf files directly. All configuration was done inside of the Web interface for trixbox itself.
Make sure you check your outbound and inbound routes to make sure they are routing the calls properly. Also, have you verified the boxes are even talking to eachother?
Stephen
Hi Stephen absolutely fabulous work!!! Whilst i have got everything to work. The only thing im struggling with is calls from the PSTN>Trixbox>CUCM. Calls from the PSTN work fine to TrixBox but i just can’t figure out how to get TrixBox to forward calls to an extension in CUCM. Any ideas?
Hey Shawn,
Glad I could help! Just curious, can you call from Trixbox extensions to your CUCM extensions? If that’s the case, everything is working and you just need to setup your ring groups, etc… For example, when someone calls my general in line, it rings 3 extensions on the tribox, and 3 extensions on the CUCM. Whoever picks up when it rings takes the call…
Now if your actually having trouble calling the CUCM extensions from a Trixbox extension, that means your having some difficulties with communication (ie, theres a bug in the config).
Let me know if this works, or your need anything else.
Thanks,
Stephen
TrixBox can talk to CUCM fine. I just adjusted the dial plan to forward to the extensions im using in CUCM. Problem is im slightly confused as to “how” i tell CUCM to receive calls from the PSTN via TrixBox. As at the moment they’re terminating at TrixBox because i have an extension there set to receive calls from the PSTN. Iv added the correct dial pattern on TrixBox for calls from the PSTN. How do i do the same thing you have done with your 3 extensions? As that’s exactly what id like to do. Albeit for just two extensions.
Actually i need it to go to a specific extension on the CUCM as i have an Exchange UM providing the voicemail. Dammit im so close but i just can’t figure it out?!!
Sorry I forgot to mention this before…
With the config I have put online, and in my specific config, my Trixbox extensions are 1XX while my CUCM extensions are 3XXX.
In order for the calls to terminate on the CUCM side, you need to setup an outbound route for trixbox that routes all calls made to 3XX and make it route through the route configured for the outbound calls.
As for the voicemail stuff, I know absolutely NOTHING about… If you got it going, then good, once you create the outbound routes on trixbox, it should handle it properly.
Iv already done that. TrixBox can talk to CUCM with no problems. PSTN calls terminate at the TrixBox and are not forwarding to the CUCM. I need to find a way to tell TrixBox to forward a dial pattern (for PSTN calls 0XXXXXXXXXX) to CUCM over the TrixBox > CUCM trunk. Almost like a dual fork similar to your setup although id prefer if the TrixBox just acted as a proxy and calls from the PSTN were just forwarded to CUCM.
Essentially what i have is PSTNSipgate>TrixBox>CUCM. Outbound is fine. Inbound is the issue.
Il figure it out eventually. Thanks for your help!
As for forwarding of all calls, I’m not sure how it would be done…
What I do, is I configured a Ring Group, and in the list, i listed all my 1XX extensions and my 3XX extensions.
Make sure on your “Outbound Route” that you have the “Intra Company Route” box checked.
Here’s something. If you create an Inbound trunk from the PSTN and have it go to a Ring Group. Make sure that any extensions that belong to your CUCM box, have a # behind them. This is needed since it’s not an extension on the system…
IE on my RingGroup I created for my PSTN line, here are the extension list:
101
102
103
301#
302#
303#
All my Cisco extensions have a # following.
Excellent that’s worked. Although the ring group doesn’t allow the call to terminate at CUCM. If only you could create an extension with the # on the end in TrixBox. As that would allow the call to go to voice mail at the CUCM which is ultimately what i need for the Exchange UM (voice mail) side in order to dial from the PSTN straight into the Exchange UM. Then again the UM works fine with VOIP calls. That said iv got calls to the PSTN Sipgate TripBox MS Exchange UM and CUPs all working and converged in a lab. Thanks in part to yourself. When im in Calgary next il buy you a beer!!
Sounds good, glad I could help…
So it’s still not terminating on CUCM? As in, when you receive a call from the PSTN, it goes through trixbox, then to CUCM, but you can’t pick it up on a Cisco Extension? That should be working… I wonder why it’s not…
Update. Got it to work! I put only one CUCM extension in the Hunt Group on TrixBox left the ring strateg at first available and destination if no answer set to voicemail on the TrixBox. This allows the call to terminate at the CUCM which in turn goes down the SIP trunk on CUCM to the Exchange UM.
So now i have calls from the PSTN via Sipgate > TrixBox > CUCM > Exchange UM and CUPs integrated to boot.
