sudo

I have followed the steps to manually provision the phone outlined in http://kiwi.pbxnsip.com/index.php/Aastra but no dice. The phone tries to register the info in the Registrar/Proxy Server fields. so the SIP registration request comes through as Request: REGISTER sip:pbx-domain.com:5060, and of course the server sends back a 404 not found. When I add the extension@domain in the Registrar/Proxy Server fields, the register request does not come at all. Can anyone at Someone grab an Aastra phone and see if it works with their system. I would assume they do because there are multiple posts in this section. I was also told that PnP was supported, but apparently that not the case either (per snomone).

Ive been trying to get an Aastra 9480i connected to snomone. Can anyone let me know how they got their Aastra connected to snomone? I have entered in the information in the Global SIP settings but the SIP registration comes through with only the domain and when I try adding extension@domain the SIP registration does not com through at all. I have the following: Basic SIP Authentication Settings: Authentication Name : extension # Password: SIP passwd Line mode: Generic Call Waiting: Enabled Basic SIP Network Settings: Registrar Server: FQDN of pbx (I have tried extension@domain, but SIP registration does not come through) Registrar Port: 5060 All other settings are default. Any info will be much appreciated. sudo

Bingo - Exported the .wav in 8 Khz, 16 bit mono and it uploaded. Thanks Auf Wiedersehen

I checked the snom one technical manual and it makes no reference to uploading audio. I would assume by the snomone webUI that you can upload audio due to the fact you can browse files on the local machine, but im not finding documentation provided by snom or anyone else that outlines the requirements of uploading audio from the webUI. A heads up from snom one letting me know if this is possible and the requirements would be nice. Good customer service...you know

I know that you can dial in to the system to upload a agent group greeting etc, But im having issues uploading audio from the web interface. Under 'Caller Setup' I browse to the .wav file for the initial greeting and save the page. But it is not uploading the audio. Could it be the bit rate of the .wav file? It does not look like the system is uploading anything. It just immediately refreshes the screen with no audio file in the greeting, wave file 1, wave file 2 etc. This should be pretty straight forward and I cannot imagine what the issue is here.

Can I just get a yes or no answer? Does the cdri log what option is selected in the Auto Attendant?

That link is not working. I get the following error: [#404] Sorry, we could not locate the page you are requesting to view. Please click here to return back to the forum's home

I have a request from a customer. They want to see how effective their Auto Attendant is. What options are being selected most often. They want to 'trim the fat' so to speak and remove any unnecessary options. When I inspect the generated .xml file in the /cdri directory, Im not seeing anything that states what option was selected. Below is the output I get when opening the .xml in excel: TLVBcid 78840151234pvopgn0@60.100.50.200d5dr20001dt18391e41fI"CID_OF_CALLER" <sip:+18885551234@pbx-zenger.voonami.com:5060;user=phone>s13335551234.868t/"F_NAME L_NAME" <sip:200@CUST_PBX_DOMAIN.com>u41ymailbox (Please let me know if I butchered this too much in editing out sensitive information) I believe there are limitations opening .xml files in excel on a Mac so there might be something lost in translation. Can anyone explain how to decipher the above cdri? http://wiki.snomone.com/index.php?title=Call_Detail_Records_%28CDRs%29 does not help in what im looking for. Many thanks in advance! ~Sudo

So whats the fix here? Is there anyway to lower the media timeout? At least the call wont be hung up for 3600 seconds.

