vicidial.org

Hello;

GoAutoDial CE 2.1, 2.6.18-238.9.1.el5.goPAE (SMP), Intel(R) Core(TM) i7-3770 CPU @ 3.40GHz, Single Machine.

I am facing a problem to let Cisco IP Phone 7942G register on Asterisk. The firmware has been downloaded from the TFTP successfully and currently I am running this load SIP42.9-3-1SR2-1S*

I feel that there is a problem in the SEPMAC.cnf.xml but really I do not know which one to be used exactly. Basically, there is some effect that appears on the Phone (for example, it is appearing the extension on the button), but the Phone is not able to register. I tried to ssh or even http or https to the phone but I can not access it. Although I configured the ssh in the SEPMAC.cnf.xml as following:

<sshUserId>admin</sshUserId>
<sshPassword>cisco</sshPassword>

Anyone tried to register Cisco 7942G on Asterisk? Which SEPMAC.cnf.xml was used?

How I can access the Phone via ssh or http to be able to see the logs and understand what is happening?

Note: I tried to see the sip debug on the CLI, but did not see any traffic is reaching from the IP Phone to asterisk, although I see the extension appears on the IP Phone button and if I changed the extension, then it is changing.

By the way: this SEPMAC.cnf.xml is existed on cisco website? Is it specialized for each Phone type (does it differs from Cisco IP Phone 7940 to 7942 to 7960)?

Appreciate the help as really I am sticked at this point and not able to move forward.

Thanks in advance.
Regards
Bilal

Sorry Bilal, this is completely unrelated to vicidial. It is entirely possible you will get a response here, but it is not very likely and if I were you I would not wait around and hope. This question would be much better asked on an Asterisk forum or a forum for cisco or tftp phones. The Vicidial system is designed to use any device that can communicate with Asterisk, of course. But the Asterisk community is huge compared to the Vicidial community and your "help base" is much larger out there. Of course, once you have your phone registered and communicating with Asterisk, we can help you with pretty much everything else ... but unless someone happens to have some experience with that particular phone, it is unlikely that there will be any help forthcoming.

That being said: if you were to post the rest of your xml file (or a link to the full text if it is large ...) with any passwords removed, we may be able to offer a guess or two. But seriously, there has to be another forum that is "on point" with this topic. It would be helpful if you found that and posted a link back here (I'm sure some others will eventually have similar issues, but I've not seen this model phone nor have my clients or I'd have direct experience with it already!).

Hello William;

I know that it is better to post on asterisk, but I posted here because u know that goautodial is using asterisk 1.4 where asterisk forum might find this old, moreover that vicidial is using its own asterisk version.

Meanwhile, I downgraded the firmware to become SIP42.8-3-5S and it is appearing on asterisk that it is register but on the phone it self, it is appearing registering !

The xml file is:

<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>admin</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>China Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>pool.ntp.org</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>192.168.10.10</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<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>g711ulaw</preferredCodec>
<natEnabled>false</natEnabled>
<phoneLabel>501</phoneLabel>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>501</featureLabel>
<proxy>192.168.10.10</proxy>
<port>5060</port>
<name>501</name>
<displayName>501</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>501</authName>
<authPassword>d0nt105</authPassword>
<sharedLine>false</sharedLine>
<contact>501</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
<loadInformation>SIP42.8-3-5S</loadInformation>
</device>

Regards
Bilal

i note that the profile has two user/pass areas. one for "user" and one for "authName" and they have different passwords. It is much safe when "unsure" of which setting will affect what to use the same user/pass everywhere and if that works, then begin modifying them to see which one is appropriate where and what it affects.

also: you stated that both the phone and asterisk show it registered ... I don't see where the problem is if that is true.

It is resolved. It was related to the nat, I have to make it nat=no and disable it from the xml file. Also, I used firmware sip 8 and not 9.

Now, I am stick at the timezone, it is delaying 3 hours.
The country is Kuwait, what should I place the settings in the xml file?

<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>Kuwait</timeZone>
<ntps>
<ntp>
<name>pool.ntp.org</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>


What should I place instead of Kuwait?

Regards
Bilal

You cannot just put a random value in the timezone. Find other users settings for this or perhaps try a GMT value. What was in the field before you modified it to Kuwait? That will give a hint as to what should be here, but it is not valid to "just plop a value in". There is a limited number of values, and you'll need to get ahold of that list if you cannot "guess" properly.

it was:

<timeZone>China Standard/Daylight Time</timeZone>

What can I place?

Kuwait time is GMT+3.

Regards
Bilal

Perhaps you could get some example time zones here (one may fit your needs ...):

http://bit.ly/147YOhJ


Look what i found there: http://www.cisco.com/en/US/docs/voice_i ... #wp1056604

bghayad wrote:It is resolved. It was related to the nat, I have to make it nat=no and disable it from the xml file. Also, I used firmware sip 8 and not 9.

Now, I am stick at the timezone, it is delaying 3 hours.
The country is Kuwait, what should I place the settings in the xml file?

<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>Kuwait</timeZone>
<ntps>
<ntp>
<name>pool.ntp.org</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>


What should I place instead of Kuwait?

Regards
Bilal

Hi Bilal,

I was wondering if you can help.

I'm trying to get my Cisco 7960 to work with Goautodial 3.0. How do I get the phone to register to one of the extensions, 8001 for example?

I've tried to create a second line on one of my Cisco 7960's which is registered to a Trixbox server on the same IP range. I enter the username, password and IP address of the Goautodial server but the extension (8001) just has an "X" next to it. I've looked in the extensions-vicidial.conf and turned NAT off but it still doesn't want to register.

In Trixbox it downloads a config file and registers with the Trixbox server via the phones MAC address.

Any ideas?

Thanks

Andy