w8sdz
join:2001-05-21
Port Orange, FL
| [VoiceStick] BYOD configurations needed VoiceStick will soon be adding BYOD support. They would like users who have successfully configured ATAs for their service to post their configurations here. Both Softphone and hardware devices are welcome, however Asterisk will not be supported at this time.
Look for future postings in this forum from one or more VoiceStick employees. They seem genuinely interested in participating. Please make them feel welcome.
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73 de w8sdz
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w8sdz
join:2001-05-21
Port Orange, FL
2 edits | Re: [VoiceStick] BYOD configurations needed I have a Sipura SPA-2000 ATA successfully working using the configuration shown here:
»www.voipconfig.com/voicestick_config.htm
After entering the settings shown in the link above, make sure you have:
NAT Keep Alive Msg: $PING
Use OB Proxy In Dialog: no
Register Expires: 3600
The same configuration should work with the Linksys PAP2.
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73 de w8sdz
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 | Quattrohead
join:2005-02-09 | Re: Sipura SPA-2000 BYOD configuration for VoiceStick I would LOVE to use them with Asterix, Axon or 3CX. So the question is, can they set the credentials required for authentication to be 'friendly' towards those programs ? | |
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| | 
dot_null
Premium
join:2004-06-28
Kennesaw, GA
 ·Callcentric
 ·Comcast
·VoiceStick
 ·AT&T Southeast
2 edits | Re: Sipura SPA-2000 BYOD configuration for VoiceStick As far as SIP configurations go, the following settings are the only settings I needed to change in a Sipura SPA-2000 reset to factory defaults to have my Voicestick account functioning correctly:
Proxy: i2telecom.com
Outbound Proxy: 206.165.50.116
Register Expires: 3600
Use Outbound Proxy: yes
User ID: Voicestick DID (including leading 1) 1xxxxxxxxxx
Password: VoiceStick Password
Voicestick recently shut down the accounts of users that used Asterisk (including mine). I have gotten approval from Voicestick customer care to use consumer level SIP devices (Sipura SPA-2000, Linksys PAP2, Grandstream GXP-2000, etc.), but they still do not allow Asterisk.
Now I wish they would just fix the problem of disconnecting incoming calls when the calling party disconnects; when the calling party hangs up, Voicestick's servers send the SIP device no indication that the calling party has hung up, thus, transmitting silence to and from Voicestick's servers until the callee (the Voicestick user) hangs up their phone. This creates BIG problems if you are using an answering machine, or an FXO card in conjunction with Asterisk.
I have also noticed that I cannot put calls on hold; For example: using the FLASH button in the case of an analog phone used in conjunction with an ATA, or using the HOLD button on an IP phone like the Grandstream GXP-2000.
If they would fix these minor issues, Voicestick would be a leading BYOD provider.  | |
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| mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA
| said by w8sdz  :NAT Keep Alive Msg: $PING I just noticed the above $PING string. On a PAP2v1, $NOTIFY is the default string. Has anyone tried this $PING string on a PAP2v1 besides the OP and how is it different from the default string? Does it send out less packets?
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Mazi (UK Non-Geo Phone: +44-703-194-2574) | |
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| | hwittenb
join:2003-12-20
·Future Nine Corpor..
