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- About johnhart
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Member since
01-15-2012
24
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01-24-2019
11:19 PM
Hi IOS SMEs, can anyone please confirm expect behaviour of the work around outlined in this bug report, which is consistent with the configuration I posted. Here is bug link: https://bst.cloudapps.cisco.com/bugsearch/bug/CSCtl58680 In the text of link it says: "This is an enhancement to just drop the packet in these cases and not the entire UDP session and child connections" My expectation of added the service-policy: <<-- snippet -->> class-map type inspect sip match-any allow-violations match protocol-violation policy-map type inspect sip allow-violations class type inspect sip allow-violations allow policy-map type inspect self->out class type inspect self->out inspect service-policy sip allow-violations class class-default drop log <<-- snippet -->> would be to not drop the packet and let it pass through. Am I wrong in this thinking ? Thank you. Regards, John.
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01-24-2019
12:31 AM
Hi Cisco IOS SMEs, there have been some prior posts on this subject, but they are very old and relate to much less recent IOS versions: here: https://community.cisco.com/t5/firewalls/zbfw-quot-sip-protocol-violations-quot/m-p/1251998 I am running IOS 15.5(3)M7 on 2901 ISR Router and am in process of trying to get SIP Trunk configured with CME on the router, but has issue with this and so started to test with SW sip user agent (Blink Pro). My trunk provider is using Cisco Broadworks SIP Server. As part of testing I have put SIP devices in DMZ with public IP address so as to avoid any potential NAT/SIP complications. Based on reading other posts relating to getting SIP_PROTOCOL_VIOLATION reports (via log) I have configured the following set of definitions into my configuration: <<start of snippet>> class-map type inspect sip match-any SIP-MESSAGE match protocol-violation class-map type inspect match-all SIP-FW-PROTOCOL match protocol sip policy-map type inspect sip SIP-ACTION class type inspect sip SIP-MESSAGE log allow policy-map type inspect POLICY-DMZ-OUT class type inspect SIP-FW-PROTOCOL inspect service-policy sip SIP-ACTION class class-default drop log <<end of snippet> So even when I have configured this I am getting the following result: 004654: Jan 24 08:25:14.844 UTC: %AIC-4-SIP_PROTOCOL_VIOLATION: SIP protocol violation (Content length invalid / Non-SIP MSG recvd) - dropping udp session 203.XX.XX.40:62509 45.XX.1XX.232:5060 on zone-pair ZP-DMZ-OUT class SIP-FW-PROTOCOL My testing has found that the violation occurs during INVITE exchange. REGISTER completes successfully, without any SIP Protocol Violations. So the documented work around for protocol violations itself does not appear to be working. Can anyone please advise. As the response is coming from Cisco Broadwork server, I am very surprised that Cisco IOS FW is not handling traffic ... Thanks in advance for any suggestions. John.
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08-29-2012
04:17 AM
Hi Peter, I was waiting for Mountain Lion aware VMWare Fusion release arrived before testing this again. With the arrival of VMWare Fusion 5, I have retested a Mountain Lion download with HTTP Inspection on and changes to buffer size as per your original note and all goes ok, downloaded 4GB without a hickup. Thanks very much for providing the corrective configuration. Regards, John.
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08-11-2012
07:23 PM
Hi Peter, thanks very much for the suggestion. I did a check on the 2901 and the config has very small (default) allocations: parameter-map type ooo global tcp reassembly timeout 5 tcp reassembly queue length 16 tcp reassembly memory limit 1024 tc p reassembly alarm off As per your suggestion I have update the sizes (the 2901 has 2GB RAM) moved HTTP application inspection back up to it overides straight tcp protocol inspection. parameter-map type ooo global tcp reassembly timeout 5 tcp reassembly queue length 512 tcp reassembly memory limit 16384 tcp reassembly alarm off I have a couple of further Apple Mac's that need to be updated to Mountain Lion so will test download again when updating these machines. Cheers, John.
