[Spce-user] Caller receives busy signal on inbound call without caller ID

[Andy Bradford]

Hi all,

Forgive me in advance - still getting my feet wet with sipwise, and have
faced a challenge in the last phase of cutting a few DIDs over to routing
through my sipwise instance. I've hit a slight snag.

I've noticed that callers from the PSTN ringing into a DID pointed to
sipwise, if they decide to "block" their caller ID by dialing *67, or not
sending CID for whatever reason from their provider, the call will go to a
busy signal. My receiving trunk on my Asterisk box that I have sipwise
connected to does not see the call at all.

I tail'd /var/log/ngcp/kamailio-proxy.log on sipwise and placed a test
call, dialing *67 before the number, and receive the following:

====

589|3aec3ab0-b7d2-4b1a-a470-afe8b6145347|1212xxxxxxx|45.79.x.x|1212xxxxxxx|159.x.x.x
< sipwise IP|4||||||||||||' - R=sip:1212xxxxxxx at 45.79.x.x:5060;transport=udp
ID=165099216a7104e914ed26600bd72b93 at 178.79.x.x:5060 UA='Asterisk PBX
11.13.1~dfsg-2+b1'
May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE:
<script>: Local user online, using alias as user -
R=sip:1212xxxxxxx at 45.79.x.x:5060;transport=udp
ID=165099216a7104e914ed26600bd72b93 at 178.79.x.x:5060 UA='Asterisk PBX
11.13.1~dfsg-2+b1'
May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE:
<script>: Setting display-name/caller_domain_userprov '"Resource matching
URI \"/cnam\" not found"@178.79.x.x' for rcv_display -
R=sip:1212xxxxxxx at 45.79.x.x:5060;transport=udp
ID=165099216a7104e914ed26600bd72b93 at 178.79.x.x:5060 UA='Asterisk PBX
11.13.1~dfsg-2+b1'
May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE:
<script>: Setting caller_cli_userprov/caller_domain_userprov '0 at 178.79.x.x'
for upn - R=sip:1212xxxxxxx at 45.79.x.x:5060;transport=udp
ID=165099216a7104e914ed26600bd72b93 at 178.79.x.x:5060 UA='Asterisk PBX
11.13.1~dfsg-2+b1'
May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE:
<script>: Setting From to '"Resource matching URI \"/cnam\" not found"
<sip:0 at 178.79.x.x>' - R=sip:1212xxxxxxx at 45.79.x.x:5060;transport=udp
ID=165099216a7104e914ed26600bd72b93 at 178.79.x.x:5060 UA='Asterisk PBX
11.13.1~dfsg-2+b1'
May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE:
<script>: Setting caller_cli_netprov/caller_domain_netprov '0 at 178.79.x.x'
for npn - R=sip:1212xxxxxxx at 45.79.x.x:5060;transport=udp
ID=165099216a7104e914ed26600bd72b93 at 178.79.x.x:5060 UA='Asterisk PBX
11.13.1~dfsg-2+b1'
May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE:
<script>: Setting P-Called-Party-ID '<sip:1212xxxxxxx at 45.79.x.x>' -
R=sip:1212xxxxxxx at 45.79.x.x:5060;transport=udp
ID=165099216a7104e914ed26600bd72b93 at 178.79.x.x:5060 UA='Asterisk PBX
11.13.1~dfsg-2+b1'
May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE:
<script>: Setting 'sip:45.79.x.x:5060' taken from R-URI as next hop after
lb for PSTN call - R=sip:1212xxxxxxx at 45.79.x.x:5060;transport=udp
ID=165099216a7104e914ed26600bd72b93 at 178.79.x.x:5060 UA='Asterisk PBX
11.13.1~dfsg-2+b1'
May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE:
<script>: Appending P-D-URI
'sip:127.0.0.1:5060;received='sip:45.79.x.x:5060;lr;transport=udp''
- R=sip:1212xxxxxxx at 45.79.x.x:5060;transport=udp
ID=165099216a7104e914ed26600bd72b93 at 178.79.x.x:5060 UA='Asterisk PBX
11.13.1~dfsg-2+b1'
May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE:
<script>: Forcing request via B2BUA 'sip:127.0.0.1:5080' -
R=sip:1212xxxxxxx at 45.79.x.x:5060;transport=udp
ID=165099216a7104e914ed26600bd72b93 at 178.79.x.x:5060 UA='Asterisk PBX
11.13.1~dfsg-2+b1'
May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE:
<script>: Request leaving server, D-URI='sip:127.0.0.1:5080' -
R=sip:1212xxxxxxx at 45.79.x.x:5060;transport=udp
ID=165099216a7104e914ed26600bd72b93 at 178.79.x.x:5060 UA='Asterisk PBX
11.13.1~dfsg-2+b1'
May 25 14:23:22 sipwise-nyc3-test (local7.err) proxy[26546]: ERROR: <core>
[parser/parse_from.c:74]: parse_from_header(): ERROR:parse_from_header: bad
from header [""Resource matching URI \"/cnam\" not found"" <sip:0 at 178.79.x.x
>;tag=as1185f930]
May 25 14:23:22 sipwise-nyc3-test (local7.err) proxy[26546]: ERROR: uac
[replace.c:744]: restore_uris_reply(): failed to find/parse FROM hdr
May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE:
<script>: NAT-Reply - S=400 - could not parse From hf M=INVITE
IP=178.79.x.x:5060 (127.0.0.1:5080)
ID=165099216a7104e914ed26600bd72b93 at 178.79.x.x:5060 UA='<null>'
May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE:
<script>: Failure route for PSTN call -
R=sip:1212xxxxxxx at 45.79.x.x:5060;transport=udp
ID=165099216a7104e914ed26600bd72b93 at 178.79.x.x:5060 UA='Asterisk PBX
11.13.1~dfsg-2+b1'
May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE:
<script>: No failover routing needed for this response code -
R=sip:1212xxxxxxx at 45.79.x.x:5060;transport=udp
ID=165099216a7104e914ed26600bd72b93 at 178.79.x.x:5060 UA='Asterisk PBX
11.13.1~dfsg-2+b1'
===

My provider winds up sending me "Resource matching URI \"/cnam\" not found"
for CNAM and "0" as the CID when they can't capture a calling party number.
I imagine that sipwise is seeing this and not knowing what to do.

I have rewrite rules for ^([1-9][0-9]{9})$ and ^\+(1[1-9][0-9]{9})$ defined
already. Should I just create one for 0, and that will correct the issue?
Is there any way to tell sipwise to pass the call off to my far end
Asterisk box with CNAM of "Anonymous" so it doesn't confuse the callee?

Thanks,

Andy
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