| SIPPhone.com delivers! First impressions. I received my two Grandstream BudgeTone-100 phones from »www.sipphone.com/ today (Friday). Not bad, considering I ordered on Wednesday, but then we do share a zipcode.
The BudgeTone phone is more 'shiny' than it looks in the pictures, maybe even feminine. The keypad buttons look like they are made of glass. I imagine some will like it, some won't. Executives with dark desks won't, for example.
The power supply is 5v DC, 400mA, and nice and slim. Ethernet cable is included.
The SIPPhone folks have kindly stickered the assigned (747) phone number and user password on the front of the phone. NO user documentation is provided, and none is on-line at sipphone.com, so make a quick stop here to pick up a PDF User Manual for the phone:
»www.grandstream.com/user_manuals···e100.pdf
Don't forget the helpful FAQ with info on configuring for FWD and doing a full factory reset:
»www.grandstream.com/FAQ.pdf
Inside you will learn the first important thing.. the default password for HTTP configuration is "admin". You're welcome.
You can do some basic config from the Menu/Display/Keypad, but it's tedious and limited. My version numbers reported:
Boot: 2003-06-19 1.0.0.7
Phone: 2003-07-19 1.0.3.78
Vocoder: 2003-06-05 0.0.0.1
HTTP: 2003-07-03 1.0.0.16
One bug I hope they fix in the HTTP code soon is the totally illegal responses to GET! The only web browser that seems to work with the BudgeTone is IE because it kindly ignores the violation.
Instead of sending back a "HTTP/1.1 200 OK" like any sane server, it just belches right into the HTML with no response prefix. I really hope they paid more attention to the SIP standards than the HTTP RFC's.
In any case, it appears the sipphone.com TFTP server is down right now, as is their main web page and NTP server. The phone can't connect to them, so it's showing a bogus date and can make no calls.
The phone is trying to download "cfg.txt" from their TFTP server and getting nowhere. It is preconfigured to use their SIP proxy and NTP server.
Here are the CODECs listed as available:
G-711u
G-711A
G-723
G-726
G-728
G-729
There is mention of G-722 (high quality) in the manual, but it doesn't appear to be available.
A firmware update is available from Grandstream, but I have not attempted an upgrade. When sipphone.com comes back, perhaps they do it automatically.
And hey, give me a call if you want: (747)669-1026
It would be nice to hear what this sounds like. |
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| Just tried calling you from FWD by dialing **7476691026 but I got a message that you were offline...Jeff |
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| reply to Slacktron
Thanks for the (attempted) call Jeff. It looks like I'm not out of the woods yet. Neither SIPPhone is correctly connecting to home base. I'm watching the packets go through the NAT/Firewall, and after the first connection to the TFTP server they make no more connections to anywhere.
They're getting their DHCP assignments (IP, Router, subnet, DNS), then try to download "cfg.txt" from 130.94.123.253 (Not found). Then.. that's it. No more connections.
Both phones act the same way, so it must be something I'm doing wrong. I've tried configuring the NAT both to forward and not-forward the usual SIP ports. Puzzling.
They CAN call each other using direct IP Number dialing, but that doesn't mean much.
Anyone have a working SIPPhone yet? |
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jarruda
join:2002-08-25
Pompano Beach, FL | reply to jeffpulver
said by jeffpulver:
Just tried calling you from FWD by dialing **7476691026 but I got a message that you were offline...Jeff
Received last night my pair also, but, SIPPHONE didn't work at first (can seem my RTP stream "leaving", but nothing back). Saved the configuration for both, reconfigured to be able to co-exist with P8, tested with FWD (one of the phones, to the amazing 411 at FWD , now that is a test of codec quality  ..
One weird thing is that, I was able to use all just fine, then I tried to change the codec to G.723, I could see my RTP stream "outbound" was G.723, but inbound was G.711 (ethereal checked). I've changed all my options to G.723, no voice path inbound  I need to study this further..
Does FWD "411 connected" media gateway does G.723 ?
PS: Placed a Libratel -> call, but was too lazy to try from actual PSTN, so went P8 -> PSTN -> Libratel -> FWD, voice quality was not exacly good , but the call had voice bearer "both ways". |
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| quote:
Does FWD "411 connected" media gateway does G.723 ?
As far as I've observed, the FWD 411 connection is using GSM.
/Jeff |
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| reply to Slacktron
said by Slacktron:
Anyone have a working SIPPhone yet?
I have the service working, but I use a Cisco 7960. |
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| said by aup2:
said by Slacktron:
Anyone have a working SIPPhone yet?
I have the service working, but I use a Cisco 7960.
If you would like to have a test call, please post your number. |
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| reply to Slacktron
Anyone experiencing problems accessing the SIPphone site today? Can't access it from Germany today. Trace is going just fine over VERIO, but page doesn't show ip, maybe the webserver is down. Anyone in here has the same problem?
