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- About jan.klepal
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Member since
03-05-2003
13
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30
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1
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Created by
jan.klepal
in IP Telephony and Phones
in IP Telephony and Phones
10-11-2019
01:03 AM
10-11-2019
01:03 AM
OK, I have CP-7841-3PCC-K9 = which was accidentally upgraded to enterprise firmware. Can I revert it back to 3PCC without any cost? Document you linked states: For customers that are not part of the Flex plan, the transition from one call control platform to another will incur a cost. We are not part of the "Flex plan".
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Created by
jan.klepal
in IP Telephony and Phones
in IP Telephony and Phones
10-10-2019
09:23 AM
10-10-2019
09:23 AM
Unfortunately not. Cisco uses their proprietary SIP with Callmanager and Asterisk SIP stack can't handle that. We use IP phones with 3PCC firmware (for example CP-7841-3PCC-K9=). And again, unfortunately you can't flash non-3PCC IP phone with 3PCC firmware without buing additional conversion licence L-CP-E2M-XXXX SKU (as mentioned by @Ashish Patel). To make things even more funny, in case you accidentaly flash 3PCC phone with enterprise firmware you can't revert it back can revert it back by buing conversion licence. Thanks to Cisco licence policy you are struck with phone working only with Callmanager even you paid them extra money for non-Callmanager firmware. Unless you pay them another extra money for "conversion" licence. I loved Cisco products, now I try to avoid them whenever I can.
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10-22-2016
06:28 AM
Yes, there are two extensions (two SIP peers in Asterisk terms), one for Alice and second for Bob. 6900 and 7900 series phones can register two or more accounts without problems.
I've tried 10.3(1) and 11.5(1) with same result. I can't downgrade below 10.3(1) because IP phone rejects firmware with message invalid load. 10.3(1) and above firmwares are twice in size of previous firmwares so I assume they changed whole system and therefore it can't be downgraded. Unfortunetaly switch to 3PCC firmware is not possible too.
I need somebody with access to CUCM to check if Callmanager ignores invalid credentials for second line on same IP phone or if Cisco changed provisioning for 7800 series phones and it just requires some additional tag in XML provisioning file.
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10-22-2016
04:22 AM
Hi Leo,
here is complete provision file. I've checked consolelog messages in the phone and there are no error messages about configuration file parsing.
If single line is configured registration works correctly.
If two or more lines are configured, phone tries to register first line and it fails because REGISTER message contains credentials from last configured account in SIP Authorization header (From, To and Contact headers contains correct username as you can see in my original post). Phone never tries to register second account because registration of first account fail. In other words phone registers alice's account with bob's credentials.
I own the Asterisk, it rejects registration with 403 Forbidden (Bad auth) and logs:
[2016-10-22 11:59:55.552] WARNING[2237] chan_sip.c: username mismatch, have <alice>, digest has <bob>
It corresponds with REGISTER SIP message, see attached sip-register.txt.
--Edit:
In case I switch SIP transport protocol from TCP to UDP, phone will register successfully second line with bob's username and bob's credentials however it still tries to register first line with alice's username and bob's credentials.
Also tried to change alice to 1111 and bob to 2222, however same result, it still uses invalid credentials: username mismatch, have <1111>, digest has <2222>.
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Created by
jan.klepal
in IP Telephony and Phones
in IP Telephony and Phones
10-21-2016
01:53 PM
10-21-2016
01:53 PM
Hi,
I am testing Cisco CP 7841 phone as replacement for 6900 series IP phones. All works fine until two or more SIP accounts are provisioned. In such case telephone sends credentials for first account registration with Authorization header having credentials from second account. In case I add more accounts then credentials from last account are used.
Phone: 7841, Protocol: SIP, Tried with enterprise (non 3PCC) firmware: 10.3(1) and 11.5(1), SIP registrar: Asterisk 11
First account "alice" registration message:
REGISTER sip:192.168.222.1 SIP/2.0 From: <sip: alice @192.168.222.1> To: <sip: alice @192.168.222.1> Contact: <sip: alice @192.168.222.41:50015;transport=tcp> Authorization: Digest username=" bob ",realm="asterisk",uri="sip:192.168.222.1",response="***",nonce="5f830b34",algorithm=MD5
This register message is rejected beacuse fromuser != authuser.
