Re: [cisco-voip] CME w/ SIP Trunk
Sometimes the “number” requires @DNSNAME or @IPADDRESS for the registrar to accept it. That’s in the normal SIP world. Not sure how CME would handle that. From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ed Leatherman Sent: Wednesday, June 24, 2015 3:28 PM To: Cisco VOIP Subject: Re: [cisco-voip] CME w/ SIP Trunk For my own sanity.. can someone tell me if I'm correct on this; the username and password for registering a SIP line with a registrar should look like : ! sip-ua credentials number X username user1 password MyPassword realm MyRealm authentication username user1 password MyPassword ! Is there any where else it could pull the credentials from? Still getting 403 and SP still just says its a PW mismatch - but i'm just pasting in what they send me (checked for stray spaces and what-not already) Thanks! Ed On Wed, Jun 24, 2015 at 2:32 PM, Ed Leatherman ealeather...@gmail.commailto:ealeather...@gmail.com wrote: Another one for posterity.. So it seems like SIP REGISTER requests are only shown using debug ccsip noncall ... however there is no output from that command unless debug ccsip messages is also running. very confusing when I only has debug noncall on and didnt see any messages, but they were present in packet cap! On Wed, Jun 24, 2015 at 11:26 AM, Ed Leatherman ealeather...@gmail.commailto:ealeather...@gmail.com wrote: Following up for posterity.. Still fighting this one but a small bit of progress. So CME was trying to register both the ephone-dn extension and the E164 expanded number of the extension (which was the correct one) - so SP was putting us in timeout for trying to register invalid numbers. I figured out how to stop that with the number xxx no-reg command in each ephone-dn. Also, I had to re-write some headers in the REGISTER requests (to/from/request-uri and Authorization fields) with sip profiles. Also - as brian suggested, they told me the wrong username, should have not had dashes in it. After all that, still getting back 403 Authentication Failed - but its at least cleaned up and I'm not getting put in timeout. onward.. On Tue, Jun 23, 2015 at 6:06 PM, Ed Leatherman ealeather...@gmail.commailto:ealeather...@gmail.com wrote: Didnt seem to help but thats a good thought. Slogging it out with SP tomorrow again. I'm really puzzled about the SIP debugs.. its like i'm not getting all my SIP messages in there On Tue, Jun 23, 2015 at 3:29 PM, Brian Meade bmead...@vt.edumailto:bmead...@vt.edu wrote: I'd try the username without the dashes first. On Tue, Jun 23, 2015 at 3:26 PM, Ed Leatherman ealeather...@gmail.commailto:ealeather...@gmail.com wrote: I did a packet cap and we are sending the SIP REGISTER, but its not showing up in sip debug?? really weird. anywhere I'm not binding SIP to my loopback address, i'm not getting SIP debugs for. So I am getting 403 back from SP after all, gonna double check username/passwords On Tue, Jun 23, 2015 at 3:16 PM, Brian Meade bmead...@vt.edumailto:bmead...@vt.edu wrote: How about connecting via telnet over 5060? You may be having a TCP issue which is why you never see the Register sent. On Tue, Jun 23, 2015 at 3:09 PM, Ed Leatherman ealeather...@gmail.commailto:ealeather...@gmail.com wrote: Brian msu-tmp-access#sho sip-ua register status Line peer expires(sec) reg survival P-Associ-URI == === 2031120001 43 no normal 2031220003 43 no normal 2031320005 43 no normal 2031420007 43 no normal .. etc .. all no I can ping the sip-server from router so it appears to be able to resolve the name OK. On Tue, Jun 23, 2015 at 2:48 PM, Brian Meade bmead...@vt.edumailto:bmead...@vt.edu wrote: What do you see for show sip-ua register status? Are you sure the gateway can resolve the sip-server via DNS? On Tue, Jun 23, 2015 at 2:32 PM, Ed Leatherman ealeather...@gmail.commailto:ealeather...@gmail.com wrote: Hello! I'm trying to get a SIP trunk out to a regional SP (Lumos) configured. I need to get CME setup to register numbers with their sip proxy, but the registration is not happening and i'm not seeing any register messages debugs from debug ccsip messages to troubleshoot from. So I think maybe CME isn't trying? What should trigger CME to try and register these numbers? My config looks like this (some ephones/ephone-dns up and registered) - authentication credentials were provided from Lumos. IOS 15.4(3)M2 msu-tmp-access#sh run | s sip-ua sip-ua credentials username 304-929-0300tel:304-929-0300 password 7 blah realm sbc.ia.ntelos.nethttp://sbc.ia.ntelos.net authentication username 304-929-0300tel:304-929-0300 password 7 blah retry register 10 registrar
Re: [cisco-voip] CME w/ SIP Trunk