Hello Dear
thanks for your good instruction
actually i do all you said and everything goes fine , i can call between Trixbox and CUCM
there is only on problem, if i call the DN on CUCM and want to forward call to another DN, the calls wait on Hold and not forward, do you have any idea??
I’m not too sure if this will help or not. But try logging on to CUCM administration and start the “Media Service” Service. (I think that’s what it’s called).
I remember I had some issues with holds, transfers, and 3-ways, and this resolved it. Let me know if this works for you…
Dear Stephen
actually i can not find the service name “Media Service” on my CUCM Serviceability page
do you know where is it?
actually i change the sip profile to use RFC 2543 Hold but it still not work
Hello,
Sorry about that, I was trying to recall it from memory and I was incorrect on that one.
Here are the services that should be on:
Cisco CallManager
Cisco Tftp
Cisco Messaging Interface
Cisco Unified Mobile Voice Access Services
Cisco IP Voice Media Streaming App
Cisco CallManager Attendant Console Server
Cisco IP Manager Assistant
Cisco WebDialer Web Service
Cisco CallManager SNMP Service
On top of all that, what I’m saying below might actually resolve your issue. I beleive you have to have a “Media Termination Point” configured…
On the Cisco Unified CM Administration interface, drop down the “Media Resources” menu and select Media Termination Point…
Create a Media Termination point. After configuring this, test it. If it doesn’t work, you may have to associate it with a device pool.
Dear Stephen
actually i found t he problem
the problem is , Trixbox use its default feature codes on transfering calls or holding thems
and cisco ip-phones do the features in the way they like (by assigning the seprate button for each of them)
so trixbox can not understand the codes which cisco says
now i’m trying to change the codes to custom ones
do have any idea about that
atrin
Hi Atrin,
I’m not sure I know what your talking about… For example all features work on my setup on both sides.
Let’s pretend here on my side… Trixbox extensions are 1XX and Cisco are 3XX. I can hit the forward button on a call on my cisco, and forward/transfer the call to a trixbox extension. This also works vice-verse, and with outbound calls.
Also, my hold music works. If a call is on the Cisco phones, and hold is pressed, it plays the Cisco hold music, whereas if someone on a Trixbox extension presses hold, it will play the hold music on the Trixbox server.
Unless I’m misunderstanding, you shouldn’t have to play with any codes whatsoever…
Dear Stephen
thank you fro your kind
i found my problem
i forgot to check the “Media Termination Point Required” on my trunk
Thank you very much! It’s very detail and accurate. Everything works well.
Hi, everyone.
This article has been a great help for me configuring Trixbox SIP Trunk and CUCM 6.1.1. As of now, I have the following configuration VoIP Trunk Trixbox CUCM. At first I only had accomplish to make calls from CUCM to VoIP Trunk (International calls) using Trixbox as a proxy. But, whenever some rang back on our DID the calls didn’t come thru CUCM. I read all the comments in this article and realize I had to configure the Ring Groups and make outbound route to CUCM. Now, I’m able to make calls from Trixbox to CUCM, from CUCM to Trixbox, from CUCM to VoIP Trunk and from VoIP Trunk to CUCM.
However, when a call comes from VoIP to an extension in CUCM (let’s say the operator) and you try to transfer that call, you could hear the Hold On Music and the call is transferred to new extension, but neither of the extensions hear a thing.
Anyone of you have a clue around this issue?
Thanks in advance.
Hi Nery,
I’m not sure what you mean by the problem your having. So does it not transfer the call? Or what is working/not working?
Well, when the call passes to the CUCM extension and this extension transfers it to another one, this doesn’t work. The person who originated the call keeps listening to the Music On Hold and the other party doesn’t hear a thing.
Let me try to explain it in another way. Someone calls 1XXX-XXX-XXXX (which is my DID), Trixbox answers the call and redirect it to an extension in CUCM (configured with Ring Groups). Let’s say is extension 1010, the person in this extension picks up the headset and talks to the original party, so far so good.
Then the original party asks to transfer his/her call to, let’s say, extension 1090. In this point, the original party starts listening to a background sound (Music on Hold), transfers the call and the person in extension 1090 picks up the phone and cannot hear a thing, meanwhile, the original caller is still listening to the music on hold.
I don’t know if this is clear now, if not; tell me that I will try to clarify it.
Thanks.
I did the same configuration but..
the calls callManager –> trixbox Ok
trixbox –>callManager doesn’t work
what’s the problem???
I’m using X-Lite softphone in Trixbox, is it relevant?
help me please!!