Accounts are correct in SEPMAC.cnf.xml:
<sipLines> <line button="1" lineIndex="1"> <featureID>9</featureID> <featureLabel>alice</featureLabel> <proxy>USECALLMANAGER</proxy> <port>5060</port> <name>alice</name> <displayName>alice</displayName> <autoAnswer> <autoAnswerEnabled>2</autoAnswerEnabled> </autoAnswer> <callWaiting>3</callWaiting> <authName>alice</authName> <authPassword>alice</authPassword> <sharedLine>false</sharedLine> <messageWaitingLampPolicy>3</messageWaitingLampPolicy> <messagesNumber>alice</messagesNumber> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <contact>alice</contact> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>true</callerNumber> <redirectedNumber>false</redirectedNumber> <dialedNumber>false</dialedNumber> </forwardCallInfoDisplay> <maxNumCalls>2</maxNumCalls> <busyTrigger>1</busyTrigger> </line> <line button="2" lineIndex="2"> <featureID>9</featureID> <featureLabel>bob</featureLabel> <proxy>USECALLMANAGER</proxy> <port>5060</port> <name>bob</name> <displayName>bob</displayName> <autoAnswer> <autoAnswerEnabled>2</autoAnswerEnabled> </autoAnswer> <callWaiting>3</callWaiting> <authName>bob</authName> <authPassword>bob</authPassword> <sharedLine>false</sharedLine> <messageWaitingLampPolicy>3</messageWaitingLampPolicy> <messagesNumber>bob</messagesNumber> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <contact>bob</contact> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>true</callerNumber> <redirectedNumber>false</redirectedNumber> <dialedNumber>false</dialedNumber> </forwardCallInfoDisplay> <maxNumCalls>2</maxNumCalls> <busyTrigger>1</busyTrigger> </line> <sipLines>
Does anybody experience same problem? Can anybody with access to CUCM make SIP communication dump and confirm this issue?
Regards,
Jan
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Labels:
08-20-2015
06:50 AM
There is no official support, however telnet can be enabled by following steps (if you do so, you will probably void your warranty, God kills a kitten and other evil things happen): Log into switch via web interface. Go to Administration -> File Management -> Download/Backup Configuration/Log Select Save Action: backup, Source File Type: Running configuration file, Sensitive Data: Exclude and hit Apply button. Open downloaded running-config.txt in text editor and just after line "username cisco ..." add line ip telnet server and save the file. Go back to switch web interface and hit Done button to get back into Download/Backup Configuration/Log. Select Save Action: Download, Source File Name: choose your modified running-config.txt, Destination File Type: Running configuration file and hit Apply button. Telnet is enabled on port 6500... so from commandline run telnet <switch ip> 6500 and enjoy CLI :-) You can verify that the telnet is enabled in web interface under Security -> TCP/UDP Services. Look for line which begins with Telnet in TCP Service Table. Port telnet server is listening on is in Local Port column. Snip of running-config.txt: username cisco password encrypted xxxxx privilege 15 ip telnet server ! interface gigabitethernet1
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Created by
jan.klepal
in IP Telephony and Phones
in IP Telephony and Phones
01-10-2014
05:35 AM
01-10-2014
05:35 AM
I am dialing numbers via IP phone URL http:// <phone_ip>/CGI/Execute. XML POST request to dial number I use: <CiscoIPPhoneExecute> <ExecuteItem URL="Dial:1234"/> </CiscoIPPhoneExecute> Phone response to this request is: <CiscoIPPhoneResponse> <ResponseItem URL="Dial:1234" Data="Success" Status="0"/> </CiscoIPPhoneResponse> However telephone display only first digit and refuses to dial requested number. Tested on CP-6921 and CP-6945 phones with SIP load version 9.3.1.5 and 9-4-1-3. I also tried to add Priority="1" to ExecuteItem tag without success (even with priority 0 or 2). Finally I checked and then simplified dialplan just to: <DIALTEMPLATE> <TEMPLATE MATCH="...." Timeout="2"/> </DIALTEMPLATE> without any luck... dialplan seems to be OK as number dial by hand works well. Nothing interesting in phone log. In case I execute same request as above on 7945 phone it is executed as expected and phone dials number 1234 correctly. Only workaround I found for 6900 phones is to use EditDial followed by "press" softkeys: <CiscoIPPhoneExecute> <ExecuteItem URL="EditDial:1234"/> <ExecuteItem URL="Key:Soft4"/> <ExecuteItem URL="Key:Soft2"/> </CiscoIPPhoneExecute> Did anybody experienced same problem? Can somebody try to execute same request on 6900 series phone with SIP load to confirm its a BUG? Or am I just missing some additional parameter for tag or request header? Cheers, Jan
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Labels:
09-15-2004
01:40 AM
is there in AXL any way how to change lineGroups? I'm only able to list line groups but i want to add and remove its members. I'm looking for something like getLineGroup :-) Any idea? Tnx
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Labels:
Created by
jan.klepal
in Other Collaboration Subjects
in Other Collaboration Subjects
08-22-2004
11:22 AM
08-22-2004
11:22 AM