Philip, Thanks for that - it looks like CME automatically adds the @dnsname (according to what SP is telling me) correctly but its nice to have some confirmation on that part from someone else :) On Wed, Jun 24, 2015 at 4:30 PM, Walenta, Philip philip.wale...@polycom.com wrote: Sometimes the “number” requires @DNSNAME or @IPADDRESS for the registrar to accept it. That’s in the normal SIP world. Not sure how CME would handle that. *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf Of *Ed Leatherman *Sent:* Wednesday, June 24, 2015 3:28 PM *To:* Cisco VOIP *Subject:* Re: [cisco-voip] CME w/ SIP Trunk For my own sanity.. can someone tell me if I'm correct on this; the username and password for registering a SIP line with a registrar should look like : ! sip-ua credentials number X username user1 password MyPassword realm MyRealm authentication username user1 password MyPassword ! Is there any where else it could pull the credentials from? Still getting 403 and SP still just says its a PW mismatch - but i'm just pasting in what they send me (checked for stray spaces and what-not already) Thanks! Ed On Wed, Jun 24, 2015 at 2:32 PM, Ed Leatherman ealeather...@gmail.com wrote: Another one for posterity.. So it seems like SIP REGISTER requests are only shown using debug ccsip noncall ... however there is no output from that command unless debug ccsip messages is also running. very confusing when I only has debug noncall on and didnt see any messages, but they were present in packet cap! On Wed, Jun 24, 2015 at 11:26 AM, Ed Leatherman ealeather...@gmail.com wrote: Following up for posterity.. Still fighting this one but a small bit of progress. So CME was trying to register both the ephone-dn extension and the E164 expanded number of the extension (which was the correct one) - so SP was putting us in timeout for trying to register invalid numbers. I figured out how to stop that with the number xxx no-reg command in each ephone-dn. Also, I had to re-write some headers in the REGISTER requests (to/from/request-uri and Authorization fields) with sip profiles. Also - as brian suggested, they told me the wrong username, should have not had dashes in it. After all that, still getting back 403 Authentication Failed - but its at least cleaned up and I'm not getting put in timeout. onward.. On Tue, Jun 23, 2015 at 6:06 PM, Ed Leatherman ealeather...@gmail.com wrote: Didnt seem to help but thats a good thought. Slogging it out with SP tomorrow again. I'm really puzzled about the SIP debugs.. its like i'm not getting all my SIP messages in there On Tue, Jun 23, 2015 at 3:29 PM, Brian Meade bmead...@vt.edu wrote: I'd try the username without the dashes first. On Tue, Jun 23, 2015 at 3:26 PM, Ed Leatherman ealeather...@gmail.com wrote: I did a packet cap and we are sending the SIP REGISTER, but its not showing up in sip debug?? really weird. anywhere I'm not binding SIP to my loopback address, i'm not getting SIP debugs for. So I am getting 403 back from SP after all, gonna double check username/passwords On Tue, Jun 23, 2015 at 3:16 PM, Brian Meade bmead...@vt.edu wrote: How about connecting via telnet over 5060? You may be having a TCP issue which is why you never see the Register sent. On Tue, Jun 23, 2015 at 3:09 PM, Ed Leatherman ealeather...@gmail.com wrote: Brian msu-tmp-access#sho sip-ua register status Line peer expires(sec) reg survival P-Associ-URI == === 2031120001 43 no normal 2031220003 43 no normal 2031320005 43 no normal 2031420007 43 no normal .. etc .. all no I can ping the sip-server from router so it appears to be able to resolve the name OK. On Tue, Jun 23, 2015 at 2:48 PM, Brian Meade bmead...@vt.edu wrote: What do you see for show sip-ua register status? Are you sure the gateway can resolve the sip-server via DNS? On Tue, Jun 23, 2015 at 2:32 PM, Ed Leatherman ealeather...@gmail.com wrote: Hello! I'm trying to get a SIP trunk out to a regional SP (Lumos) configured. I need to get CME setup to register numbers with their sip proxy, but the registration is not happening and i'm not seeing any register messages debugs from debug ccsip messages to troubleshoot from. So I think maybe CME isn't trying? What should trigger CME to try and register these numbers? My config looks like this (some ephones/ephone-dns up and registered) - authentication credentials were provided from Lumos. IOS 15.4(3)M2 msu-tmp-access#sh run | s sip-ua sip-ua credentials username 304-929-0300 password 7 blah
Re: [cisco-voip] CME w/ SIP Trunk
