Cisco 8861 on Asterisk

I have a Cisco 8861 loaded with the SIP firmware which I can get to work with Asterisk for one line but cannot get it to work for more than one line. It tries to register and can receive inbound calls but cannot make outbound calls. Has anyone got this phone working with Asterisk/Elastix?

can receive inbound calls but cannot make outbound calls
Can receive inbound calls but not make outbound calls?
Check your inbound and outbound proxy configurations in your Asterisk box. Your inbound proxy details should be exactly the same as your outbound proxy details.

Similar Messages

Configuration of Cisco 2911 for Asterisk

Hi all
I use Cisco 2911 for Asterisk phone system communicate with external.
However, sometime I can make call in and out. Sometime, just call in or out. Sometime, cannot make any call.
I think it is the NAT, PAT and ACL in Cisco 2911 problem. This Cisco is also a gateway to internet for users.
Please any advice
Thanks a lot
Here is the configuration:
Router#show run
Building configuration...
Current configuration : 1981 bytes
! Last configuration change at 20:06:06 UTC Thu Nov 14 2013
! NVRAM config last updated at 15:04:59 UTC Tue Nov 5 2013
! NVRAM config last updated at 15:04:59 UTC Tue Nov 5 2013
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
enable secret 5 xxxxx
no aaa new-model
memory-size iomem 20
no ipv6 cef
ip source-route
ip cef
multilink bundle-name authenticated
crypto pki token default removal timeout 0
license udi pid CISCO2911/K9 sn FTX1603AH9C
interface Embedded-Service-Engine0/0
no ip address
interface GigabitEthernet0/0
description internal-LAN
ip address 172.x.x.x 255.255.0.0
ip nat inside
ip virtual-reassembly in
duplex auto
speed auto
interface GigabitEthernet0/1
no ip address
duplex auto
speed auto
interface GigabitEthernet0/1.1
encapsulation dot1Q 11
ip address 172.16.x.x 255.255.240.0
interface GigabitEthernet0/2
description internet
ip address 50.240.x.x 255.255.255.240
ip nat outside
ip virtual-reassembly in
duplex auto
speed auto
ip forward-protocol nd
no ip http server
no ip http secure-server
ip nat inside source list 100 interface GigabitEthernet0/2 overload
ip route profile
ip route 0.0.0.0 0.0.0.0 50.240.x.x
ip route 0.0.0.0 0.0.0.0 172.10.0.30 name ROUTE-VPN-REMOTE
ip route 172.16.240.0 255.255.254.0 172.10.x.x
access-list 100 permit ip 172.10.0.0 0.0.255.255 any
access-list 100 permit ip 172.16.240.0 0.0.0.255 any
access-list 100 permit udp any any range 5004 5090
access-list 100 permit udp any any range 10000 20000
control-plane
line con 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
login
transport input all
scheduler allocate 20000 1000
end

There are some VSP where they do the NAT. If your VSP (like mine) do the NAT, then you need to globally disable NAT in your Asterisk.
My VSP also recommends I disable ALG on my router.
So you need to ask you VSP.

Connecting Cisco 7906 to asterisk

Hi
I have cisco cucm ,
There are cisco phones (7906) i just want to try with asterisk server . so i created two vlans , one for endpoints other for asterisk server .
What is the best practice running cucm and other pbx in an enterprise network .
Do i need to add switcport voice vlan x for the endpoint .
What is exactly switch port voice means ?
Thanks

can you confirm the below setup is ok ?
for cucm server
interface Vlan50
ip address 192.168.50.2 255.255.255.0
standby 50 ip 192.168.50.1
standby 50 priority 200
standby 50 preempt
for asterisk server
interface Vlan40
ip address 192.168.40.2 255.255.255.0
standby 50 ip 192.168.40.1
standby 50 priority 200
standby 50 preempt
on distribution side
for cisco phone to connect to cucm
interface Vlan50
ip address 10.0.50.2 255.255.255.0
standby 50 ip 10.0.50.1
standby 50 priority 200
standby 50 preempt
for cisco phone to connect to asterisk server
interface Vlan40
ip address 10.0.40.2 255.255.255.0
standby 50 ip 10.0.40.2
standby 50 priority 200
standby 50 preempt
Access layer
for asterisk phone
interface GigabitEthernet3/3
switchport access vlan 3
switchport voice vlan 40
storm-control broadcast level 40.00
storm-control action shutdown
for cisco phone
interface GigabitEthernet3/3
switchport access vlan 3
switchport voice vlan 50
storm-control broadcast level 40.00
storm-control action shutdown
Thanks

