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I have an Asterisk system which has been functioning perfectly well for about six months and now we want to add incoming caller id.

We got the service from our phone company; we see the caller ID arrive in the Cisco logs (debug vpm sig). We see the calls arrive from the Cisco to the Asterisk (with tcpdump); but those with caller ID are simply ignored and no ringing happens.

In tpdump we see that the difference between the calls is whether they are from sip:ip@ipaddress or sip:callerid@ipaddress and Asterisk accepts the former and ignores the ones with callerid -- no ringing, nothing.

There must be something missing on the sip.conf, but I haven't been able to find out what.

Can some kind soul advise how to make Asterisk accept these calls?

Versions

  • Asterisk 14.7.5 on Ubuntu 16.04.3 LTS (server)
  • About 20 phones, all Cisco SPA504G
  • Cisco 2811 with VIC2-4FXO and PVDM2-32 running IOS 15.1(4)M10

Network

 telephone lines
 | | | |   4FXO interfaces
  2811
   |.254
   |                 172.31.20.0/24
===+=====+=============+=====+=====
         |.224         |.1   |.2
      asterisk       phone phone ...

Please just say if other versions, logs or config needed.

Asterisk config

This is the bulk of sip.conf:

[general]
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=0.0.0.0
tcpenable=no
callcounter=yes

; this is the Cisco with the 4FXO interface
[172.31.20.254]
context=LocalSets
type=friend
host=172.31.20.254
disallow=all
allow=all

Working

Working config on the Cisco ("working=calls go through, but say from unknown")

Call comes in as seen in tcpdump

From: "unknown" <sip:172.31.20.254>;tag=4C2B58F4-1422

Cisco config

dial-peer voice 1000 voip
 description link to pbx
 preference 10
 destination-pattern [1-8]..
 session protocol sipv2
 session target ipv4:172.31.20.224:5060
 session transport udp
 voice-class h323 1
 dtmf-relay rtp-nte
 codec g711ulaw
 clid strip
 no vad

Failing

The failing behaviour (failing="caller id sent to Asterisk, Asterisk ignores whole call, no ringing")

Calls come in as seen in tcpdump

From: <sip:07777123456@172.31.20.254>;tag=4C2F4350-71D

Same config except no clid strip, instead

dial-peer voice 1000 voip
 ...
 clid network-provided
2
+50

Just idea but did you try also SIP debug inside Asterisk? The tries you have described seems to me more about network traffic then signalling processing... This doesn't seem to me like network issue. This may be more processing part.

Try one of the following:

sip set debug on
sip set debug peer 172.31.20.254

And to disable

sip set debug off

Then you can check also core debug (including also e.g. dialplan processing detail):

core set debug ...

You have set focus on From but what would be interested is also what is To or what is default / whatever in your dialplan. It can also happen that there is also other (for you so far) not visible change cause not fitting the dialplan entry so the session may be processed but the result may be just not ringing any branch.

1
  • Many thanks Kamil, I'll try your suggestions as soon as I'm on client site again. I agree it's probably not networking and almost certainly Asterisk config, but that's where I have the least knowledge.
    – jonathanjo
    Nov 26 '18 at 23:32

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