SIAX - Native SIP/IAX client for iPhone

[allanet]

Where is the log file and how do u get to it?

[tha5150]

just caught up on this app.

my conclusion is that its awsome for people with 2.1 and 2.2 it doesnt work. i am in diar need of a good VOIP app cause i work from home. if someone could steer me to the yellow brick road i would be very greatful.

[bmxtreme]

Where is the log file and how do u get to it?

under /private/var/mobile
Don't be surprise about the "open source" thing you will see. I still think the dev deserve credits to bring this onto the iphone.

There's an update out. I hope this will work for you guys on 2.2. I also hope it won't go bust in 2.1 since I am not planning to upgrade

[tha5150]

is there a way to set this up to call on 3g? maybe edit the voipover3g app that came out? also is there a way to set it up to when you call someone, your name comes up and looks like your calling from your regular cell number? again, again with this can you have someone call your cell number and siax rings with this runnning in the backround?

sorry for the questions, just trying to understand.

[bmxtreme]

is there a way to set this up to call on 3g?

This works over 3G OOB, you don't have to rely on voipover3gs

[tha5150]

Oob? Thanks for the info. Guess in the morning I will read back on how to set this up and check it out. What about the calling people? Will they see my name on their caller id? When someone calls me, could I see their name?

[bmxtreme]

Oob? Thanks for the info. Guess in the morning I will read back on how to set this up and check it out. What about the calling people? Will they see my name on their caller id? When someone calls me, could I see their name?

Ok, so you're saying if you can set your own number to be send out when making an SIP call. This depends on your SIP provider if they allow you to do that (like Truphone), though most don't really support this. If you have Vonage or other high-end VOIP service, you probably have this feature already working.

[skiller]

The chrash after start Siax, is an issue when your trial period is expired. The trial version is working for 7 days.
-------------------------------------------------------------

Does somebody know the settings for freephonie.net and babytel ?

---------------------------------------------------------------
Version 1.05, release notes:

(1) Fixed bug related to stun server.
Note: after changed stun server, we must restart the app.

(2) Added out bound proxy server for sip

(3) Added proxy server for per-account for sip

(4) I changed the Icon, so it looks better

(5) Fixed problem for trial version to input serial number

(6) Fixed long sip name problem

(7) other small fixes.

[tha5150]

Thanks again for the help but unless I can call out or receive calls with name showing this won't work for me. Wish there was another way. Hmmm.

You know actually I can just use this to make calls to offices or places and cut back on general calls when needed to call a friend I can use my regular phone app

[allanet]

(2) Added out bound proxy server for sip

(3) Added proxy server for per-account for sip

Sorry for being dense, but what's the difference between (2) and (3)? Also, what is the difference between "Domain" and "Proxy Server" for per-account setup?

By the way, the damn hardcoded prefix is starting to annoy me than I originally thought. It's great for Betamax voip services, but as a PRO user I have other sip accounts, which use different prefix than "00". So, now I'm forced to dial manually if I'm not using Betamax, as opposed to dialing from contacts (a PRO-version feature!!) Hoping that ver 1.06 will have this fixed.

under /private/var/mobile
Don't be surprise about the "open source" thing you will see. I still think the dev deserve credits to bring this onto the iphone.

Doh! THAT log! hehe. Thanks!

"Open source" is wonderful if you can take it and transform into something usable. I can't and as such have no issue with it. Although, I'm so overriding ALL of the graphics in SIAX!! I'll post some screenshot over the next weekend.

I also hope it won't go bust in 2.1 since I am not planning to upgrade

Really, why not? Seems to be more stable and faster.

[phonemd]

stupid question, but this can't receive calls right

[allanet]

stupid question, but this can't receive calls right

Yes, you can receive calls via SIAX provided your SIP/IAX service allows incoming calls. You can even set it to run in background for continuos operation. However, as I mentioned in one of the previous posts,

On an incoming call, the phone only vibrates and makes a faint beeping sound; there is no way to set a ringtone or adjust the volume.

