[Asterisk-Users] grandstream sip phone to analog not working
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 to connect directly, and are failing to do so. Has anyone else had this problem? is there a work around? thanks for any help. Yours cheerfully, Andrew Bush ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone to analog not working
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 to connect directly, and are failing to do so. Has anyone else had this problem? is there a work around? Lots of people have systems working with the above plus a lot more. Your problem is very likely related to not understanding the configuration necessary to accomplish the goal. No one is going to be able to even suggest anything if you don't show us the appropriate pieces of your *.conf files. Since you haven't told us how you connected the analog phones to asterisk, we can't even tell you which *.conf files are needed. Taking a guess, show us the sip.conf entries for the devices you're trying to use, extensions.conf, and if you're using the TDM card then the zapata.conf pieces used to define the analog phones. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone calling Zap/1 on TDM20B rings and answers but not hear voice
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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] grandstream sip phone calling Zap/1 on TDM20Brings and answers but not hear voice
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 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 - I answer and I dont hear any voice. for the grandstream I have tried allow=all for the codes but made no difference. Any ideas on what I am missing? Thanks, Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] grandstream sip phone calling Zap/1 on TDM20B rings and answers but not hear voice
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 - I answer and I dont hear any voice. for the grandstream I have tried allow=all for the codes but made no difference. Any ideas on what I am missing? Thanks, Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
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 - Original Message - From: Greg Renouf [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 17, 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 - Original Message - From: marrandy [EMAIL PROTECTED] To: [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 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 are in the $95-$105 range ??? Regards...Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
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;t have a digital display- but it works flawlessly. I just looked on dlink's site and the only one I can find is the DHP-100. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
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 Europe, the US or Australia. It's not a case of DLink dumping old stock to the developing world, is it. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone (NTP)
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, create an NTP mirror which allows access from machines on 192.168.10.X by adding the following line to the ntp.conf file: restrict 192.168.10.0 mask 255.255.255.0 notrust nomodify notrap The IP range and netmask arguments are obvious. The 3 option flags tell the ntp daemon that none of the machines that might communicate over this subnet are to be trusted as time servers, none of them are to be allowed to update the ntp daemon running on the asterisk server, and none of them will be able to use the trap service for logging purposes. Finally, I also like to set up a different (from the one used by the phones for SIP and RTP) IP address for the NTP server (so the * box has 2 addresses on the 192.168 net). It goes without saying that the asterisk box must also have a public IP address so that it can synchronize itself with a remote time server. In my setup, I have one net card for the public address, while the 2 192.168 addresses are on a second card. -- Stephen R. Besch, Ph.D. SachsLab 320 Cary Hall SUNY at Buffalo Buffalo, NY 14214 (716) 829-3289 x106 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
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) Please contact our reseller (Ovislink/dgtimes) regarding your sample purchase. James @ Ovislink/dgtimes can be reached at tel: (626) 854-1805 or fax: 626.854.0835 and [EMAIL PROTECTED] Their web site is at: www.ovislink.com On Wed, 16 Jul 2003, 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 ? What about the prices ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
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 who use them. Join #asterisk on irc.freenode.net (or .com don't remember) and ask around. What about the prices ? $85 for the 102 regards Marian Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
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 the date.. This issue I am sure will be solved in the near future.. I also have SNOM200's which are awesome phones but they are over twice the price including shipping of the GS phones to the UK.. Without shipping I would be able to get nearly 3 GS phones for the price of one SNOM200... Unfortunately the GS phone does not have a GSM codec but it does support just about every other codec out there and supports just about every feature you could want for a standard desktop phone.. If you want to know more let me know.. 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: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 time correctly from an NTP server that is not on the local network so the display always shows 1900-XX-XX for the date.. This issue I am sure will be solved in the near future.. 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. Could be a Firewall issue, maybe? It works on every firmware since .58, for me. -- Stefano ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 are in the $95-$105 range ??? Regards...Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
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 Firewall issue, maybe? No its not the firewall becasue I have no problems setting time on variout PC's using NTP and public servers.. Thanks for the thoughts anyway.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
- 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. I have tried many public ntp too. It worked ONLY with the default one. Check for this if you haven't done yet. At the present I don't remember (Sorry) the exact server which worked for me, since I've installed an ntp service on my server. -- Stefano ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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 - Original Message - From: marrandy [EMAIL PROTECTED] To: [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 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 are in the $95-$105 range ??? Regards...Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
