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

[Andy Bradford]

Thanks, Raúl. Could you elaborate a little more on how to change this
within the config file? I'm still new to sipwise, so have started to look
through, but wasn't sure how to proceed.

I'd appreciate any guidance.

Andy

On Friday, May 27, 2016, Raúl Alexis Betancor Santana <
rabs at dimension-virtual.com> wrote:

> Not with rewrite rules ... you should do it at .cfg level, add a check of
> the 0@[IP] and then erase the From Display part ... or better, fix the
> whole request and put it as it should be.
>
>
> ------------------------------
>
> *De: *"Andy Bradford" <andy.bradford.cms at gmail.com
> <javascript:_e(%7B%7D,'cvml','andy.bradford.cms at gmail.com');>>
> *Para: *"Raúl Alexis Betancor Santana" <rabs at dimension-virtual.com
> <javascript:_e(%7B%7D,'cvml','rabs at dimension-virtual.com');>>
> *Enviados: *Jueves, 26 de Mayo 2016 12:34:27
> *Asunto: *Re: [Spce-user] Caller receives busy signal on inbound call
> without caller ID
>
> Understood - I know it's their problem, but something they haven't fixed
> yet. While I wait for them to do so, is there any way for me to rewrite it
> so sipwise will pass it along to my Asterisk system?
>
> Andy
>
> On Thu, May 26, 2016 at 2:28 AM, Raúl Alexis Betancor Santana <
> rabs at dimension-virtual.com
> <javascript:_e(%7B%7D,'cvml','rabs at dimension-virtual.com');>> wrote:
>
>> 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
>> <javascript:_e(%7B%7D,'cvml','andy.bradford.cms at gmail.com');>>
>> *Para: *spce-user at lists.sipwise.com
>> <javascript:_e(%7B%7D,'cvml','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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>>
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>
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