·callwithus
·Callcentric
| Re: [VoiceStick] BYOD configurations needed said by mazilo :I just noticed the above $PING string. On a PAP2v1, $NOTIFY is the default string. Has anyone tried this $PING string on a PAP2v1 besides the OP and how is it different from the default string? Does it send out less packets? If you look at a packet trace, the effect is pretty much the same, at least to VoiceStick. The number of packets sent depends on the NAT Keep Alive Intvl setting on the SIP tab. The default is 15-seconds. In either case the adapter sends a Sip Notify or a Sip Ping packet every 15-seconds. VoiceStick returns a packet that says Sip Bad Request. At least this is how it works on my SPA1001. The purpose, of course, is to keep your router from closing the sip signalling port. If your router doesn't close your port you probably don't need to send the packets at all. | |
|
slow mo
join:2002-03-19
USA
4 edits | I use D-Link DVG-1402S:
Proxy: i2telecom.com
Domain: i2telecom.com
Initial Unregistration: Disabled
Register Expires: 3600
Outbound Proxy: Enabled
Outbound Proxy: 206.165.50.118
Outbound Proxy Port: 5060
Authentication ID: Voicestick DID 1xxxxxxxxxx
Password: VoiceStick Password
Would be nice to be able to set rings before VM kicks in. As of now, it's fixed at 4. | |
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|
mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA
| said by w8sdz :VoiceStick will soon be adding BYOD support. They would like users who have successfully configured ATAs for their service to post their configurations here. Both Softphone and hardware devices are welcome, however Asterisk will not be supported at this time. This is definitely a good added feature VS will be doing to compete with other VoSPs. However, VS still faces some problems to provide its on-line signup procedures for new users. I have some UK friends who live here in the US and one of them told me that he's still unable to sign up for a free DID offered by VS. He's trying to get a free DID and put it on his ATA device in UK so he can call home for free from here locally. Here is what he had said:
•The sign up process requires his private phone number. To make this happens, I told him to use my UK non-geo number and it looked like VS has no problem with it.•It looks like his CC is no good with VS auto checking systems that every time he wanted to open for a new account the system responded with the following error message:
This transaction cannot be processed.
Please try and correct the error above. If you need additional
assistance you may contact Customer Service at 1-866-410-3892. - He tried to settle this problem by calling the above toll-free number provided by VS and always got a recording message (no live personnel to greet him): Thank you for calling our IT Telecom. No one is available to take your call at this moment. So, please leave your message and we will return your call ASAP.
Honestly, if VS wants to compete with other VoSPs, it should resolve this problem in a time manner, let alone to require a CC. Stanaphone doesn't even require a CC and will give out its free DID #s.
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Mazi (UK Non-Geo Phone: +44-703-194-2574) | |
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| rcilink
Premium
join:2003-12-15
Manchester, NH
| Re: [VoiceStick] BYOD configurations needed said by mazilo :•It looks like his CC is no good with VS auto checking systems that every time he wanted to open for a new account the system responded with the following error message... Could it be possible that they refuse non-US credit cards? | |
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w8sdz
join:2001-05-21
Port Orange, FL
1 edit | said by mazilo :It looks like his CC is no good with VS auto checking systems that every time he wanted to open for a new account the system responded with the following error message... Most web sites that accept credit cards check the address you specified during your sign-up against the mailing address the credit card company uses when they send your monthly statement to you. If these addresses do not match the transaction is denied.
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73 de w8sdz
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| | mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA
| Re: [VoiceStick] BYOD configurations needed said by w8sdz :Most web sites that accept credit cards check the address you specified during your sign-up against the mailing address the credit card company uses when they send your monthly statement to you. If these addresses do not match the transaction is denied. The addresses on the account and VS to be account are the same. The problem here is there isn't any attendant on the toll-free number to have the problem intervened and been fixed.
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Mazi (UK Non-Geo Phone: +44-703-194-2574) | |
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| | | stufried
Premium
join:2003-10-13 | Re: [VoiceStick] BYOD configurations needed Some providers have the capacity to put in a "virtual address." Amex fixed my card so that I could set my home or my office as my billing address. | |
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| | | BruceN
Hi
join:2006-11-17
Roswell, GA
·Future Nine Corpor..
 ·AT&T U-Verse
·Comcast
| We are having a in house debate about the CC reject. Seems that a small handful of banks need at least a $1 transaction to allow us to address verify. Only happens in a small number of attempts.
The fraud that we see is INSANE, and to avoid giant problems we use the CC to verify a new customer. There is no completed charge (unless you select to purchase something). I could write 3 books about fraud.
As to 24/7 service. We are in the middle of making some giant changes. We had a top dollar service doing level 1 support. Had them for a long time, but due to a service of this type having a giant crew and huge turnover they seemed to solve very little. We got lots of hot hate email!!!
We have brought in house the tech support and we can handle without a referral all level 1 and 2 issues. Level 3 issues are again done in house with our own network people.
What does this all mean. You will not be connected to a live person.
What we are heading towards is a 100% call back system where a customer leaves their account number and issue (or if new, leave just an email)
This will allow use to pull up the account, study the issue, and have a solid answer for our call back.
This means no waiting on line, or punching a ton of menu commands. It means having someone that knows what to do to handle the issue. A person that is empowered to solve, not just say no.
