cpendl
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Posts posted by cpendl
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Oh the file was probably too big for even downloading it. If you can get file system access, just use tar to grab the file system content (tar cvfz backup.tgz snomONE) and restore it on the target machine. After all, even the SoHo is just a Linux machine. Just make sure that re make a copy of the pbxctrl executable on the target machine, so that you don't overwrite it with the wrong architecture.
Thanks for the good info. I'm a little lost. Can you recommend anyone who can help me do this? I have file system system access on the Mac Mini that's currently hosting this.
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First of all, SoHo or mini? If you are using the SoHo, be careful as the SoHo has no factory reset and if you screw up the network config, you have a dead brick.
How big is your file? You might have to temporarily change the upload file limit (admin settings) so that the PBX takes it.
It's a SoHo.
When I look at the PBX.tar file, it is "zero bytes". If I unarchive it, I get a whopping 415 bytes. Would that size prohibit uploading the config?
I also have a custom AA greeting. How do get that file over to the SoHo?
Got the network config to work, but thanks for the heads up.
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I'm running Snom One on a Mac Mini and want to migrate over to the Soho. When I try to upload the config file through the UI, nothing happens. How can I ssh and what exactly to I need to copy?
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I recently purchased the Snom Soho. I want to transfer the configuration files from my Snom One PBX running on a Mac Mini (version 2011-4.2.1.4025). I've saved the configuration file (pbx.tar), but when I try to update and save it to the Snom Soho (version 2011-4.2.1.4025), nothing happens. What am I missing? I'm using Safari for my browser when managing the PBX through the web interface.
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!([0-9]{3}$)!\1!t!700 not sure if 700 is your AA? you can change that.
This example always uses the last 3 digits for example 19543767319= 7319 of the number, regardless of how long the number is.
This example assumes that the number of digits is always the same.
Here's what I currently have in the send to extension field. It's still sending the 7319 number to 701 (the conference server) and sending 3180 to 700 (the AA), but not sending 2067450870 to 500. Any ideas?
Thanks!
!+19543767319!701! !([0-9]{3}$)!\1!t!700
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I currently have the following expression to send calls directly to the AA and the Conference Server. I have another DID that needs to go an extension. What expression do I need to create to keep the existing routing while sending all calls from the new DID to extension 500?
Current Expression:
!+19543767319!701! 700
New DID
2067450870
New Extension
500
Where can I read more information about this?
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i've tried that a few times - including restarting the computer. can anyone recommend a premium support option to get this fixeD?
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i just upgraded from pbxnsip 3.x to snom one on a mac server running 10.6.5.
I already have a service running on port 80, so i changed the pbx.xml file with to port 81, restarted the service and server..and I still get a "can't load page error" in safari.
I've also tried stopping the service and changing back to port 80 without success.
Does anyone have any ideas as to what's happening?
Thanks.
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If you set the trunk ANI PBX should send that number (provided, the extension or the domains do not have the DID set) for outgoing calls
Where does the name come from?
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How do I set the caller ID on the trunk? When I place outgoing calls now, they show up as unavailable. Is this done on the trunk or the extension?
Using PBXNSIP 3.4
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Currently I have the PBX configured with one domain. I'm using the auto attendant and domain-specific hold music. I want to create another "identity" within the PBX so I can call out and have a different caller ID and separate DID for incoming calls. What is the easiest way to achieve this? Do I create another extension or another domain?
Also, how is the outgoing caller ID passed on? It is through the trunk or extension?
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The phone does not answer the challenge it seems. Try a newer firmware version, the one you are using might be buggy.
Found out what was going on here...the firewall was rejecting the packets because the headers were too large and contained all of the codecs. I removed the ones that weren't being used, and the problem has been solved.
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There are couple of things you need to consider here.
- As you guessed, create 3 or 4 digit extensions/accounts.
Domain_>Settings:"Default PnP Dialplan Scheme:" - choose proper option from the drop-down, if you want the phone to automatically dial out. If you do not want the phone to automatically dial out, then use "user must press enter" option here.