Yes, you are right, there is a chnage in RP registration. Thanks for hint, also thanks to Roger Reed from convergedsoftware.com who mailed me directly a piece of code. It should be useful for all migrating from CCM 3.x to 4.x, so here is it: CiscoAddress ciscoAddress = (CiscoAddress)ciscoProvider.getAddress(routePointString); Terminal[] terminals = ciscoAddress.getTerminals(); if(terminals == null){ throw new Exception("Route point address " + routePointString + " terminal array is null"); } ciscoRouteTerminal = (CiscoRouteTerminal)terminals[0]; ciscoRouteTerminal.register(null, CiscoRouteTerminal.NO_MEDIA_REGISTRATION); Thank you all for your support :)
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Created by
jan.klepal
in Other Collaboration Subjects
in Other Collaboration Subjects
08-16-2004
10:48 AM
08-16-2004
10:48 AM
We are using CCM 3.3 and I was testing if we can upgrade to CCM 4.0 We use application witten in Java for call control via JTAPI a CTI Route Points. Application runs properly with CCM 3.3. After upgrade to CCM 4.0(2)a all CTI Route points are out of service. When I list them in CcmAdmin, all has status "Not Found" and empty field where IP address normally appears. No exeption in our application happen, I can login, fetch devices registered with CCM etc. Just when the application try to register CTI Route point via registerRouteCallback() method, I can see in jtapi log file "%JTAPI-JTAPIIMPL-7-UNK:Address [Distributor_999:999(1,71)] out of service". - jtapi.jar is downloaded from CCM 4.0(2a) plugins - I tryed remove CTI Route Point, add new associate it with user and restart CCM without any luck - CTI Manager trace log files doesn't contain any messages about CTI Route Point errors or any clue even when I set log level to Detailed - Event viewer on CCM doesn't contain any errors or warnings about CTI or CCM service I am out of ideas where I should look for more clues... pleeease help :) Cheers, Jan
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Labels:
Created by
jan.klepal
in Other Collaboration Subjects
in Other Collaboration Subjects
03-07-2003
11:40 AM
03-07-2003
11:40 AM
I have tryed 56k modem connection trough ATA-186 in standard configuration; modem could connect only at 14.400, unfortunetaly real transfer speed was about 500 bytes per second. If you don't have modem passthrough feature, try to configure some parameters on voice port. Use G711 codec, disable echo cancellation, disable comfort noise generation, disable voice activity detection, set playout-delay to 200ms and make sure you have correctly configured QoS on IP network between voice gateways which modem call goes through.
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Created by
jan.klepal
in Other Network Architecture Subjects
in Other Network Architecture Subjects
03-05-2003
09:14 AM
03-05-2003
09:14 AM
There is no way how to check how many computers are behind NAT device. To check what MAC addresses are used on the line between your and customer's router simply type "show ip arp". But there will be only MAC addresses which are on segments directly connected to your router. By sniffing traffic on the line you can only guess how many computers are behind NAT device. For example count simulateous connections from NAT device to internet with destination port 80. Unfortunetaly it is very inaccurate...
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03-05-2003
07:45 AM
There is a very useful tool: http://www.cisco.com/cgi-bin/Support/Cmdlookup/home.pl All commands you mentioned should be in standard IP IOS image.
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10-11-2019
OK, I have CP-7841-3PCC-K9 = which was accidentally upgraded to
enterprise firmware. Can I revert it...
10-10-2019
Unfortunately not. Cisco uses their proprietary SIP with Callmanager and
Asterisk SIP stack can't ha...
10-22-2016
Yes, there are two extensions (two SIP peers in Asterisk terms), one for
Alice and second for Bob. 6...
10-22-2016
Hi Leo, here is complete provision file. I've checked consolelog
messages in the phone and there are...
Created by
jan.klepal
in IP Telephony and Phones
10-21-2016
10-21-2016
Hi, I am testing Cisco CP 7841 phone as replacement for 6900 series IP
phones. All works fine until ...
08-20-2015
There is no official support, however telnet can be enabled by following
steps (if you do so, you wi...
Created by
jan.klepal
in IP Telephony and Phones
01-10-2014
01-10-2014
I am dialing numbers via IP phone URL http:///CGI/Execute. XML
POST request to dial number I use: Ph...
09-15-2004
is there in AXL any way how to change lineGroups? I'm only able to list
line groups but i want to ad...
Created by
jan.klepal
in Other Collaboration Subjects
08-22-2004
08-22-2004
Yes, you are right, there is a chnage in RP registration. Thanks for
hint, also thanks to Roger Reed...
Created by
jan.klepal
in Other Collaboration Subjects
08-16-2004
08-16-2004
We are using CCM 3.3 and I was testing if we can upgrade to CCM 4.0 We
use application witten in Jav...
Created by
jan.klepal
in Other Collaboration Subjects
03-07-2003
03-07-2003
I have tryed 56k modem connection trough ATA-186 in standard
configuration; modem could connect only...
03-05-2003
There is no way how to check how many computers are behind NAT device.
To check what MAC addresses a...
Public Statistics
| Date Registered | 03-05-2003 07:27 AM |
| Date Last Visited | 10-15-2019 10:52 AM |
| Total Messages Posted | 13 |
| Total Helpful Votes Received | 30 |
Helpful Votes From
| User | Helpful Count |
|---|---|
| 5 | |
| 5 | |
| 5 | |
| 5 | |
| 5 |
