error i getting when sip client try to register with asterisk server
[Mar 14 15:41:21] NOTICE[6429]: chan_sip.c:28073 handle_request_register: Registration from ‘sip:rajen@192.168.0.17’ failed for ‘192.168.0.18:35132’ - Wrong password
help me out.
Thanks,
rajen
Seems to be a case of wrong password.
And if you have changed the password in config, did you reload the Asterisk?
–Satish Barot
satish4asterisk@gmail.com
I’m getting the same error for a Cisco 7962 running on the AsteriskNow v3.0.1 x64 distro (FreePBX v2.11.x and Asterisk 11). My phone is running the SIP42.8-4-4S firmware, and I shall attach my XMLDefault.cnf.xml and SEP$mac.cnf.xml configs below. (Of course, the two Digium D40’s work flawlessly with minimal to no configuration needed).
Also, I have another issue with any lines containing the “{$ext.line.1}” variable never get populated. I can manually edit the setions in the specific SEP"MAC".cnf.xml file as its created for each Cisco extension, but that kinda defeats the point of using the FreePBX gui to provision phones. On of the sections is near the end of the file:
{$ext.line.1}
{$ext.line.1}
Everything else populates correctly from the variable placeholders.
I’m to the point I hate Cisco phones, regardless of how many other peeps have them implemented and running. Any assistance with this would be GREATLY appreciated, especially if it boils down to me missing something in the configs that worked fine under AsteriskNow v1.7.x x64. Thanks in advance.
XMLDefault.cnf.xml - 7962
–>
2000
2427
2428
192.168.1.85
P0S3-8-12-00
P0S3-8-12-00
SIP41.9-2-3S
SIP41.8-5-2S
SIP41.8-5-2S
SIP41.8-5-2S
SIP42.8-4-4S
SIP42.8-4-4S
SIP45.8-4-2S
SIP45.8-4-2S
SIP70.8-4-2S
SIP70.8-0-3S
SCCP70.8-3-1S
SEP$mac.cnf.xml - 7962
SIP
admin
cisco
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>192.168.1.85</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
SIP42.8-4-4S
false false 0 1 0 0 0 0 <webAccess>1</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
United_States
<networkLocaleInfo>
<name>United_States</name>
<version>1.0.0.0-1</version>
</networkLocaleInfo>
1
http://192.168.1.85/cisco/services/authentication.php
http://192.168.1.85/xmlservices/PhoneDirectory.php
http://192.168.1.85/xmlservices/index.php
http://192.168.1.85/xmlservices/index.php
96
0
96
4
0
3804
false
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x--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>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled></natEnabled>
<natAddress></natAddress>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<phoneLabel>{$displayname.line.1}</phoneLabel>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>{$ext.line.1}</featureLabel>
<name>{$ext.line.1}</name>
<displayName>{$displayname.line.1}</displayName>
<contact>{$ext.line.1}</contact>
<proxy>192.168.1.85</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>{$ext.line.1}</authName>
<authPassword>{$secret.line.1}</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
Rajen,
If you are running either the AsteriskNOW v2.x.x or v3.x.x distro, under Applications/Extensions, select your specific extension associated with your Cisco 79xx phone. Scroll down to the Device Options section, modify the “secret” that is auto-generated when creating the extension to be NO MORE THAN NINE (9) CHARACTERS IN LENGTH (alpha-numeric is fine). Perform a hard-reset of the handset, and it should become operational at this point.
If it does not, you may need to modify the specific SEP$mac.cnf.xml file associated with the extension that resides in your /tftpboot/ folder. Seek out the line in the XML file that contains password, and then soft-reset the phone. Hope this helps, it worked so far for me.
And in my own biased opinion, Cisco phones suck ass. Get a pair of Digium’s phones. I have two and it was essentially close to “zero” effort provisioning these pieces of hardware. The only reason I trouble myself with these Cisco clunkers is by proxy of what my employer wishes to deploy in our organization.
Best of luck to you sir.
[quote=“butlerkevind”]
And in my own biased opinion, Cisco phones suck ass. Get a pair of Digium’s phones. I have two and it was essentially close to “zero” effort provisioning these pieces of hardware. [/quote]
I shared this with some others internally who work on phones, and we all had a good laugh. Thanks 
Malcolm,
Outstanding! Glad I brought some comedic pause to the crew there in Huntsville.
I am still new to the whole Asterisk/VoIP/Linux community (Microsoft weaned and siloed originally). In my year and a half travel seeking help, at times it seems akin to asking the firing order of cylinders in an engine, without specifically being shown EXACTLY how one must go about implementing such a task, but having all the tools present to do the job and not fully understanding their specific roles in completing the task.
And as I exclaimed prior, working with the Digium phones was a breeze. Almost as easy as plug and play in the Microsoft ecosystem, if and when they get it perfected. May want to pass that type of knowledge onward to Redmond. 