For my own sanity.. can someone tell me if I'm correct on this; the username and password for registering a SIP line with a registrar should look like : ! sip-ua credentials number X username user1 password MyPassword realm MyRealm authentication username user1 password MyPassword ! Is there any where else it could pull the credentials from? Still getting 403 and SP still just says its a PW mismatch - but i'm just pasting in what they send me (checked for stray spaces and what-not already) Thanks! Ed On Wed, Jun 24, 2015 at 2:32 PM, Ed Leatherman ealeather...@gmail.com wrote: Another one for posterity.. So it seems like SIP REGISTER requests are only shown using debug ccsip noncall ... however there is no output from that command unless debug ccsip messages is also running. very confusing when I only has debug noncall on and didnt see any messages, but they were present in packet cap! On Wed, Jun 24, 2015 at 11:26 AM, Ed Leatherman ealeather...@gmail.com wrote: Following up for posterity.. Still fighting this one but a small bit of progress. So CME was trying to register both the ephone-dn extension and the E164 expanded number of the extension (which was the correct one) - so SP was putting us in timeout for trying to register invalid numbers. I figured out how to stop that with the number xxx no-reg command in each ephone-dn. Also, I had to re-write some headers in the REGISTER requests (to/from/request-uri and Authorization fields) with sip profiles. Also - as brian suggested, they told me the wrong username, should have not had dashes in it. After all that, still getting back 403 Authentication Failed - but its at least cleaned up and I'm not getting put in timeout. onward.. On Tue, Jun 23, 2015 at 6:06 PM, Ed Leatherman ealeather...@gmail.com wrote: Didnt seem to help but thats a good thought. Slogging it out with SP tomorrow again. I'm really puzzled about the SIP debugs.. its like i'm not getting all my SIP messages in there On Tue, Jun 23, 2015 at 3:29 PM, Brian Meade bmead...@vt.edu wrote: I'd try the username without the dashes first. On Tue, Jun 23, 2015 at 3:26 PM, Ed Leatherman ealeather...@gmail.com wrote: I did a packet cap and we are sending the SIP REGISTER, but its not showing up in sip debug?? really weird. anywhere I'm not binding SIP to my loopback address, i'm not getting SIP debugs for. So I am getting 403 back from SP after all, gonna double check username/passwords On Tue, Jun 23, 2015 at 3:16 PM, Brian Meade bmead...@vt.edu wrote: How about connecting via telnet over 5060? You may be having a TCP issue which is why you never see the Register sent. On Tue, Jun 23, 2015 at 3:09 PM, Ed Leatherman ealeather...@gmail.com wrote: Brian msu-tmp-access#sho sip-ua register status Line peer expires(sec) reg survival P-Associ-URI == === 2031120001 43 no normal 2031220003 43 no normal 2031320005 43 no normal 2031420007 43 no normal .. etc .. all no I can ping the sip-server from router so it appears to be able to resolve the name OK. On Tue, Jun 23, 2015 at 2:48 PM, Brian Meade bmead...@vt.edu wrote: What do you see for show sip-ua register status? Are you sure the gateway can resolve the sip-server via DNS? On Tue, Jun 23, 2015 at 2:32 PM, Ed Leatherman ealeather...@gmail.com wrote: Hello! I'm trying to get a SIP trunk out to a regional SP (Lumos) configured. I need to get CME setup to register numbers with their sip proxy, but the registration is not happening and i'm not seeing any register messages debugs from debug ccsip messages to troubleshoot from. So I think maybe CME isn't trying? What should trigger CME to try and register these numbers? My config looks like this (some ephones/ephone-dns up and registered) - authentication credentials were provided from Lumos. IOS 15.4(3)M2 msu-tmp-access#sh run | s sip-ua sip-ua credentials username 304-929-0300 password 7 blah realm sbc.ia.ntelos.net authentication username 304-929-0300 password 7 blah retry register 10 registrar dns:sbc.ia.ntelos.net:5060 expires 120 sip-server dns:sbc.ia.ntelos.net:5060 ! msu-tmp-access#sho run | s voice service voice service voip ip address trusted list ipv4 216.12.114.195 address-hiding allow-connections sip to sip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip bind control source-interface GigabitEthernet0/2 bind media source-interface GigabitEthernet0/2 registrar server options-ping 60 -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net
[cisco-voip] Android Jabber Client doesn't respond while asleep
Still testing out Jabber (10.6), finding some odd things here and there. In this case, when I put the Moto G (OS 4.4.4) to sleep by sleep button or cover, it does not respond any longer to inbound calls. Behaviour is different on calling phone, sometimes you hear ring back, sometimes you don't. Most of the time nothing happens, but sometimes you get a half ring at the very end, close to when it's supposed to send call to voicemail, but I think that's the missed call notification, because I get something similar if I call the Moto G but hang up. It's very weird. Anyone seen anything like this? Lelio --- Lelio Fulgenzi, B.A. Senior Analyst, Network Infrastructure Computing and Communications Services (CCS) University of Guelph 519‐824‐4120 Ext 56354 le...@uoguelph.ca www.uoguelph.ca/ccs Room 037, Animal Science and Nutrition Building Guelph, Ontario, N1G 2W1 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Mixed mode revert to unsecure
