Best SIP client for connecting to asterisks pbx?

[kuzew]

Good day all:

A group of lugers and I have started a podcast called The TUX Podcast (or TUX-Cast), and we have setup, a nice asterisks server for guests and for the recording of the actual show. Because we are all in remote areas, we can't meet up and do the production in person.

Therefore, I was wondering what SIP client works best under GNU/Linux and with asterisks? What do you recommend to your guests that don't use a POT line to connect? What audio codec would be best for SIP?

I think that pretty much covers it. Thank you for your time.

Cheers!

-- kuzew
2007-03-20

[riddlebox]

I use xten-lite, it is easy to setup, I think Ekiga is decent too, but I thought it was a little confusing to use

[allix]

A group of lugers and I have started a podcast called The TUX Podcast (or TUX-Cast),

whats the url ?

[riddlebox]

http://www.xten.com/index.php?menu=download

[kuzew]

Good evening all:

Got home from school and tried what you recommended, riddlebox. When connecting, it beeps at me. ): I don't know if its configuration error on my part both server or client, I'll begin troubleshooting tomorrow morning. I just got my FON router free in the mail today and just hacked it, so now its running DD-WRT, so that took up my evening. Thank you for the recommendation, riddlebox.

As for the release of the URL, http://tuxpod.kuzew.net/. Have fun! d: We have not even done a full episode, hence the 0.1. d: That was with a very tricky system done over TeamSpeak and a lot of clients for phone/audio. Reason why we sacked that and went for asterisks. So, give sometime for a real episode to be released sometime this weekend (maybe?). Anywho, I must be off.

Cheers!

--kuzew
2007-03-21

[riddlebox]

if you want you can IM me and I will help you connect to the asterisk server
I am
clpackageman = aol
james2dope = yahoo
jctsphone = msn

I may be on skype but I cannot remember what my account is

[Patrick]

Gizmo is actually a pretty decent SIP client. Dan Dennedy was connected to the Asterisk server with it and he sounded pretty decent. You can make calls via the Gizmo network or via Asterisk.

[kuzew]

Afternoon all:

Thank you both Patrick and riddlebox for your help. I'm currently very busy with school and haven't had time to play around with the server and the different clients. I will give them a try during my spring break and definitely release a new show. Thanks again and looking forward to the show tonight.

Cheers

--kuzew
2007-03-28

[kuzew]

Afternoon all once again:

Well, I've started to play with Gizmo, lovely project, great UI. Any who, I call into my server, and well, I just hangs up after one ring. I looked in the logs and found something interesting:

Code: Select all

[Mar 29 14:19:16] WARNING[3431] chan_sip.c: username mismatch, have <trunk_1>, digest has <17471121157>
[Mar 29 14:19:16] NOTICE[3431] chan_sip.c: Failed to authenticate user <sip:17471121157@proxy01.sipphone.com>;tag=4db8556e

I understand the warning, but I don't know how to fix it! Anyone want to give me a hand? Thanks.

Cheers
--kuzew
2007-03-29[/i]
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EDIT: Haha, nevermind all that, the OpenWengo client works great for me. Still would like to know why Gizmo was having problems connecting, find that quite odd.

[liamgbf]

Ok i should say that this could be a better SIP client for connecting to asterisks pbx... awesome find man for the software... is it free??

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[dann]

I have been really digging linphone lately, especially since it has a command line client. Other than that I typically stick with SJPhone.

[eddie]

http://www.voip-info.org/wiki/view/VOIP ... ders%2BB2B