Hi Stephen,
Thanks for your great work. Could I ask you a simple questions?
1) From Asterisk ext: xxxx-xxxx (8100-1000) call to ext behind Cisco (8200-1000), could we pass this ext number from the original call from Asterisk to shown in IP-Phone of Cisco (ext:8200-1000) and vice versa?
2) Do you have any solution for questions from Nery No.38 in this website?
3) my experience is the same problem with Nery but for conference call, from cisco side if the initiate the conference call, they call always disconnect to ext behind Asterisk.
thanks
Harto
Hi Harto,
1. The caller Id should work. You just have to make sure you have caller ID enabled on your Asterisk box. Also, when configuring CUCM, make sure that any “pass caller ID” options are enabled. This works on my setup, and I don’t think I even had to do anything to get it to work, I do vaguely remember a checkbox on the CUCM configuration regarding passing caller ID, which I enabled.
2. As for comment 38, I believe this has to do with the “Media Termination Point” service not being enabled, I could be wrong though. It might be worthwhile to make sure the media termination point service is enabled, and also configured inside of CUCM.
3. Just make sure you have the Media Termination service on, and that you configure media termination.
I had a lot of issues when I first started playing with this stuff with hold music, conference calls, etc… Handoffs for certain VoIP actions weren’t being done correctly until I got the media termination service and media termination point configured, now hand-offs of all different types of VoIP services function perfectly.
thanks a lot Stephen,
COuld I have your personal email address ?
thanks
Hi Stephen,
I have another questions,
Do you ever try to get all address books from Cisco Call manager to Asterisk LDAP ?
If yes, by manually or automatically (I have doubt we can do it).
I this scenario, we are using Asterisk to connect to Cisco Call Manager. and all The latest updated Address books is in Cisco Call Manager
Many thanks for your kindly advise.
Harto
Hi Harto,
I have to apologize, but I’ve never played with address books. I actually don’t even run CUCM in production nor use it regularly. I’ve only done work for clients connecting CUCM to Asterisk, and that’s about it.
Sorry,
Stephen
Lets assume that CUCM in your local network and it has local ip and asterisk has public ip ,so asterisk will send some calls to CUCM local phones .
Can CUCM register to Asterisk using by username and password on created sip trunk ?
Hi Jeroen,
I’m pretty sure there is a way to do this using a user/pass authentication method, but I’m not exactly sure how. Maybe someone else who reads the blog will be able to comment and suggest a method?
If not, worst case scenario is you could configure it the way listed above. Then instead of leaving it unsecured, you could configure a firewall rule to only allow communication to/from your CUCM box specifically only to the IP address of the asterisk box. Since the IP addresses are already defined inside of Asterisk inside of the trunk, the Asterisk box as far as I know wouldn’t be a security concern.
Hi Stephen,
My call flow is that Asterisk send calls to CUCM and CUCM route these calls to cisco ip phones,so not bidirectional.
I have read CUCM documents and note that there is no registeration provided to sip trunks by CUCM, I need registraion because of the CUCM is behind firewall. But we can match asterisk with route-pattern as IP address .
If you create a SIP trunk that you expected match incoming calls which are sent by Asterisk, you should also create a sip route pattern to match asterisk’s IP and then create calling search space.
I tried to just create a sip trunk and route-pattern (for local registered cisco phone) it was not worked out.
I also read your first post and note that outgoing trunk settings on Asterisk is configured without send auth. or any other that is the configuration which I need at asterisk side.
My first attempt was successfull, I configured a x-lite softphone as a sip gw and could sent calls to CUCM’s internal phones.Next step is that sending calls from asterisk but I m not sure firewall can handle this nat.
I’m still working on this configuration and share with yo when it is done.
This has been great, but I took it a little further and using Google Voice I have setup a trunk to the PBX to it. The calls outbound from my CUCM works great, calls in however, come in and can be answered but the outside sender never stops ringing, while the IP phone thinks he has the call.
I have setup the media termination point and I am starting to wonder if it is a NAT issue?
Any suggestions would be welcomed!
HELP…ONE WAY COMMUNICATION from CUCM EXTENSIONS TO ASTERISK BUT NOT REVERSE. What could be wrong? .251 is Voice Gateway and .240 is CUCM.
eb 8 19:48:29 NOTICE[71138224]: sched.c:221 sched_settime: Request to schedule in the past?!?!
