Flavio
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Posts posted by Flavio
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Hello all.
I don't know if it's the correct section to post, anyway: I'd like to know if there is the chance to get the standard/default PBX audio messages also spoken in other languages. Is there any location they could be downloaded from?
Kind regards,
F.
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Hello MR X!
This is what I got in the "Call Log":
2011/04/14 11:29 My Secretary (42) Me Myself (41)(41) 00:27 0911234567
This is the corresponding "Logfile":
[5] 2011/04/14 11:29:52: Identify trunk (line match) 1
[6] 2011/04/14 11:29:52: Sending RTP for qBdMwBMEmhK.qH4IvBtJPGsHjcIryJuq to 172.16.16.17:4000, codec not set yet
[5] 2011/04/14 11:29:52: Global trunk 0911234567@pbx.piramide.local sends call to 72 in domain pbx.piramide.local
[3] 2011/04/14 11:29:52: Hunt group 72 wants to add 4 members
[6] 2011/04/14 11:29:52: Codec pcmu/8000 is chosen for call id qBdMwBMEmhK.qH4IvBtJPGsHjcIryJuq
[6] 2011/04/14 11:29:59: Codec pcmu/8000 is chosen for call id fd630615@pbx
[6] 2011/04/14 11:29:59: Sending RTP for fd630615@pbx to 172.16.16.231:52088, codec pcmu/8000
[5] 2011/04/14 11:29:59: INVITE Response 487 Request Terminated: Terminate 4ef1b0c7@pbx
[5] 2011/04/14 11:29:59: INVITE Response 487 Request Terminated: Terminate 91b1e516@pbx
[5] 2011/04/14 11:29:59: INVITE Response 487 Request Terminated: Terminate 5223328d@pbx
[5] 2011/04/14 11:29:59: INVITE Response 487 Request Terminated: Terminate 773ca7ef@pbx
[6] 2011/04/14 11:30:08: Codec pcmu/8000 is chosen for call id fd630615@pbx
[6] 2011/04/14 11:30:09: Sending RTP for 3c6960323099-0iponi9rshnz to 172.16.16.231:64890, codec not set yet
[6] 2011/04/14 11:30:09: Codec pcmu/8000 is chosen for call id 3c6960323099-0iponi9rshnz
[6] 2011/04/14 11:30:13: Codec pcmu/8000 is chosen for call id 90b3ae3d@pbx
[6] 2011/04/14 11:30:13: Sending RTP for 90b3ae3d@pbx to 172.16.16.230:61804, codec pcmu/8000
[5] 2011/04/14 11:30:19: Call fd630615@pbx: Last request not finished
[6] 2011/04/14 11:30:19: Codec pcmu/8000 is chosen for call id 90b3ae3d@pbx
[5] 2011/04/14 11:30:19: BYE Response: Terminate 3c6960323099-0iponi9rshnz
[5] 2011/04/14 11:30:19: Tuning to new SSRC
[5] 2011/04/14 11:30:19: BYE Response: Terminate fd630615@pbx
[5] 2011/04/14 11:30:26: BYE Response: Terminate qBdMwBMEmhK.qH4IvBtJPGsHjcIryJuq
Is it correct that the CallerID is missing for the incoming call?
Thanks,
F.
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Hello everybody.
I noticed missing registrations in the "Call log". Following is the scenario:
1) external call coming in;
2) secretary (ext 42) picking up, talks to the remote part;
3) secretary transfers the call to my m9 by means of the BLF-keys (Function keys are configured with BLF for our extensions);
4) I answer the phone, talk to my secretary;
5) she hangs up;
6) call gets transferred;
7) I talk with the remote part;
8) call is terminated.
If I then look at the "Call log", I just see the call between the two extensions, the external caller number does not appear!
Is there any parameter I missed to set/configure?
Thanks and kind regards,
F.
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On the phone, there is a "Number Display Style" option under "Preferences" page. Maybe you should try different options to see if it displays "Anonymous" along with the IP address. You do not have to reboot the phone for this.
I've tried every single option but it didn't remove the IP address showing up.
Again: may this depend on the VoIP provider's settings?
Kind regards,
F.
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Well, here is the problem: There is no username in the SIP URI of the From header. So the phone falls back to display the IP address.
IMHO not beautiful, but what else should it do...
