Re: [Asterisk-Users] grandstream sip phone to analog not working

2005-07-03 Thread Rich Adamson
> Ive got 3 analog phones and 2 grandstream sip phones working with > asterisk, the problem is that although the analog phones can talk to > each other and the sip phones can talk to each other the two types dont > seem to be able to cross communicate. > It looks as though the SIP phones are se

[Asterisk-Users] grandstream sip phone to analog not working

2005-07-03 Thread Andrew Bush
Hi all, Ive got 3 analog phones and 2 grandstream sip phones working with asterisk, the problem is that although the analog phones can talk to each other and the sip phones can talk to each other the two types dont seem to be able to cross communicate. It looks as though the SIP phones are set

[Asterisk-Users] grandstream sip phone calling Zap/1 on TDM20Brings and answers but not hear voice

2005-01-23 Thread Jerry Geis
Here is the console screen. Starting simple switch on Zap/1-1 Executing Dial("Zap/1-1", "SIP/403") in new stack Called 403 SIP/403-9c60 is ringing SIP/403-9c60 answered Zap/1 Spawn extension (smvoice-incoming, 403, 1) exited nonzero on Zap/1-1 Hangup Zap/1 I have a grandstream 101

Re: [Asterisk-Users] grandstream sip phone calling Zap/1 on TDM20B rings and answers but not hear voice

2005-01-23 Thread timebandit001
Could you give us the output of the console when you try the call ? That would help us to point you in the right direction. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSC

[Asterisk-Users] grandstream sip phone calling Zap/1 on TDM20B rings and answers but not hear voice

2005-01-22 Thread Jerry Geis
I have a grandstream 101 that is calling an extension on Zap/1 of a TDM20B. The grandstream 101 can call another grandstream 101 at a different extension- that works fine. The two phones on TDM 20B can call each other.- no problem.When I call the TDM20B Zap/1 from the grandstream phone it rings -

Re: [Asterisk-Users] grandstream sip phone (NTP)

2003-07-17 Thread Stephen R. Besch
I have solved the time server problem with the Grandstream by having my * box's NTP service mirror a public NTP server. I had to do this because my phones are all on the 192.168 subnet, which is non-routable. For example, assuming that the NTP service is configured and running on your * box, c

Re: [Asterisk-Users] grandstream sip phone

2003-07-17 Thread Dave Cotton
On Thu, 2003-07-17 at 08:40, Rainer Jochem wrote: > There's also a DPH-80: > http://www.dlink.co.in/dlink/Products/voip/dph80.htm > > (Found with google) But without a VoIP system it'll probable cost more than the phone itself in phone bills to convince a DLink India reseller to send one to Euro

Re: [Asterisk-Users] grandstream sip phone

2003-07-16 Thread Rainer Jochem
> I just looked on dlink's site and the only one I can find is the > DHP-100. There's also a DPH-80: http://www.dlink.co.in/dlink/Products/voip/dph80.htm (Found with google) -- http://graphics.cs.uni-sb.de/VoIP/ ___ Asterisk-Users mailing list [EMAI

Re: [Asterisk-Users] grandstream sip phone

2003-07-16 Thread Dave Cotton
On Thu, 2003-07-17 at 08:17, Kelvin Chua wrote: > do you have any technical specification of this dlink sip phone? or > pictures? links? i can't seem to find any related literature on this. thanks > > > Dlink has the dhp-90 (currently in limited release like Grandstream) for > > $60-70. It doesn;

Re: [Asterisk-Users] grandstream sip phone

2003-07-16 Thread Kelvin Chua
7, 2003 5:18 AM Subject: Re: [Asterisk-Users] grandstream sip phone > Dlink has the dhp-90 (currently in limited release like Grandstream) for > $60-70. It doesn;t have a digital display- but it works flawlessly. > > There are a few others- you just need to look around... > > -GSR

Re: [Asterisk-Users] grandstream sip phone

2003-07-16 Thread Greg Renouf
lt;[EMAIL PROTECTED]> Sent: Wednesday, July 16, 2003 10:02 PM Subject: Re: [Asterisk-Users] grandstream sip phone > On Wednesday 16 July 2003 03:52 pm, Greg Renouf wrote: > > > Grandstream can improve the quality of their 'user interface' (many others > > have a

Re: [Asterisk-Users] grandstream sip phone

2003-07-16 Thread Stefano Finetti
- Original Message - From: "WipeOut ." <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 16, 2003 10:19 PM Subject: Re: [Asterisk-Users] grandstream sip phone > > I have tried many public NTP servers and all have the same result.. > Wait

Re: [Asterisk-Users] grandstream sip phone

2003-07-16 Thread WipeOut .
> Have you tried to mantain the default ntp server on your phone? (the *.gov > one) > > I normally use internal ntp servers but in a particular context i've used > that ntp server and it worked perfectly. I have tried many public NTP servers and all have the same result.. > > Could be a Firewal

Re: [Asterisk-Users] grandstream sip phone

2003-07-16 Thread marrandy
On Wednesday 16 July 2003 03:52 pm, Greg Renouf wrote: > Grandstream can improve the quality of their 'user interface' (many others > have already accomplished this goal,) I can see very few situations where > the $10-20 cost saving will make the quality sacrifice worthwhile. What other phones a

Re: [Asterisk-Users] grandstream sip phone

2003-07-16 Thread Greg Renouf
D]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 16, 2003 8:49 PM Subject: Re: [Asterisk-Users] grandstream sip phone > I'm working greatly with 40+ Grandstream phones. Audio quality is good > enough for production environment, the cost is really low and the > configuration

Re: [Asterisk-Users] grandstream sip phone

2003-07-16 Thread Stefano Finetti
I'm working greatly with 40+ Grandstream phones. Audio quality is good enough for production environment, the cost is really low and the configuration is *Really* easy. But a little answer to Wipeout is: > The only issue that I still have is that the phone does not seem to be able to pickup the t

Re: [Asterisk-Users] grandstream sip phone

2003-07-16 Thread WipeOut .
I have been testing a couple of them for about 2 weeks now.. They are very good for the price.. The only issue that I still have is that the phone does not seem to be able to pickup the time correctly from an NTP server that is not on the local network so the display always shows 1900-XX-XX for

Re: [Asterisk-Users] grandstream sip phone

2003-07-16 Thread Patrick
On Wed, 2003-07-16 at 15:44, Marian Danisek wrote: > hello, > > i found in list archives some notes about grandstream sip voip phones. > Does anybody succesfuly tested those phones with asterisk ? Mark ? They seem to work with asterisk. I don't yet have a couple myself but on irc there are people

Re: [Asterisk-Users] grandstream sip phone

2003-07-16 Thread Steve Creel
I asked [EMAIL PROTECTED] the other day. They wrote back: > US list retail price of BudgeTone SIP phones: > Model 101 $75/ea (available now) > Model 102 $85/ea (available now) > > US list retail price of HandyTone VoIP analog telephone adaptor: > $75/ea (available in late July 2003) > >P

[Asterisk-Users] grandstream sip phone

2003-07-16 Thread Marian Danisek
hello, i found in list archives some notes about grandstream sip voip phones. Does anybody succesfuly tested those phones with asterisk ? Mark ? What about the prices ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-