Who else on the list is using them, particularly in a hosted environment? We've just decided to transition to them as our primary recommendation instead of the Cisco SPA series. We did it because of the value and feature set, like having an inexpensive phone with a small BLF, which a lot of customers asked for. I'm wondering if others have tips they've learned along the way, or any advice they want to offer. Also anyone using the advanced features like the browser for anything useful? For those who haven't tried them, or who like us, didn't like their older models, take another look. We have been surprised at the value they give us. The prices are low, but the functionality and quality are high. They aren't Polycom 600s to be sure, but they are nice phones that have a huge set of features for a great price. Customers are liking them a lot. Has anyone used the new DECT phone? We currently use the Panasonic DECT phones but they are a nightmare to configure. If anyone wants to get in touch with them, our Grandstream contact is Dennis Ryan, dryan at grandstream.com . -- Carlos Alvarez TelEvolve 602-889-3003 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120816/70010950/attachment.htm>
Carlos, I am waiting for my Grandstreams to arrive too. Similar reasons. Great feature set, reasonable price. My primary interest is security. Grandstream claims their intermediate and higher-end models support TLS and SRTP. I am really tired of trying to make Cisco phones to communicate securely with Asterisk. Cisco has a great security model but one has to have their provisioning server for it to function. Regards, Vladimir On 8/16/2012 7:07 PM, Carlos Alvarez wrote:> Who else on the list is using them, particularly in a hosted > environment? We've just decided to transition to them as our primary > recommendation instead of the Cisco SPA series. We did it because of > the value and feature set, like having an inexpensive phone with a > small BLF, which a lot of customers asked for. I'm wondering if > others have tips they've learned along the way, or any advice they > want to offer. Also anyone using the advanced features like the > browser for anything useful? > > For those who haven't tried them, or who like us, didn't like their > older models, take another look. We have been surprised at the value > they give us. The prices are low, but the functionality and quality > are high. They aren't Polycom 600s to be sure, but they are nice > phones that have a huge set of features for a great price. Customers > are liking them a lot. > > Has anyone used the new DECT phone? We currently use the Panasonic > DECT phones but they are a nightmare to configure. > > If anyone wants to get in touch with them, our Grandstream contact is > Dennis Ryan, dryan at grandstream.com <mailto:dryan at grandstream.com> . > > -- > Carlos Alvarez > TelEvolve > 602-889-3003 > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120817/9e8cc9c9/attachment.htm>
On Fri, 2012-08-17 at 09:30 -0700, Carlos Alvarez wrote:> On Fri, Aug 17, 2012 at 9:08 AM, Vladimir Mikhelson > <vlad at mikhelson.com> wrote: > My primary interest is security. Grandstream claims their > intermediate and higher-end models support TLS and SRTP. I am > really tired of trying to make Cisco phones to communicate > securely with Asterisk. Cisco has a great security model but > one has to have their provisioning server for it to function. > > > > We've never had customers ask for this, but if doing so is fairly easy > we would look at it as just another feature we push. Do let me know > how it works out for you. >We are very happy with most Grandstream products we work very closely with their engineering and support guys they are very good about getting us new features and bug fixes, and yes like most VOIP hardware they do have lot's of bugs but they actually fix most of them. The only current products I don't recommend from them right now are the GXV-3175 (Slow processor and very buggy firmware), GXV-3000 (Has a handset mic overdrive issue. Have to add a resistor to the inside of the handset). I am holding on the new Dect DP715 for the next firmware rev (it is a good product at a fair price but several firmware bugs are show stoppers. We are awaiting beta fixes sometime next week and then we will start shipping them) Hope this helps. zktech -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120817/8ba56bd9/attachment.htm>
Vladimir We are testing the DP715 very aggressively. We have been please with the units for the most part, but we too have been working bugs with Grandstream. We have several in so far and a number of feature requests as well. I deal directly with several of the support engineers and they bring in the developers when necessary. I would be open to working with you on your issue. If I can create validation tests for your items and reproduce the issue I have had great success getting them to take note and address issues they really do want to address issues. In less than two weeks they have given me test builds address two of our issues and they are working on several others. Because of the cooperation of Grandsteam we are close to being able to offer the DP715 phones to our customers. Even then they will have more items to address to allow for full feature deployments but they are serious about the DP715 product. Thanks Bryant Zimmerman (ZK Tech Inc.) ---------------------------------------- From: "Vladimir Mikhelson" <vlad at mikhelson.com> Sent: Friday, August 31, 2012 9:07 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Grandstream VoIP phones Carlos, So far the experience with DP715 is extremely negative. It all starts with the WEB interface which is only served on port 80, no https, period. There is no login name, just password. The phone worked as expected with insecure SIP and RTP. As I started playing with security the phone started acting up. It randomly took calls, then stopped. It placed calls, then stopped. Following is a sample of a corrupted SIP message Asterisk receives from DP715 (pay attention to Call-ID: 477744485-5061-8 at BHC.BH.BDH.HB): [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/TLS 172.17.137.11:5061;branch=z9hG4bK2f5ce157;rport=5061 [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 2 [ 57]: From: <sip:*97 at pbx.int.mikhelson.com:5061>;tag=as50c4dc59 [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 3 [ 54]: To: <sip:471 at pbx.int.mikhelson.com:5061>;tag=436538044 [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 4 [ 39]: Call-ID: 477744485-5061-8 at BHC.BH.BDH.HB [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 6 [ 51]: Contact: <sip:471 at 172.17.137.71:5061;transport=tls> [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 7 [ 43]: Supported: replaces, path, timer, eventlist [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 8 [ 37]: User-Agent: Grandstream DP715 1.0.0.5 [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 9 [ 80]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 10 [ 17]: Content-Length: 0 According to RFC 3261, "Call-ID contains a globally unique identifier for this call, generated by the combination of a random string and the softphone's host name or IP address." Interestingly, the problem is intermittent. Some calls go through. Asterisk must be able to process these calls from time to time. Which is strange on its own. On top of everything Grandstream's support organization does not seem to exist for all practical purposes. I opened the case on 08/22/2012. Today, 08/31/2012, I finally received a response, "Sorry for missing your call yesterday. We checked the syslog you sent to us and seems the TLS is shut down. I just got some TLS internal test accounts today and will do a quick test. I'll let you know soon. It took them 9 days to start looking into the issue. I will update this thread with progress. Regards, Vladimir On 8/17/2012 11:30 AM, Carlos Alvarez wrote: On Fri, Aug 17, 2012 at 9:08 AM, Vladimir Mikhelson <vlad at mikhelson.com> wrote: My primary interest is security. Grandstream claims their intermediate and higher-end models support TLS and SRTP. I am really tired of trying to make Cisco phones to communicate securely with Asterisk. Cisco has a great security model but one has to have their provisioning server for it to function. We've never had customers ask for this, but if doing so is fairly easy we would look at it as just another feature we push. Do let me know how it works out for you. -- Carlos Alvarez TelEvolve 602-889-3003 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120831/6e53cc1a/attachment.htm>
On 9/1/2012 8:27 AM, Patrick Lists wrote:> On 01-09-12 04:14, Vladimir Mikhelson wrote: > [snip] >> * Ability to send host name or other CN not equal to the phone IP in >> TLS negotiation > > Afaik you usually put alternative CNs in SubjectAltName in the > certificate. Have you tried that? > > Regards, > Patrick >Patrick, Thank you for the hint. I will try and report whether DP715 supported this. -Vladimir> > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >