I have a running asterisk server with 2 Linksys SPA3102’s connected to 2 phones, I am able to get the physical phones to call each other. I want to use a soft phone for additional testing. I have tried using Linphone and Jami but neither have been able to register with asterisk. Is there a recommended soft phone app to use for testing that would be simple for a new asterisk user? Ive searched for similar posts on the forum but cant anything that was posted recently. Or, if someone could point me in the direction of an up to date tutorial for testing asterisk with a soft phone? Also, I am running ubuntu 22.04 if that is of any importance.
On Wednesday 10 June 2026 at 19:56:53, harrisonM wrote:
I have tried using Linphone and Jami but neither have been able to register
with asterisk. Is there a recommended soft phone app to use for testing that
would be simple for a new asterisk user?
Zoiper definitely works with Asterisk.
However, you might learn more, about both Asterisk and SIP, if you were to
debug why you can’t get Linphone (which I believe works fine) or Jami (which
I’ve never heard of) to register.
I suggest you install sngrep on the Asterisk server, and use it to watch the
network traffic when your softphones try to register.
a) it will almost certainly be instructive for you
b) you might even get them to work
Antony.
–
Numerous psychological studies over the years have demonstrated that the
majority of people genuinely believe they are not like the majority of people.
I’m partial to a physical desk phone. I use an old Polycom 501. Being able to hit that redial button or ‘end call’ without mousing about or losing focus on my monitors rocks. I tried Zoiper but my muscle memory is with my old Poly. Physical is faster for me.
sngrep is a wonderful tool. It, along with wireshark, tcpdump, and pcapsipdump have satisfied my packet level debugging needs.
Note that you don’t need to run sngrep on your Asterisk server, but it is convenient. You can use tcpdump or pcapsipdump to collect packets and copy the pcaps to another host for analysis.
pcapsipdump is great because it can separate every call (SIP and RTP) into a separate file so when a customer complains you have a complete record.
You can get logs out of LinPhone, but you either need to access them using an Android development environment, or accept that they are staged through a server run by Linphone’s developers, and not end to end encrypted.
Thanks for the response! I was able to get 2 jami softphones set up on 2 separate laptops and they can now call eachother. I found out there was an issue in my pjsip file that was preventing registration. I want to try and get linphone working next. I am confused how to input the server info, like its address, I have seen some sources say you have to enter it with the format “sip:name@XXX.XXX.X.XX” but others say just putting the IP is fine. I also currently have this for the SIP server address: “sip:XXX.XXX.X.XX;transport=udp.” The route is “sip:XXX.XXX.X.XX;transport=udp.”
Linphone works with URIs, but note that transport is optional for UDP, and, as you can set it elsewhere, probably shouldn’t be included even if not UDP.
Also, simple Asterisk configurations don’t need a proxy to be specified. I’m not sure where you are getting Route from, but that should not appear in the INVITE, for normal use of Asterisk.
I’m using Linphone with an ITSP, not with Asterisk, but it is working, in that case, with server as a URI, without transport, and with no Proxy set.