Make all accounts the same length e.g., if you decide to do 3 digit extensions, make the AA also 3 digits.
Before making these changes, I would suggest to make a backup of the configuration (just in case if you need to refer back any data)
Thanks! Should I stay away from any numbers that might interfere with other phone functions? Also, what's the best practices as far as where to start? Right now, the system starts with 40 for extensions. Can I start these there digit extensions at 100?
- As you guessed, create 3 or 4 digit extensions/accounts.
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I had to reload the saved configuration file for the PBX and my VM greetings and name announcements got blown out. Is there a way to fix?
Using 3.4.0.3201 (Darwin)
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I'm currently using the PBX (3.4.0.3201 (Darwin)) in the default configuration using two digit extensions. How do I change this to three or four digit dialing? Is it just a matter of editing the extension number in the accounts list in the domain? Do I need to change anything else?
Are there any numbers I should stay away from?
Thanks!
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What version of the software are you using?
Version: 3.4.0.3201 (Darwin)
Here is the email i receive about the error.
The call between sip:8002752273@10.10.0.9;user=phone and
sip:40@10.10.0.9 has been disconnected because no media session was
establised (source=10.10.1.90:1024)
Here is the log from attempting to make a call where it does not find trunk.
[9] 2010/01/28 10:47:46: Resolve 70672: aaaa udp 10.10.1.90 1024
[9] 2010/01/28 10:47:46: Resolve 70672: a udp 10.10.1.90 1024
[9] 2010/01/28 10:47:46: Resolve 70672: udp 10.10.1.90 1024
[9] 2010/01/28 10:47:46: Resolve 70673: aaaa udp 10.10.1.90 1024
[9] 2010/01/28 10:47:46: Resolve 70673: a udp 10.10.1.90 1024
[9] 2010/01/28 10:47:46: Resolve 70673: udp 10.10.1.90 1024
[9] 2010/01/28 10:47:47: Resolve 70674: aaaa udp 10.10.1.90 1024
[9] 2010/01/28 10:47:47: Resolve 70674: a udp 10.10.1.90 1024
[9] 2010/01/28 10:47:47: Resolve 70674: udp 10.10.1.90 1024
[5] 2010/01/28 10:47:50: SIP Rx udp:10.10.1.90:1024:
INVITE sip:8002752273@10.10.0.9;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.1.90:1024;branch=z9hG4bK-tit99cgtbg9q;rport
From: "Chris Pendl" <sip:40@10.10.0.9>;tag=dhytveg316
To: <sip:8002752273@10.10.0.9;user=phone>
Call-ID: 3c34d70a133c-l437z7ee50bv
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:40@10.10.1.90:1024;line=68fy0mcb>;reg-id=1
X-Serialnumber: 000413400B70
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom820/8.2.11
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 450
v=0
o=root 1646706789 1646706789 IN IP4 10.10.1.90
s=call
c=IN IP4 10.10.1.90
t=0 0
m=audio 49264 RTP/AVP 0 8 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:MrIgPoS2Z9P6mumzlBeH59we70t32m94rtIYR48k
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[8] 2010/01/28 10:47:50: Could not find a trunk (2 trunks)
[9] 2010/01/28 10:47:50: Resolve 70675: aaaa udp 10.10.1.90 1024
[9] 2010/01/28 10:47:50: Resolve 70675: a udp 10.10.1.90 1024
