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

Thu May 26 02:28:04 EDT 2016

It's your provider fault ... it should send you an 'annonymous' call when they are not able to do the cnam query, not "Resource matching URI \"/cnam\" ... because that violates the From parsing, that's the problem you have. 

Ask them to send you a correct annonymous call when they send it to you. 

> De: "Andy Bradford" <andy.bradford.cms at gmail.com>
> Para: spce-user at lists.sipwise.com
> Enviados: Jueves, 26 de Mayo 2016 4:08:15
> Asunto: [Spce-user] Caller receives busy signal on inbound call without caller
> ID

> 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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