Vladimir
I have been working with Grandstream on the DP715 firmware. Can you give me
screen shots of your configs or a download of it, and possibly some
asterisk config examples of how your system is set so I can try your
configs in our test env. We have the DP715 units working with the new
firmware. Also are you dealing with US support or other country? I would
like to offer your feed back to the US project engineers for the product,
and any info such as ticket numbers and support agent names would be
helpful.
Your experience of thier support is not the Grandstream I know, and I would
like to get to the bottom of the issue.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
----------------------------------------
From: "Vladimir Mikhelson" <vlad at mikhelson.com>
Sent: Saturday, September 22, 2012 2:55 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Grandstream VoIP phones
Quick update.
Grandstream finally released the first update to theirDP715 firmware, new
v. 1.0.0.8.
Here are the differences:
I can receive calls over secure SIP and RTP No outgoing calls go
through
What I observed the phone replies from a different port compared to a port
it receives SIP messages on. As a result Asterisk becomes confused. For
example, "sip set debug peer 999" would only track messages to the
phone.
Grandstream's support is beyond the level of criticism. It takes them 10
days to reply to a posted message. It seems their only goal is to close
the case. So far I am still to see a single bit of help from them.
I will continue updating this thread.
-Vladimir
On 8/31/2012 8:07 PM, Vladimir Mikhelson wrote:
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
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-- _____________________________________________________________________ --
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Vladimir
DP715 Phone Only.
Also another point of note. At present I would not promote the DP715 as an
executive level advanced feature phone at best it is a residential grade
unit with the current firmware. This last firmware release fixed some major
issues but crippled the unit from four concurrent calls to two if you are
using g729. This is a big kick in the paints and shows some possible
engineering shortcomings of the units. We are talking to the engineers to
see what their product will look like once the firmware is closer to
production ready. At current we have downgraded our release state from
production to beta on our network. Several customers are very please with
the units but it has failed to meet the expectations of others.
Take a close look at the release notes for the DP715 this will infer some
of what was wrong in the first release and give you a kind of idea of where
the product still needs to go.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
----------------------------------------
From: "Vladimir Mikhelson" <vlad at mikhelson.com>
Sent: Saturday, September 22, 2012 2:55 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Grandstream VoIP phones
Quick update.
Grandstream finally released the first update to theirDP715 firmware, new
v. 1.0.0.8.
Here are the differences:
I can receive calls over secure SIP and RTP No outgoing calls go
through
What I observed the phone replies from a different port compared to a port
it receives SIP messages on. As a result Asterisk becomes confused. For
example, "sip set debug peer 999" would only track messages to the
phone.
Grandstream's support is beyond the level of criticism. It takes them 10
days to reply to a posted message. It seems their only goal is to close
the case. So far I am still to see a single bit of help from them.
I will continue updating this thread.
-Vladimir
On 8/31/2012 8:07 PM, Vladimir Mikhelson wrote:
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
-- _____________________________________________________________________ --
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
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