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07-26-2012
06:54 PM
Ciscoers, as suspected this appears to be a problem with ZBFW. As a work around I have moved HTTP inspection down to the end of my policy list, so TCP protocol policy take priority of HTTP application policy and now people can download again. So the work around for the time being appears to be to disable HTTP inspection. I am very surprised that I appear to be the first person who has reported a problem here, as this is a major web site that is having a problem with HTTP inspection. I hope that cisco responds with a patch or particular configuration resolution. Regards, John.
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07-26-2012
03:57 PM
Cisoers, I have a repating 2901 router failure when people attempt to download Apple Mac OS X Moutnain Lion upgrade from App Store. The 2901 just hangs following getting a series of ZBFW packet drop failures: 001928: Jul 26 22:37:18.783 UTC: %APPFW-4-HTTP_PROTOCOL_VIOLATION: HTTP protocol violation (0) detected - session 192.168.223.109:49310 184.25.254.67:80 on zone-pair ZP-PRIVATE-OUT class ccp-protocol-http appl-class ccp-http-blockparam 001929: Jul 26 22:37:20.871 UTC: %APPFW-4-HTTP_DEOBFUSCATION: Deobfuscation signature (15) detected - session 192.168.223.109:49369 66.235.138.44:80 on zone-pair ZP-PRIVATE-OUT class ccp-protocol-http appl-class ccp-http-blockparam 001930: Jul 26 22:37:22.779 UTC: %FW-6-DROP_PKT: Dropping tcp session 192.168.223.130:49217 184.31.204.244:443 on zone-pair ZP-PRIVATE-OUT class ccp-insp-traffic due to Stray Segment with ip ident 0 The failure results in the ACT Light stopping to blink and the SYS Light remains on solid Green and the entire router hangs. I cannot SSH to it, all logging to console stops and the only way I can recover the router is by powering it off and on again. This is very alarming as this is a very common download site and I am finding router is hanging consistently and repeatly when people go there. Does anyone have any suggestions? This looks like a major bug in IOS. Regards, John.
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Created by
johnhart
in IP Telephony and Phones
in IP Telephony and Phones
07-07-2012
06:55 PM
07-07-2012
06:55 PM
Hi Paulo, I know that INVITE is sip specific mechanism to establish a "call" so while COR is not applied to sip operations I assume it does apply to logical call operations. I presume, from the fact that no-one has provided a definitive response to my posting, that there is no way to define a default COR list with CME... Which in turn means that CME has a security hole so big that it is not viable for use as a general and publiclly exposed SIP Proxy... It looks like it is back to "opensips" for public sip gateway. Cheers, John.
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Created by
johnhart
in IP Telephony and Phones
in IP Telephony and Phones
07-05-2012
05:10 PM
07-05-2012
05:10 PM
Hi, the configuration area you should be looking at is called "Class of Restriction" or COR. The COR implements a lock/key model where phone and dial-peers are given keys/locks definitions (essentially matching outgoing/incoming COR identifier). I found the most understandable description of this in the (Cisco Press) book: "Cisco Voice Gateways and Gatekeepers" - Chapter 12. This provides a much clearer description of how COR works that the ios example on Cisco site.... which are very hard to follow. I am currently trying to solve a simillar problem via COR, so cannot provide definitive answer, but believe this the right place to start digging. Cheers, John.
