<html><body><div style="font-family: arial, helvetica, sans-serif; font-size: 12pt; color: #000000"><div>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.<br></div><div><br data-mce-bogus="1"></div><div>Ask them to send you a correct annonymous call when they send it to you.<br data-mce-bogus="1"></div><div><br></div><hr id="zwchr" data-marker="__DIVIDER__"><div data-marker="__HEADERS__"><blockquote style="border-left: 2px solid #1010FF; margin-left: 5px; padding-left: 5px; color: #000; font-weight: normal; font-style: normal; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-size: 12pt;" data-mce-style="border-left: 2px solid #1010FF; margin-left: 5px; padding-left: 5px; color: #000; font-weight: normal; font-style: normal; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-size: 12pt;"><b>De: </b>"Andy Bradford" <andy.bradford.cms@gmail.com><br><b>Para: </b>spce-user@lists.sipwise.com<br><b>Enviados: </b>Jueves, 26 de Mayo 2016 4:08:15<br><b>Asunto: </b>[Spce-user] Caller receives busy signal on inbound call without caller ID<br></blockquote></div><div data-marker="__QUOTED_TEXT__"><blockquote style="border-left: 2px solid #1010FF; margin-left: 5px; padding-left: 5px; color: #000; font-weight: normal; font-style: normal; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-size: 12pt;" data-mce-style="border-left: 2px solid #1010FF; margin-left: 5px; padding-left: 5px; color: #000; font-weight: normal; font-style: normal; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-size: 12pt;"><div dir="ltr"><div><div><div><div><div><div>Hi all,<br><br></div>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.<br><br>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. <br><br></div>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:<br><br>====<br><br>589|3aec3ab0-b7d2-4b1a-a470-afe8b6145347|1212xxxxxxx|45.79.x.x|1212xxxxxxx|159.x.x.x < sipwise IP|4||||||||||||' - R=sip:1212xxxxxxx@45.79.x.x:5060;transport=udp ID=165099216a7104e914ed26600bd72b93@178.79.x.x:5060 UA='Asterisk PBX 11.13.1~dfsg-2+b1'<br>May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE: <script>: Local user online, using alias as user - R=sip:1212xxxxxxx@45.79.x.x:5060;transport=udp ID=165099216a7104e914ed26600bd72b93@178.79.x.x:5060 UA='Asterisk PBX 11.13.1~dfsg-2+b1'<br>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@45.79.x.x:5060;transport=udp ID=165099216a7104e914ed26600bd72b93@178.79.x.x:5060 UA='Asterisk PBX 11.13.1~dfsg-2+b1'<br>May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE: <script>: Setting caller_cli_userprov/caller_domain_userprov '0@178.79.x.x' for upn - R=sip:1212xxxxxxx@45.79.x.x:5060;transport=udp ID=165099216a7104e914ed26600bd72b93@178.79.x.x:5060 UA='Asterisk PBX 11.13.1~dfsg-2+b1'<br>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@178.79.x.x>' - R=sip:1212xxxxxxx@45.79.x.x:5060;transport=udp ID=165099216a7104e914ed26600bd72b93@178.79.x.x:5060 UA='Asterisk PBX 11.13.1~dfsg-2+b1'<br>May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE: <script>: Setting caller_cli_netprov/caller_domain_netprov '0@178.79.x.x' for npn - R=sip:1212xxxxxxx@45.79.x.x:5060;transport=udp ID=165099216a7104e914ed26600bd72b93@178.79.x.x:5060 UA='Asterisk PBX 11.13.1~dfsg-2+b1'<br>May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE: <script>: Setting P-Called-Party-ID '<sip:1212xxxxxxx@45.79.x.x>' - R=sip:1212xxxxxxx@45.79.x.x:5060;transport=udp ID=165099216a7104e914ed26600bd72b93@178.79.x.x:5060 UA='Asterisk PBX 11.13.1~dfsg-2+b1'<br>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@45.79.x.x:5060;transport=udp ID=165099216a7104e914ed26600bd72b93@178.79.x.x:5060 UA='Asterisk PBX 11.13.1~dfsg-2+b1'<br>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@45.79.x.x:5060;transport=udp ID=165099216a7104e914ed26600bd72b93@178.79.x.x:5060 UA='Asterisk PBX 11.13.1~dfsg-2+b1'<br>May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE: <script>: Forcing request via B2BUA 'sip:<a href="http://127.0.0.1:5080" target="_blank" data-mce-href="http://127.0.0.1:5080">127.0.0.1:5080</a>' - R=sip:1212xxxxxxx@45.79.x.x:5060;transport=udp ID=165099216a7104e914ed26600bd72b93@178.79.x.x:5060 UA='Asterisk PBX 11.13.1~dfsg-2+b1'<br>May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE: <script>: Request leaving server, D-URI='sip:<a href="http://127.0.0.1:5080" target="_blank" data-mce-href="http://127.0.0.1:5080">127.0.0.1:5080</a>' - R=sip:1212xxxxxxx@45.79.x.x:5060;transport=udp ID=165099216a7104e914ed26600bd72b93@178.79.x.x:5060 UA='Asterisk PBX 11.13.1~dfsg-2+b1'<br>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@178.79.x.x>;tag=as1185f930]<br>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<br>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 (<a href="http://127.0.0.1:5080" target="_blank" data-mce-href="http://127.0.0.1:5080">127.0.0.1:5080</a>) ID=165099216a7104e914ed26600bd72b93@178.79.x.x:5060 UA='<null>'<br>May 25 14:23:22 sipwise-nyc3-test (local7.notice) proxy[26546]: NOTICE: <script>: Failure route for PSTN call - R=sip:1212xxxxxxx@45.79.x.x:5060;transport=udp ID=165099216a7104e914ed26600bd72b93@178.79.x.x:5060 UA='Asterisk PBX 11.13.1~dfsg-2+b1'<br>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@45.79.x.x:5060;transport=udp ID=165099216a7104e914ed26600bd72b93@178.79.x.x:5060 UA='Asterisk PBX 11.13.1~dfsg-2+b1' <br>===<br><br></div>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. <br><br></div>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? <br><br></div>Thanks,<br><br></div>Andy <br></div><br>_______________________________________________<br>Spce-user mailing list<br>Spce-user@lists.sipwise.com<br>https://lists.sipwise.com/listinfo/spce-user<br></blockquote></div></div></body></html>