Here is the sip trace: 2012/11/26 12:00:36 Tx: tcp:192.168.1.1:13886 (1017 bytes) INVITE sip:200@10.1.1.1;transport=tcp SIP/2.0 Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-2b12b9dccafb724fe8e2c614ebfa276a;rport From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 To: <sip:900@cust_domain.com> Call-ID: 3236d169@pbx CSeq: 24571 INVITE Max-Forwards: 70 Contact: <sip:200@cust_ip_add:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/2011-4.5.0.1050 Coma Berenicids Content-Type: application/sdp Content-Length: 382 v=0 o=- 325608696 325608696 IN IP4 cust_ip_add s=- c=IN IP4 cust_ip_add t=0 0 m=audio 56626 RTP/AVP 0 8 9 18 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no 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 2012/11/26 12:00:36 Rx: tcp:192.168.1.1:13886 (446 bytes) SIP/2.0 100 Trying Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-2b12b9dccafb724fe8e2c614ebfa276a;rport From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 CSeq: 24571 INVITE Call-ID: 3236d169@pbx Contact: <sip:200@10.1.1.1;transport=tcp> User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439 Accept-Language: en Content-Length: 0 2012/11/26 12:00:37 Rx: tcp:192.168.1.1:13886 (512 bytes) SIP/2.0 180 Ringing Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-2b12b9dccafb724fe8e2c614ebfa276a;rport From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 CSeq: 24571 INVITE Call-ID: 3236d169@pbx Contact: <sip:200@10.1.1.1;transport=tcp> User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439 Allow-Events: talk,hold,conference Accept-Language: en Require: 100rel RSeq: 8193 Content-Length: 0 2012/11/26 12:00:37 Tx: tcp:192.168.1.1:13886 (446 bytes) PRACK sip:200@10.1.1.1;transport=tcp SIP/2.0 Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-0bb1d0d9c3f074fe2fbe2e5ccd5a1fbd;rport From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 Call-ID: 3236d169@pbx CSeq: 24572 PRACK Max-Forwards: 70 Contact: <sip:200@cust_ip_add:5060;transport=tcp> RAck: 8193 24571 INVITE Content-Length: 0 2012/11/26 12:00:37 Rx: tcp:192.168.1.1:13886 (441 bytes) SIP/2.0 200 OK Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-0bb1d0d9c3f074fe2fbe2e5ccd5a1fbd;rport From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 CSeq: 24572 PRACK Call-ID: 3236d169@pbx Contact: <sip:200@10.1.1.1;transport=tcp> User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439 Accept-Language: en Content-Length: 0 2012/11/26 12:00:43 Rx: tcp:192.168.1.1:13886 (766 bytes) SIP/2.0 200 OK Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-2b12b9dccafb724fe8e2c614ebfa276a;rport From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 CSeq: 24571 INVITE Call-ID: 3236d169@pbx Contact: <sip:200@10.1.1.1;transport=tcp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439 Accept-Language: en Content-Type: application/sdp Content-Length: 193 v=0 o=- 1353881399 1353881399 IN IP4 10.1.1.1 s=Polycom IP Phone c=IN IP4 10.1.1.1 t=0 0 m=audio 2236 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 2012/11/26 12:00:43 Tx: tcp:192.168.1.1:13886 (417 bytes) ACK sip:200@10.1.1.1;transport=tcp SIP/2.0 Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-a35f384c9690c01159e767ce6f4dcee1;rport From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 Call-ID: 3236d169@pbx CSeq: 24571 ACK Max-Forwards: 70 Contact: <sip:200@cust_ip_add:5060;transport=tcp> Content-Length: 0 2012/11/26 12:01:35 Rx: tcp:192.168.1.1:13886 (864 bytes) INVITE sip:200@cust_ip_add:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bK58b41ef7F5AAB32 From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 CSeq: 1 INVITE Call-ID: 3236d169@pbx Contact: <sip:200@10.1.1.1;transport=tcp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 205 v=0 o=- 1353881399 1353881400 IN IP4 10.1.1.1 s=Polycom IP Phone c=IN IP4 10.1.1.1 t=0 0 a=sendonly m=audio 2236 RTP/AVP 0 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 2012/11/26 12:01:35 Tx: tcp:192.168.1.1:13886 (843 bytes) SIP/2.0 200 Ok Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bK58b41ef7F5AAB32;rport=13886;received=192.168.1.1 From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 Call-ID: 3236d169@pbx CSeq: 1 INVITE Contact: <sip:200@cust_ip_add:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/2011-4.5.0.1050 Coma Berenicids Content-Type: application/sdp Content-Length: 237 v=0 o=- 325608696 325608696 IN IP4 cust_ip_add s=- c=IN IP4 cust_ip_add t=0 0 m=audio 56626 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=recvonly 2012/11/26 12:01:35 Rx: tcp:192.168.1.1:13886 (557 bytes) ACK sip:200@cust_ip_add:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bKd2d05b4862B5D81B From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 CSeq: 1 ACK Call-ID: 3236d169@pbx Contact: <sip:200@10.1.1.1;transport=tcp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439 Accept-Language: en Max-Forwards: 70 Content-Length: 0 2012/11/26 12:01:38 Rx: tcp:192.168.1.1:13886 (568 bytes) REFER sip:200@cust_ip_add:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bK7bb88c1e4E0E8A51 From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 CSeq: 2 REFER Call-ID: 3236d169@pbx Contact: <sip:200@10.1.1.1;transport=tcp> User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439 Accept-Language: en Refer-To: sip:210@cust_domain.com:5060;user=phone Referred-By: <sip:200@cust_domain.com> Max-Forwards: 70 Content-Length: 0 2012/11/26 12:01:38 Tx: tcp:192.168.1.1:13886 (431 bytes) SIP/2.0 202 Accepted Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bK7bb88c1e4E0E8A51;rport=13886;received=192.168.1.1 From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 Call-ID: 3236d169@pbx CSeq: 2 REFER Contact: <sip:200@cust_ip_add:5060;transport=tcp> User-Agent: snomONE/2011-4.5.0.1050 Coma Berenicids Content-Length: 0 Any Thoughts? I think I remember reading somewhere that Polycoms have issues with the REFER method. Dont know if its the same for PRACK? Thanks again - Sudo