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Created by
johnhart
in IP Telephony and Phones
in IP Telephony and Phones
07-05-2012
04:47 PM
07-05-2012
04:47 PM
Hi IOS and CME users, I have got a simple CME configuration up and running on 2901, which supports: 1. REGISTER & INVITE of SIP phones within LAN 2. International Calling via external SIP Service Provider using "dial-peer" to SIP Trunk 3. National & Local Calling via Linksys/Cisco SPA3102 using "dial-peer" to SIP Trunk. I am still having issues with getting access to this from WAN side of router, but before I solve this problem, I need to ensure that I have correctly configured "Class of Restriction" configuration in place to avoid "Toll Fraud" (ie internet user sending in INVITE which goes through dial-peer and thus allows anyone to make international or local calls at my expense). I have added cor definitions to both phones and dial-peers: <Sample COR Config> dial-peer cor custom name international name national name local name emergency name toll-free name internal name private ! dial-peer cor list authourised member international member national member local member emergency member toll-free member internal member private ! dial-peer cor list national member national member local member emergency member toll-free ! dial-peer cor list internal member internal member private ! dial-peer cor list international member international ! dial-peer cor list external member internal ! dial-peer cor list staff member local member emergency member toll-free member internal member private ! dial-peer cor list private member private ! dial-peer cor list emergency member emergency ! dial-peer cor list local member local ! dial-peer cor list toll-free member toll-free ! voice register pool 1 id mac 0000.0002.0003 number 1 dn 1 core incoming 1 authourised 1 615555 <- Setup COR to allow full access core outgoing internal 1 615555 voice-class code 1 username frogb password XXXX ! voice register pool 4 id mac 0000.0001.0002 number 1 dn 4 cor incoming internal 1 6117777 <- Setup COR to constrain access cor outgoing private 1 6117777 voice-class codec 1 username froga password XXXX ! ! dial-peer voice 13 voip corlist outgoing national <- Setup COR so only incoming with "national" key can access this dial peer description national-61Nxxxxxxxx translation-profile outgoing outbound-national preference 5 destination-pattern 61[2-9]........ b2bua session protocol sipv2 session target ipv4:200.30.200.30:5061 voice-class codec 2 voice-class sip localhost dns:spa.FROGHOP.COM voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte ! <<End of Config Example>> However I have not been able to find where in CME do you provide a default COR definition, which would apply to someone doing a call in (INVITE) to CME via internet ie: INVITE: From: 999@ANY-DOMAIN.com To: 615555@FROGHOP.com Where FROGHOP.com is CME. As the INVITE does not come from Registered User, ther COR is empty and so CME will let the call request through, irrespective of what COR definitions are on the "dial-peers" or "voice register pool" defined numbers. I would like to have a default COR which restricts access to "internal" only. I know that with SRST this could be achived via: call-manager-fallback cor outgoing internal default <- Make "internal" the default outgoing cor cor incoming internal default <- Make "internal" the default ingoing core How can I acheive a simillar default configuration with CME?? Thanks in advance for any help. John.
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Created by
johnhart
in IP Telephony and Phones
in IP Telephony and Phones
06-28-2012
01:12 AM
06-28-2012
01:12 AM
Hi Adam, thanks for response. I do not have any "voice source-group" items in my configuration. I have since done more testing and played around with the ZBFW configuration. I found that to get SIP in and out network you need to have: SELF -> OUT-ZONE sip inspect OUT-ZONE -> SELF sip inspect OUT-ZONE -> IN-ZONE sip inspect (where I have Loopback0 terminator for SIP & NAT) PRIVATE-ZONE -> OUT-ZONE sip inspect (this is where NAT'ed 192.168.x.x network is) I am now able to Register over the Internet but, only from some networks... For example if I got into an office which as Class B addresses on its Intranet then I can Register from there via the Blink client but not the iPhone Groundwire/Acrobits client. If I go into iPhone Groudwire/Acrobits client and configure the Account/Hacks Settings to set the sending port to 5060 then it works ok. If I connect my Mac to network via iPhone tethering then I cannot Register via Blink client from Mac. So it appears that the problem may be with NAT or filtering within Service Provder network.. All these variable make it very hard to diagnose the problem. Now the problem I am having is that while I can do register from some external (Internet) networks... I cannot get INVITE to work... I will post some more into after I have done some more testing. John.
