I am trying to accomplish something but I need someone to at least point me in the right direction to get started. Here is my equipment: a 3g iphone running a decent SIP client, a wrt54gs linksys router running asterisks, and a skpye account.
I want to be able to call my asterisk server from anywhere (my phone is always connected to internet and therefore sip enabled via 3g network) and have my calls fowarded by the asterisk server to my skypeout account.
I doesn’t sound too complicated to me but I can find any concrete information on how to configure the Asterisk server to do this. ANYONE??
I would wait a few weeks and use the skype client for the iphone thats due out. Other than that I dont know of any of the sip-skype apps that will run on your hardware.
I can’t find anything about that app. I read a post on the official Skype forums about a potential and a demand for many people needing such an app but the post is fairly old and there is no other mention of one being in the works. I am really interested in finding a way to voip with my iphone without all the quirks and inferior quality of all the other proxy apps I’ve tried. This is why I wanted to set up my own SIP server - to cut out out the “middle man” by becoming the middle man. I just need a way to route my calls to regular POTS and mobile lines. I thought Asterisk was supposed to be perfect for this because of its ability to be super customizable and open source. THanks for the reply.
Um… well for the sake of argument, you wouldn’t be taking out the middle man, so much as adding additional middle men.
Why do you need skype at all? Are you just using it to terminate to the POTS? Because if you are, You could just connect from your Iphone to the Asterisk server and then jump out to the POTS via some termination company that’s not skype.
I mean, by the time you routed yourself from your phone, through the mobile network, through the data network, to your house, through the asterisk server and back out again, let alone over skype… well, you might notice some latency.
Yes, actually that’s exactly what I’m trying to accomplish - I just need a termination to POTS from Asterisk. I am new to this and find it very interesting but need start points. I won’t have a problem once I get on the right track and find where to research. I would like to think I am fairly competent with tech and networks as I have a MS in IT net administration.
I know Skype does this ok, that is why I was wanting to use it. Right now I have my phone using a cheap SIP provider using a SIP client app. The quality is decent but no incoming calls and fairly large lag which is annoying. Also it seems as if the connection is half duplex as only one speaker can go at a time.
I love playing with and hacking to make things work the way I want. I love Linux and opensource and think it’s great how customizable things are. When I started reading about Asterisk I was amazed at its capabilities and really want to dig in. The topic we are discussing now is the most applicable and useful reason I could think of for investing time into and learning Asterisk. Plus, it will allow me to make calls with no deduction from my wireless plan minutes.
What should I search for to find a service that will allow my server to terminate to a POTS service? I thought that was what Skype and SIP services were? I could always build a Asterisk box, buy the appropriate hardware and get to POTS from my home but I am trying to save money and I don’t have POTS service at home - I just use my mobile phones.
Alright, if I was going your route I would do the following:
Get a DID (direct inward dialing) number.
You can get this from a whole lot of different providers, I use Link2voip.com
You want to get yourself a chunk of minutes. You could use pretty much any place that does termination such as Link2voip.com, call-labs, voipjet whatever.
3a) Setup your asterisk server so that incoming calls to your DID pop through to your SIP client on your phone. (Simpler than it sounds.)
3b) Use a low bandwidth codec, (something free like gsm@13kbps, avoiding stuff like ulaw@64kbps)
3c) Remember to use a “qualify” statement for your sip phone, otherwise the conveluted cellular data network is going to have no idea where to send your data.
Make calls.
Remind me that I’m missing a lot of steps by coming back to the forums and asking more direct questions!
Alright, if I was going your route I would do the following:
Get a DID (direct inward dialing) number.
You can get this from a whole lot of different providers, I use Link2voip.com
You want to get yourself a chunk of minutes. You could use pretty much any place that does termination such as Link2voip.com, call-labs, voipjet whatever.
3a) Setup your asterisk server so that incoming calls to your DID pop through to your SIP client on your phone. (Simpler than it sounds.)
3b) Use a low bandwidth codec, (something free like gsm@13kbps, avoiding stuff like ulaw@64kbps)
3c) Remember to use a “qualify” statement for your sip phone, otherwise the conveluted cellular data network is going to have no idea where to send your data.
Make calls.
Remind me that I’m missing a lot of steps by coming back to the forums and asking more direct questions!
Ok, I will do as you say. Sounds like a good setup. How is your sound quality? I am using a SIP provider now (voipcheap.com) and the delay is pretty bad. I am using Siphon SIP app.
At this point, I know very little about Asterisk and this is one of the main reasons for my mission here. I understand about using a low bandwidth codec and making the appropriate changes to the proper .conf files. That’s something I think I can figure out pretty easily with a little research. Only thing I don’t quite grasp is the use of a “qualify” statement you mention in 3c. If you want to expand on this point, great. If not, I’m sure I’ll figure it out with some reading.
