Cisco 9971 phone configuration working example with setup tips

FreePBX Tips and Tricks
81

I have a 7942 and cant get it to register to FreePBX. Anyone with working configs or tips onhow they got their’s working?

regards

82

I Have Two Cisco 9971 IP Phone that is working Fine with CUCM 8.6. But Now want to configure with FreePBX. I already took it all files from CUCM Manager and Put it external TFTP server. When i saw logs of TFTP server there All files start fine and completed fine except CTLSEP.tlv

Now Problem is When CUCM is working then Phones is registered but when i shutdown Communication Manager then Phones goes to searching mode and try to register But not succeeded. Please Someone tell me How can i register Phone with Free PBX ?

I intalled AsteriskNOW Software PBX 3.0.0 latest version using this link
http://www.asterisk.org/downloads But don’t know how it work and How add Ip Phones this exchange.

waqas Hussain

83

This phone is supported by the Commercial End Point manager. You need to purchase it then read the manual.

I also suggest you install the FreePBX distro if you want support from us AsteriskNow is distributed by a third party. They are a partner but we don’t know all the changes they do to their distribution.

Key things you need to do is to make sure FreePBX in voice VLAN. Also your DHCP server will have to point Option 66 and 150 to FreePBX. For this reason I don’t think you should run the FreePBX and CUCM in same VLAN. I would use the DHCP in the distro and setup a new VLAN for FreePBX. Then configure the VLAN in phone manually as your switches are probably configured to automatically send the voice VLAN ID to the phones.

84

Thanks a lot for your reply Skyking,

I already bought Two Cisco IP Phones and testing on it In Lab Environment. Before i tested on CUCM 8.6 (Vedio Call and all other features) and Now I want to test on FreePBX.

I installed FreePBX 5.2 with asterisk 1.8 using this link
http://schmoozecom.com/distro-download.php

SkyKing Please Tell me what i need to do next for Video call to each other, As i write my previous post i am working in lab environment that why i not configured DHCP server, All configuration manually On IP Phones, FreePBX and TFTP Server but in same network and able to ping.

85

So i finally bit the bullet and started moving away from Trixbox to FreePBX. I had my 9971’s working on trix with video, Visual VM, etc. I installed FPBX from distro (2.11 Asterisk 11.6) and started setup. Setup a few softphones, got video working, moved on to ISoftPhone (IPAD App) got that working with video. All seemed good. Then I started on 9971. Moved my config over from trix, made necessary changes and to my surprise 9971’s registered and basic functions were working…including video. So I moved over files for Visual VM, SugarCRM database to list Directory info…minor tweaks, but all working. Then I moved on to Presence and Conference calling. I updated to Asterisk 11.7 along with the presence patch, no issues. Made the updates in sip_custom_post.conf, saved, reset the phone…and Presences and Conference worked. But now video doesnt work anymore. So i went to my other 9971 which i had not added to the sip_custom_post.conf yet and reset it (first time to do so after Asterisk update / patch). Video still working, but no presence or conference…which is what i expected. Added the info for that extension to sip_custom_post.conf and reset the phone. Presence / Conf working, but no video now. The odd thing is that i never noticed when i started this whole move…the Extensions for both 9971 were set to UDP only and transport for both phone configs were set to 2. Video was working without the patch and without TCP. Now, I have set the extension to TCP Only and updated the config to 4 for transport and i cannot get the phones to register at all. I know I am missing something here and with so many changes from the move, I am hoping I am just overlooking something. Anyone have any ideas at all?

1 Like
86

If i remove cisco_usecallmanager=yes from sip_custom_post and and run amportal restart…Video works but obviously Conference doesnt. If I add cisco_usecallmanager=yes, conferencing works, but video stops. Anyone have any ideas?