== CDR updated on SIP/10.10.2.250-e027
— Executing SetGroup(“SIP/10.10.2.250-e027”, “sip-kenbo”) in new stack
— Executing CheckGroup(“SIP/10.10.2.250-e027”, “10”) in new stack
— Executing Wait(“SIP/10.10.2.250-e027”, “1”) in new stack
— Executing Playback(“SIP/10.10.2.250-e027”, “pls-hold-while-try”) in new stack
— Playing ‘pls-hold-while-try’ (language ‘en’)
— Executing Dial(“SIP/10.10.2.250-e027”, “SIP/[email protected]”) in new stack
— Called [email protected]
— Got SIP response 400 “Bad Request – ‘Malformed CC-Diversion/Diversion/CC-Redirect Header'” back from 10.128.11.240
== No one is available to answer at this time
— Executing Playback(“SIP/10.10.2.250-e027”, “an-error-has-occured”) in new stack
— Playing ‘an-error-has-occured’ (language ‘en’)
— Executing Playback(“SIP/10.10.2.250-e027”, “pls-try-call-later”) in new stack
— Playing ‘pls-try-call-later’ (language ‘en’)
— Executing Hangup(“SIP/10.10.2.250-e027”, “”) in new stack
== Spawn extension (mainmenu, 61160, 8) exited non-zero on ‘SIP/10.10.2.250-e027’
Hi Pamiller,
Did you configure the CUCM to make outbound SIP over UDP? (The default on CUCM is SIP over TCP).
I’d also double check all your SIP settings on your CUCM. I think it’s not liking your asterisk for some odd reason…
Hope this helps! Let me know how you make out!
Stephen
j’ai un problème:
j’ai fait une trunk pour les appels vers le GSM dans le fichier sip.conf
[appels-gsm]
type=friend
secret=azerty
context=appels-entrants
host=dynamic
disallow=all
allow=ulaw
nat=yes
qualify=yes
mais quand quelqu’un m’appelai depuis son phone 3cx l’appel n’arrive pas
Hi Mariem,
I have to apologize, but I don’t speak french. I tried using a translator, but I couldn’t really make out what your question is.
Would you be able to post it in English? Is there a chance someone may be able to translate it for you?
Stephen
ALL CIRCUITS BUSY
Thanks Stephen for the tutorial. I did have the All Circuits Busy issue and I followed Jepes response regarding changing the Calling Search Space in the Inbound calling section for the Sip Trunk on CUCM. I updated it to match the calling search space on my phone line registered on CUCM.
This configuration worked great. With an internal trunk setup this way. Is it possible to due an “asterisk phonebook” lookup on the number that comes from the Call Manager? I tested this with an external call going through the call manager and then over to the trixbox and was hoping there was a way to get the trixbox to peform a phonebook lookup on the call.
Since this is a trunk I don’t believe it comes into an “inbound route” to perform the phonebook lookup.
Thanks,.
Thanks for the post. I have a question. How can we configure a sip trunk between Cisco Call manager and Trixbox if CUCM is on private ip and Trixbox is on public ip.
Is it just simply NATing which needs to be done at the CUCM side ? or is it something more ?
Do i need to set up port forwarding at the CUCM side ?
Also can i have Asterisk to be used as a NATing device ?
So setup might look like this
CUCM ( private ip ) – > Asterisk ( private ip ) – Asterisk ( public ip ) – Trix box public ip ?
Will this work ?
Pleaes advice
Hey Stephen,
Thanks a ton, you saved me quite a bit of time in getting calls working from CUCM 8.6.2 –> AsteriskNOW.
Cheers
Taran
Hi,
I have followed this setup, and got the connection to work, except Trixbox –> CUCM is a fast busy tone. CUCM –> Trix box works fine. I have check & recheck CSS for incoming calls, and I am not have any luck. Suggestions?
Hi Andrew,
Somewhere inside of my post I mentioned configuring the SIP protocal on the CUCM side. Is there a chance that your CUCM is listening on TCP for SIP, instead of UDP for SIP? This could be the problem.
If not, it might have to do with the route configuration on CUCM.
Thanks so much for sharing this information.
Hello All
Do you know configuration of CME 7.1 for same setup?
Thanks
Dear All,
I have Elastix 2.4 and I need to create a sip trunk with cisco call manger 8.6 and this is the trunk configuration.
when I dial the extension number I get error message all circuits are busy now.
Can you help me please in this issue.