I did this at a company I owned for a decade and customers loved it. After all, you are just trying to get an issue taken care of. You don't want to have a phone in your ear all day long. Would it not be nice when you got sick that you could call the Doctor and later the Doctor calls you back with your cure?
And hey.... I am one of the key people in the company and I am silly enough to post my email up here.
So, I think overall, we are on the right track to get customers great service. Will it be perfect, heck no, but it is the goal.
As to BYOD, I can use every bit of help I can get. I used customers before in my last business, and they were my best source. | |
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| | | | sokhapkin
Premium
join:2003-05-08
Somerset, NJ
| Re: [VoiceStick] BYOD configurations needed said by BruceN :And hey.... I am one of the key people in the company and I am silly enough to post my email up here. OK, let me summarize complaints about your service scattered in this forum, I hope you will pay an attention.
1. Why SIP registration interval should be more than 1 hour? The client might be offline but your server will not notice it.
2. If inbound DID is called and calling party hang up, the server do not send BYE to the SIP client and called party gets dead air. The behavior is incompatible with voicemail and answering machines.
3. Unneeded REINVITE on outbound calls. When the outgoing call is placed, your server responds with
SIP/2.0 200 OK
c=IN IP4 206.165.50.116
t=0 0
m=audio 35898 RTP/AVP 0 101
Session progress audio use this RTP port. But when the call is connected, your server does REINVITE and wants different RTP port:
INVITE
c=IN IP4 206.165.50.116
t=0 0
m=audio 35900 RTP/AVP 0
What is the reason for reinvite if the audio server IP address did not change? Moreover, some SIP clients have problems with reinvite. I can understand REINVITE to forward the voice path to another server in order to not run the traffic through your SIP server, but see no any reason in REINVITE to the same server.
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»www.callwithus.com | |
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| | | | | BruceN
Hi
join:2006-11-17
Roswell, GA
·Future Nine Corpor..
·AT&T U-Verse
·Comcast
| Re: [VoiceStick] BYOD configurations needed As to the technical questions. I am way out of the loop on this.
I will send the link over to our network expert.
But do bear in mind that we are interested in learning from this group. Just because we run a company does not make us right!
Thanks for the input, and I will contact Mark about all this. | |
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| | | | | | sokhapkin
Premium
join:2003-05-08
Somerset, NJ
| Re: [VoiceStick] BYOD configurations needed said by BruceN :But do bear in mind that we are interested in learning from this group. Just because we run a company does not make us right! Thanks for the input, and I will contact Mark about all this. Well, I submitted a similar complaint using "Contact Us" form on your web site 2 months ago and did not get any response...
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»www.callwithus.com | |
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| | | | | | | BruceN
Hi
join:2006-11-17
Roswell, GA
·Future Nine Corpor..
·AT&T U-Verse
·Comcast
| Re: [VoiceStick] BYOD configurations needed Sokhapkin, 2 months ago... I bet you are 100% right.
We turned the place upside down about 3 weeks ago.
If you saw the extreme amount of money we were spending only to end up with canceled accounts, angry email, and all the other problems..
It was INSANE!
I was hired in to fix the place.
Marketing is more then just putting on a pretty face, and customer retention is even more important then finding new customers.
Sorry to you and any others that have had problems.
Call in or email customercare@i2telecom.com and see what is working now.
BTW, we are going to change the incoming message and work a small menu program I hope this next week.
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| | | | | jdl20
join:2006-11-19
Wilmington, MA
| said by sokhapkin :3. Unneeded REINVITE on outbound calls. When the outgoing call is placed, your server responds with SIP/2.0 200 OK c=IN IP4 206.165.50.116 t=0 0 m=audio 35898 RTP/AVP 0 101 Session progress audio use this RTP port. But when the call is connected, your server does REINVITE and wants different RTP port: INVITE c=IN IP4 206.165.50.116 t=0 0 m=audio 35900 RTP/AVP 0 What is the reason for reinvite if the audio server IP address did not change? Moreover, some SIP clients have problems with reinvite. I can understand REINVITE to forward the voice path to another server in order to not run the traffic through your SIP server, but see no any reason in REINVITE to the same server. I have a cisco ata 186 configured for voicestick, I can't make outgoing calls because of this unnecessary reinvite. It seems like this reinvite thing is something that VS implemented a few weeks ago, because I used to be able to make outgoing calls. I can only call my voicemail at this point because there is no reinvite when I call voicemail.