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Created by
johnhart
in IP Telephony and Phones
in IP Telephony and Phones
06-25-2012
05:05 AM
06-25-2012
05:05 AM
IOSers & VoIPers, I have done some more testing on the remote phone and have found that the remote phone does not appear to be getting any response back from CME. I getting the input messagse and have configured ZBFW with SIP inspect from OUT-ZONE to SELF as I could see that UDP messages to port 5060 where getting dropped. What is the behvaior of ZBFW for the return SIP message? If a SIP REGISTER or INVITE is received on port 5060 does ZBFW inspect mean that the port is then opened to allow the return reponse which can be on any UDP port to get past the ZBFW and back to the OUT-ZONE? Also I addedb 'bind into sip configuration as documentation seems to indicate that this is needed: < > ! voice service voip ip address trusted list ipv4 0.0.0.0 0.0.0.0 allow-connections sip to sip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip bind control source-interface Loopback0 registrar server expires max 600 min 60 localhost dns:FROGSTOMP.com ! < > Here is lastest SIP Message trace: < > 000308: Jun 25 11:40:19.510 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: REGISTER sip:FROGSTOMP.com SIP/2.0 Via: SIP/2.0/UDP 10.169.174.195:2526;branch=z9hG4bKFZvENjUgCu9E2als;rport Contact: ;expires=0 Max-Forwards: 70 From: "Joe Toad" 613333@FROGSTOMP.com >;tag=B6C4A9291D65343D263E88C1D1ECC345 Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY, MESSAGE Supported: replaces, path User-Agent: Acrobits Softphone Business/2.3 To: "Joe Toad" 613333@FROGSTOMP.com > Expires: 0 Call-ID: 2100FCC8292E2B193740F66A68E0B56D9ACE8E9A CSeq: 1163 REGISTER Content-Length: 0 000309: Jun 25 11:40:19.510 UTC: //78/50DB48C6803E/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.169.174.195:2526;branch=z9hG4bKFZvENjUgCu9E2als;rport;received=110.150.231.102 From: "Joe Toad" 613333@FROGSTOMP.com >;tag=B6C4A9291D65343D263E88C1D1ECC345 To: "Joe Toad" 613333@FROGSTOMP.com >;tag=404F4C-13C0 Date: Mon, 25 Jun 2012 11:40:19 GMT Call-ID: 2100FCC8292E2B193740F66A68E0B56D9ACE8E9A Server: Cisco-SIPGateway/IOS-12.x CSeq: 1163 REGISTER Content-Length: 0 < > Thank you to any ZBFW/SIP guru who can help. John.
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Created by
johnhart
in IP Telephony and Phones
in IP Telephony and Phones
06-24-2012
05:19 PM
06-24-2012
05:19 PM
Hi VoIPers & Sippers, After a false start I have now got a very basic CME configuration (with 2901) going which: 1. Provides Internet Facing SIP Proxy 2. Supports generic SIP clients (Mac OS X Blink & iPhone Acrobit/Groundwire and Grandstream GXV3175) 3. Allows me to call SIP extensions within an internal 192.168.X.X network However I cannot get my iPhone to REGISTER successfully from outside (via public Internet) Here is the core of my config: <ios VoIP Config> voice service voip ip address trusted list <-- I have deliberately turned off "ip address trusted list" as I am using ZBFW to handle security ipv4 0.0.0.0 0.0.0.0 allow-connections sip to sip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip <- I have not put a bind command, as SIP Proxy is acting as application gateway btw public/private network and I want it to listen on both interfaces registrar server expires max 600 min 60 localhost dns:FROGHOP.com ! voice class codec 1 <- I want the negoitation of CODECs to be done by endpoints via SDP as this reduces load on gateway and is also more flexible codec preference 1 transparent ! voice class codec 2 <- This set of CODECS will be used for B2BUA gatewaying to external SIP Service Providers and my own SPA3102 ATA for PSTN codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729br8 codec preference 4 g729r8 codec preference 5 g723ar53 codec preference 6 g723ar63 codec preference 7 g723r53 codec preference 8 g723r63 codec preference 9 g726r16 codec preference 10 g726r24 codec preference 11 g726r32 ! ! voice register global mode cme max-dn 20 