configuring Cisco 3911 SIP phones

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configuring Cisco 3911 SIP phones

Postby calldrop » Tue Jul 16, 2013 2:03 pm

Hi everyone, im pretty new to the whole Asterisk community.

I have already setup the server with a digium analog gateway and have configured Sip clients on my computer to make outbound calls ... (so everything works :) )

I have purchased 5 Sip phones (Cisco 3911), but cannot edit the configuration (Asterisk IP address, Sip name, ect ...)

Please advice me as to what i need to do.

thank you
calldrop
Newsterisk
 
Posts: 11
Joined: Tue Jul 16, 2013 1:59 pm

Re: configuring Cisco 3911 SIP phones

Postby calldrop » Tue Jul 16, 2013 3:17 pm

update: I have uploaded the 2 files:

SIPDefault.cnf
SIP001AA11B8254.cnf (the MAC of my ip phone is: 001AA11B8254)

to the tftpboot folder (i have check the tftp and it is working, i downloaded files to my computer from it)

the content of the file are as such:

SIP001AA11B8254.cnf
; phone-specific configuration file sample
line1_name : 0011
line1_authname : 0011
line1_password : qazxsw123456


SIPDefault.cnf
; sip default configuration file
#Image Version
image_version:SIP3951.8-1-2;
#Proxy server address
proxy1_address: 192.168.1.222;
proxy_register: 1;
#Default Codec
preferred_codec :g711ulaw

does everything look okay? am i missing anything?
calldrop
Newsterisk
 
Posts: 11
Joined: Tue Jul 16, 2013 1:59 pm

Re: configuring Cisco 3911 SIP phones

Postby calldrop » Wed Jul 17, 2013 6:09 am

I realised that i could telnet to the phone ...

and got these traces from the phone

CFGMGR[3477]: exec tftp_config_get process "192.168.1.222" XMLDefault.cnf.xml

[TFTP] TRANSFER_FILE_ERROR
CFGMGR: [3543]download filename "XMLDefault.cnf.xml" file not found



CFGMGR[3477]: exec tftp_config_get process "192.168.1.222" SEP001AA11B8254.cnf.xml

[TFTP] TRANSFER_FILE_ERROR
CFGMGR: [3543]download filename "SEP001AA11B8254.cnf.xml" file not found


turns out the filenames that i was using were wrong ...

and now it seems that the content of these files are also wrong ...

I need to find out the correct format of what goes into the config files ...

anyone out there can guide me in the right direction?
calldrop
Newsterisk
 
Posts: 11
Joined: Tue Jul 16, 2013 1:59 pm

Re: configuring Cisco 3911 SIP phones

Postby calldrop » Thu Jul 18, 2013 7:44 am

I have really tried EVERYTHING :(

nothing seems to work ... if anyone has a sample of working files, please share them.
-XMLDefault.cnf.xml
-SEPMAC.cnf.xml

I was able to upgrade to the latest version by telnetting to the phone and then TFTP to my server and getting the files (i couldnt upgrade the Load File though :( ) I'm now running on SIP3951.8-1-4-0

I would really appreciate someone's help here :?
calldrop
Newsterisk
 
Posts: 11
Joined: Tue Jul 16, 2013 1:59 pm

Re: configuring Cisco 3911 SIP phones

Postby calldrop » Fri Jul 26, 2013 4:34 am

OK, there has been some progress but I still haven't been able to get the phone registered :(

here is my SEPMAC.cnf.xml

<device xsi:type="axl:XIPPhone" ctiid="1566023366">
<deviceProtocol>SIP</deviceProtocol>

<devicePool>
<dateTimeSetting>
<dateTemplate>D-M-YA</dateTemplate>
<timeZone>Arabian Standard Time</timeZone>
<ntps>
<ntp>
<name>192.168.1.222</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>

<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>192.168.1.222</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>

<sipProfile>
<sipProxies>
<backupProxy></backupProxy>
<backupProxyPort></backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort></emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort></outboundProxyPort>
<registerWithProxy>true</registerWithProxy>
</sipProxies>

<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x-cisco-serviceuri-cfwdall</callForwardURI>
<callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
<callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
<meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
<abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
<rfc2543Hold>false</rfc2543Hold>
<callHoldRingback>2</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
</sipCallFeatures>

<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<timerRegisterExpires>3600</timerRegisterExpires>
<timerRegisterDelta>5</timerRegisterDelta>
<timerKeepAliveExpires>120</timerKeepAliveExpires>
<timerSubscribeExpires>120</timerSubscribeExpires>
<timerSubscribeDelta>5</timerSubscribeDelta>
<timerT1>500</timerT1>
<timerT2>4000</timerT2>
<maxRedirects>70</maxRedirects>
<remotePartyID>false</remotePartyID>
<userInfo>None</userInfo>
</sipStack>

<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>

<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>

<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>0011</featureLabel>
<proxy>192.168.1.222</proxy>
<port>5060</port>
<name>0011</name>
<displayName>0011</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>0011</authName>
<AuthUserName>0011</AuthUserName>
<authPassword>qazxsw123456</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>0011</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>

<line button="2">
<featureID>9</featureID>
<featureLabel>0022</featureLabel>
<proxy>192.168.1.222</proxy>
<port>5060</port>
<name>0022</name>
<displayName>0022</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>0022</authName>
<AuthUserName>0022</AuthUserName>
<authPassword>qazxsw123456</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>0011</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>