If VS is serious about their BYOD plans, they should consider getting rid of this unecessary reinvite. | |
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| | | | | | BruceN
Hi
join:2006-11-17
Roswell, GA
·Future Nine Corpor..
·AT&T U-Verse
·Comcast
| Re: [VoiceStick] BYOD configurations needed I doubt that the reinvite is doing you in. We have made no system changes. I wonder about your account. No money in it? bad CC date, or some other issue.
This is most often the cause of this sort of problem.
Let me know if you want me to look at your account.
Bruce | |
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| | | | | | | Quattrohead
join:2005-02-09
·VOIPo
| Re: [VoiceStick] BYOD configurations needed Hi Bruce,
As this is turning into a VS help fest 
I obtained a local number through you with a PAYG package.
If I ever gave up my account with you, can I keep the number using LNP ?
Also, please consider opening up to * or 3CX | |
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| | | | | | | | BruceN
Hi
join:2006-11-17
Roswell, GA
·Future Nine Corpor..
·AT&T U-Verse
·Comcast
1 edit | Re: [VoiceStick] BYOD configurations needed I would love to do Asterisk, but we got hit BAD by some BAD guys. I don't dare publish what they were doing. Asterisk is a good thing in the hands of good people.
As to the DID, it is my understanding that some numbers have been ported off our system, but I am not 100% sure. We 'rent" the DID's from Level3 and also use them for top quality incoming termination. | |
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| | | | | | | jdl20
join:2006-11-19
Wilmington, MA
| said by BruceN :I doubt that the reinvite is doing you in. We have made no system changes. I wonder about your account. No money in it? bad CC date, or some other issue. This is most often the cause of this sort of problem. Let me know if you want me to look at your account. Bruce There is money in the account, CC is legit. I even get billed 1 or 2 minutes for the call. My ata is simply not able to handle the reinvite. the ata continues to send packets to the first port that VS's server said it was expecting them on. I can provide packet traces of this to you offline if you want. | |
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| | | | | | | | BruceN
Hi
join:2006-11-17
Roswell, GA | Re: [VoiceStick] BYOD configurations needed Well I guess this is back to my point. We do not have hardly any info on using other devices on our service. Any of you smart people have any thoughts? | |
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| | | | | | | | | im_chandave
join:2005-07-28
Cleveland, OH
| Re: [VoiceStick] BYOD configurations needed said by BruceN :Well I guess this is back to my point. We do not have hardly any info on using other devices on our service. Any of you smart people have any thoughts? Well, I WAS providing support for VS to asterisk users...I think I was the first one to publish complete SIP info so people didn't have to use your branded SJ Phone...
And, I WAS working on a way for Asterisk to determine when your end terminates the call since you don't send a proper SIP BYE sequence...
And, I WAS working to get the Nortel and my company's SIP phones (both WIFI and desktop models) to work with it...
Key term here is WAS until you guys cut my account (still with $5 of my own money I deposited into it) and never replied to any of my inquires I sent through the Web form and via email.
Makes me almost want to edit my May Voxilla posting (the one most Asterisk and Trixbox users reference) and comment on your lack of support and indifference.
See ya...
d.c. | |
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| | | | | | | | | BruceN
Hi
join:2006-11-17
Roswell, GA
·Future Nine Corpor..
·AT&T U-Verse
·Comcast
| Re: [VoiceStick] BYOD configurations needed Dave sorry that you got run over in our Asterisk CRISIS!!
Any email about this was being sent to my attention so I could answer ir personaly. There were maybe 20 accounts.
Not sure why your email did not get to me....
I dare not publish what the Asterisk BAD guys were doing as it would be like Microsoft when they talk about a security hole. It just invites problems.
I can assure you it was gawd awful, and that is why we went on a hunting mission.
IM me if you need.
Thanks for your past support.
