Hi Guys,
I am using FreePBX-4.211.64-7 and having trouble where the CallerID won’t display.
I’ve got it set at the extension, the trunk and the outgoing route to display my local number, but I still can’t get it to work. The syntax I’m using is <##########> and the phone system calls out from a private number.
I’ve checked with my provider, and they’re not blocking callerID. can you help, please? I’ve been working on this for days and days now.
TIA,
edooze.
What type of trunks?
BF
Sorry, must’ve forgotten.
They’re SIP trunks. Do you think there might be arguments I can add to the PEER details that might assist?
TIA,
edooze.
And what do you mean “and the phone system calls out from a private number.”
BF
Well, it comes up as a private number on receiving phones. No caller ID displayed. I’m not sure what else it may be called in other parts of the world.
There must be something I’m missing, but for the life of me I can’t find any other locations within freepbx to stipulate that I want callerID, and the number I want displayed. I’ve set it in the Trunk, Outgoing route and Extension details so far.
TIA,
edooze.
I’ve also tried adding /########## after the register string within the trunk options, as well as CallerID=########## in the PEER details section.
With regards to the syntax of the CallerID itself, I’ve selected to force the Trunk CID, and tried ##########, <##########>, “”<##########>, “Name”<##########>, “” <##########>, and “Name” <##########>.
The only thing I can think of is that either there’s an option I don’t know about that I haven’t configured, or there’s some argument I need to send in the PEER details section that isn’t currently present.
Hope this helps shed some light.
Thanks for any and all help,
edooze.
Maybe your provider has something provisioned wrong. If you do a 'sip set debug peer (insert your provider trunk name) you will see the invite with the CID go out.
Don’t forget to turn it off. ‘sip set debug off’
Is this what you mean?
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('HANGUP',{ts '2013-11-05 17:50:09'},'CID:0********6','0********2','','','','','macro-dialout-trunk','SIP/S*****l-0******8','AppDial','(Outgoing Line)',3,'','13******83.456','13******81.455','','','')] Scheduling destruction of SIP dialog '***randommd5hashgoeshere***@xxx.xxx.xxx.xxx:5060' in 32000 ms (Method: INVITE)
As you can see, I've not studied these logs much to make sense of them...
No, clearly you didn’t do the debug as I suggested. The CID is clearly in the CDR you posted (I assume you redacted it). I have never seen a case where Asterisk doesn’t send the CID. Have you spoken with your provider?
Well, you’ll have to forgive me, I don’t really know what I’m looking at here; there’s a lot of information in this report that I’m trying to sift through.
Does the debug output into the asterisk CLI? That is where this is copied from, as I always keep an eye on it while I’m calling. Otherwise, maybe there’s another location I can grab the debug output from?
Have checked with the provider, they say the callerID is functioning correctly from their end. They even placed a call to me from my own number, so it seems it’s not a problem at their end.
I’m in the process of setting up another trunk with another company to test it also, so I can find out definitively whether it’s their service or my phone system causing the problem.
Thanks,
edooze.
/var/log/asterisk/full
You will see CID in md5 hash in SIP invite like below:
Reliably Transmitting (NAT) to 192.168.xx,xxx:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ;branch=z9hG4bK245dd4e5;rport
Max-Forwards: 70
From: “5999” sip:5999@xxxxx;tag=as172db4c4
To: sip:[email protected]
Contact: sip:5999@:5060
Call-ID: 2ce657d63e3dfdb9200008e032b47ab0xx.x.xx.:5060
CSeq: 102 INVITE
User-Agent: PBXact-2.10.1(1.8.21.0)
Date: Wed, 06 Nov 2013 06:22:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 239
[2013-11-06 23:52:07] VERBOSE[7407][C-0000011b] chan_sip.c: Reliably Transmitting (NAT) to 20x.xxx.xxx.xx0:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 12x.xxx.xxx.xxx:5060;branch=z9hG4xxxxx1281;rport
Max-Forwards: 70
From: sip:[email protected];tag=as04d4e933
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.5.1)
Date: Wed, 06 Nov 2013 13:52:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 288