[9] 2010/01/28 10:47:50: Resolve 70675: udp 10.10.1.90 1024
[5] 2010/01/28 10:47:50: SIP Tx udp:10.10.1.90:1024:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.1.90:1024;branch=z9hG4bK-tit99cgtbg9q;rport=1024
From: "Chris Pendl" <sip:40@10.10.0.9>;tag=dhytveg316
To: <sip:8002752273@10.10.0.9;user=phone>;tag=2526568c61
Call-ID: 3c34d70a133c-l437z7ee50bv
CSeq: 1 INVITE
Content-Length: 0
[9] 2010/01/28 10:47:50: Resolve 70676: aaaa udp 10.10.1.90 1024
[9] 2010/01/28 10:47:50: Resolve 70676: a udp 10.10.1.90 1024
[9] 2010/01/28 10:47:50: Resolve 70676: udp 10.10.1.90 1024
[5] 2010/01/28 10:47:50: SIP Tx udp:10.10.1.90:1024:
SIP/2.0 401 Authentication Required
Via: SIP/2.0/UDP 10.10.1.90:1024;branch=z9hG4bK-tit99cgtbg9q;rport=1024
From: "Chris Pendl" <sip:40@10.10.0.9>;tag=dhytveg316
To: <sip:8002752273@10.10.0.9;user=phone>;tag=2526568c61
Call-ID: 3c34d70a133c-l437z7ee50bv
CSeq: 1 INVITE
User-Agent: pbxnsip-PBX/3.4.0.3201
WWW-Authenticate: Digest realm="10.10.0.9",nonce="2acca235cefaa3239a09cefca18b5721",domain="sip:8002752273@10.10.0.9;user=phone",algorithm=MD5
Content-Length: 0
[5] 2010/01/28 10:47:50: SIP Rx udp:10.10.1.90:1024:
ACK sip:8002752273@10.10.0.9;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.1.90:1024;branch=z9hG4bK-tit99cgtbg9q;rport
From: "Chris Pendl" <sip:40@10.10.0.9>;tag=dhytveg316
To: <sip:8002752273@10.10.0.9;user=phone>;tag=2526568c61
Call-ID: 3c34d70a133c-l437z7ee50bv
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:40@10.10.1.90:1024;line=68fy0mcb>;reg-id=1
Content-Length: 0
[9] 2010/01/28 10:47:51: Resolve 70677: aaaa udp 10.10.1.90 1024
[9] 2010/01/28 10:47:51: Resolve 70677: a udp 10.10.1.90 1024
[9] 2010/01/28 10:47:51: Resolve 70677: udp 10.10.1.90 1024
[9] 2010/01/28 10:47:52: Resolve 70678: aaaa udp 10.10.1.91 5060
[9] 2010/01/28 10:47:52: Resolve 70678: a udp 10.10.1.91 5060
[9] 2010/01/28 10:47:52: Resolve 70678: udp 10.10.1.91 5060
[9] 2010/01/28 10:48:01: Resolve 70679: aaaa udp 10.10.0.9 26964
[9] 2010/01/28 10:48:01: Resolve 70679: a udp 10.10.0.9 26964
[9] 2010/01/28 10:48:01: Resolve 70679: udp 10.10.0.9 26964
[9] 2010/01/28 10:48:01: Resolve 70680: aaaa udp 10.10.1.90 1024
[9] 2010/01/28 10:48:01: Resolve 70680: a udp 10.10.1.90 1024
[9] 2010/01/28 10:48:01: Resolve 70680: udp 10.10.1.90 1024
[9] 2010/01/28 10:48:02: Resolve 70681: aaaa udp 10.10.0.9 63136
[9] 2010/01/28 10:48:02: Resolve 70681: a udp 10.10.0.9 63136
[9] 2010/01/28 10:48:02: Resolve 70681: udp 10.10.0.9 63136
[9] 2010/01/28 10:48:06: Resolve 70682: aaaa udp 10.10.1.91 5060
[9] 2010/01/28 10:48:06: Resolve 70682: a udp 10.10.1.91 5060
[9] 2010/01/28 10:48:06: Resolve 70682: udp 10.10.1.91 5060
[9] 2010/01/28 10:48:09: Remote site closed the connection
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For outbound calls, the log message probably comes from the call that initiates the call (from the extension). That is of course okay. This message is only interesting for inbound calls from a trunk to the PBX.
The problem is, I have to make every outgoing call twice to get one to connect. How do I fix?