</sipLines>

<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>


</sipProfile>

<webAccess>1</webAccess>


<loadInformation>SIP3951.8-1-4a</loadInformation>

<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>

<certHash></certHash>
<encrConfig>false</encrConfig>


</device>


the phone is getting the right configuration now (when i telnet to it and do a "sip show all") here is what i get:

[admin]# sip show all

callerid_blocking : 2
call_manager1_addr : 192.168.1.222
call_manager2_addr :
call_manager3_addr :
call_manager4_addr :
call_manager5_addr :
call_manager1_sip_port : 5060
call_manager2_sip_port : 5060
call_manager3_sip_port : 5060
call_manager4_sip_port : 5060
call_manager5_sip_port : 5060
conMonDur : 120
line1_pkid : 0011
line2_pkid : 0011
line1_name : 0011
line2_name : 0022
line1_displayname : 0011
line2_displayname : 0022
line1_authname : 0011
line2_authname : 0022
line1_password : qazxsw123456
line2_password : qazxsw123456
messages_uri : *97
remote_party_id : 0
timer_invite_expires : 180
timer_register_expires : 3600
timer_keepalive_expires : 120



[admin]#


The issue now is that when i check the Console logs from the phone itself (use the phone's IP address on a browser to view this) I get this error:
0000211695 - SipCallAuthenticate()@807 missing AuthUserName or AuthPassword
0000211702 - SipCallAuthenticate()@807 missing AuthUserName or AuthPassword
0000211703 - nmm_ssmu_dn_state_register_fail()@939: Only one available server, phone doesn
't do failover or fallback operation
0000211703 - nmm_ssmu_dn_state_register_fail()@939: Only one available server, phone doesn
't do failover or fallback operation


So it seems that the phone thinks it doesn't have the AuthUserName or AuthPassword even though it shows up on the configuration.

Does anyone have any suggestions?

Any help what so ever will be greatly appreciated :D
calldrop
Newsterisk
 
Posts: 11
Joined: Tue Jul 16, 2013 1:59 pm

configuring Cisco 3911 SIP phones

Postby calldrop » Sun Aug 18, 2013 2:31 am

:!: :!:

(found a work around)

:!: :!:

I finally got the Cisco3911 phone to work with the Asterisk server!
I took such a long time because I had to order an old "dumb" hub so that i could sniff all the packets going in and out of the phone/Asterisk using wireshark---> it turned out that the phone was just not sending out the authentication password to Asterisk (even though it was available in its configuration).
The work-around is to basically remove the password from the extension configuration in Asterisk (which I have to mention is highly NOT recommended, since it means that any phone can connect to the network just by configuring the extension number).
Since this setup is currently just being used in my house, i am not too worried about it having such bad security. (i am not connected to an external network).
The next step is to now find out why the phone is not sending the password to Asterisk and get it to send it ... I'll keep you posted.
calldrop
Newsterisk
 
Posts: 11
Joined: Tue Jul 16, 2013 1:59 pm

Re: configuring Cisco 3911 SIP phones

Postby calldrop » Wed Aug 21, 2013 2:13 am

I finally got it to work ...

Cisco IP Phone fully registered and functional (with all features working)
Voice mail working including the VMI (voice mail indicator - red light on the phone)

All this with NO call manager or BVSM ... You just got to love ASTERISK.
Attachments
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Functional 3911 on Asterisk
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calldrop
Newsterisk
 
Posts: 11
Joined: Tue Jul 16, 2013 1:59 pm

Re: configuring Cisco 3911 SIP phones

Postby subvip7 » Wed Oct 09, 2013 1:54 am

calldrop - what was it that you did to fix the password login issue?
subvip7
Newsterisk
 
Posts: 1
Joined: Wed Oct 09, 2013 1:48 am

Re: configuring Cisco 3911 SIP phones

Postby calldrop » Fri Oct 25, 2013 2:29 pm

subvip7 wrote:calldrop - what was it that you did to fix the password login issue?


Hi subvip7, i didnt really fix the issue, its more of a "work-around" ... a really ugly work-around.

I basically removed the password from the configuration, so that the phone only asks to register the phone number. The disadvantage in this is that any phone can connect to the phone network without being authenticated. I am using these phones at home, and the network does not access the internet, so this method works well for me.

If you do find out how to resolve this issue, please do let me know :)
calldrop
Newsterisk
 
Posts: 11
Joined: Tue Jul 16, 2013 1:59 pm

Re: configuring Cisco 3911 SIP phones

Postby digivoip » Sat Apr 19, 2014 9:26 am

Hi All

I have the same issue with cisco cp-3911 it only works with asterisk if i remove the secret from sip conf!

Any news about this issue?
digivoip
Newsterisk
 
Posts: 1
Joined: Sat Apr 19, 2014 9:24 am

Re: configuring Cisco 3911 SIP phones

Postby rulgar19 » Fri Aug 29, 2014 4:19 am

Hi digivoip,

How did you registry you 7911g against Asterisk? I can made calls form 7911g to a softphone but not from a softphone to the 7911g.

the problem i have is with the Reg Contact, the extension sip softphone is correctly.
Reg. Contact : sip:333@192.168.10.6:59081;rinstance=04291c45c66e3ba0;transport=UDP


But the 7911g is emprty. How did it?

Thanks in advance.

regards
rulgar19
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Posts: 4
Joined: Thu Aug 28, 2014 12:41 am


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