Bruce | |
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| | | | | | gimp55
join:2003-08-16
Ironton, OH
| said by jdl20 :said by sokhapkin :3. Unneeded REINVITE on outbound calls. When the outgoing call is placed, your server responds with SIP/2.0 200 OK c=IN IP4 206.165.50.116 t=0 0 m=audio 35898 RTP/AVP 0 101 Session progress audio use this RTP port. But when the call is connected, your server does REINVITE and wants different RTP port: INVITE c=IN IP4 206.165.50.116 t=0 0 m=audio 35900 RTP/AVP 0 What is the reason for reinvite if the audio server IP address did not change? Moreover, some SIP clients have problems with reinvite. I can understand REINVITE to forward the voice path to another server in order to not run the traffic through your SIP server, but see no any reason in REINVITE to the same server. I have a cisco ata 186 configured for voicestick, I can't make outgoing calls because of this unnecessary reinvite. It seems like this reinvite thing is something that VS implemented a few weeks ago, because I used to be able to make outgoing calls. I can only call my voicemail at this point because there is no reinvite when I call voicemail. If VS is serious about their BYOD plans, they should consider getting rid of this unecessary reinvite. Same issue here, money in account, worked fine with ata186 till recently, incoming calls work fine.
I have to keep switching over to my sellvoip account to make outgoing calls, would rather use VS but cant.
here is what i got back from support, settings worked fine till lately:
Your ATA is your responsibility as we do not support this device since it is not a certified product on our network. Here is our SIP configuration for your ATA.. double check it as something might be incorrectly set up...
The information you need to provision your 3rd party IAD is the following:
SIP Proxy IP: 206.165.50.116
SIP Proxy Port: 5060
SIP Registration Domain: i2telecom.com
STUN: Disabled
SIP Username/Password: Your DID #/ your password
Be advised that we do not provide any troubleshooting or maintenance support for 3rd party IAD’s. Also, the provisioning information can change at any time.
The i2 Telecom Customer Care Team | |
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| | | | | | | BruceN
Hi
join:2006-11-17
Roswell, GA
·Future Nine Corpor..
·AT&T U-Verse
·Comcast
| Re: [VoiceStick] BYOD configurations needed said by gimp55 :said by jdl20 :said by sokhapkin :3. Unneeded REINVITE on outbound calls. When the outgoing call is placed, your server responds with SIP/2.0 200 OK c=IN IP4 206.165.50.116 t=0 0 m=audio 35898 RTP/AVP 0 101 Session progress audio use this RTP port. But when the call is connected, your server does REINVITE and wants different RTP port: INVITE c=IN IP4 206.165.50.116 t=0 0 m=audio 35900 RTP/AVP 0 What is the reason for reinvite if the audio server IP address did not change? Moreover, some SIP clients have problems with reinvite. I can understand REINVITE to forward the voice path to another server in order to not run the traffic through your SIP server, but see no any reason in REINVITE to the same server. I have a cisco ata 186 configured for voicestick, I can't make outgoing calls because of this unnecessary reinvite. It seems like this reinvite thing is something that VS implemented a few weeks ago, because I used to be able to make outgoing calls. I can only call my voicemail at this point because there is no reinvite when I call voicemail. If VS is serious about their BYOD plans, they should consider getting rid of this unecessary reinvite. Same issue here, money in account, worked fine with ata186 till recently, incoming calls work fine. I have to keep switching over to my sellvoip account to make outgoing calls, would rather use VS but cant. here is what i got back from support, settings worked fine till lately: Your ATA is your responsibility as we do not support this device since it is not a certified product on our network. Here is our SIP configuration for your ATA.. double check it as something might be incorrectly set up... The information you need to provision your 3rd party IAD is the following: SIP Proxy IP: 206.165.50.116 SIP Proxy Port: 5060 SIP Registration Domain: i2telecom.com STUN: Disabled SIP Username/Password: Your DID #/ your password Be advised that we do not provide any troubleshooting or maintenance support for 3rd party IAD’s. Also, the provisioning information can change at any time. The i2 Telecom Customer Care Team gimp55
I asked Mark our network guy if we had made any changes to ANYTHING in the last few weeks and he assures me that we have not.
Anyone want to get a free MG3 from me and trade me (or lend me) you Cisco 186 device.
We don't have one, and no one seems to have an answer.
Or if someone comes up with an answer, I will pay some free telcom time.
Anyone have any other thoughts?