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When I try to place outgoing calls I get a "SIP Trunk Not Found" error when I check the logs. If I have up and call again (within 2 seconds) the call goes through. Any idea what might keep causing this? Is there a "keep alive" funtion on the trunk that would help with this?
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Could be a simple phone problem. It should try again after getting challenged. Try http://dms.snom.com, username "beta", password "beta".
That website didn't work. No server found. It happens both on my M3 and 820.
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When I try to place an outgoing call, it get an unconnected call error. If I hang up and make that same call again, it works. Incoming calls are not a problem either. Is there a setting to keep the SIP truck connection alive? Am I missing something else?
Here's the long from the unsuccessful call.
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INVITE sip:8002752273@10.10.0.9;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.1.90:1024;branch=z9hG4bK-y62slugvngnt;rport
From: "Chris Pendl" <sip:40@10.10.0.9>;tag=9fzyeie025
To: <sip:8002752273@10.10.0.9;user=phone>
Call-ID: 3c27971aaaaf-ht87o80ori4d
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:40@10.10.1.90:1024;line=68fy0mcb>;reg-id=1
X-Serialnumber: 000413400B70
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom820/8.2.11
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 450
v=0
o=root 1553854432 1553854432 IN IP4 10.10.1.90
s=call
c=IN IP4 10.10.1.90
t=0 0
m=audio 49738 RTP/AVP 0 8 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:3izJ80c3vuOv0TgF1qqsZ0ZZSE9uhHNOHONCvP3j
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[8] 2010/01/18 09:34:43: Could not find a trunk (2 trunks)
[9] 2010/01/18 09:34:43: Resolve 11901: aaaa udp 10.10.1.90 1024
[9] 2010/01/18 09:34:43: Resolve 11901: a udp 10.10.1.90 1024
[9] 2010/01/18 09:34:43: Resolve 11901: udp 10.10.1.90 1024
[9] 2010/01/18 09:34:43: SIP Tx udp:10.10.1.90:1024:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.1.90:1024;branch=z9hG4bK-y62slugvngnt;rport=1024
From: "Chris Pendl" <sip:40@10.10.0.9>;tag=9fzyeie025
To: <sip:8002752273@10.10.0.9;user=phone>;tag=debb4b288a
Call-ID: 3c27971aaaaf-ht87o80ori4d
CSeq: 1 INVITE
Content-Length: 0
[9] 2010/01/18 09:34:43: Resolve 11902: aaaa udp 10.10.1.90 1024
[9] 2010/01/18 09:34:43: Resolve 11902: a udp 10.10.1.90 1024
[9] 2010/01/18 09:34:43: Resolve 11902: udp 10.10.1.90 1024
[9] 2010/01/18 09:34:43: SIP Tx udp:10.10.1.90:1024:
SIP/2.0 401 Authentication Required
Via: SIP/2.0/UDP 10.10.1.90:1024;branch=z9hG4bK-y62slugvngnt;rport=1024
From: "Chris Pendl" <sip:40@10.10.0.9>;tag=9fzyeie025
To: <sip:8002752273@10.10.0.9;user=phone>;tag=debb4b288a
Call-ID: 3c27971aaaaf-ht87o80ori4d
CSeq: 1 INVITE
User-Agent: pbxnsip-PBX/3.4.0.3201
WWW-Authenticate: Digest realm="10.10.0.9",nonce="d2b05148da523dc9c39c0a1cf2b8bae9",domain="sip:8002752273@10.10.0.9;user=phone",algorithm=MD5
Content-Length: 0
[9] 2010/01/18 09:34:44: SIP Rx udp:10.10.1.90:1024:
ACK sip:8002752273@10.10.0.9;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.1.90:1024;branch=z9hG4bK-y62slugvngnt;rport
From: "Chris Pendl" <sip:40@10.10.0.9>;tag=9fzyeie025
To: <sip:8002752273@10.10.0.9;user=phone>;tag=debb4b288a
Call-ID: 3c27971aaaaf-ht87o80ori4d
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:40@10.10.1.90:1024;line=68fy0mcb>;reg-id=1
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I have a DID that I want to use to ring to conference server (extension 71), but when I call the DID, nothing happens. It just rings. I can access extension 71 locally, but not with the DID.