So you are telling me that it should rather look like
From: "Anonymous" <sip:USERNAME;user=phone>;tag=1204182133
If yes, is this depending on the VoIP provider or on my snomOne PBX?
Thanks,
F.
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You need to turn the SIP logging on if you want to see the INVITE request, it is a SIP request.
Hello, back again.
Here it goes:
INVITE sip:46@192.168.1.220:1034;line=b0ac8vw4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-a8348b8044f65b804662e8b425dafa9b;rport
From: "Anonymous" <sip:192.192.192.192;user=phone>;tag=1204182133
To: <sip:41123456789@192.192.192.192;user=phone>
Call-ID: fd24d171@pbx
CSeq: 10787 INVITE
Max-Forwards: 70
Contact: <sip:46@192.168.1.200:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/4.2.0.3959
Alert-Info: <http://127.0.0.1/Bellcore-dr4>
Content-Type: application/sdp
Content-Length: 337
v=0
o=- 642798310 642798310 IN IP4 192.168.1.200
s=-
c=IN IP4 192.168.1.200
t=0 0
m=audio 49768 RTP/AVP 8 0 9 2 3 101
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
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Hello everybody.
When I'm out of office and nobody is there, I'd like to have my calls directly transferred to my cellphone.
With traditional PSTN phones, I would do it with *210791234567, how may I achieve this same behaviour with snomOne PBX?
Many thanks,
F.
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Yes, in the SIP trace, the INVITE request. There is a From-header, it would be interesting to see what is there.
Hello.
I got this one now, from the Call log:
[5] 2011/03/01 14:03:08: Identify trunk (line match) 1
[6] 2011/03/01 14:03:08: Sending RTP for bef1776-31dd02fe-73ed13a0-8112@62.65.128.50 to 195.190.166.249:54602, codec not set yet
[5] 2011/03/01 14:03:08: Global trunk [12345678]@pbx.piramide.local sends call to 72 in domain pbx.piramide.local
[3] 2011/03/01 14:03:08: Hunt group 72 wants to add 4 members
[6] 2011/03/01 14:03:08: Codec pcmu/8000 is chosen for call id bef1776-31dd02fe-73ed13a0-8112@62.65.128.50
[5] 2011/03/01 14:03:22: INVITE Response 487 Request Terminated: Terminate 08914055@pbx
[5] 2011/03/01 14:03:22: INVITE Response 487 Request Terminated: Terminate ee0bf877@pbx
[5] 2011/03/01 14:03:22: INVITE Response 487 Request Terminated: Terminate e91673ea@pbx
[5] 2011/03/01 14:03:22: INVITE Response 487 Request Terminated: Terminate bcc69cc1@pbx
I didn't pick up the incoming call.
F.
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Check out the auto attendant and the service flag. The auto attendant contains the message that should be played; and the service flag is used to define when that message should be played.
Hello there!
I recorder my IVR on my Auto Attendant (ext 70).
I created ext 65 as a Service Flag account as follows:
Mode: manual
State: clean
Confirmation tone: Service active/inactive
Extensions that may change status: 41
Then I put 65 in the Auto Attendant IVR corresponding to my previously recorded message.
If I put on my trunk the destination to ext 70 and I enable from ext 41 the Service Flag, I get to hear my message.
What I now would like to do, is that first of all the phones extensions 40-43 should ring, then after say 20 seconds, the Service Flag should intervene.
How to achieve this?
Thanks,
F.
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Whats in the From-header? Something like "<sip:12.23.45.56>"? Then that's what the Caller-ID looks like...
Where should I look for "from-header"? In the "Call Log" I see "From" = "Anonymous()"
Kind regards,
F.
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Hy there.
I'd like to know how to set up some sort of "out-of-office" message, which I would like to be able to activate/deactivate manually.
What's the best procedure to achieve this? I'm quite new to PBX in general, thus I'd need some help...
Kind regards,
F.
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Hello all.
When an anonymous call (without Caller-ID) comes in, my snom 820 connected to snomOne shows the IP address of the sip provider.
Is there any way to change this behaviour to have "anonymous" shown?
Thanks,
F.
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Hello all!
I would like to know if there's a way to have a view of the call list as they come in (and go out) without having to log into the admin GUI of snomone. This would be of use for our users to be able to see what phone calls are coming in.
Thanks for any suggestions.
F.
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Hy there,
is there any howto/guide to set up syslogd redirection?