You just need to run utils ctl set-cluster non-secure-mode on the publisher, then run file delete tftp CTLFile.tlv on each node (may not be needed, can't remember), then restart Cisco CallManager service and Cisco TFTP service on each node. Then you'll be able to re-configure auto-registration. On Wed, Jun 24, 2015 at 2:57 AM, Alessandro Bertacco bertacco.alessan...@alice.it wrote: Hi guys, for mistake I have enabled my 10.5 cucm in mixed mode using Cli command and with out using the token. Customer don't have any capf profile defined and no one use encryption, but obviously auto registration now is disabled, and for sure I've many other complications not needed. Can I easily revert back to Unsecure mode? Thanks. Alessandro ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Problems with incoming ISDN call / CME does not process the call - any clou? q931 debug inside
For the first one, you're getting a facility message instead of a setup message which is strange. Are you sure you caught the beginning of the call? I'm not aware of any ISDN switch-types that would use a Facility message for initial call setup. On Wed, Jun 24, 2015 at 11:41 AM, Florian Mühl f...@trinitynetworks.de wrote: Hello, I really invested a lot of time in this case and do not have any idea what is going on. Probably there is someone who really can help me. Otherwise I am lost :D I have a cisco uc540 installed with two bri isdn interfaces which is working like a charme. 12 telephones with separate dld, auto attended and so on works well. did range is from 00 to 99. This week I was advised to install a separate did 33 for incoming faxes. Typically no problem but…. I wanted to configure another did 33 as fax onramp service and I am not able to get any signal to this did. I tried to reconfigure the working did 05 to did 33 and even the could not get any „answer“. I reconfigured from 33 to 05 again and everything works. I even reloaded the old config with 05 as fax an did a debug isdn q931 and have differnt outputs for unconfigured dids (unconfigured did 33 and unconfigured did 34 are completly different). There are no dialpeers or extensions with these numbers bit I get different q931 debug output for two unconfigured did. debug info: q931 debug for unconfigured did 33 025448: Jun 24 13:14:53.225: ISDN BR0/1/0 Q931: RX - FACILITY pd = 8 callref = N/A Facility i = 0x91A1150202D409020124300C300AA1053003020100820103 Protocol Profile = Remote Operations Protocol 0xA1150202D409020124300C300AA1053003020100820103 Component = Invoke component Invoke Id = 54281 Operation = AOCECharging Unit Called Party Number i = 0xC1, '7879338' Plan:ISDN, Type:Subscriber(local) 025449: Jun 24 13:14:53.225: ISDN BR0/1/0 **ERROR**: host_facility_invoke: HOST_FAC_INV: B-channel = D-channel call id 0x q931 debug for unconfigure did 34 00: Jun 24 07:21:35.343: ISDN BR0/1/1 Q931: RX - SETUP pd = 8 callref = 0x01 Sending Complete Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0x89 Exclusive, B1 Progress Ind i = 0x8183 - Origination address is non-ISDN Calling Party Number i = 0x2183, '8945244290' Plan:ISDN, Type:National Called Party Number i = 0xC1, '7879334' Plan:ISDN, Type:Subscriber(local) 004445: Jun 24 07:21:35.359: ISDN BR0/1/1 Q931: TX - CALL_PROC pd = 8 callref = 0x81 Channel ID i = 0x89 Exclusive, B1 Why is there any difference in the q931 debug information? Both lines are unconfigured I am completly clueless. Thank you very much Flo P.S: I even tried to get a Cisco SPA phone to work with did 33 - no way. Configured new did 35 and everything works. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Problems with incoming ISDN call / CME does not process the call - any clou? q931 debug inside