Outgoing Settings
PEER Details:
type=friend
qualify=yes
nat=no
insecure=very
host=10.12.23.33
fromdomain=10.12.23.33
dtmf=rfc2833
disallow=all
context=from-internal
canreinvite=no
allow=ulaw
and
Incoming Settings
type=friend
qualify=yes
nat=no
insecure=very
host=10.12.23.33
fromdomain=10.12.23.33
dtmf=rfc2833
disallow=all
context=from-internal
canreinvite=no
allow=ulaw
Super helpful thank you! I’ve been fighting with CUCM and CUC for the past 2 weeks!
Fantastic Post! Congratulations! No have documentation about this theme in the web. The way of u explained it´s so easy! Tnk´s a lot! Nice job!
I have tried this configuration and it looks like if everything worked ok, wondering if anyone has had any issue like this.
I was able to successfully configure CUCME and Asterix
all extensions on CUCME call each other ok, all extension on Asterix call each other ok
all CUCME call Asterix and vice versa with 2 way audio
CUCME can make outbound calls to land and mobiles with 2 way audio
but Asterix makes outbound calls to mobile and land line the only problem I have is with Asterix extensions the audio is only one way ( I can hear them but they cannot hear me) On Asterix log / Report it say No Answer . Meaning Asterix doesn’t recognise that call was even answered thereby not sending audio to the receiver. How can i solve this. Please can anyone help.
Hi Kenneth,
Quick question, are the trunks to PSTN configured on your Asterisk PBX?
And just to confirm, CUCM is working fine, it’s only asterisk based outbound calls that are having the one way audio issues?
Stephen
Also Kenneth,
Is your Asterisk behind a NAT firewall?
I’m trying to set this up on CUCM v8.6.2 with TrixBox. Is there anyone who has done this who can help me out?
hi, guys I do have the same issue, I trying to connect my Cisco Callmanager express 4.1 to trixbox system, the communication from a trixbox to cisco Callmanager express is working but from the callmanager express to trixbox I am getting the message all line are busy please try later, can somebody help me to make it work? It has been a while I am trying to fix this! I will really Appreciate your help.
Hi there,
I know this is an old post but it’s the best I”ve found for setting this up. I have the two systems talking perfectly, however what I’d like is to be able to call my Asterisk box and route the calls to my CUCM. Is this possible and if so how would I set this up?
Thanks,
Troy
Hi Troy,
Glad to hear the post helped!
Are you referring to a PSTN call coming in to the Asterisk box? If so, that’s exactly what I did in this guide! If you have extensions on both the Asterisk and Cisco side, and they can call each other and it works, then most of the work is done.
All you need to do is configure the inbound routes. In my case, I have a default inbound route, I setup a ring group, and then I added each extension (extensions on the asterisk side, as well as extensions on the CUCM side). I had to suffix the CUCM extensions with a # (example 123#) since they are on a remote system (uses the trunk to the CUCM).
Also, one other thing, is that since the Asterisk box is routing the calls, you may need to find a setting on the CUCM side. Make sure “Media Termination Point Required” is enabled.
Let me know how you make out, or if you need any more assitance!
Cheers
Hi
Can u tell me network connection scenario and ip address of the pbx u have used?
Hello Qasim,
In my scenario I used a local single subnet, sized as a /24. No routing occurs between the PBX or the Cisco CUCM instance in my case.
Please keep in mind that this is just because of my network layout. Depending on your situation you may be more complicated, in which case you may be using multiple subnets, require routing, and/or have your SIP protocol configured for NAT traversal (and firewalls).
Cheers,
Stephen
I have checked everyting and cucm is integrated with Asterisk and call are working fine.But we are facing issue issue with callerid.
When calling from Cisco to Asterisk ,Cisco phones are not showing the asterisk extension name on Cisco phone’s display.
Asterisk team confirmed that ,they are getting calls from cisco as anonymous.
Any suggestion pls
Hi Anees,
The Called ID function won’t go backwards. If you’re calling from a Cisco phone, to Asterisk, on the Cisco phone it will simply show the number you dialed (possibly the shortcut name if you configured the shortcut buttons).
As for the Asterisk side. The receiving Asterisk extensions should be showing the CallerID of the Cisco phone that is calling them.
I don’t have access to this environment to check anymore, but I would check your Asterisk settings to allow the incomming CallerID, but I’d also check your Cisco CUCM settings to make sure that CallerID is passed on outbound calls over the truck.
Make sure the context on the Asterisk side for the trunk is “from-internal”, and the Call classification on the CUCM is OnNet on the trunk and Route Pattern.
Cheers,
Stephen
What about configure only outgoing call from asterisk to CUCM?
Do i need to configure trunk on cucm ? or only on asterisk ?