Bruce i2telecom | |
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| | | | | | | | gimp55
join:2003-08-16
Ironton, OH
3 edits |  Re: [VoiceStick] BYOD configurations needed
my ata186 for VS | 
my ata186 for VS | 
my ata186 for VS |
I got my outgoing voice working with the ata186,
I had done a factory reset when i used my sellvoip account and it worked fine.
Then i remembered on my VS account i had put in a NAT server and without it the ata186 will call out but no one can hear on the other end,
NATIP:206.165.50.101
NatServer:206.165.50.101
Add these to the 186 and it will work fine.
Anyway for those with a ata186 it works great, the free incoming DID is great and i will be adding more money to my account or go with the unlimited.
I have a Ironton,Oh DID, noticed You don't have them anymore, but was wondering if and when you will have them for Ashland,Ky. Most of the voip DIDs there start with 606-393-XXXX.
Thanks again for a great product
P.S. do i get any free telcom time, HAHA!!!. | |
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| | | | | | | | | BruceN
Hi
join:2006-11-17
Roswell, GA | Re: [VoiceStick] BYOD configurations needed gimp55
Yes, you win the free telcom time. IM me your VS DID and I have a nice suprise for you.
As to the DID, I am doing a cleanup of dead accounts and hope to have a ton of DID's up in about 10 days.
Thanks for the answer. | |
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See 12 replies to this post |
|
| | | | mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA
| said by BruceN :So, I think overall, we are on the right track to get customers great service. Yup, I just couldn't disagree with you anymore, except definitely you are on the right track to lose customers because the great service you refer to only looks good from your point of views and really sucks for customers (wanna be)!
Will it be perfect, heck no, but it is the goal. Depends on how you view it, nothing is perfect and there is always some flaws. So, if you can fix your imperfection, then you can achieve your goals even better with more profits! Remember, the customers are always first! If you don't listen to the customers about their complains, your service is of no good.
BTW, in practice you may think what the heck with my friend who failed trying to sign up for a free DID; however, what you don't see in a bigger picture is how many UK friends he can bring in as your customers. Nah, that will be considered as a big lost to you because you have no live CSR to handle the services.
--
Mazi (UK Non-Geo Phone: +44-703-194-2574) | |
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| | | | | BruceN
Hi
join:2006-11-17
Roswell, GA
·Future Nine Corpor..
·AT&T U-Verse
·Comcast
| Re: [VoiceStick] BYOD configurations needed mazilo, I hear what you are saying BUT we are on the right track.
I am not a young buck with silly ideas. I am an old war horse when it comes to internet marketing and retail.
Yes, we have a problem on sign ups in rare cases. We will find a fix and get it fixed for sure.
I have used this method of customer contact in the past in a far larger company I owned. There are many problems to doing service right. The customer does not want to spend an hour on the phone, wants a person that can fix the problem to speak or deal with.
We were spending 25% of gross income on support. Not that there are that many problems at all, but to have the man power just sit there 95% of the time, waiting for a call...
Then worse, the manpower did not fix the problem (shared call center) they just kicked it to us hours or days later.
Call in now and see how long it is before you get a call back and see who it is that calls you back.
Companies can only spend limited funds on support, and I don't think what we had was the answer, and I know sending it abroad is not the answer.
We MUST use our own people for this. Who is going to show more love to the customer? A temp at a shared call center or ourselves.
We are in a terrible business where EVERYTHING is price! We are never going to win by price as the other guys will just match us. We need to have more loving service, and have features in our product that the other guys do not have.
I am happy to report that business is good, and we might end up being one of the few (maybe the only) that will turn a profit.
And if profit sounds like a bad word, it is a measure of a company doing the right stuff, and a company that will be around for awhile.
Thanks for your input, and sorry for the long answer. | |
|
Test99
Premium
join:2003-04-24
San Jose, CA
·DSL EXTREME
·InPhonex
| Settings for the eyeBeam soft phone.