Hello, I really invested a lot of time in this case and do not have any idea what is going on. Probably there is someone who really can help me. Otherwise I am lost :D I have a cisco uc540 installed with two bri isdn interfaces which is working like a charme. 12 telephones with separate dld, auto attended and so on works well. did range is from 00 to 99. This week I was advised to install a separate did 33 for incoming faxes. Typically no problem but…. I wanted to configure another did 33 as fax onramp service and I am not able to get any signal to this did. I tried to reconfigure the working did 05 to did 33 and even the could not get any „answer“. I reconfigured from 33 to 05 again and everything works. I even reloaded the old config with 05 as fax an did a debug isdn q931 and have differnt outputs for unconfigured dids (unconfigured did 33 and unconfigured did 34 are completly different). There are no dialpeers or extensions with these numbers bit I get different q931 debug output for two unconfigured did. debug info: q931 debug for unconfigured did 33 025448: Jun 24 13:14:53.225: ISDN BR0/1/0 Q931: RX - FACILITY pd = 8 callref = N/A Facility i = 0x91A1150202D409020124300C300AA1053003020100820103 Protocol Profile = Remote Operations Protocol 0xA1150202D409020124300C300AA1053003020100820103 Component = Invoke component Invoke Id = 54281 Operation = AOCECharging Unit Called Party Number i = 0xC1, '7879338' Plan:ISDN, Type:Subscriber(local) 025449: Jun 24 13:14:53.225: ISDN BR0/1/0 **ERROR**: host_facility_invoke: HOST_FAC_INV: B-channel = D-channel call id 0x q931 debug for unconfigure did 34 00: Jun 24 07:21:35.343: ISDN BR0/1/1 Q931: RX - SETUP pd = 8 callref = 0x01 Sending Complete Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0x89 Exclusive, B1 Progress Ind i = 0x8183 - Origination address is non-ISDN Calling Party Number i = 0x2183, '8945244290' Plan:ISDN, Type:National Called Party Number i = 0xC1, '7879334' Plan:ISDN, Type:Subscriber(local) 004445: Jun 24 07:21:35.359: ISDN BR0/1/1 Q931: TX - CALL_PROC pd = 8 callref = 0x81 Channel ID i = 0x89 Exclusive, B1 Why is there any difference in the q931 debug information? Both lines are unconfigured I am completly clueless. Thank you very much Flo P.S: I even tried to get a Cisco SPA phone to work with did 33 - no way. Configured new did 35 and everything works. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CME w/ SIP Trunk
Following up for posterity.. Still fighting this one but a small bit of progress. So CME was trying to register both the ephone-dn extension and the E164 expanded number of the extension (which was the correct one) - so SP was putting us in timeout for trying to register invalid numbers. I figured out how to stop that with the number xxx no-reg command in each ephone-dn. Also, I had to re-write some headers in the REGISTER requests (to/from/request-uri and Authorization fields) with sip profiles. Also - as brian suggested, they told me the wrong username, should have not had dashes in it. After all that, still getting back 403 Authentication Failed - but its at least cleaned up and I'm not getting put in timeout. onward.. On Tue, Jun 23, 2015 at 6:06 PM, Ed Leatherman ealeather...@gmail.com wrote: Didnt seem to help but thats a good thought. Slogging it out with SP tomorrow again. I'm really puzzled about the SIP debugs.. its like i'm not getting all my SIP messages in there On Tue, Jun 23, 2015 at 3:29 PM, Brian Meade bmead...@vt.edu wrote: I'd try the username without the dashes first. On Tue, Jun 23, 2015 at 3:26 PM, Ed Leatherman ealeather...@gmail.com wrote: I did a packet cap and we are sending the SIP REGISTER, but its not showing up in sip debug?? really weird. anywhere I'm not binding SIP to my loopback address, i'm not getting SIP debugs for. So I am getting 403 back from SP after all, gonna double check username/passwords On Tue, Jun 23, 2015 at 3:16 PM, Brian Meade bmead...@vt.edu wrote: How about connecting via telnet over 5060? You may be having a TCP issue which is why you never see the Register sent. On Tue, Jun 23, 2015 at 3:09 PM, Ed Leatherman ealeather...@gmail.com wrote: Brian msu-tmp-access#sho sip-ua register status Line peer expires(sec) reg survival P-Associ-URI == === 2031120001 43 no normal 2031220003 43 no normal 2031320005 43 no normal 2031420007 43 no normal .. etc .. all no I can ping the sip-server from router so it appears to be able to resolve the name OK. On Tue, Jun 23, 2015 at 2:48 PM, Brian Meade bmead...@vt.edu wrote: What do you see for show sip-ua register status? Are you sure the gateway can resolve the sip-server via DNS? On Tue, Jun 23, 2015 at 2:32 PM, Ed Leatherman ealeather...@gmail.com wrote: Hello! I'm trying to get a SIP trunk out to a regional SP (Lumos) configured. I need to get CME setup to register numbers with their sip proxy, but the registration is not happening and i'm not seeing any register messages debugs from debug ccsip messages to troubleshoot from. So I think maybe CME isn't trying? What should trigger CME to try and register these numbers? My config looks like this (some ephones/ephone-dns up and registered) - authentication credentials were provided from Lumos. IOS 15.4(3)M2 msu-tmp-access#sh run | s sip-ua sip-ua credentials username 304-929-0300 password 7 blah realm sbc.ia.ntelos.net authentication username 304-929-0300 password 7 blah retry register 10 registrar dns:sbc.ia.ntelos.net:5060 expires 120 sip-server dns:sbc.ia.ntelos.net:5060 ! msu-tmp-access#sho run | s voice service voice service voip ip address trusted list ipv4 216.12.114.195 address-hiding allow-connections sip to sip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip bind control source-interface GigabitEthernet0/2 bind media source-interface GigabitEthernet0/2 registrar server options-ping 60 -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip -- Ed Leatherman -- Ed Leatherman -- Ed Leatherman -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CME w/ SIP Trunk