87

I had similar problems when I first setup the patch. It’s been a long time now so i’m having trouble remembering specifics of what I did to fix it though!! I do know you’re missing something in your config. I asked for help here: https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel

As far as I can tell, it’s these two steps that solved the problems for me:

  1. Set the sip.conf option maxcallbitrate to 1000000 (or more) to fix video quality issues. In FreePBX, Go to Asterisk SIP Settings and change the Max Bit Rate for Video Codecs to 1000000. I think the default there is usually 384.
  2. and should be set to 1 in in SEPMAC.cnf.xml as well as Auto Transmit Video enabled in in settings menu to automatically send video.
88

I added the maxcallbitrate entry. I checked and confirmed the other entries were already present in the config as well as auto transmit video being enabled. I read thru the post from your link and nothing really stood out. If you can think of anything else, let me know. Thanks.

89

some of the setting on the first post are old so try the following:

  • download the latest firmware for your phone.
  • patch and compile then install asterisks using the patch from:
    https://issues.asterisk.org/jira/browse/13996
  • on sip_custom_post.conf make sure you didn’t miss the (+) next to the extension:

you only need:
extension_number
cisco_usecallmanager=yes

have a read also on this link:
http://sourceforge.net/p/raspbx/discussion/general/thread/ea8dae4d/

1 Like
90

Here is the thing thats confusing me. I can only get this phone to register when the config has 2. Everything i have seen says that is UDP and i should have 4 for TCP, but i cant get that to work. When the phone registers and i have cisco_usecallmanager=yes, presence and conference works…video doesnt. When i have it set to cisco_usecallmanager=no, video works, presence and conferencing does not. Cant get all of it going at the same time so far with FPBX.

91

keep it on 4
go to the extention settings and check following settings:
nat = no - RFC3581
transport = TCP only

then go to Settings ===> Asterisk SIP Settings
make sure Video is enabled
in other SIP Settings right at the bottom add the following feilds:
tcpenable=yes
subscribe=your_ext_#

92

I finally got the tcp / udp issue sorted out. I had a second extension setup on the phone for autoanswering my Visual Voicemail setup and I never set that extension to TCP. So now the phone is set to 4 for transportlayerprotocol and both extensions are TCP Only. My original problem still remains. I cant get Video and Presence to work at the same time. cisco_usecallmanager=yes enables Presence, Conf., Forwarding, etc. but stops video calls. cisco_usecallmanager=no fixes video but stops presence and the other features mentioned. I removed my second extension (autoanswer for Visual VM) from sip_custom_post.conf completely and left the primary set to cisco_usecallmanager=yes. I can make video calls from the second extension no problem, and receive all of the presence info on the primary. If anyone can think of a reason i cant get Video and Presence to work on the same extension at the same time when cisco_usecallmanager=yes is enabled, I would greatly appreciate it. Thanks.

93

With cisco_usecallmanager=yes call another device capable of getting video. Initially, you will not have video as you state. Press mute on your handset and then take it off mute a few seconds later and tell me if video now starts up. I’m asking because I looked up my notes and that was the problem I was having with my setup. If your problem is the same, I could go lookup all of the settings on my Freepbx box and we’ll compare them all to see what’s different.

Also, can you confirm the firmware version on your phone? I’m using 9-3-2SR1-1 on my 9971.

94

Mute doesnt seem to do the same for me. Its funny that you say that because I had a similiar work-a-round with receiving video calls from non cisco devices and I would just put the call on hold for a few seconds, take it off and the video would work. That was only when i received calls from non cisco devices. When i initiated the call from 9971 to any phones, video was fine. Im actually using version 9-4-1-9 right now. I have been using 9-2-3-27 for a long time. I did confirm that I had the issue with both. I could download and try the same version you have…worth a try.

95

Downgrading to 9-3-2SR1-1 didnt resolve the issue.

96

I’m on 9-4-1-9 and everything is woking well.
did you patch astrisk with gareth patch?
which version of asterisk are you using?

97

I did. I was previously at 11.6, used gareth-11.7.0.patch to go to version 11.7. No issues with the applying the patch or make install. I know the patch is working because it resolved presence and not being able to 3-way conference.

98

Asterisk SIP settings -
Video Support Enabled
Codecs - h264, h263
Max Bit Rate - 100000

Reinvite Behavior - No
Registration Maxexpiry - 900

Advanced Settings
Bind Address - 0.0.0.0
Call Events - Yes
Other Settings
tcpenable = yes
callcounter = yes
udpbindaddr = 0.0.0.0
tcpbindaddr = 0.0.0.0
buggymwi = yes
allowsubscribe = yes
notifyringing = yes
alwaysauthreject = yes
pedantic = no

Extension Settings -
canreinvite = yes
nat = no
qualify = yes (I’ve seen some suggestions to set this to no)
transport = All - TCP Primary

extensions_custom.conf

[from-internal-custom]
; CISCO Custom Stuff
; Strip the x-cisco-serviceuri- prefix
exten => _[x]-cisco-serviceuri-.,1,Goto(${EXTEN:19},1)

; Enable forwarding
exten => _cfwdall-.,1,Answer
same => next,Set(SIPPEER(${CHANNEL(peername)},callforward)=${EXTEN:8})
same => next,Hangup(normal_clearing)

; Disable forwarding
exten => cfwdall,1,Answer
same => next,Set(SIPPEER(${CHANNEL(peername)},callforward)=)
same => next,Hangup(normal_clearing)

; Call Pickup
exten => _blfpickup-.,1,Answer
exten => _blfpickup-.,n,Dial(Local/**${EXTEN:10}@from-internal)
exten => _blfpickup-.,n,Hangup(normal_clearing)

sip_custom_post.conf

7101
ciscounified=yes
cisco_usecallmanager=yes
cisco_multiadmin_conference=yes
cisco_keep_conference=yes
dndbusy=no
subscribe=7101
subscribe=7102
subscribe=7106
subscribe=7607101
subscribe=*68

sip_notify_custom.conf

[clear-mwi]
Event=>message-summary
Content-type=>application/simple-message-summary
Content=>Messages-Waiting: no
Content=>Message-Account: sip:[email protected]
Content=>Voice-Message: 0/0 (0/0)
Content=>

; Cisco

[cisco-check-cfg]
Event=>check-sync

[cisco-restart]
Event=>service-control
Subscription-State=>active
Content-Type=>text/plain
Content=>action=restart
Content=>RegisterCallId={${SIPPEER(${PEERNAME},regcallid)}}
Content=>ConfigVersionStamp={00000000-0000-0000-0000-000000000000}
Content=>DialplanVersionStamp={00000000-0000-0000-0000-000000000000}
Content=>SoftkeyVersionStamp={00000000-0000-0000-0000-000000000000}
Content=>FeatureControlVersionStamp={00000000-0000-0000-0000-000000000000}

[cisco-reset]
Event=>service-control
Subscription-State=>active
Content-Type=>text/plain
Content=>action=reset
Content=>RegisterCallId={${SIPPEER(${PEERNAME},regcallid)}}
Content=>ConfigVersionStamp={00000000-0000-0000-0000-000000000000}
Content=>DialplanVersionStamp={00000000-0000-0000-0000-000000000000}
Content=>SoftkeyVersionStamp={00000000-0000-0000-0000-000000000000}
Content=>FeatureControlVersionStamp={00000000-0000-0000-0000-000000000000}

99

Are you using the stock FreePBX distro or something else?
I setup all of this originally on the stock distro, with appropriate patch applied manually. A few days back I have moved to a Raspberry PI using the RASPBX distro. Even though all of my settings are the same on the new platform, I do have some weirdness - for example, when I call from a Bria soft phone to my 9971 the video does NOT work. If I call from the 9971 to the Softphone the Video DOES work. I’m having a lot of other problems as well with BLFs and hangups not working so the Video issue is something I’ve not been able to look at yet.

One thing that stands out is that on my original install, in extension settings I had restricted the codecs to h264,ulaw,alaw and things worked just fine. However, on RASPBX this doesn’t work - I get an error in the logs that the h264 codec is not recognized. And that’s a cue for you to start looking through logs and see what comes up. I’m not too good at that so perhaps someone else can point at what specifically to look for.

100

Just to test, i went back in and added a duplicate maxcallbitrate setting via FreePBX and set that to a small value. That killed the video for me. To be sure you don’t have that problem, check sip_general_additional.conf and make sure that there is only 1 line for maxcallbitrate and that its set to 1000000