Tnx
stefan |
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| reply to Slacktron
It does work again. Never mind. |
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| reply to Slacktron
I have no problems viewing the phones HTML pages with Mozilla/Netscape/etc ... I recommend getting the latest HTML code from the Grandstream TFTP server at 4.3.153.56 ...
Im having no problems with my phone ... tested it with FWD setup, MCI setup, and others ... works fine .. Wish it came in a matte black finish ... and whats up with the red lights behind the key area |
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jarruda
join:2002-08-25
Pompano Beach, FL | reply to jeffpulver
said by jeffpulver:
quote:
Does FWD "411 connected" media gateway does G.723 ?
As far as I've observed, the FWD 411 connection is using GSM.
/Jeff
Humm..You mean a SIP call from FWD to 411 goes "GSM" VOIP end-to-end ? (not taking NAT 'tricky' with someone in the middle into account ?)
I'm not an expert, but I was able to send/receive voice from these Grandstream budgetone-100, and it was G.711 inbound, and G.711 outbound (with the default grandstream config).
I've verified this with ethereal.
How FWD "talk" to the 411 "beast" ? Is via PSTN (with a media gateway from the FWD side) ? Or is VOIP all the way ?
In either case, the "other end" (from my P.O.V) does speak G.723 ? |
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| reply to Slacktron
jarruda: Great idea, this calls for ethereal
I see this boot sequence:
DHCP query/response
TFTP connection to 130.94.123.253
Request cfg.txt (not found)
Request bootload.bin, sipp.bin, voc.bin, html.bin (all aborted after start, probably because they're the same version)
ring.bin (not found)
then DHCP again.
Then DNS query for ntp01.sipphone.com
The response is interesting. The name resolves to 192.43.244.18, a non-routable address! How am I supposed to connect to THAT? Does it get tunneled? So I pointed it to a more local NTP server.
After that, it does a DNS query for proxy01.sipphone.com, and gets 130.94.123.252. Then it gets weird.. it does repeated ARP requests for 130.94.123.252! Even though the subnet and router are correct, it's trying to ARP it as if it was on the local LAN.
I tried hardcoding the IP config instead of DHCP -- same thing. What am I doing wrong?
[text was edited by author 2003-08-09 16:53:10] |
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| reply to jarruda
quote:
Is via PSTN (with a media gateway from the FWD side)...
FWD is just software. It owns no switches. FWD is an end-to-end SIP network.
For a hint of how this works, just see:
»listserv.pulver.com/cgi-bin/wa.e···P=R11756 |
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jarruda
join:2002-08-25
Pompano Beach, FL
| reply to Slacktron
said by Slacktron:
jarruda: Great idea, this calls for ethereal
I see this boot sequence:
DHCP query/response
TFTP connection to 130.94.123.253
Request cfg.txt (not found)
Request bootload.bin, sipp.bin, voc.bin, html.bin (all aborted after start, probably because they're the same version)
ring.bin (not found)
then DHCP again.
Then DNS query for ntp01.sipphone.com
The response is interesting. The name resolves to 192.43.244.18, a non-routable address! How am I supposed to connect to THAT? Does it get tunneled? So I pointed it to a more local NTP server.
After that, it does a DNS query for proxy01.sipphone.com, and gets 130.94.123.252. Then it gets weird.. it does repeated ARP requests for 130.94.123.252! Even though the subnet and router are correct, it's trying to ARP it as if it was on the local LAN.
I tried hardcoding the IP config instead of DHCP -- same thing. What am I doing wrong?
[text was edited by author 2003-08-09 16:53:10]
192.168.0.0/16 is in the RFC 1918, others "192" are not.
Your default gateway is set correctly ? This kind of "arp for all" is quite common when the IP stack has default gw == his own IP address. It would require Proxy ARP in the router to make it work in this case.
I do get REGISTER/OK fine with the config that came with my 2 SIPPHONEs, my problem is with the voice bearer it seems (I've not studied the SIP invite to check what is going on, too lazy/sleepy right now) |
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jarruda
join:2002-08-25
Pompano Beach, FL | reply to jeffpulver
said by jeffpulver:
quote:
Is via PSTN (with a media gateway from the FWD side)...
FWD is just software. It owns no switches. FWD is an end-to-end SIP network.
For a hint of how this works, just see:
»listserv.pulver.com/cgi-bin/wa.e···P=R11756
So, the "411" is an actual SIP endpoint, not a media gateway into PSTN...interesting..any others SIP "services" tied to FWD ? |
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| reply to jarruda
jarruda: You're quite right; my mistake. I thought all of 192 was special, but it's just these:
10.x.x.x
172.16.x.x - 172.31.x.x
192.168.x.x
So sipphone's NTP server is just fine.
I haven't got to the root of the problem yet, but I did find that if I lie about my netmask, I can make the SIPPhone work! For some reason, a netmask of 255.255.255.0 causes it to never send packets to the router, even for clearly external addresses.
If I set the netmask to 255.255.255.252, it finally routes the packets to the router.