Finally, is it even necessary to use an Asterisk server in this scenario since the voip provider is the SIP server? In this case, aren’t I simply just routing the traffic through my server at the cost of additional latency? Simply put, what’s the benefit of my Asterisk server instead of using the provider direct? Thanks for all you comments, they are greatly appreciated!
Well, if your SIP provider will provide you with SIP URI, I guess you could bypass the Asterisk server. But, I am not sure that they do.
Another thing to note is the ping time to voipcheap’s sip server. Of course, my pings are from Montreal… it really depends where you are pinging from. But never the less…
Pinging sip.voipcheap.com [194.221.62.198] with 32 bytes of data:
Reply from 194.221.62.198: bytes=32 time=164ms TTL=231
Now lets ping the servers I use, just as a point of comparison:
Pinging sw1.link2voip.com [74.52.109.90] with 32 bytes of data:
Reply from 74.52.109.90: bytes=32 time=92ms TTL=42
Pinging sw2.link2voip.com [74.52.109.140] with 32 bytes of data:
Reply from 74.52.109.140: bytes=32 time=94ms TTL=42
Pinging west.voipjet.com [208.72.186.66] with 32 bytes of data:
Reply from 208.72.186.66: bytes=32 time=69ms TTL=43
Pinging east.voipjet.com [208.72.186.66] with 32 bytes of data:
Reply from 208.72.186.66: bytes=32 time=68ms TTL=43
Does that 70ms make a difference? Well, ever little bit is just another strain on the system so… YES.
If you have some sort of ping app, you might want to try sending out some pings from your phone to your sip servers and see what kind of ping times your getting to start!
The qualify thing is used so that you can run behind NAT/Firewall, correct me if I am wrong, but I believe it just keeps the connection from the SIP client to the SIP server alive, so that if there is a call to a DID and it comes in through the asterisk server, then it has an idea of where to send it rather then it just getting obliterated by a NAT/Firewall.
Please don’t get me wrong - I really want to use a learn Asterisk. That is 50% of the reason for me doing all this foot work and research…but…the other 50% is to figure a way to use my iphone as well as other phones from home and not have to use up all my cell minutes and not have to pay another 40 per month to the local telco for “digital” phone service that probably uses voip somewhere along the way.
Yes, on a “jailbroken” phone I can communicate via voip (sip) via 3G, but in my case is not detrimental because most of my converstions, on this phone anyway, take place at home. If I’m at home I would like to use my iphone through a voip service for much cheap - which I am doing now but the SIP provider doesn’t offer DID and the lag is too much to handle - not to mention the seemingly one way at a time (half duplex) transmission of voice.
I will try to ping different servers (affordable SIP providers) from my phone as I have installed a command prompt. That is, of course, if this will be a good indicator of sound quality and delay. I thought, however, it had to with more complicated factors such as type of servers used, codec usage, among other attributes. what are your recommendations for best call quality vs. price?
ADDED: I just pinged voipcheap.com from my location and it was 139ms as compared to link2voip server in TX of 37ms. Link2voip icosts more as compared to free from voipcheap but at least, as far as it looks, the quality will be much more bearable and I can get a DID number.
I am a new member and don’t have much experience with Asterisk. I have just a concept of how cool it is after attending the Digium Asterisk Exhibit in San Jose last year.
What I would like to do is utilize several iPhones for wireless voIP locally over the 5.8 kHz Skypilot Ethernet radio network we have on the West side of Cook Inlet. We have very limited GSM and CDMA cell phone coverage there as we are in “the Bush”. I plan to use at least two Multi-tech analog/GSM Cellular gateways and an Asterisk PBX at one of the “hot spots” within the fifteen-mile radius of our network.
The iPhone is a perfect fit as we can provide Ethernet wireless AP’s at any of our dozen facilities on the network and will have this access to the web on the phone for other purposes.
It appears there are a lot of Apps for iPhone voIP “services” via the web. What we need is an App we can configure to connect to our own system. So far I have only found the RingFree App that even mentions utilizing your own PBX.
If you find a SIP phone app for your iPhone, you should be able to register with whatever SIP server you like. This sounds like an interesting project, and I would be interested in helping any way I can
go to the app store and download fring, This works fine over wifi connecting to asterisk. There are others to, as well as sip softphones for the iphone, fring works well though.
or use gizmo for its skype to sip forwarding and also sip to skype calling
The problem with fring is that everything, i.e. all voice traffic, has to pass through fring’s server. To connect directly to your asterisk server, you need a sip client. I use iTalk, http://italk-iphone.com/index.html, which works well and is available on iTunes. You have to keep the application open to receive calls, but that’s a limitation of the iPhone.
Skype and open source don’t go well together. As far as I know the only way of interfacing to Skype is to run Skype’s code on the gateway and use a virtual back to back sound card (or two real sound cards back to back) for the audio, with Skype API for the call control.