.

[skiller]

Meantime, is that possible to collect more information on those crashed phones?

Like to attach the log file (/var/mobile/sipphone.log) on to that crash bug?

We are still working on it, thanks

Please sent per PM

Thanks
Skiller

[bmxtreme]

About the ringtone coming out of the earpiece, can you have that coming out from the speakers? That would be awesome and I could hear it even if I am far away from my phone when it rings.

-------------------------------------------------------------

Does somebody know the settings for freephonie.net and babytel ?

---------------------------------------------------------------

I responded to someone with Babytel before. I believe this could be due to a different port that they use, maybe you should make the port configurable (other than the default 5060).
http://www.modmyi.com/forums/native-...ml#post2674021

Plus, port for STUN server, they could be different, currently I have my stun server as "stun.voipcompany.net:5895", I am not sure if that works

[allanet]

I responded to someone with Babytel before. I believe this could be due to a different port that they use, maybe you should make the port configurable (other than the default 5060).
http://www.modmyi.com/forums/native-...ml#post2674021

I don't use babytel.ca myself, but I know that they use 5060 for sip port and 5065 (as stated in your post) for outbound proxy. With SIAX 1.05, the following settings might work. I wish SIAX didn't crash on Cookiemosta, so he could check if this does the trick:

Domain: sip.babytel.ca
Username: 15141111111
Password: your pass
Outbound proxy: sip:nat.babytel.ca:5065

Perhaps, one of Skiller's clients can give it a go?..

On a separate note, I'm trying to set MagicJack up on SIAX. So far, I've been able to get it to register whereas before ver. 1.05 I wasn't even able to do that. Now, though, when I make a call, it seems to receive 400 (Bad Request) for its INVITE. Any ideas?

[skiller]

Thanks to Bmxtreme and Allanet and all other users again.

We found the problem "Crashing Siax after start"

In a few hours is the new build available.

Still we have problems with the Providers Babytel and freephonie.net
hopefully its fixed next week. We got some accounts from Siax users to try it out.

Cheers
Skiller

[votepedro]

I have noticed that SIAX crashes at the point where you enable a SIP account and it attempts to register with the SIP server.

First on 2.2, it would just crash on startup (using my old settings). I deleted my SIAX logs/plist files and it started fine.

However, when I re-entered my SIP account details, and turned it from "off" to "on", it crashes immediately. Now, it will start back up, but I can't enable the SIP account without it crashing.

Looking forward to a fix! Great app functionality wise, interface could use some work but I'm sure that will come with time and further development.

[skiller]

Dear votepedro,

this matter is, cause the trial version is expired. The trial version is running for 7 days.

If you want I double check it, provide me your UDID and send a mail to
support at siax . org

Thanks
Michael

[webitos]

Before few day i put my hands on brand new iPhone 3G,
as i'm in VoIP business myself i tried Fring as SIP client
everything worked instantly, but from both sides of conversation you could hear echo.

So i decided to look for another product that i can use for sip calls, thaks to CYDIA i found SIAX.

i purchased Pro version and on version 1.05. i couldn't do any call trough my asterisk voip pbx, after upgrading to 1.06 i was able to use SIP for few hours and then it stopped working.

So as I'm now paying customer I e-mailed for help tech guys, and support didn't give up on me and all my e-mails until they resolved problem that i have.

For rest of you if you have your own VoIP PBX you need to setup

context from "context=default" to "context=from-internal"

I'm using this opportunity to say that SIAX support is great, product is awesome, and I hope that they will continue to give support like this further.

And for you tech guys from SIAX you will hear more from me, as I'm now become user who is using your product and i will try to help you guys to develop SIAX in even better product as he is today.

One usefull tip for you would be to put somewhere in a program, when you register product that you now that you are registered user

Ok, enough from me for now!

Big thanks goes to Michael!

Zvonimir

[briangriffin]

Has anyone got this working with a CommuniGate SIP server? (CommuniGate Systems)

Steve