Another one for posterity.. So it seems like SIP REGISTER requests are only shown using debug ccsip noncall ... however there is no output from that command unless debug ccsip messages is also running. very confusing when I only has debug noncall on and didnt see any messages, but they were present in packet cap! On Wed, Jun 24, 2015 at 11:26 AM, Ed Leatherman ealeather...@gmail.com wrote: Following up for posterity.. Still fighting this one but a small bit of progress. So CME was trying to register both the ephone-dn extension and the E164 expanded number of the extension (which was the correct one) - so SP was putting us in timeout for trying to register invalid numbers. I figured out how to stop that with the number xxx no-reg command in each ephone-dn. Also, I had to re-write some headers in the REGISTER requests (to/from/request-uri and Authorization fields) with sip profiles. Also - as brian suggested, they told me the wrong username, should have not had dashes in it. After all that, still getting back 403 Authentication Failed - but its at least cleaned up and I'm not getting put in timeout. onward.. On Tue, Jun 23, 2015 at 6:06 PM, Ed Leatherman ealeather...@gmail.com wrote: Didnt seem to help but thats a good thought. Slogging it out with SP tomorrow again. I'm really puzzled about the SIP debugs.. its like i'm not getting all my SIP messages in there On Tue, Jun 23, 2015 at 3:29 PM, Brian Meade bmead...@vt.edu wrote: I'd try the username without the dashes first. On Tue, Jun 23, 2015 at 3:26 PM, Ed Leatherman ealeather...@gmail.com wrote: I did a packet cap and we are sending the SIP REGISTER, but its not showing up in sip debug?? really weird. anywhere I'm not binding SIP to my loopback address, i'm not getting SIP debugs for. So I am getting 403 back from SP after all, gonna double check username/passwords On Tue, Jun 23, 2015 at 3:16 PM, Brian Meade bmead...@vt.edu wrote: How about connecting via telnet over 5060? You may be having a TCP issue which is why you never see the Register sent. On Tue, Jun 23, 2015 at 3:09 PM, Ed Leatherman ealeather...@gmail.com wrote: Brian msu-tmp-access#sho sip-ua register status Line peer expires(sec) reg survival P-Associ-URI == === 2031120001 43 no normal 2031220003 43 no normal 2031320005 43 no normal 2031420007 43 no normal .. etc .. all no I can ping the sip-server from router so it appears to be able to resolve the name OK. On Tue, Jun 23, 2015 at 2:48 PM, Brian Meade bmead...@vt.edu wrote: What do you see for show sip-ua register status? Are you sure the gateway can resolve the sip-server via DNS? On Tue, Jun 23, 2015 at 2:32 PM, Ed Leatherman ealeather...@gmail.com wrote: Hello! I'm trying to get a SIP trunk out to a regional SP (Lumos) configured. I need to get CME setup to register numbers with their sip proxy, but the registration is not happening and i'm not seeing any register messages debugs from debug ccsip messages to troubleshoot from. So I think maybe CME isn't trying? What should trigger CME to try and register these numbers? My config looks like this (some ephones/ephone-dns up and registered) - authentication credentials were provided from Lumos. IOS 15.4(3)M2 msu-tmp-access#sh run | s sip-ua sip-ua credentials username 304-929-0300 password 7 blah realm sbc.ia.ntelos.net authentication username 304-929-0300 password 7 blah retry register 10 registrar dns:sbc.ia.ntelos.net:5060 expires 120 sip-server dns:sbc.ia.ntelos.net:5060 ! msu-tmp-access#sho run | s voice service voice service voip ip address trusted list ipv4 216.12.114.195 address-hiding allow-connections sip to sip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip bind control source-interface GigabitEthernet0/2 bind media source-interface GigabitEthernet0/2 registrar server options-ping 60 -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip -- Ed Leatherman -- Ed Leatherman -- Ed Leatherman -- Ed Leatherman -- Ed Leatherman ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] R: Mixed mode revert to unsecure