I'd like to configure it to log to localhost (OSX Snow Leopard Server) redirecting to a separate /var/log/snomOne.log file.
Eventually, also intergrating some rotation would be nice.
And, most important: which facility is used by snomOne to log?
TIA.
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I have to say that changing the ports is mentioned somewhere in the instructions, but it is not made sufficiently clear or even easy to do since you have to delve into pbx.xml which is inconveniently totally unformatted.
In fact, IMO they should supply it with a default setting that will work and then advise users to change them back to std. if that is what they need. At least with the PBX running this can be easily done via the web interface.
Still, glad I could help.
Well, a thing that I'd like to see implemented is that during the installation process, some script detects which ports are in use and prompts for their change.
Cheers,
F.
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Have you set HTTP and LDAP away from the std. ports? If you haven't done that it WILL crash on startup. Just change them to ports NOT being used by OSX Server.
Great!
UKenGB, you saved my life!
It changed ports:
80-->90
443-->444
389-->390
Now it is *running*.
Thanks again!
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Here is the latest binary file (not the installer) - http://downloads.snom.net/snomONE/mac/pbxc...2011-4.2.0.3959.
Steps to follow (as root):
1. Stop the PBX (/Library/StartupItems/PBX/PBX stop), if it is running.
2. Download and copy above file to /Library/snomone
3. Apply the executable permission to this file (chmod a+x pbxctrl-darwin9.0-2011-4.2.0.3959)
4. Delete the pbxctrl-darwin9.0 link (rm pbxctrl-darwin9.0)
5. Recreate the link to the new binary (ln -s pbxctrl-darwin9.0-2011-4.2.0.3959 pbxctrl-darwin9.0)
6. Restart the PBX (/Library/StartupItems/PBX/PBX start)
BTW, PBX is working on a mac-mini (Darwin Kernel version 9.7.0 i386) even without this change. Also, we know that there are few customers running this on Mac for a while (not sure of their OS version though)
Hello there.
I'm also facing the crash problem on OSX Server 10.6.5 (all updates applied as of today, 31.12.2010).
I've gone through this thread, downloaded both 3959 and 3962 executables, tried both of them but *still* getting crashes!
Any help will be appreciated, but first I'd like to know which user is allowed to start/stop that PBX service: I'm actually using "sudo" for each command you wrote above...
Here some infos:
server:snomone admin$ cd /Library/snomone/
server:snomone admin$ ls -la
total 66912
drwxr-xr-x 10 admin staff 340 Dec 31 10:44 .
drwxrwxr-t+ 65 root admin 2210 Dec 31 09:55 ..
-rw-r--r--@ 1 admin staff 6148 Dec 31 09:26 .DS_Store
drwxr-xr-x 4 root staff 136 Dec 28 17:07 PBX
drwxrwxr-x 371 root admin 12614 Dec 31 10:15 audio_en
drwxrwxr-x 21 root admin 714 Dec 31 10:15 audio_moh
lrwxr-xr-x 1 root staff 33 Dec 31 10:44 pbxctrl-darwin9.0 -> pbxctrl-darwin9.0-2011-4.2.0.3962
-rwxrwxr-x 1 root admin 11405576 Oct 15 18:54 pbxctrl-darwin9.0-2011-4.2.0.3958
-rwxrwxr-x@ 1 root staff 11413996 Dec 31 10:41 pbxctrl-darwin9.0-2011-4.2.0.3959
-rwxrwxr-x@ 1 root staff 11422196 Dec 31 10:41 pbxctrl-darwin9.0-2011-4.2.0.3962
server:snomone admin$ sudo chmod a+x pbxctrl-darwin9.0-2011-4.2.0.3959
server:snomone admin$ rm pbxctrl-darwin9.0
server:snomone admin$ sudo ln -s pbxctrl-darwin9.0-2011-4.2.0.3959 pbxctrl-darwin9.0
server:snomone admin$ sudo /Library/StartupItems/PBX/PBX start
Starting the PBX Service
Changing to directory /Library/snomone
server:snomone admin$ /Library/StartupItems/PBX/PBX start
Starting the PBX Service
Changing to directory /Library/snomone
Attached is the crash report... pbxctrl-darwin9.0-2011-4.2.0.3959_2010-12-31-104720_server.crash.txt
Thanks in advance!!!
F.
PBX audio messages in other languages?
in Call Treatment
Posted
Many many thanks!
F.