My network is a pretty standard 192.168.x.x Class-C, and there are 54 other machines that never had any problem with my DHCP server settings over three years, but for some reason, I have to special-case the SIPPhone and put in an incorrect netmask to force it to stop ARPing everything. |
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jarruda
join:2002-08-25
Pompano Beach, FL | said by Slacktron:
I haven't got to the root of the problem yet, but I did find that if I lie about my netmask, I can make the SIPPhone work! For some reason, a netmask of 255.255.255.0 causes it to never send packets to the router, even for clearly external addresses.
If I set the netmask to 255.255.255.252, it finally routes the packets to the router.
My network is a pretty standard 192.168.x.x Class-C, and there are 54 other machines that never had any problem with my DHCP server settings over three years, but for some reason, I have to special-case the SIPPhone and put in an incorrect netmask to force it to stop ARPing everything.
My dhcpd.conf:
subnet 192.168.1.0 netmask 255.255.255.0 {
# default gateway
option routers 192.168.1.1;
option subnet-mask 255.255.255.0;
option domain-name "xxxx.xxxx";
option domain-name-servers 192.168.1.1;
range dynamic-bootp 192.168.1.16 192.168.1.253;
default-lease-time 86400;
max-lease-time 86400;
}
So, you may say is a dumb setup like yours.
But I do see my sipphone working fine.
One thing worth mention, both SIP Phone "behind" the NAT won't work, unless you change the UDP ports numbers (5060, 3478 and 5004) to something else. I've "moved" my others sip devices (the other GS from the pair, set to talk with FWD and a Packet8 DTA310), to leave the default UDP ports "free" for sipphone "trial" device.
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| jarruda: That's almost exactly my dhcp.conf. I'm on 192.168.42.0, same netmask. Nothing special -- boilerplate DHCP template. I don't think there's anything special about 192.168.42.x, and fifty other machines of all flavors seem happy with it.
Most importantly, the same behavior (ARP-all, ignoring router) happens if I hard-code the IP information in the phone and do not use DHCP at all. I really think it's a bug in the phone firmware somehow, but I don't see why it's just happening to me and nobody else.
Again, if I lie and force the SIPPhone netmask to 255.255.255.252, then all is well for both SIPPhone and FWD. Almost. They can't call each other, but I can call 411.
UPDATE: I tried the SIPPhone with 100% as-shipped config on a second unrelated LAN it does not work there either. Netmask was 255.255.252.000. I don't have a packet trace, but no calls could be completed. The Vonage ATA worked 100% fine (incoming and outgoing) on both networks without any changes.
Color me unimpressed. Has anyone got a SIPPhone with default config to work? pattimcc, are you still around?
Grandstream needs a debug feature like NPrintf.
[text was edited by author 2003-08-11 17:37:10] |
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jarruda
join:2002-08-25
Pompano Beach, FL
| said by Slacktron:
jarruda: That's almost exactly my dhcp.conf. I'm on 192.168.42.0, same netmask. Nothing special -- boilerplate DHCP template. I don't think there's anything special about 192.168.42.x, and fifty other machines of all flavors seem happy with it.
Most importantly, the same behavior (ARP-all, ignoring router) happens if I hard-code the IP information in the phone and do not use DHCP at all. I really think it's a bug in the phone firmware somehow, but I don't see why it's just happening to me and nobody else.
Again, if I lie and force the SIPPhone netmask to 255.255.255.252, then all is well for both SIPPhone and FWD. Almost. They can't call each other, but I can call 411.
UPDATE: I tried the SIPPhone with 100% as-shipped config on a second unrelated LAN it does not work there either. Netmask was 255.255.252.000. I don't have a packet trace, but no calls could be completed. The Vonage ATA worked 100% fine (incoming and outgoing) on both networks without any changes.
Color me unimpressed. Has anyone got a SIPPhone with default config to work? pattimcc, are you still around?
Grandstream needs a debug feature like NPrintf.
[text was edited by author 2003-08-11 17:37:10]
Another friend took one of my GS to his home, and it got it working with FWD in the same "painless" way.
The only thing I can think of is that in sipphone.com setup, the STUN server is being used. Maybe this has some weird interaction with the IP stack ?
My friend has a Contivity 100 at his home (my NAT/FW is a linux/iptables), so you may say we had a "fair sample".
We've tested also G.723, and the results where good (he forced the CES 100 to dialup), the latency was quite high, but still good enough, I guess that further "distance" would make the latency worse (not packet loss). |
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| jarruda: When you have FWD working on the BudgeTone, are you using fwdnat.pulver.com:5082 proxy, or connecting directly (no proxy, with STUN for external IP address)?
Have you tried to use the phone at all with SIPPhone's default config? One key difference between Vonage and SIPPhone is that SIP doesn't proxy. FWD supports both ways, of course.
linux/iptables is probably more capable and configurable than my Linksys NAT box, but they should amount to the same thing. |
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