cpendl

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

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.

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.

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?

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.

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

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?

i've tried that a few times - including restarting the computer. can anyone recommend a premium support option to get this fixeD?

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.

Where does the name come from?

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

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?

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.

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?

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)

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!

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

The problem is, I have to make every outgoing call twice to get one to connect. How do I fix?

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?

That website didn't work. No server found. It happens both on my M3 and 820.

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. ===== 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

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

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.

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?

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.