From my years ago attempts to use Cisco phones, it was obvious that Cisco’s implementation of SIP was non-standard. Avaya took the same approach with their SIP implementation. Even loading the 3PCC firmware on the Cisco phones does not give all the features that current open-source SIP phones can provide. You get some basic functions but not like an SCCP phone running on CUCM and not like a native SIP phone like a Yealink or Grandstream. Chan_sccp-b was a kluge of stuff that allowed a pretty decent implementation tied into chan_sip which allowed all the Cisco exceptions to work with Asterisk. It even supported bridged call appearances. I believe the developer just has no plans to update the package to run with chan_pjsip. In my case, chan_sccp-b would frequently cause seg faults and crash Asterisk.
The Cisco phones dumped on the market at bottom dollar prices has generated huge interest in FreePBX, VitalPBX and 3CX being able to support them. It could be a winning proposition if a native, fully functional SCCP module could be developed for FreePBX and Asterisk, even if it was only on FreePBX-17. It would have to be done for core Asterisk and then developed on FreePBX.
From the chan_sccp project: " * We are implementing all code in this project under Sect. 1201 (f) Reverse Engineering exception of the DMCA for the sole purpose of writing interoperable software"
The limitations of these models from Cisco are well known. My question was more related to the it works with chan_sip and not chan_pjsip because without the patch the limitations are the same. The basic functions like registering and making/receiving a call should work regardless of which driver in Asterisk is being used.
They do. Other things like programmable buttons, proper CLID display, distinctive ring, BLF keys, etc. Don’t. Even with an Avaya “sip” phone, you can do basic stuff. The whole idea of that sccp channel was to bring all CUCM features for those phones to Asterisk without using pure SIP.
Has VoIP not reached a point yet where there are proper SIP surplus phones for those afraid of investment. Why in 2024 are we still on these junk Cisco phones.
Claim: We don’t have a budget for more expensive phones.
Truth: The only way these phones are cheaper is if your time is worth nothing. If you have to fight something for hours those cost add up they just aren’t directly in the price tag.
All that said if you’re this motivated to use old obsolete junk why update your FreePBX? Go find an old trixbox build and live the simple life.
The comment that I originally replied to stated that phones or gateways refuse to work with pjsip. I was asking for clarification of what isn’t working, such as registration. If it is something like registration or issues with incoming calls, etc. that is most likely a configuration issue at some level.
The lower feature Cisco phones like the non-video phone 6921s and such don’t have a lot of buttons on them and the “enterprise SIP” code is adequate for those as basic extensions on pjsip
I emailed the callmanager patch author, Gareth Palmer, on 3/30/24 on this issue. I asked him if he was going to merge into the interlinked1 fork of chan_sip and his response was:
“his fork has additional changes to chan_sip that prevents the callmanager patch from applying as they modify similar locations in the file”
I then asked him what did he intend on doing and gave him some suggestions and he said it would be something like forking chan_sip and applying his patches then keeping the fork buildable on current Asterisk, the same way that interlink1 intends on doing and the same way we have been discussing.
I have NOT seen anything on his website that says he did that.
The last sentence he sent in that email communication stated the following answer to another one of my questions regarding porting the patch to pjsip:
“chan_pjsip implements the SIP protocol using multiple modules which may not all be loaded (eg: MWI, subscriptions etc.), the phones expect all those features to be available”
I’m going to see what I can do with this and Asterisk 21
Me: Doctor, after my accident when I bought a lot of cheap Cisco’s, will I be able to play the violin?
Doctor: Well, after many hours of therapy and $$$ , then probably yes.
Me: That’s good because I couldn’t before!
Those modules would need to be loaded regardless if it is a Cisco phone or a standard SIP phone for anyone expecting to use MWI, BLF, etc… Not sure how that is a stopper item.
Unfortunately the CALLMANAGER patch only applies cleanly to Asterisk 20 It would have to be manually applied to the chan_sip driver from Asterisk 21 It ALMOST applies to the chan_sip driver right before it was deprecated. I’m going to experiment more with this.
I CAN report though that Asterisk 21 DOES compile just fine with the interlink1 chan_sip using the following instructions:
git clone -b 21 https://github.com/asterisk/asterisk asterisk-21
cd asterisk-21
wget https://raw.githubusercontent.com/InterLinked1/chan_sip/master/chan_sip_reinclude.sh && chmod +x chan_sip_reinclude.sh && ./chan_sip_reinclude.sh
contrib/scripts/get_mp3_source.sh
contrib/scripts/install_prereq install
./configure
make menuselect
make menuselect.makeopts
make menuselect
menuselect/menuselect --enable chan_sip menuselect.makeopts
make
make install
make samples
make config
ldconfig
Well, if you can buy a Cisco for $5 that takes immense effort to make it work and then just kind lamely, why in god’s green acres would you not pay $6 for a Polycom, Yealink or Grandstream that JFW and admit to not listening to all that extant ‘caveate emptor’ advice?
Oh well , we will get these posts for years until someone works out how to add 4Mb of memory to them to support a real SIP stack, but that’s just not going to happen soon, if you buy them, then spring for buying UCM also
Because when you have a site that has 300 desk phones all in place spread out over multiple sites the Polycom phone actually costs $6+labor to hire the crew to replace every phone. So it’s much much more than $6 a phone.
The 3PCC firmware is required by Cisco if you move your deskphones to Webex Calling (the Cisco’s cloud PBX product) Cisco touts this product as a drop in replacement for a CUCM. I’d be very interested in what features you found are missing.