max-pool 25 load 8961 8961/sip8961.9.2-2SR1-9 <--- These are just here because the documentation indicate the should be. load 7942 7942-62/SIP42.9-2-1S <-- My network is 100% SIP with no SCCP and no firmware downloading/management authenticate register authenticate realm FROGHOP.com timezone 47 time-format 24 date-format D/M/Y tftp-path flash: create profile sync 0014049064062229 ! voice register dn 1 <- This is the sample roaming DN, which I want to beble to use to get/place SIP calls while: at home, in office or travelling number 613333 allow watch name Joe Toad label jt-sip ! voice register dn 2 number 6113333 allow watch name LineA Home label linea ! voice register dn 3 number 6123333 allow watch name LineB Home label lineb ! …. ! voice register pool 1 id mac BEE0.BEE1.BEE2 <-- Dummy mac address for the roaming case. number 1 dn 1 voice-class codec 1 username joe password XXXXXXXX ! voice register pool 2 id mac 000A.9255.9097 number 1 dn 2 voice-class codec 1 username linea password XXXXXXXX ! voice register pool 3 id mac 000B.8233.9B86 number 1 dn 3 voice-class codec 1 username lineb password XXXXXXX ! .... <End VoIP Config> I turned on "debug ccsip messages" and got the following trace: Which hightlights the problems is that CME is returning a "404 Not Found" error, when the REGISTER is coming in via internet while it works fine when coming from the internal PRIVATE 192.168.X.X network. I presume that I am missing something very basic here, that I need to do to ensure that the REGISTER from the Internet goes through ok. <SIP Log TRACE> Received: REGISTER sip:FROGHOP.com SIP/2.0 Via: SIP/2.0/UDP 10.197.31.18:2525;branch=z9hG4bKfaUSJlXb1d982vqa;rport Contact: <sip:613333@10.197.31.18:2525;rinstance=50094A52>;expires=0 Max-Forwards: 70 From: "Joe Toad" <sip:613333@FROGHOP.com>;tag=8AB69B07421347565B0F51ACE8E78AB1 Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY, MESSAGE Supported: replaces, path User-Agent: Acrobits Softphone Business/2.3 To: "Joe Toad" <sip:613333@FROGHOP.com> Expires: 0 Call-ID: 2100FCC8292E2B193740F66A68E0B56D9ACE8E9A CSeq: 962 REGISTER Content-Length: 0 001110: Jun 24 10:24:44.746 UTC: //109/99899ED2805F/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.197.31.18:2525;branch=z9hG4bKfaUSJlXb1d982vqa;rport;received=1.140.3.65 From: "Joe Toad" <sip:613333@FROGHOP.com>;tag=8AB69B07421347565B0F51ACE8E78AB1 To: "Joe Toad" <sip:613333@FROGHOP.com>;tag=7DA624-977 Date: Sun, 24 Jun 2012 10:24:44 GMT Call-ID: 2100FCC8292E2B193740F66A68E0B56D9ACE8E9A Server: Cisco-SIPGateway/IOS-12.x CSeq: 962 REGISTER Content-Length: 0 001111: Jun 24 10:24:48.066 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: REGISTER sip:FROGHOP.com SIP/2.0 Via: SIP/2.0/UDP 10.197.31.18:2525;branch=z9hG4bKfaUSJlXb1d982vqa;rport Contact: <sip:613333@10.197.31.18:2525;rinstance=50094A52>;expires=0 Max-Forwards: 70 From: "Joe Toad" <sip:613333@FROGHOP.com>;tag=8AB69B07421347565B0F51ACE8E78AB1 Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY, MESSAGE Supported: replaces, path User-Agent: Acrobits Softphone Business/2.3 To: "Joe Toad" <sip:613333@FROGHOP.com> Expires: 0 Call-ID: 2100FCC8292E2B193740F66A68E0B56D9ACE8E9A CSeq: 962 REGISTER Content-Length: 0 001112: Jun 24 10:24:48.066 UTC: //109/99899ED2805F/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.197.31.18:2525;branch=z9hG4bKfaUSJlXb1d982vqa;rport;received=1.140.3.65 From: "Joe Toad" <sip:613333@FROGHOP.com>;tag=8AB69B07421347565B0F51ACE8E78AB1 To: "Joe Toad" <sip:613333@FROGHOP.com>;tag=7DA624-977 Date: Sun, 24 Jun 2012 10:24:48 GMT Call-ID: 2100FCC8292E2B193740F66A68E0B56D9ACE8E9A Server: Cisco-SIPGateway/IOS-12.x CSeq: 962 REGISTER Content-Length: 0 <End of SIP TRACE> Could the ZBFW be causing a problem with external access? Currently I have policy for: OUT-ZONE -> SELF == SIP Inspect (to allow incoming calls to hit CME) SELF -> OUT-ZONE == SIP Inspect (to allow CME to establish call with external SIP Service Providers) Thanks in advance to anyone who can help. Cheers, John.
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06-23-2012
05:37 AM
Thanks Chris, yes I did a bit more reading and as you indicated I just needed to accept the license prompt. Previously I set up my CME configuration via Cisco Configuration Professional and it set things up as "mode cme". But as I have had problems with this configuration and the SIP REGISTER, I wiped the configuration and started again via ios command line. With the command line configuration I did not do a "mode cme" and so the configuration seems to have defaulted to "mode cme-srst" which has resulted in reason for prompt. My reading indicates that as long as I am within limit of "max-pool 25" then I should be ok license wise, however I have since set "mode cme" as I believe this reflects what I am trying to do, which is to create a SMB SIP hub which handles REGISTER and local SIP network as well as allowing me to create SIP trunks out to SIP Service Providers and my own Cisco SPA for PSTN integration. Thanks for the assurance. Cheers, John.
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06-22-2012
06:12 PM
Hi Cisco Unified Comms, I have a Cisco 2901 for which I have recently purchased and installed a SL-29-UC-K9 License PAK. However when I go into voice register global configuration I get the following license notice: # voice register global # max-pool 20 PLEASE READ THE FOLLOWING TERMS CAREFULLY. INSTALLING THE LICENSE OR LICENSE KEY PROVIDED FOR ANY CISCO PRODUCT FEATURE OR USING SUCH PRODUCT FEATURE CONSTITUTES YOUR FULL ACCEPTANCE OF THE FOLLOWING TERMS. YOU MUST NOT PROCEED FURTHER IF YOU ARE NOT WILLING TO BE BOUND BY ALL THE TERMS SET FORTH HEREIN. Use of this product feature requires an additional license from Cisco, together with an additional payment. You may use this product feature on an evaluation basis, without payment to Cisco, for 60 days. Your use of the product, including during the 60 day evaluation period, is subject to the Cisco end user license agreement http://www.cisco.com/en/US/docs/general/warranty/English/EU1KEN_.html If you use the product feature beyond the 60 day evaluation period, you must submit the appropriate payment to Cisco for the license. After the 60 day evaluation period, your use of the product feature will be governed solely by the Cisco end user license agreement (link above), together with any supplements relating to such product feature. The above applies even if the evaluation license is not automatically terminated and you do not receive any notice of the expiration of the evaluation period. It is your responsibility to determine when the evaluation period is complete and you are required to make payment to Cisco for your use of the product feature beyond the evaluation period. Your acceptance of this agreement for the software features on one product shall be deemed your acceptance with respect to all such software on all Cisco products you purchase which includes the same software. (The foregoing notwithstanding, you must purchase a license for each software feature you use past the 60 days evaluation period, so that if you enable a software feature on 1000 devices, you must purchase 1000 licenses for use past the 60 day evaluation period.) Activation of the software command line interface will be evidence of your acceptance of this agreement. ACCEPT? [yes/no]: Given that I have a valid UC license why is it telling me I need additional license? Prior to purchasing the UC license to enable the UC evaluation licence so I could start using UC and establish a correct configuration. Is a license issue causing me to have this problem with getting SIP phone to REGISTER: https://supportforums.cisco.com/message/3665941#3665941 Here is the show license result from router: License UDI: ------------------------------------------------- Device# PID SN ------------------------------------------------- *0 CISCO2901/K9 XXXXXXXXX Technology Package License Information for Module:'c2900' ----------------------------------------------------------------- Technology Technology-package Technology-package Current Type Next reboot ------------------------------------------------------------------ ipbase ipbasek9 Permanent ipbasek9 security securityk9 Permanent securityk9 uc uck9 Permanent uck9 data None None None Thanks in advance. Regards, John.