In Properties, under the Account tab:
Display name: your name
User name: Voicestick DID (including leading 1) 1xxxxxxxxxx
Password: VoiceStick password
Authorization user name: Voicestick DID (including leading 1) 1xxxxxxxxxx
Domain: i2telecom.com
Register with domain: check
Send outbound via: proxy. Address: 206.165.50.116
Leave other settings at their default values, except if needed:
Under the Topology tab:
STUN Server: Use specified server: sip.xten.com
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50775@fwd.pulver.com | |
|
meister_sd
Premium
join:2006-01-29
La Mesa, CA | Bruce, it seems there are alot of enthusiastic people on this board who can help make up some of the manpower that you may/maynot be looking for. Take advantage of some of the people here. I for one am happy with the voicestick account I have. | |
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PhilAIV
join:2002-02-16
Carrollton, GA
clubs:
·AT&T Southeast
| If you end up not supporting asterisk at all we can at least hope that you'll have forwarding to a Sip URL as an option.
Also, Good luck on your house cleaning and you might not want to take Mazilo's advice lightly because he's been around the voip community long enough to know what works and what doesn't.
PhilAIV
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My Blog | |
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markosjal
join:2005-08-06
Mexico
| Just my two cents about VoiceStick.
I tested VoiceStick as I have many other providers. I always need to know what the competition is up to!
My main complaint is that on International calls I often received no call progress tones, just silence till the line was answered! If the called number was busy, I would hear nothing!
I recently re-tested with the same results!
Also I had sent a message to support, through your web portal, noting that the billed amount of at least one destination was different than the posted rate. I believe it was Guadalajara Mexico 52 33 XXX XXXX . I received no reply to that message. Accurately billing according to the posted rate is a basic customer need (and I would say a right!). It is a shame it was not replied to.
I do not want to sound too harsh here, but I need to post what I feel is the truth. | |
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christcorp
Premium
join:2001-05-21
Cheyenne, WY
1 edit | Here's mine for voicestick. Actually, it's the default for my PAP2 I believe. I put it in for my RT31P2 voicestick, and it works fine.
(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
Later... Mike.... | |
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| MannyL
join:2002-10-22
Toms River, NJ | Re: [VoiceStick] BYOD configurations needed I'm trying slo mo's dial plan. Of course now is when my cordless decides the battery level is too low and won't work. I hope to have everything up an running tonight so i can cut the ties to V_N_G_ at the end of my month. | |
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| | MannyL
join:2002-10-22
Toms River, NJ | Re: [VoiceStick] BYOD configurations needed couldn't wait plugged in a wired phone and it works (almost) I can call any number I tried including free information (1-800-Free-411).
The only thing that does not work is calling my voice mail. I am unable to dial 0000 or my own number | |
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| | | slow mo
join:2002-03-19
USA
2 edits | Re: [VoiceStick] BYOD configurations needed You will always get a busy signal when dial your own number. That's how telephone works.
If you want to dial 0000, just add it to the dial plan:
(*xx|0000|{:1aaa}[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|{:1}[2-9]xx[2-9]xxxxxxS0|011xx.)
aaa - area code
Replace { and } with (less than symbol) and (more than symbol).
EDIT: I don't use voice mail on Voicestick so I never put in 0 or 00 or 0000. | |
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| | | | MannyL
join:2002-10-22
Toms River, NJ
| Re: [VoiceStick] BYOD configurations needed said by slow mo :You will always get a busy signal when dial your own number. That's how telephone works. If you want to dial 0000, just add it to the dial plan: (*xx|0000|{:1aaa}[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|{:1}[2-9]xx[2-9]xxxxxxS0|011xx.) aaa - area code Replace { and } with (less than symbol) and (more than symbol). Normaly from a phone that has voicemail I've been able to dial my own number and it sends me right into voicemail. I added 0000 to the dialing plan and am able to connect into my voice mail box. It doesn't ask me for a password It then ignore any options I press from my phone. I did call into my cell phone and that provider does work with my touchtones. I'll have to just call in from my other line to setup the voicemail. Also it is not sending my callier ID info (shows as unavailable number) when I call out. is this normal? EDIT: I don't use voice mail on Voicestick so I never put in 0 or 00 or 0000. | |
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| | | | | BruceN
Hi
join:2006-11-17
Roswell, GA | Re: [VoiceStick] BYOD configurations needed As to the 800 number. A couple weeks ago we had an issue that turned out to be a carrier problem. You can IM me the 800 number if you want and I can check it out to make sure it is not us.
