Tom Waterman

Ok I have 2 of these Aastra 57iCT phones and I need them to be able to record a call with out the other party hearing the DTMF tones. On the Snom 360s I have it is easy you push the record button and you are all set. Can this be doen with the Aastra or am I stuck using the start codes? Tom

Ok here is one of the voicemails. All you hear is the sound at the begining and the rest of the voicemail is dead air. msg576.wav

This was fixed with the new release of pbxnsip.

Thank you, now I know I am not going nuts. But what is happening to the voicemails that are being left? DOes PBXnSIP have a fix for this?

I have 2 users each in a seperate location on 2 different PBXs. Each user will let me know that occasionaly they will get voice mails that sound exactly like a couple of LOUD farts. One of them happens to be the owner of our company. We are running the newest version in both locations however this was happening on the older version as well. There have been no spikes in CPU usuage or anything out of the ordinary and it is just one user in each location. I have also deleted there accounts and tried again with no luck. Any help would be greatly appreciated. Tom

In the 360 to get it to provision I did the following: On the Security Tab I have the usernameof: twaterman@md-911 and my account passowrd Next on the update tab I have: update automatically the settings url of thge server is: http://ip address/provisioning/snom360.htm refresh timer = 0 subscribe conf = off PnP config = off reboot and the magic happened. Now as far as drilling down down through the directory. Can you explain how this is done? I get the first 32 number no problem. If I push 8 I get some different number but all of our company extension begin with 8. Thanks for the help. Tom

OK I got this to work. I just have one more question about the address book. When I press the directory button it gives me the first 32 numbers but that is it. Is there a way to make it fetch more of the list from the server? Thank you! Tom

I am trying to auto provision a snom 360 with firmware 7.3.14. My phone has a DHCP address of 172.16.x.x my server has na ip address of 172.31.x.x The server sits in the DMZ. I can see the phone requesting the file snom360.htm. I have the file below named as such and it is in the tftp and html directories. How ever each time the phone requests the file I see an authentication error. In the PBX under settings-->PNP I have the snom admin password set to 1234. Then under the domain settings--->provisioning parameters I have the password set to 1234. In the phone under security I have the password set to 1234 with a blank username, I have also tried using my account username and password here as well, no dice. Under the update section on the snom I have it seto to update automatically. The setting URL is http://172.31.x.x/provisioning/snom360.htm Subscribe and PnP config are both off. And I get an authorization error in Wireshark everytime. Any help would be great. here is my snom360 config <?xml version="1.0" encoding="utf-8" ?> - <settings> - <phone-settings e="2"> <redirect_event perm="">none</redirect_event> <setting_server perm="">172.31.3.10</setting_server> <subscribe_config perm="">off</subscribe_config> <ip_adr perm="RW">172.16.2.30</ip_adr> <netmask perm="RW">255.255.255.0</netmask> <update_server perm="RW">172.31.3.10</update_server> <dns_domain perm="RW">mdgn.microdatagis.com</dns_domain> <dns_server1 perm="RW">172.16.3.1</dns_server1> <dns_server2 perm="RW">172.16.3.3</dns_server2> <gateway perm="RW">172.16.2.254</gateway> <utc_offset perm="">-18000</utc_offset> <ntp_server perm="RW">172.16.1.1</ntp_server> <http_pass perm="">*****</http_pass> <dst perm="">3600 03.02.07 02:00:00 11.01.07 02:00:00</dst> <timezone perm="">USA-5</timezone> <admin_mode perm="">on</admin_mode> <tone_scheme perm="">USA</tone_scheme> <logon_wizard perm="">on</logon_wizard> <update_policy perm="">auto_update</update_policy> <firmware_version perm="">snom360-SIP 7.3.14</firmware_version> <http_client_pass perm="">*****</http_client_pass> <use_hidden_tags perm="">on</use_hidden_tags> <uboot_version perm="">1.1.3-m</uboot_version> <user_name idx="1" perm="">858</user_name> <user_host idx="1" perm="">srpbx01.mdgn.microdatagis.com</user_host> <user_pass idx="1" perm="">858</user_pass> <user_uid idx="1" perm=""><urn:uuid:0b2f8928-4af7-4214-a4a5-b20f12d91c65></user_uid> </phone-settings> <functionKeys e="2" /> <tbook e="2" /> </settings> Also where is the proper placement of this file! Thank you all and by the way we are running pbxnsip version 3.3.1.3177 (Win32)

Awesome that was the answer I was looking for!!

OK so here is the issue. I have powered on the phone and just entered my user name and password and IP info into the phone. I am registered fine. My buttons that do not work are Conference, DND, Menu. Again these are the regular buttons on the Snom 360. My firmware is 7.3.10a. I did not auto provision these. They are 100 phones we put in a customers site last month and they had no DHCP server at the time. Please help. Tom

Thank you! It appears that this will be corrected in the next firmware release.

