[Solved] CID of incoming calls not displayed on end device

[vapiano]

Version: 2.14-858a
Build: 220513-0819
Asterisk 13.38.2
Cloud VPS, 4 vCPU Cores, 8 GB RAM, 200 GB SSD
ViciBox_v10.x86_64 10.0.1.iso

Hi,

my problem is that the number (callerID) of incoming calls are not shown on my mobile phone.

I use call forwarding by setting the DID Route to EXTEN and using a mobile number as extension. The call forwarding is working but the callerID of the incoming call is not shown on my mobile phone (i tried it with different phones).

- When using In Group as DID Route instead of EXTEN, the callerID is shown. I have this problem only when using call forwarding. Some months ago the callerID of incoming calls were shown when using call forwarding. But now I do not know what happened that it doesnt work anymore like before
- When calling directly to the mobile phone the callerID is shown on phone display

I did not find any setting for that. Maybe someone can help here. Thanks.

[mflorell]

Might be a carrier issue, many carriers stopped allowing unapproved CID on outgoing calls in the last year. This happened to one of our carriers as well.

[vapiano]

Might be a carrier issue, many carriers stopped allowing unapproved CID on outgoing calls in the last year. This happened to one of our carriers as well.

Okay thank you for this information. I will contact my carrier provider and then inform you what they say.

Edit: I have contacted the carrier support (Telnyx) and they told me that they dont block the callerID when using call forwarding. They can see the callerID of incoming calls and the problem of not displaying the callerID of a forwarded call is not because of them.

So any ideas where the problem comes from?

[mflorell]

Next step is for you to post the screenlog.0 output from one of these calls. Make sure you have the Server set to use AGI Output to BOTH.

[vapiano]

Next step is for you to post the screenlog.0 output from one of these calls. Make sure you have the Server set to use AGI Output to BOTH.

I hope this one is the right peace of code. I had to figure out how to do the screenlog. Again, thank you for your help.

49408XXXXXX0 = phone number where customers can call me
491791XXXXX1 = with this number I did the test call to this 49408XXXXXX0 number
49176595XXXX5 = to this number the calls get forwarded


[Feb 8 20:11:19] == Using SIP RTP CoS mark 5
[Feb 8 20:11:19] > 0x7fc8cc01e0b0 -- Strict RTP learning after remote address set to: 185.246.41.170:19418
[Feb 8 20:11:19] -- Executing [49408XXXXXX0@trunkinbound:1] AGI("SIP/telnyx-00000005", "agi-DID_route.agi") in new s
tack
[Feb 8 20:11:19] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
2023-02-08 20:11:19|agi-DID_route.agi||INSERT INTO vicidial_did_log SET uniqueid='1675883479.11',channel='SIP/telnyx-00000005',server_ip='XXX.XX.XXX.88',call
er_id_number='491791XXXXX1',caller_id_name='491791XXXXX1',extension='49408XXXXXX0',call_date='2023-02-08 20:11:19',did_id='6',did_route='EXTEN';|
2023-02-08 20:11:19|agi-DID_route.agi|-- DID LOG : |1|INSERT INTO vicidial_did_log SET uniqueid='1675883479.11',channel='SIP/telnyx-00000005',server_ip='XXX.XX.XXX.88',caller_id_number='491791XXXXX1',caller_id_name='491791XXXXX1',extension='49408XXXXXX0',call_date='2023-02-08 20:11:19',did_id='6',did_route='EX
TEN';|
2023-02-08 20:11:19|agi-DID_route.agi|-- CALL LOG : |1|INSERT INTO call_log SET uniqueid='1675883479.11', channel='SIP/telnyx-00000005', channel_group='DI
D_INBOUND', server_ip='XXX.XX.XXX.88', type='SIP', extension='494085XXXXX0', number_dialed='494085XXXXX0', caller_code='491791XXXXX1', start_time='2023-02-08
20:11:19', start_epoch='1675883479';|
2023-02-08 20:11:19|agi-DID_route.agi|exiting the DID app, transferring call to 949176595XXXX5 @ default
[Feb 8 20:11:19] -- <SIP/telnyx-00000005>AGI Script agi-DID_route.agi completed, returning 0
[Feb 8 20:11:19] -- Executing [949176595XXXX5@default:1] AGI("SIP/telnyx-00000005", "agi://127.0.0.1:4577/call_log")
in new stack
[Feb 8 20:11:19] -- <SIP/telnyx-00000005>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Feb 8 20:11:19] -- Executing [949176595XXXX5@default:2] Dial("SIP/telnyx-00000005", "SIP/telnyx/176595XXXX5,60,tTor
") in new stack
[Feb 8 20:11:19] == Using SIP RTP CoS mark 5
[Feb 8 20:11:19] -- Called SIP/telnyx/176595XXXX5
[Feb 8 20:11:20] -- SIP/telnyx-00000006 is ringing
[Feb 8 20:11:24] > 0x7fc914015320 -- Strict RTP learning after remote address set to: 185.246.41.167:31216
[Feb 8 20:11:24] -- SIP/telnyx-00000006 is making progress passing it to SIP/telnyx-00000005
[Feb 8 20:11:34] == Spawn extension (default, 949176595XXXX5, 2) exited non-zero on 'SIP/telnyx-00000005'
[Feb 8 20:11:34] -- Executing [h@default:1] AGI("SIP/telnyx-00000005", "agi://127.0.0.1:4577/call_log--HVcauses--PRI---
--NODEBUG-----0-----CANCEL---------------SIP 183 Session Progress)") in new stack
[Feb 8 20:11:34] -- <SIP/telnyx-00000005>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -------SIP
183 Session Progress) completed, returning 0

[mflorell]

Nothing looks suspect there, time to test with asterisk sip debugging enabled, which is a lot of data, but that will show what exactly was sent to the carrier.

[vapiano]

Nothing looks suspect there, time to test with asterisk sip debugging enabled, which is a lot of data, but that will show what exactly was sent to the carrier.

I contacted again the carrier to check the issue. Because now I forward the call by the carrier settings, so the call does not go through Vicidial anymore. But the number is still not displayed on my end device. So I think the issue must be on the carrier site. After they have checked it I will give an update here. Thank you so far.

[vapiano]

Update: The carrier support did a test call and concluded the following: "This test was to rule out the local carrier we use for Germany calls potentially causing issue but this call connected over international routes so I am now fairly certain the Private Number display is surrounding some kind of config on your PBX/environment side".

I wanted to post the asterisk cli with sip debug enabled but getting the hint that "Your message contains too many URLs. The maximum number of URLs allowed is 5". I have already changed the sip.telnyx.eu to sip(dot)telnyx(dot)eu but get still this message.

[GenXOutsourcing]

I am having a similar issue, however my problem is that the DID is not showing in the lead modification page. And somehow this is causing the Inbound traffic to go randomly to different Ingroups.

Not every call, but most



https://imgur.com/a/mG5eeUs

[vapiano]

UPDATE:

Now, after my sip trunk provider contaced my sim card provider I know why the number is not displayed when call forwarding.

Due to §120 of German Telecommunications Law the carrier has to anonymize all Germany incoming traffic on interconnection trunks from abroad that carry a German A-Number (FROM header or PAI) from Dec. 1st 2022.

For all germans this information could be interesting.

[mflorell]

Thanks for the postback!