Cisco7942 - Background beep tone when no incoming audio

Hello,
I m using Cisco7942 in SIP Mode, after capturing SIP messages and RTP audio streams, I found that:
-When Cisco7942 doesnt receive audio (example: when its leaving a message on a VoiceMail Box), it automatically plays a "background beep", like "beep .... beep .... beep ..... I confirm this "beep" audio tones, come from the phone, they arent on the RTP streams.
¿Is there any way to disable this feature, may be a parameter in the SEP...xml file ?
I ve googled without answer ...
Many Thanks in advance !

Hi Marcos,
Ok ... sorry, i m new on this community, I thought it covers all Cisco products and related inter operability like Cisco 5xx and Asterisk (i saw some doc about that on this community).
If it is not supported, ok :)
Thanks for answering by the way ...

Connect Cisco IP Phone 7941G-GE to asterisk

I have a cisco IP PHONE 7941G-GE already with SIP firmware, but when i upload the config file with asterisk extension, etc the phone keeps saying "Registering" and never finish registering.
My SIP Configuration File : http://pastebin.com/SB6XPiXD
There are any log that i cant use to see if asterisk is blocking the phone or something like that ?
Can anyone help me out ?
Regards ;)

Note - you started the thread in SPA IP Phones / XML Phone Applications
But 7941G is not member of SPA product line and your's question has nothing to do with XML Application at all. So you are off-topic here. Please consider drop of this thread here (red button on top of page) and create it in more appropriate forum (you should add information mentioned by Leo Laohoo as well as information mentioned by me in next paragraph).
According the question itself - best way to analyze registration problem is packet capture. You need to catch SIP REGISTER packets sent from your phone to Asterisk and responses sent in opposite direction. Use a packet catcher (tcpdump, wireshark or so) or display them on Asterisk's console (sip set debug peer ...)

Cisco Asterisk auto provisioning

Hi all,
I have setup asterisk pabx and the extension phones (SPA504g) have configured to auto provision by asterisk server.
The system was working fine, but today one extension phone got unregistered. I have reset the phone and test it again, it gives "Checking DNS". Other phones are working fine.
Do anyone has an idea about this issue?
Thanks,
Yasiru

Hi yasiru and welcome to the Cisco Home Community!
The SPA504G is handled by the Cisco Small Business Support Community.
For discussions about this product, please go here. https://supportforums.cisco.com/community/netpro/small-business
cheers!
OnnagokorO

Cisco 7962G phone is not gettting register or send any traffic sip to register !!

Hi Guys i have cisco 7962 G phone that is not going to get register !
i have the conf working on asteriks when asterisk in the same subnet.
but with this phone its not registering to destination outside the network !
im sure no firewall and it can reach 8.8.8.8 and dns okay and flashed correctly with sip firmware .
=========
so i will paste here the config files :
<Default>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>xxx.0.220</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation434 model="Cisco 7942">SIP42.9-2-1S</loadInformation434>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>
and here is sepmac file :
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D.M.Y</dateTemplate>
<timeZone>W. Europe 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>192.168.0.220</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>true</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>g711alaw</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled>false</natEnabled>
<natAddress></natAddress>
<phoneLabel>103</phoneLabel>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>103</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>103</name>
<displayName>103</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>103</authName>
<authPassword>abc123</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>103</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>true</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
<commonProfile>
<phonePassword>1234</phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP42.9-2-1S</loadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>1</pcPort>
<settingsAccess>1</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<sshAccess>0</sshAccess>
<sshPort>22</sshPort>
<webAccess>0</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<versionStamp></versionStamp>
<userLocale>
<name>English_United_Kingdom</name>
<langCode>en</langCode>
</userLocale>
<networkLocale>Germany</networkLocale>
<networkLocaleInfo>
<name>Germany</name>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL></servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>2</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
</device>
as u see the ip is private
if i set it to external pbx , the phone dont register
how can u help me with enable loggin to debug on dat phone ??
cheers

phone has public ip and didnt work !
we use softphone and get register but the cisco .....not working !!
about my problem
u can see here :
http://www.voip-info.org/boards/view/topic/38915
it has no solution so far .
i do debug and tcpdump on the pbx and i cant see any incoming resgteration trials.
cheers

How to change the background image on IP Phone 8861

Hi Concern,
I face some problem on changing background image on IP Phone 8861. I prepare
.png file as per below doc.
Full size image - 800 pixels (width) X 480 pixels (height).
Thumbnail image - 159 pixels (width) X 109 pixels (height).
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/8811_8841_8851_8861/10_5/english/adminguide/P881_BK_A92A3B94_00_adminguide-8811_41_51_61-10_5/P881_BK_A92A3B94_00_adminguide-8811_41_51_61-10_5_chapter_01010.html#P881_RF_C8927D88_00
I upload the png and List.xml on cucm 10.5 as per above doc.
List.xml Example
<CiscoIPPhoneImageList>
<ImageItem Image="TFTP:Desktops/800x480x24/TN-Fountain.png" URL="TFTP:Desktops/640x480x24/p1-800X480.png"/>
<ImageItem Image="TFTP:Desktops/800x480x24/TN-FullMoon.png" URL="TFTP:Desktops/640x480x24/p2-159X109.png"/>
</CiscoIPPhoneImageList>
Now i go to phone set and check new background picture not shows it says "no wallpaper server" something like that.
Please help on the issue.
Thanks
Afzal

Hi Jaime,
Thanks is done ..
On dx650 phone background change is same procedure or different.
File size is different i know but list.xml is same or different ?
Or i can upload only one list.xml file just line add for dx650 phone.
Thanks
Afzal

Third party Video conference device conferencing with Cisco SX80

Hi ALL,
We have Cisco BE6K (Call Manager 10.5), CTS-SX80-IPST60-K9, DX650, 7861, 8861 IP phone, Cisco 2901 Voice Getaway, 3750 Switch and ASA. internet was connected on ASA. SIP connectivity (For DID) connected to Cisco 2901 Voice Getaway.
I need to know cisco best practice and implementation guide on below question:-
1. How Third party Video conferencing units (LIFESIZE and Polycom) can conference and call with Call Manager IP Phones and SX80. Third party device will not register on MY CUCM.
2. We have two site. On Sit-1 has own Call Manager 10.5 and Site-2 has own Call Manager 8.X both are reachable. Now How they make a call?
Thanks
Gazi Afzalur Rahman

HI Ayodeji,
Thank you for valuable update.
1. On B2B solution :-
a) Are Third party device needs to register on CUCM or not. What the B2B solution says because that device not register on CUCM?
b) Which connectivity i need Internet and SIP or Only Internet?
c) On other site are they need SIP connectivity or Internet ?
2. Sip trunk between the two clusters are we need SIP connectivity on both site or need only reachability? On this scenario are we need VOICE RT because we have different data connectivity router for both site.
Can You refer any CVD guide of implementation ..
Thanks In Advance
Gazi Afzalur Rahman

SPA504G SCCP to Asterisk - SEP$mac.cnf.xml - Define CallManager entry

Hi
I am using the SPA504G and the asterisk chan_sccp_b module to connect the phone to asterisk
The communications works well.
My problem is the options of SEP$mac.cnf.xml vs the XMLDefault.cnf.xml
Almost all the doco is about zerotouch etc is for SIP, but I am using SCCP
In the sccp environment if a SEP$mac.cnf.xml exists for the phone, it reads nothing else. (But I ant work out how to set the callmanager server), so it just loops reloading.
If I set the callmanager entries in XMLDefault.cnf.xml it connects but I cant set the ntp server/timezone.
I would prefer to create individual SEP$mac.cnf.xml files, can anyone tell me how to set the callmanager entry ?
Thanks in advance
Mark

Hi Godfrey,
Could you please collect packet capture from the back of the phone when it tries to register using the link:
https://supportforums.cisco.com/document/44741/collecting-packet-capture-cisco-ip-phone
If the IP phone port is not spanned, then you have to collect the sniffers from the switchport where the phone is connected to. You have to span the switch port and collect the sniffer captures
~Amit

!!cisco ATA 186, back to back.SOS

hi there, just enlighten me on cisco ATA 186 back to back config, my network is like:
phone---ATA----Switch---ATA---Phone,wnat to know the sample config, if this not possible, fine ,then let me know what shud i configure to get a dial tone,as im getting a busy tone than a dial tone, or lastly what shud i do to reset the ATAs to factory default settings,
cheers
shukky

If your ATA is running skinny right now then I think it is not possible (because no callmanager in the picture).
But if you load your ATA with a SIP image then in the SIP proxy filed, you can specify the IP address of the other SIP ATA and then you should be able to call from one ATA to another.
I haven't tried that with the ATA, but done with the 7940 phones and it worked.
To get the dial-tone, you must configure a number for the ATA first. (UID)
To reset the ATA for factory defaults
http://lists.digium.com/pipermail/asterisk-users/2003-August/017474.html
Hope this would help.
Thanks

What does asterisk mean in show clock

Hello,
I noticed there is an asterisk(*) in show clock. No idea what it means. I think there should be no asterisk here normally.
sensor# sh clock
*09:09:28 GMT+08:00 Mon May 10 2010
Does anybody has the information about that?
Thanks,
-Alejin

The "asterisk" means that you have not configured NTP clock for your IPS and the clock has been configured manually.
Here is the command for your reference:
http://www.cisco.com/en/US/docs/security/ips/7.0/command/reference/crCmds.html#wp485412
Hope that answers your question.

Cisco SPA502 G - DHCP & Call problem

    Good day.
     I have many cisco spa 502G phone in my network which working correctly. But one phone doesn't working via DHCP. If I using static IP, I can ping phone, or login in on web-interface, but when i was trying upgrade firmware through Firmware Updater, I see error - upgrade failed, No Response From Spa. Phone firmware version is 7.4.3a .
     End the most important problem is that phone can't work correctly with my asterisk-server. If i receive call from anybody, i can do this only system keys nearly phone-screen. I can't call from this telephone ((
Please, community help, and sorry for my english

Hello Sergey,
To wich firmware do you want to upgrade? For some firmwares you have to do this in steps, you can not go directly to 7.5.4 form the frimware that you are using.
When you put the phone back to DHCP can it be that the phone tries to get an ip adres that is allready in use. Chek the DHCP pool etc to check the leases. At our office some phones say that they are initializing network for a long time and when we check the problem is that the phone wants to get an ip adres that is allready in use.
You can check your other phones web ui to see if an duplicated ip adres is causing the problem that you can not use the phone with DHCP.
Kind Regards
Richard

Configure BEKEM36 on a 8861

Hi community,
HOW can i configure the buttons on the BEKEM36 on a 8861 phone with CUCM 10.5?
On the KEMs with 89xx this was quite easy, i just used directly the phone configuration page on the callmanager.
What i read actually is that i have to use an account an access the self service portal, so i can set it up as a end user by myself.
Even if i set this up, and access that page, i cannot see an option there for the KEM on that phone. Only one to configure speeddials.
Is cisco kidding? Shall really the enduser configure that? Or do i miss a configuration option? I think the last one...
Thanks for any input,
Andreas

whenever you need to trunk to a switch and be able to have more than the limit of physical interfaces. For instance an ASA 5510 allows you to have 100 VLAN interfaces.
Whenever you need to setup more than on DMZ.

SPA3102 to Asterisk IVR

Hello,
i have this setup, everything works OK but when i introduce IVR to the setup i can't get IVR to work. The problem is probably in dailing rule that passes PSTN calls to asterisk. Atm on SPA i have (S0<:[email protected]>) so when phone rings call is transfered to 205 extension. In dialing rule in asterisk i tryed setting up the DID based incoming rule where 12345 was DID number and dialing plan on SPA was (S0<:12345>), but had no sucess on picking the call by IVR nor extension if i put in dialing rule. Some help would be appriciated
Regards!

I am not too familiar with setting up asterisk server with SPA devices but I see already a problem in your settings. I don’t see the relation of your SPA unit with regards with the number 12345? Is the DID number 12345 also relating with the extension 205 or with another extension or any extension in your VoIP network? What should happen when someone calls in or calls out DID 12345? What should happen when someone calls in or calls out DID 205? I think you have also a problem with the dial plan you made for the SPA unit. You only fix this if you will know the relation of the DID with regards to extensions. The dial plan (S0<:12345>) clearly is a dead end dial plan since you only substituted all possible dial option of the IP phone to be DID 12345. You didn’t indicate what happens after this. You can effectively make the dial plan if you know the relation of each component.
Please discuss the call path you wanted to achieve and let me see if we can point out the problem in you dial plan configurations.
Other than that I suggest contacting Cisco Tech support to further look into your concern. I believe this unit belongs to the business series devices that Cisco is now supporting. Try to go to this link for the other business series devices and the site where you can get hold of Cisco for support: http://www.cisco.com/web/products/linksys/index.html

Cisco 8861 on Asterisk

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