09-28-2016 01:38 PM - edited 03-18-2019 12:08 PM
Hi guys,
Disclaimer - all I know about VoIP I've learned from Google and poking around in my CUCM 10.5. That said, I'm stuck, and I need some help.
I'm attempting to set up CFA to a user's external number (cell phone). I had it working and then I broke it somehow, but I can't figure out what I did to break it.
What happens now is this: Phone A calls Phone B. Phone B forwards to cell phone. Cell phone rings, but when answered, it beeps three times and disconnects. Phone A is still ringing and the cell phone will ring a second time, but when answered, same results. Then Phone A gets the busy signal.
What I have done so far:
In the page for the DN, I have set the Forward All destination to the mobile number (8AAABBBCCCC). I can dial this number Phone A and Phone B, and it works. The CSS is set to "CF_External_CSS". That CSS has a single partition selected, "Local_PT". This partition is used for local 7 and 10 digit dialing. Only 10 digit dialing is relevant for this problem. There are two Route Patterns associated with this partition, one for 7 digit and the other for 10 digit. When I was modifying the 10 digit pattern, all that I changed was deselect "Use Calling Party's External Phone Number Mask". I saved it and then the CFA stopped working. After clicking on Save, a notification pops up to say "The Authorization Code will not be activated. Press OK if you want to proceed...etc." That should be fine, we don't use FACs. A second notification pops up that says "Any update to this Route Pattern automatically resets the associated gateway or Route List." I clicked OK, and it saved. So since I didn't really change anything significant in the Route Pattern, I'm assuming this reset of the gateway or Route List could have also caused it to break. Only I don't know if that's correct, and if it is, why.
If anyone has any idea what I have done and can help me fix this, I would greatly appreciate it.
Thanks to any respondents!
Steve
Solved! Go to Solution.
09-29-2016 12:52 PM
Try adding this line under 'dial-peer voice 999 voip':
voice-class codec 1
Also, go to the trunk configuration in CUCM and note the name of the SIP Profile at the bottom of the page. Then go to Device > Device Settings > SIP Profile and find the one that was configured on the trunk. Find the "SIP Rel1XX Options" setting and change it from Disabled to Send PRACK if 1xx contains SDP. Save and Reset. Note that this will drop all active calls to/from PSTN, so you might want to do it after hours.
09-28-2016 05:29 PM
What is the protocol between CUCM and the PSTN gateway, and then to service provider? For example, CUCM ---> SIP ---> VG ---> SIP ---> ITSP
09-28-2016 07:03 PM
I'm afraid I don't know enough to answer that confidently. Can you direct me to figure it out?
09-29-2016 06:17 AM
On the Route Pattern in question, see what you have in "Gateway/Route List". It could be a Route List containing a Route Group containing a gateway, or it could point to a gateway directly. Your gateway will be at either Device > Gateway or Device > Trunk. You should be able to see if it's SIP, H323, or MGCP gateway then.
09-29-2016 07:27 AM
Thanks for the explanation. I was able to follow those steps easily.
The route pattern shows a route list "PSTN_PRI_RL"
That route list has one group called "Standard Local Route Group". If I'm correct, this means the Device Pool defines which route group will be used, in which case it is HQ_PSTN_PRI_RG (found in the Device Pool Configuration". That is the only route group in my system. In that route group, there is one selected member called "HQ_VG01_PSTN_PRI_1_ST (All Ports)". That's my SIP trunk.
09-29-2016 07:40 AM
Try to make a test call (forwarded) and collect these debugs on the gateway:
debug ccsip messages
debug isdn q931
Attach these debugs along with a sanitized 'show run' from the router.
Also, have you tried enabling back "External Phone Number Mask" on the route pattern?
09-29-2016 08:07 AM
09-29-2016 11:14 AM
From the call flow it seems like you dialed the external (DID) number of the forwarded phone from an inside phone. It goes out to PSTN and PSTN sends it back inside to the forwarded phone, which then forwards it to the cell phone out to PSTN again.
When you called from Phone A (internal) to Phone B (internal, forwarded to cell phone) did you dial 4 digits or 10 (with the 8 access code)?
09-29-2016 11:24 AM
Exactly right. I dialed the 10 digit number with the 8 access code from internal Phone A. Internal Phone B forwarded out to my cell phone.
09-29-2016 11:33 AM
Does it work if you dial from Phone A to Phone B using 4 digits? You should test 2 scenarios and see if it works:
Scenario 1. Phone A (internal) dials Phone B (forwarded) using internal dialing of 4 digits.
Scenario 2. An outside PSTN phone dials the DID of the forwarded phone.
09-29-2016 11:36 AM
Scenario 1 works, dialing from internal Phone A with 4 digits.
Scenario 2 does not work, dialing in from an external phone.
09-29-2016 11:50 AM
Please collect the same debugs for the non-working scenario.
09-29-2016 12:30 PM
09-29-2016 12:52 PM
Try adding this line under 'dial-peer voice 999 voip':
voice-class codec 1
Also, go to the trunk configuration in CUCM and note the name of the SIP Profile at the bottom of the page. Then go to Device > Device Settings > SIP Profile and find the one that was configured on the trunk. Find the "SIP Rel1XX Options" setting and change it from Disabled to Send PRACK if 1xx contains SDP. Save and Reset. Note that this will drop all active calls to/from PSTN, so you might want to do it after hours.
09-29-2016 01:58 PM
Thank you! Adding the codec line fixed it. Would you advise I complete the rest of the steps you included?
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide