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17 posts


# 176907 15-Jul-2015 22:19
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I have a Cisco 7912 with SIP firmware. I have tried configuring this for 2Talk but I can't dial out. I get a continuous beep beep beep.

Does anyone have one of these phones working with 2Talk who could supply a sample config that works?


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  # 1345078 15-Jul-2015 23:42
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My experience with one of these is you have to be a Cisco config file master to make them work.

Seriously, they were never made to do SIP in the first place. Go and get an SPA5xx and be done with it. Or even better, one of the Yealink T4x phones.

If you feel like googling for days though I would suggest Search for 'cisco 7912 on Asterisk' - seems to give a few promising results.

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  # 1345116 16-Jul-2015 08:02
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As harsh as this sounds I have one piece of advice for you...Go and buy a new proper SIP phone.

These phones are NOT designed to work with anything other than Cisco CUCM even with the SIP firmware. Yes they can be made to work, but there are so many issues that can and will occur with these they're simply not worth touching in a non CUCM environment. My day job is VoIP and I won't touch the things..


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  # 1345710 16-Jul-2015 19:17
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I got this working. I could see it wasn't registering with 2talk at their end in the cloud PBX portal. There's really very little in the way of diagnostics on the 7912 that I could find easily find, so I used internet connection sharing on my windows laptop, plugged it into the LAN port and captured the traffic with wireshark. I was getting 401 and 403 errors back from the server. Tried the phone number as the login ID without the -1 on the end (2talk one number feature) and it worked OK. Surprisingly, through two NAT gateways!

So actually a support ticket for 2talk rather than the phone config.

Now I can judge whether these really are a piece of rubbish for what we'll use them for. I'll take anyone's word for it they are but my requirements are very basic.

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  # 1345742 16-Jul-2015 20:04
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I had some configured for a while. Found they were much worse at actually getting calls ringing on it compared to a junky bugetone one on the same connection, same config. Plenty of times calling the number would have the 7912 sit there and do nothing. Im assuming its nat entry was expiring, but I had the same expiration and stuff as the other phones, which all kept it open fine and got incoming calls fine.

Your milage may vary, but the only reason I persisted at the time (long time ago) was so I could put the 24 ring tone on the phone which my other one couldnt do.


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  # 1345767 16-Jul-2015 20:41
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I too managed to get one of these working eventually on 2Talk. Fine for basic home use, seemed just like an ordinary landline. It was a bit tricky getting it flashed with the correct firmware though. 

Once it was set up I don't think I had any issues with it on 2Talk so not sure why they get so much hate. I did have mine set up with its own number, and it was the only phone connected to that number, so I guess if it was set up with an extension then maybe that would be whats causing issues that some people seem to have?

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  # 1347365 19-Jul-2015 21:06
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Sorry to drop into this thread off topic but i have 7942's that we want to setup with 2talk, and they come up login and we can make outgoing calls, and 2 talk console shows them as logged in but alas we can not receive calls, have a cisco router as a fw and turned it off and does not seem the make a change, and yes its natted 
any chance anyone can post a running config ?

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  # 1347371 19-Jul-2015 21:08
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These phones are a nightmare for NAT as they were never designed to ever operate in a NAT environment. There are plenty of limitations including port ranges.


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  # 1347375 19-Jul-2015 21:14
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undoubtedly but we have 50ish so i kinda need to solve it, i the bit that really has my panties in a bunch is that i have solved this before about a year ago i got them going good as gold and have misplaced the config (feel free to lambaste, i know i know) but someone else must have solved this also ? my config....  

#The following lines contain the authenication details
line1_name: REMOVED
line1_authname: "REMOVED"
line1_displayname: "Sip Phone new"
line1_password: "REMOVED"
messages_uri: 028890000
line1_shortname: "REMOVED"
line1_displayname: "REMOVEDe -- "
phone_label: "REMOVED ----- "
image_version: P0S3-08-6-00
proxy_register: 1
timer_register_expires: 600
preferred_codec: g729a
tos_media: 5
dtmf_inband: 0
dtmf_outofband: avt_always
dtmf_db_level: 3
timer_t1: 500
timer_t2: 4000
sip_retx: 11
sip_invite_retx: 7
timer_invite_expires: 180
dial_template: dialplan
sntp_server: ; SNTP Server IP Address
sntp_mode: unicast ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: NZST ; Time Zone Phone is in
dst_offset: 1 ; Offset from Phone’s time when DST is in effect
dst_start_month: October ; Month in which DST starts
dst_start_day: "0" ; Day of month in which DST starts
dst_start_day_of_week: Sunday ; Day of week in which DST starts
dst_start_week_of_month: 1 ; Week of month in which DST starts
dst_start_time: 02:00 ; Time of day in which DST starts
dst_stop_month: March ; Month in which DST stops
dst_stop_day: "0" ; Day of month in which DST stops
dst_stop_day_of_week: Sunday ; Day of week in which DST stops
dst_stop_week_of_month: 3 ; Week of month in which DST stops 8=last week of month
dst_stop_time: 2:00 ; Time of day in which DST stops
dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment
dnd_control: 0
callerid_blocking: 0
anonymous_call_block: 0
dtmf_avt_payload: 100
date_format: D/M/Y
time_format_24hr: 1
nat_enable: 1
nat_address: REMOVED

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