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- Tags:
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- requirements
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Created by
johnhart
in IP Telephony and Phones
in IP Telephony and Phones
06-22-2012
03:25 AM
06-22-2012
03:25 AM
Hi Mohammed, thanks for your suggestions. Even when I updated the SIP config: voice service voip allow-connections sip to sip no supplementary-service h450.2 no supplementary-service h450.3 no supplementary-service h450.7 no supplementary-service sip moved-temporarily fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip bind control source-interface Loopback0 registrar server expires max 600 min 60 ! ! voice register global mode cme source-address 200.30.200.190 port 5060 max-dn 20 max-pool 35 load 8961 8961/sip8961.9-2-2SR1-9 load 7942 7942-62/SIP42.9-2-1S authenticate register authenticate realm frogdrop.com time-format 24 date-format D/M/Y tftp-path flash: create profile sync 0002689782282013 ! Changes include: # voice service voip # supplementary-service sip refer # sip # bind control source-interface Loopback0 I am still getting the REFER & 500 Internal Server Error generated, so am unable to register. Also as I added in the bind, which result in binding to the public facing Loopback0 interface, I had to change the Blink SIP client to do register on this interface (as it could not longer contact the PRIVATE interface on 192.168.X.X network. To your question about the ZBFW setting, I have configured SIP inspection and can have no problems getting response from the interfaces, as long as I ensure that the Blink SIP Proxy settings are always the same as the "bind control source-interface" setting. As per my earlier post, I do not realy want to do a "bind control source-interface" as this means that I cannot have a SIP listener on both the public and the private interfaces, which I need for: 1. Allowing general SIP INVITE calls to enter SIP via Public Interface 2. Allow remote users to do SIP REGISTER via Public Interface 3. Allow communications with external SIP provider for international calls 4. Allow my internal SIP users to use the PRIVATE interface to do SIP REGISTER 5. Allow my internal SIP clients to do intra-office calls by directly calling local SIP numbers. NOTE: that the REFER message has the following contents: notify_display Max Phones Exceeded 10 0 1 Does the "Max Phones Exceeded" have any significance??? How can I have exceeded by maximum no of phones, when I have not even managed to get a single phone to register? Thanks for your help. Regard, John.
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01-24-2019
Hi IOS SMEs, can anyone please confirm expect behaviour of the work
around outlined in this bug repo...
01-24-2019
Hi Cisco IOS SMEs, there have been some prior posts on this subject, but
they are very old and relat...
Created by
johnhart
in IP Telephony and Phones
07-07-2012
07-07-2012
Hi Paulo,I know that INVITE is sip specific mechanism to establish a
"call" so while COR is not appl...
Created by
johnhart
in IP Telephony and Phones
07-05-2012
07-05-2012
Hi,the configuration area you should be looking at is called "Class of
Restriction" or COR.The COR i...
Created by
johnhart
in IP Telephony and Phones
07-05-2012
07-05-2012
Hi IOS and CME users,I have got a simple CME configuration up and
running on 2901, which supports:1....
Created by
johnhart
in IP Telephony and Phones
06-28-2012
06-28-2012
Hi Adam,thanks for response.I do not have any "voice source-group" items
in my configuration.I have ...
Created by
johnhart
in IP Telephony and Phones
06-25-2012
06-25-2012
IOSers & VoIPers,I have done some more testing on the remote phone and
have found that the remote ph...
Created by
johnhart
in IP Telephony and Phones
06-24-2012
06-24-2012
Hi VoIPers & Sippers,After a false start I have now got a very basic CME
configuration (with 2901) g...
Created by
johnhart
in TelePresence and Video Infrastructure
06-24-2012
06-24-2012
Hi VoIPers & Sippers,I have moved this to here:
https://supportforums.cisco.com/thread/2156306As que...
06-23-2012
Thanks Chris,yes I did a bit more reading and as you indicated I just
needed to accept the license p...
Public Statistics
| Date Registered | 01-15-2012 01:29 PM |
| Date Last Visited | 02-01-2019 03:43 PM |
| Total Messages Posted | 24 |
| Total Helpful Votes Received | 3 |