Bruce
Voicestick | |
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caseydoug
join:2001-08-14
Seattle, WA
| I thought I'd add my two cents to this thread and report that VoiceStick's SIP implementation also will not support a UTStarcom iAN-02EX or an AzaCall 200 (these are essentially the same adapter).
The problem is the same one reported by several others above. Outgoing calls ring, but there is no ringback tone and no audio. Syslog indicates that after the adapter sends an Invite, it receives an Invite, and then the log reports "Local0.Warning state[5] found." I'm not certain what state5 is, but I assume it means that the adapter is already engaged and cannot respond. In any event, when I review calls on my Lingo service, I do not encounter instances where the adapter receives an invite from the server after sending an invite.
This would explain why only outgoing calls are affected, and why the call successfully rings, but does not have ringback tones or audio.
I am not a VoiceStick customer, but was trying to help another user to set up his UTStarcom. | |
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| BruceN
Hi
join:2006-11-17
Roswell, GA | Re: [VoiceStick] BYOD configurations needed I will get this info over to the network guys | |
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| | jdl20
join:2006-11-19
Wilmington, MA
| Re: [VoiceStick] BYOD configurations needed I've taken packet captures when using VS softphone. The second reinvite is sent when making any outgoing calls with the exception of Voicemail. It looks like the softphone adapts to the second reinvite but other atas can't. Why is a reinvite even necessary in VS's SIP implementation? | |
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BruceN
Hi
join:2006-11-17
Roswell, GA | Any of you ever used Open Wengo ? | |
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chichi
Premium
join:2003-12-18
Calgary, AB | Voicestick incoming is free on the next to nothing plan | |
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BruceN
Hi
join:2006-11-17
Roswell, GA | Looking to hear of any usage with open source soft phones? Anyone get any of them to work on Voicestick?
Bruce | |
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devil24
Premium
join:2002-06-28
Houston, TX | I'm having probs with DTMF tones.
Using a PAP2T here. Can you guys post your DTMF settings, please? | |
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| bakermd
join:2007-01-06
Alpharetta, GA | Re: [VoiceStick] BYOD configurations needed Devil24: We support rfc2833 as the DTMF signaling. Please use this instead of Inband signaling.
Thanks. | |
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flroots
join:2007-02-05
Madison, SD | Does anyone have a Grandstream configuration working with VoiceStick? I have the Handytone HT-496, but suspect that any Grandstream configuration would be helpful. Thanks. | |
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| DSLrgm
Premium,MVM
join:2002-08-22
Oak Park, MI
| Re: [VoiceStick] BYOD configurations needed said by flroots :Does anyone have a Grandstream configuration working with VoiceStick? I have the Handytone HT-496, but suspect that any Grandstream configuration would be helpful. Thanks. I will try and look into this. I have 4 HTs, 2 386s and 2 488s.
All for T.38 testing. Which is going too slow... | |
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hfrisch
join:2007-02-13
Holmdel, NJ
| At Bruce's request, here is the configuration for the UTStarcom F1000 and F3000 WiFi phones (these are the same with a small change for the F3000 at the bottom):
The settings for the F1000/G are:
Press Menu -> WiFi Settings -> Signal Protocol -> SIP
Go through the items on that menu one at a time (from the top):
SIP Register Server Mode = DNS (not IP)
SIP Register Server Domain = i2telecom.com
SIP Register Server IP Address = 0.0.0.0 (not used if DNS)
SIP Register Server Port = 5060
SIP Outbound Proxy = IP (not DNS)
SIP Outbound Server Domain = (not used if IP)
SIP Outbound Server IP Address = 206.165.50.116
SIP Outbound Server Port = 5060
SIP User Name = The user Name that you get from i2telecom
SIP Authentication String = Same as SIP user name - enter here as well
SIP Password = your SIP password - not necessarily the account management login password (I don't have an account so I can't verify this)
To administer the SIP Password on the F1000, you need to enter the security code for the phone (default is 888888)
For the F3000, to get to the initial menu, you would start with:
Menu - WiFi - Signal Protocol - SIP - (Select the profile that you want, out of 3 available) - Set - then you are in the same place as on the F1000. The F3000 also has an additional field, which is the Profile Name - this is used in the menus as well as to show on the screen when the phone is registered but idle. It is not used at all in SIP processing.
I'll also include the F3000 screen here. | |
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