This is starting to get a lot of folks very angry. It is tough to troubleshoot because it seems to be random but once it starts on a base the only thing that will fix it is a restart of the base. Tom

Opps I had the Domain blank, I should have had callcentric.com Once I put that in there it registered fine. Tom

I just set up a set line with callcentric. if I use my x-lite softphone it works fine. However I created the trunk in PBXnSIP to register and it fails with the following error message: 8] 2009/03/18 15:36:00: Trunk 4 (callcentric) has outbound proxy udp:204.11.192.23:5080 udp:204.11.192.35:5060 udp:204.11.192.37:5080 [5] 2009/03/18 15:36:00: Registration on trunk 4 (callcentric) failed. Retry in 60 seconds Anyone have any ideas? Tom

Of course I did, however I get nothing when I try it. I am going to recreate it in the lab and post the logs.

They are looking for just 3 way conferencing. Just the caller and 2 office phones. By the way I set up aconference extension here at our office the other day and it was awesome! Tom

I have a rather silly question. I have 9 remote phones they are Snom 370s and the customer is reporting that the conference feature does not work. I am assuming they should just push the conference button followed by the extension and the check mark to send the call. But apparently this does nothing. Any help would be great. They are running the newest firmware that Kevin and I setup on the trip there. Tom

I am using ver 1.22 and the DTMF will work for a week or so on the base then just crap out. Restarting the base works but I have 20 of them. Any word on this?

Where is that setting on the snom M3 I can't seem to find it. Also we are getting terrible feedback with the speaker on these M3s. I know it is the speaker feedbacking into the MIC and this causes the speaker to cut out. Any ideas? Tom

We just turned it off completly. We had "Disconnect call on silence detection" turned off already, I then went in and "SIlence Detection Method" to none on the audiocodes gateway and rebooted them. I will report back. Tom

OK I just got a Syslog and checked it out. It looks like there is a silent disconnect release message. Here is the log. SIP/2.0 Via: SIP/2.0/UDP 172.31.3.11;branch=z9hG4bKac1025123861 Max-Forwards: 70 From: <sip:18776713355@172.31.3.11;user=phone>;tag=1c707751253 To: "Jessica Darling" <sip:807@pbx.mdgn.microdatagis.com>;tag=54922 Call-ID: 0abecac0@pbx CSeq: 1 BYE Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.40A.027.001 Reason: Q.850 ;cause=16 ;text="Silence Disconnect" Content-Length: 0 [Time: 8:49:40] 08:48:04.213 : 172.31.3.11 : NOTICE : ( sip_stack)(125741 ) UdpRtxMngr::Transmit 1 BYE Rtx Left: 6 Dest: 172.31.3.10:5060 CallID: (0abecac0@pbx) [Time: 8:49:40] 08:48:04.214 : 172.31.3.11 : NOTICE : ( sip_stack)(125742 ) SIPCall(#17) changes state from Connected to Disconnected [Time: 8:49:40] 08:48:04.214 : 172.31.3.11 : NOTICE : ( lgr_stk_ses)(125743 ) <SESSION #8> SendToCall - event: RELEASE_ACK m_Call = 31513336 [Time: 8:49:40] 08:48:04.215 : 172.31.3.11 : NOTICE : ( 08:48:04.202 : 172.31.3.11 : NOTICE : BYE lgr_flow)(125744 ) | | #8:RELEASE_ACK:(0abecac0@pbx) [Time: 8:49:40] 08:48:04.216 : 172.31.3.11 : NOTICE : ( sip_stack)(125745 ) AcSIPStackAPI::FreeCallAPI - #8 [Time: 8:49:40] 08:48:04.216 : 172.31.3.11 : NOTICE : ( sip_stack)(125746 ) Setting ApplicationCall of AcSIPCall #17 to NULL [Time: 8:49:40] 08:48:04.217 : 172.31.3.11 : NOTICE : ( lgr_stk_mngr)(125747 ) Resource StackSession <#8> Deleted [Time: 8:49:40] 08:48:04.217 : 172.31.3.11 : NOTICE : ( lgr_flow)(125748 ) | | #8:RELEASE:(0abecac0@pbx) [Time: 8:49:40] 08:48:04.218 : 172.31.3.11 : NOTICE : ( lgr_flow)(125749 ) | | #8:Call changing states from:DisconnectingState to:DisconnectingState [Time: 8:49:40] 08:48:04.218 : 172.31.3.11 : NOTICE : ( lgr_flow)(125750 ) | #6:RELEASE RELEASE_BECAUSE_SILENCE_DISC : (0abecac0@pbx) [Time: 8:49:40] 08:48:04.218 : 172.31.3.11 : NOTICE : ( lgr_flow)(125751 ) | #6:Close voice Channel [Time: 8:49:40] 08:48:04.219 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125752 ) #6:StopRTP_RTCP on channel 6 [Time: 8:49:40] 08:48:04.219 : 172.31.3.11 : NOTICE : ( lgr_flow)(125753 ) | #6:RELEASE_ACK (send) : (0abecac0@pbx) [Time: 8:49:40] 08:48:04.219 : 172.31.3.11 : NOTICE : ( lgr_flow)(125754 ) | | #8:RELEASE_ACK:(0abecac0@pbx) [Time: 8:49:40] 08:48:04.219 : 172.31.3.11 : NOTICE : ( lgr_callf)(125755 ) Call #8 deleted [Time: 8:49:40] 08:48:04.220 : 172.31.3.11 : NOTICE : ( lgr_psbrdex)(125756 ) InsertBoardEvent- event 105 inserted channel 6 [Time: 8:49:40] 08:48:04.220 : 172.31.3.11 : NOTICE : ( lgr_flow)(125757 ) #6:RELEASE_BECAUSE_IP_TIMER_EXPIRED_EV [Time: 8:49:40] 08:48:04.221 : 172.31.3.11 : NOTICE : ( lgr_flow)(125758 ) | #6:RELEASE_BECAUSE_IP_TIMER_EXPIRED_EV [Time: 8:49:40] 08:48:04.221 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125759 ) #6:cpDigitMapHndlr_Stop - Stoped (0) [Time: 8:49:40] 08:48:04.222 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125760 ) #6:CloseChannel: ChannelNum=6 [Time: 8:49:40] 08:48:04.223 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125761 ) Open channel: IsVoiceOn: 1, IsT38On: 0, IsVbdOn: 0, IsVideoOn: 0 [Time: 8:49:40] 08:48:04.223 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125762 ) #6:OpenChannel:on Trunk -1 BChannel:6 CID=6 with VoiceCoder: g711Ulaw64k20 VbdCoder: InvalidCoder255 DetectorSide: 0 FaxModemDet NO_FAX_MODEM_DETECTED [Time: 8:49:40] 08:48:04.224 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125763 ) #6:OpenChannel VoiceVolume= 1, DTMFVolume = -11, InputGain = 0, RTPRedundancyDepth = 0 FlashHookPeriod = 700 AgcCmd = 0x13180000 [Time: 8:49:40] 08:48:04.224 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125764 ) OpenChannel, CoderType = 1, Interval = 3, M = 1 [Time: 8:49:40] 08:48:04.225 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125765 ) #6:FAXTransportType = 1 [Time: 8:49:40] 08:48:04.225 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125766 ) #6:ConfigFaxModemChannelParams NSEMode=0, CNGDetMode=0, FAXTranType=1, VxxTranType=2, VoiceVol= 1, DTMFVol=-11, InGain=0, RTPRedDepth=0, ECE=1, SCE=1, ECNlpMode=0, DJBufMinDelay=10, DJBufOptFac=10) [Time: 8:49:40] 08:48:04.226 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125767 ) Detectors: Amd:0, Ans:0 En:0 IBScmd:0xa1 [Time: 8:49:40] 08:48:04.229 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125768 ) Turn ringer OFF for channel 6 [Time: 8:49:40] 08:48:04.230 : 172.31.3.11 : NOTICE : ( lgr_flow)(125769 ) | #6:FXO Release Line [Time: 8:49:40] 08:48:04.231 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125770 ) #6:PSOSBoardInterface::StopPlayTone- Called [Time: 8:49:40] 08:48:04.232 : 172.31.3.11 : NOTICE : ( lgr_flow)(125771 ) ---- Incoming SIP Message from 172.31.3.10:5060 ---- [Time: 8:49:40]

Can I upgrade form 3.2 to 3.3 by just stopiing the service and then dropping in the the new controller? And which version should I use I have 3.3.0.3147 and 3.3.0.3152 Thank you

I have tried to turn on the email notification and this does not work. I have upgraded to version 3.2.0.3144. I did this a couple of days ago. Yesterday I had a couple reports of dropped calls in mid sentence and I have been in the office for 1.5 hours and have 10 reports already. If this does not get fixed soon we may need to pull the plug on this and revert back to our old system.

I have noticed that if the caller leaving the voice mail presses # then the voicemail is delivered to my email as well as staying new so if I call my extension to get my voicemail there it is as well as it showing up on my Snom M3 handset. However if I do not press # and simply hang up the voicemail is deliverd to my email but it does not even stay in my voice mailbox(marked as new or read). This is an issue that must be corrected as I have now about 100 accounts with this issue and 30 more are coming on line in 30-45 days. I am using ver 3.2.0.3144 Tom

Yes I have the same issue with 3.1.2.3120. And I have not updraged to 3.2 yet.