Great. Thank you again. Regards Alessandro - Messaggio originale - Da: Brian Meade bmead...@vt.edu Inviato: 24/06/2015 20:12 A: Alessandro Bertacco bertacco.alessan...@alice.it Cc: cisco-voip@puck.nether.net cisco-voip@puck.nether.net Oggetto: Re: [cisco-voip] Mixed mode revert to unsecure I've reverted on pre-10.x clusters fine with no issues. I haven't tried rolling back 10.x but it should work the same way on the backend. On Wed, Jun 24, 2015 at 2:10 PM, Alessandro Bertacco bertacco.alessan...@alice.it wrote: Hi Brian, and thank you very much, I will do that tomorrow night. Last question, have you tried that revert before, and did you note some strange behaviour after rebooting server, or all works fine? Thank you regards Alessandro *Da:* bmead...@gmail.com [mailto:bmead...@gmail.com] *Per conto di *Brian Meade *Inviato:* mercoledì 24 giugno 2015 16:34 *A:* Alessandro Bertacco *Cc:* cisco-voip@puck.nether.net *Oggetto:* Re: [cisco-voip] Mixed mode revert to unsecure You just need to run utils ctl set-cluster non-secure-mode on the publisher, then run file delete tftp CTLFile.tlv on each node (may not be needed, can't remember), then restart Cisco CallManager service and Cisco TFTP service on each node. Then you'll be able to re-configure auto-registration. On Wed, Jun 24, 2015 at 2:57 AM, Alessandro Bertacco bertacco.alessan...@alice.it wrote: Hi guys, for mistake I have enabled my 10.5 cucm in mixed mode using Cli command and with out using the token. Customer don't have any capf profile defined and no one use encryption, but obviously auto registration now is disabled, and for sure I've many other complications not needed. Can I easily revert back to Unsecure mode? Thanks. Alessandro ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] R: Mixed mode revert to unsecure
Hi Brian, and thank you very much, I will do that tomorrow night. Last question, have you tried that revert before, and did you note some strange behaviour after rebooting server, or all works fine? Thank you regards Alessandro Da: bmead...@gmail.com [mailto:bmead...@gmail.com] Per conto di Brian Meade Inviato: mercoledì 24 giugno 2015 16:34 A: Alessandro Bertacco Cc: cisco-voip@puck.nether.net Oggetto: Re: [cisco-voip] Mixed mode revert to unsecure You just need to run utils ctl set-cluster non-secure-mode on the publisher, then run file delete tftp CTLFile.tlv on each node (may not be needed, can't remember), then restart Cisco CallManager service and Cisco TFTP service on each node. Then you'll be able to re-configure auto-registration. On Wed, Jun 24, 2015 at 2:57 AM, Alessandro Bertacco bertacco.alessan...@alice.it mailto:bertacco.alessan...@alice.it wrote: Hi guys, for mistake I have enabled my 10.5 cucm in mixed mode using Cli command and with out using the token. Customer don't have any capf profile defined and no one use encryption, but obviously auto registration now is disabled, and for sure I've many other complications not needed. Can I easily revert back to Unsecure mode? Thanks. Alessandro ___ cisco-voip mailing list cisco-voip@puck.nether.net mailto:cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Mixed mode revert to unsecure
I've reverted on pre-10.x clusters fine with no issues. I haven't tried rolling back 10.x but it should work the same way on the backend. On Wed, Jun 24, 2015 at 2:10 PM, Alessandro Bertacco bertacco.alessan...@alice.it wrote: Hi Brian, and thank you very much, I will do that tomorrow night. Last question, have you tried that revert before, and did you note some strange behaviour after rebooting server, or all works fine? Thank you regards Alessandro *Da:* bmead...@gmail.com [mailto:bmead...@gmail.com] *Per conto di *Brian Meade *Inviato:* mercoledì 24 giugno 2015 16:34 *A:* Alessandro Bertacco *Cc:* cisco-voip@puck.nether.net *Oggetto:* Re: [cisco-voip] Mixed mode revert to unsecure You just need to run utils ctl set-cluster non-secure-mode on the publisher, then run file delete tftp CTLFile.tlv on each node (may not be needed, can't remember), then restart Cisco CallManager service and Cisco TFTP service on each node. Then you'll be able to re-configure auto-registration. On Wed, Jun 24, 2015 at 2:57 AM, Alessandro Bertacco bertacco.alessan...@alice.it wrote: Hi guys, for mistake I have enabled my 10.5 cucm in mixed mode using Cli command and with out using the token. Customer don't have any capf profile defined and no one use encryption, but obviously auto registration now is disabled, and for sure I've many other complications not needed. Can I easily revert back to Unsecure mode? Thanks. Alessandro ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Making a message appear anonymous/unknown via single inbox
i'm not sure if this is the right way to handle this, but I wrote a SIP
normalization script to remove the info on the CUCM side before the call is
sent over to CUC and in initial testing this seems to do the job.
for the archives...
M = {}
function M.outbound_INVITE(msg)
if msg:getHeader(Diversion)
then
local privacy = msg:getHeaderValueParameter(Remote-Party-ID, privacy)
if string.find(privacy, full)
then
msg:applyNumberMask(Remote-Party-ID, 0)
local rpid = msg:getHeader(Remote-Party-ID)
local rpiduri = string.match(rpid, (.+))
msg:modifyHeader(Remote-Party-ID, rpiduri)
msg:applyNumberMask(P-Asserted-Identity, 0)
local pai = msg:getHeader(P-Asserted-Identity)
local paiuri = string.match(pai, (.+))
msg:modifyHeader(P-Asserted-Identity, paiuri)
end
if string.find(privacy, name)
then
local rpid = msg:getHeader(Remote-Party-ID)
local rpiduri = string.match(rpid, (.+))
msg:modifyHeader(Remote-Party-ID, rpiduri)
local pai = msg:getHeader(P-Asserted-Identity)
local paiuri = string.match(pai, (.+))
msg:modifyHeader(P-Asserted-Identity, paiuri)
end
if string.find(privacy, uri)
then
msg:applyNumberMask(Remote-Party-ID, 0)
msg:applyNumberMask(P-Asserted-Identity, 0)
end
end
end
return M
On Tue, Jun 23, 2015 at 11:47 AM, Justin Steinberg jsteinb...@gmail.com
wrote:
in addition to the PSTN, they also want to have caller id blocked on calls
on-net to other Cisco phones. So we have certain CSS on Cisco phones that
route calls through translation patterns to set the restricted CLID flag
before the call rings the other Cisco phones.When we do this, the Cisco
phone called party does see 'Private' but if they don't answer the call and
it goes into voicemail then the voicemail message will still show the
calling party number.
On Tue, Jun 23, 2015 at 11:12 AM, Ryan Huff ryanh...@outlook.com wrote:
Justin,
For clarification, you have a user(s) that makes an outbound call from
call manager to the pstn (via a sip trunk to an itsp?) And some of them
want to block their Caller ID or mask it to anon?
Thanks,
Ryan
Original Message
From: Justin Steinberg jsteinb...@gmail.com
Sent: Tuesday, June 23, 2015 11:06 AM
To: Ted Nugent tednugen...@gmail.com
Subject: Re: [cisco-voip] Making a message appear anonymous/unknown via
single inbox
CC: Cisco VoIPoE List cisco-voip@puck.nether.net
Apologies for resurrecting an old thread here, but I was curious if there
was fixed in more recent versions of Unity Connection. I'm trying this
with 10.5(2)su1 with SIP trunk between UCM and UCXN and having the problem
where UCXN ignores the restriction settings configured on CUCM.
I'm thinking maybe I could make a LUA script on the UCM sip trunk to
UCXN, but looking for an easier solution.
In my situation, I have certain callers that want to have their CLID
blocked. So it is based on the placing the call, not the phone that
receives the call and I think this prevents me from using the two options
discussed here earlier.
Justin
On Thu, Jan 24, 2013 at 4:28 PM, Ted Nugent tednugen...@gmail.com
wrote:
Thanks Chris
I like option 2 as well since they only have 24 ports as it stands. I'll
give it a shot and let you know how it turns out. Thanks again!
On Thu, Jan 24, 2013 at 3:16 PM, Chris Ward (chrward) chrw...@cisco.com
wrote:
Hi Ted,
Seems like you are hitting some known limitations. The settings in CUCM
for restricting Calling Party Information don’t really eliminate it, it
seems to just mark it as “Restricted” so the information remains and CUC
seems hell-bent on finding calling party information.
I did find 2 solutions for you, both of which use SIP:
1) Setup an additional integration between CUCM and CUC that is
meant only for this anonymous line. In the SIP trunk config, you have to
uncheck the “Remote-Party-Id” under the “Call Routing Information”
section and make sure “Calling Line ID Presentation” is set to
Restricted under the “Outbound Calls” section. This will result in an
“Unknown CallerID” message in Outlook.
2) Setup a Loopback SIP trunk for calls to this specific
Anonymous DID. You only need one SIP trunk to point to CUCMs own IP
address, it acts as the outbound and inbound SIP trunk in this scenario.
Under the “Outbound Calls” section, you would need define a Calling Party
Transformation CSS that would have access to a transformation pattern that
matches all Calling parties (remembering that only calls to this anonymous
DID are affected) and mask the calling party with some obscure mask like
or you can put a . at the end of the pattern you match and then drop
all pre-dot if you want it blank. On the route pattern that points to this
loopback SIP trunk, you would modify the called party so that once the call
re-enters CUCM, it will be destined for the *7999 number (pulled from you
example before) and continues through the normal process to be forwarded to
voicemail.
Neither solution
