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Messages - brantn

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151
Mitel MiVoice Business/MCD/3300 / Re: SIP Trunk locking up/not releasing calls
« on: July 01, 2011, 07:51:39 PM »
SIP BUSY PEER "Name of Peer" and SIP RTS PEER "Name of Peer" will at least fix it being that you only have two trunks I would probably setup a capture even if you filter it to SIP to see if there is a signaling issue. I would also look into session timer on the sip trunks that should tear it down.

152
Mitel MiVoice Business/MCD/3300 / Re: Backup Fails
« on: June 28, 2011, 11:49:51 AM »
As bobcheese says any error messages or logs? Trusted sites? Clear History on browser? I know those sound dumb but they have affected backups for myself.

153
Mitel MiVoice Business/MCD/3300 / Re: Bluetooth
« on: June 28, 2011, 11:48:04 AM »
I have had almost all of them split after a couple months of use then again a telecom takes a lot of calls.

154
Mitel MiVoice Business/MCD/3300 / Re: Bluetooth
« on: June 27, 2011, 03:35:47 PM »
Not heard of that. Would be really nice as their current headsets are not very good quality.

155
Mitel MiVoice Business/MCD/3300 / Re: No ip lease on mitel 5312 phones
« on: June 25, 2011, 09:19:28 PM »
On vlan 3 enter option 128 and 66 or the easier option on the hp do

conf t
Vlan 4
Voice

Then phones should auto discover with lldp. Make sure no options are in dhcp of the data scope just phone scope. Better advice let the 3300 be the dhcp server on the phone network.

Sent from my SCH-I500 using Tapatalk

156
Mitel Software Applications / Re: prairieFyre Software...question
« on: June 24, 2011, 01:20:42 PM »
It depends on the sync in question but in your question the sync you are referring to is in YSE that imports the user config such as extensions, agents, and trunks if you do not enable this yes you can import and build everything manually.

But PF does pull SMDR and ACD thru their 3300 tap.

So in Evans situation he is not syncing the database but has everything built manually in YSE if I am understanding correctly. Obviously Call Accounting is taping the SMDR stream.

157
Mitel MiVoice Business/MCD/3300 / Re: Fax over SIP
« on: June 23, 2011, 10:11:45 AM »
Not sure on nupoint as I have not done much on that platform with SIP as far as on the 3300 if you are using T.38 you would need T.38 licenses, compression licenses, and dsp II module.

158
Mitel MiVoice Business/MCD/3300 / Re: Paging through loudspeakers and handsets at the same time
« on: June 23, 2011, 10:09:08 AM »
Yeah done it that way as well.

159
Mitel MiVoice Business/MCD/3300 / Re: Fax over SIP
« on: June 22, 2011, 09:30:07 PM »
T.38 is the recommendation on all faxing over SIP. It allows compression which is a lot more reliable than uncompressed when it comes to faxing. Mitel 3300 site to site must be configured for T.38.

160
Mitel MiVoice Business/MCD/3300 / Re: Paging through loudspeakers and handsets at the same time
« on: June 22, 2011, 06:19:55 PM »
Depending on Viking but I have hooked up several of them for this purpose. Add the IP Paging Unit to the same Page Group as the phones, connect the paging unit to the viking and you are good to go. Also of note you need to setup a dhcp reservation for these units and have them untagged on the phone vlan.

161
Mitel MiVoice Business/MCD/3300 / Re: Paging through loudspeakers and handsets at the same time
« on: June 22, 2011, 04:36:26 PM »
Yes you can do this with an ip paging unit. Not sure of cost on them but do allow simultaneous paging as well as talk back paging.

162
Mitel SX200, SX2000, and older SX platforms / Re: Mtel Reconfig - Password Required
« on: June 20, 2011, 01:05:56 PM »
Also this would have to be transferred at your request to a new dealer for several hundred dollars in order for them to be able to get the options password. As far as administration passwords there are some defaults but any good dealer will change them. So a default of the system might have to be done.

163
Mitel MiVoice Business/MCD/3300 / Re: SIP Trunking. Call transfer, can't force SDP in INVITE from 3300
« on: June 20, 2011, 12:47:46 PM »
What are you using for a pstn gateway? The Mitel does send an sdp with the private address, dns address, or 0.0.0.0 in some occasion unless prohibited.

Here is an edited copy of a call with no transfer ...

Initial invite:

INVITE sip:blank@1.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4c88f7a994f2db72f130519551c5fedc.0
From: WIRELESS CALLER <sip:blank@blank:5060>;tag=A7C2.9B55
To: <sip:blank@blank:7061>
Call-ID: 0026.9E9B.B4F2.4DD2.A7C2.1B5A
CSeq: 8738 INVITE
Max-Forwards: 69
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4DD2.A7C2.9B558738.A48B6018.00001B95.zCp3O4Kz4K+KbJ34qyDi3Q__
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4DD2.A7C2.9B558738.-aN6AgMoWp89KmEvOe6hxw__
Contact: <sip:e4coEs7GA1Rbz1_ILEA3EVBH-hVHznloebVs7mglKAoOk-M9uL0tNDEQTg1c2kZ7r@1.2.3.4>
Session-Expires: 1800;refresher=uac
Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE
Supported: timer,100rel
Content-Type: application/sdp
Content-Length: 252
Record-Route: <sip:7c5669fe4b5976c9@1.2.3.4;lr>

v=0
o=BroadSoft 25576 25576 IN IP4 1.2.3.4 (Public SDP Owner)
s=M6 Call
c=IN IP4 1.2.3.4
t=0 0
m=audio 58238 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:40

200 OK:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4c88f7a994f2db72f130519551c5fedc.0,SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4DD2.A7C2.9B558738.A48B6018.00001B95.zCp3O4Kz4K+KbJ34qyDi3Q__,SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4DD2.A7C2.9B558738.-aN6AgMoWp89KmEvOe6hxw__
Record-Route: <sip:7c5669fe4b5976c9@1.2.3.4;lr>
From: WIRELESS CALLER <sip:blank@blank:5060>;tag=A7C2.9B55
To: <sip:blank@blank:7061>;tag=0_2334818960-92952406
Call-ID: 0026.9E9B.B4F2.4DD2.A7C2.1B5A
CSeq: 8738 INVITE
Supported: timer
Contact: <sip:blank@1.1.1.1:5060;transport=udp>
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
Content-Type: application/sdp
User-Agent: Mitel-3300-ICP 10.0.1.21
Content-Length: 162

v=0
o=- 9119 9119 IN IP4 1.2.3.4 (Private SDP Owner Proxies' 200 OK Shows Public)
s=-
c=IN IP4 1.2.3.40
t=0 0
m=audio 50278 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

I have found thru a lot of SIP work it is very wise to use an Ingate proxy as it helps to fix some of the signalling issues that you see between pbxs and soft switches. It also does the nat translation or routing which some firewalls even ones claiming to be sip aware fail on.

164
Mitel MiVoice Business/MCD/3300 / Re: SIP Trunking. Call transfer, can't force SDP in INVITE from 3300
« on: June 20, 2011, 12:11:02 PM »
So it is no longer sending an invalid sdp. What is the problem you are having besides try to dechiper sip signalling? I would be more than happy to look at the wireshark if you want to pm me.

165
Mitel MiVoice Business/MCD/3300 / Re: SIP Trunking. Call transfer, can't force SDP in INVITE from 3300
« on: June 20, 2011, 10:54:50 AM »
I would just set one of the early answer options as well as since I believe MCD and maybe before there is an option to prevent the 0.0.0.0 in sdp.

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