Setting up follow me ... problems ...

[mrdom]

Hi everybody!

I'm setting up an extension for a new user, and he'd like the system to ring the extension, then forward to his cell phone. I tried to accomplish this using the "Follow Me" settings. The user's extension is in the box, along with their cell number. I do have the # sign after the number. When I try to initiate the call to the cell phone, I'm getting this error in the log:

[pre]
[2023-11-27 15:16:41.476] VERBOSE[801517][C-0000005a] pbx.c: Executing [s@macro-dialout-trunk:23] Dial("Local/16081234567@from-internal-0000001c;2", "SIP/SIP19256/16081234567,300,t") in new stack
[2023-11-27 15:16:41.477] VERBOSE[801517][C-0000005a] netsock2.c: Using SIP RTP TOS bits 184
[2023-11-27 15:16:41.477] VERBOSE[801517][C-0000005a] netsock2.c: Using SIP RTP CoS mark 5
[2023-11-27 15:16:41.480] VERBOSE[801517][C-0000005a] app_dial.c: Called SIP/SIP19256/16081234567
[2023-11-27 15:16:41.603] VERBOSE[1512][C-0000005a] chan_sip.c: Got SIP response 604 "Does not exist anywhere" back from 216.86.44.107:5060
[2023-11-27 15:16:41.604] VERBOSE[801517][C-0000005a] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2023-11-27 15:16:41.604] VERBOSE[801517][C-0000005a] pbx.c: Executing [s@macro-dialout-trunk:24] NoOp("Local/16081234567@from-internal-0000001c;2", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 1") in new stack
[/pre]

We have four channels of our main SIP line, so there's definitely available channels to place the call. The call came in on one channel - is the SIP provider not allowing me a second channel to allow the system to dial the cell phone?

We also have a second SIP number with only one channel and a specific prefix is needed to use it. I tried to have the system make the call on that channel, and it worked. However, I'm not sure how to set the destination as the cell phone rang, but then the system came back on with a busy signal (that's what I have the destination set to). If he doesn't answer at his desk, I want the system to send the call to the cell phone and then drop off. Can I do that? I did consult the emetrotel guide, but I didn't find any helpful info. in there.

Could someone lend a hand to help me set this up? Thanks everyone!

-Michael

 

[ucxguy]

Based on the log, the SIP trunk provider rejects the call with the message
604 "Does not exist anywhere"
You will need to contact the SIP trunk provider to determine why they reject the call to the phone number 16081234567.

Once the external call is dialed successfully, you should not be getting the immediate busy after the extension dial timeout.

 

[mrdom]

Thanks for the help ucxguy. I'll work it out with the SIP provider.

I was able to set up a miscellaneous destination, and have the system transfer the call that way. That seemed to work, so we'll go with that for right now.

Many thanks!

 

[mrdom]

Hi everyone:

I did receive a note back from my sip provider (Fusion/MegaPath), and they provided a solution, but I'm not quite sure what I need to change in order to do what they are asking. I'm assuming it's something I have to change in the trunk settings, but am not sure. Could someone help decipher for me:

[pre]

Typically, a redirect/diverted call into MegaPaths network will be forbidden if the required SIP headers are not included.
Diverted calls originating from a customers PBX should be configured to send a reINVITE back to MegaPath and must include:
Diversion SIP Header
Diversion TN is the DID that diverted the call (customer number that received the call first) in your case its the 6081234567

The Diversion Header is required whenever there's a change to the initial destination request.
This is also known as a deflection or redirection.

For example; if a call is placed from an off-net number 703-555-1000 to a destination DID 703-
555-2000 and the DID has a forward set to send calls to off-net number 703-555-3000, the redirect call flow will change both the Request URI and To headers to the final destination number of 703-555-3000.

Diversion Example:

INVITE sip:7035553000@megapathvoice.com:5060 SIP/2.0

From: <sip:7035551000@megapathvoice.com:5060;transport=udp>;tag=SDiltac01-3bb1

To: sip:7035553000@megapathvoice.com:5060

Diversion:
<sip:7035552000@megapathvoice.com>;reason=unconditional;counter=1;privacy=off;screen=yes

Contact: sip:7035551000@172.25.222.2:5060;transport=UDP

The Diversion Header will outline who diverted the call, why the call was diverted, the total number of diversions, as well as a privacy and screen indicator flag.
By default, MegaPaths Broadsoft trunk-group settings will block unscreened calls, so if the screen field is set to no or not listed altogether, the BroadSoft Application Server will refuse the call with a 604 SIP Error (Does Not Exist Anywhere).

Screen Field Example:

Diversion:
<sip:7035552000@megapathvoice.com>;reason=unconditional;counter=1;privacy=off;screen=no
</sip:7035552000@megapathvoice.com></sip:7035552000@megapathvoice.com></sip:7035551000@megapathvoice.com:5060;transport=udp

[/pre]

For reference, here are the pertinent trunk settings we have right now:

[pre]
username=608XXXXXXX
type=peer
secret=XXX
qualify=yes
insecure=port,invite
host=(MegaPath host)
fromdomain=(MegaPath Host)
dtmfmode=rfc2833
disallow=all
canreinvite=no
allow=ulaw
[/pre]

I'm guessing it has somethign to do with the "canreinvite" setting, but am not sure that's all I need to change? Thanks for the help!

 

[ucxguy]

Go to PBX Configuration - Advanced Settings and in the section Dialplan and Operational set the Generate Diversion Headers option to True.

 

[mrdom]

I've enabled the diversion headers in Advanced Settings, but unfortunately, it's still giving me the same error - SIP response 603. Is there something else I need to do or tweak?

 

[mrdom]

I've heard back from the SIP provider, and they are telling me that I need the @DOMAIN included in the diversion header. With only the diversion headers option enabled, it's apparently not passing along the correct information. Here are what the diversion headers look like in the log:

[pre]
[2023-12-06 13:47:35.152] VERBOSE[103599][C-00000036] pbx.c: Executing [s@sub-diversion-header:1] Set("Local/123456789@from-internal-0000000c;2", "DIVERSION_REASON=no-answer") in new stack
[2023-12-06 13:47:35.153] VERBOSE[103599][C-00000036] pbx.c: Executing [s@sub-diversion-header:2] SIPAddHeader("Local/123456789@from-internal-0000000c;2", "Diversion: <tel:2345678>;reason=no-answer;screen=no;privacy=off") in new stack
[2023-12-06 13:47:35.153] VERBOSE[103599][C-00000036] pbx.c: Executing [s@sub-diversion-header:3] Return("Local/123456789@from-internal-0000000c;2", "") in new stack
[/pre]

Another FreePBX user seemed to be having this same issue - the forum post is located at:

https://community.freepbx.org/t/adding-a-custom-diversion-header/23959

Basically, he's being told to edit extension_custom.conf to include the following:

[pre]
[macro-dialout-trunk-predial-hook]
exten => s,1,SIPAddHeader(Diversion:>sip:+PHONENUMBER@TRUNKPROVIDER>;privacy=off;reason=unconditional)
[/pre]

I checked my extension_custom.conf file, and there is no section like this at all. I'm not sure if I should add it or not. Anyone? This is crazy to simply get the system to outdial using another one of our available channels!

Thanks for the continued help