This was working, but I have not used it in a while, and now it's not.
I have !9543767319!71! 70 in the "send call to extension" field in the truck information.
Any ideas?
3.4.0.3201 (Darwin)
Logfile
9] 2009/09/30 18:05:37: Resolve 131475: aaaa udp 10.0.10.90 1024
[9] 2009/09/30 18:05:37: Resolve 131475: a udp 10.0.10.90 1024
[9] 2009/09/30 18:05:37: Resolve 131475: udp 10.0.10.90 1024
[9] 2009/09/30 18:05:40: Resolve 131476: aaaa udp 10.0.10.3 41562
[9] 2009/09/30 18:05:40: Resolve 131476: a udp 10.0.10.3 41562
[9] 2009/09/30 18:05:40: Resolve 131476: udp 10.0.10.3 41562
[9] 2009/09/30 18:05:41: Resolve 131477: aaaa udp 10.0.10.91 5060
[9] 2009/09/30 18:05:41: Resolve 131477: a udp 10.0.10.91 5060
[9] 2009/09/30 18:05:41: Resolve 131477: udp 10.0.10.91 5060
[9] 2009/09/30 18:05:41: Resolve 131478: aaaa udp 10.0.10.90 1024
[9] 2009/09/30 18:05:41: Resolve 131478: a udp 10.0.10.90 1024
[9] 2009/09/30 18:05:41: Resolve 131478: udp 10.0.10.90 1024
[9] 2009/09/30 18:05:45: Remote site closed the connection
[9] 2009/09/30 18:05:52: Last message repeated 2 times
[9] 2009/09/30 18:05:52: Resolve 131479: aaaa udp 10.0.10.3 4440
[9] 2009/09/30 18:05:52: Resolve 131479: a udp 10.0.10.3 4440
[9] 2009/09/30 18:05:52: Resolve 131479: udp 10.0.10.3 4440
[9] 2009/09/30 18:05:55: Resolve 131480: aaaa udp 10.0.10.91 5060
[9] 2009/09/30 18:05:55: Resolve 131480: a udp 10.0.10.91 5060
[9] 2009/09/30 18:05:55: Resolve 131480: udp 10.0.10.91 5060
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I have tested this scenario (no phone registered and simultaneous ringing a mobile) on version 4 and it works.
Nice. Where does the call terminate? At the PBX assuming that the cell phone VM does not grab the call? Right now I have a softphone running on the server with the extensions where users are not actually registered on the PBX. They're only using cellphones, and I want the calls to terminate here.
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I like the fix in 3.3. that allows you to call into pbx with cell phone to place an outbound call while carrying the caller ID of the PBX instead of my cellphone. I noticed to now when calls are forked to my cell phone, it also carriers the caller ID of the PBX.
Is there a way to change this so when forking incoming calls to cellphone, I can actually see who is calling me instead of the PBX?
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The customers Domain is set up the same (as far as i can see) as on my domain
on my domain when i call in i get the option "to place an outbound call press 1" but when the customer calls his DID it doesnt give him the option
what could i have set up wrong?
what do i have to look at?
Take a look at how the number is being formated when calls come in, and it that matches how the cell phone number is entered for the extension. You might have to play with the country code formatting on the truck, or you can try adding (or removing) a "+1" in front the of cell phone number in the extension.
High Quality Audio Codec for Broadcast?
in General Setup
Posted
Hello fellow snomers. I want to explore using VOIP as an alternative to expensive broadcast lines for reporters when they're in the field.
What codecs will get wide band, high quality sound? (I'm trying to avoid the "over the phone" sound for radio broadcast.) I currently use Bria and SIP Station for Trunks. Do I have the best set up?
I'm in Seattle where LTE is widely available.
Thanks!
Chris