Version: 2011-4.5.0.1050 Coma Berenicids (CentOS64)

Im getting a similar report: The call between sip:5555551234@cust_domain:5060;user=phone and sip:900@cust_domain has been disconnected because of media timeout (3600 seconds), 1036/2178 packets have been received/sent I get a few of these everyday. "900@cust_domain" is the Auto Attendant. I should mention that this customer is using Polycoms (IP331's mostly and PnP supported with a pbxnsip license). Any suggestions on getting these alerts cleaned up? Is this a setting on the Auto Attendant? Maybe its not acknowledging the BYE signal? Is this a compatibility issues with the polycoms? (I have had RTP timeouts on all snom device domains as well, though not nearly as frequent). Should I lower the media timeout to something smaller, like 60 seconds? Thanks in advance. Sudo

How long is the registration expiration period? Where can I modify it?

Maybe we are not understanding each other. In the usr_core_redirection.htm I can modify the webpage, but this is not what I need. I need to increase the amount of time it rings the cell phone. Im no wiz when it come to coding, but I can only modify the settings on the "Redirection" page. And none of these settings let me set the amount of time the cell phone rings. The only time based feature I can modify are "Call forward no answer timeout" and "Include the cell phone in calls to extension". Both of these do not control the amount of time the cellphone rings. I need to increase the amount of time the cell rings or remove the redirection message.

Can you elaborate? This is on a linux box and at the risk of sounding like an idiot, Im not finding a html dir or pnp.xml file on the system.

Back to the php.xml template. I have modified the one line as follows: <http> <ua vendor="Polycom">Polycom-FileManager/.*</ua> <ua vendor="Polycom">FileTransport PolycomSoundStationIP-.*</ua> <ua vendor="snom">snom-m3-SIP/.*</ua> </http> Now I need to get it to work for both models (SoundStation IP6000 and SoundPoint IP331). I have tried adding an additional line as you suggested: <http> <ua vendor="Polycom">Polycom-FileManager/.*</ua> <ua vendor="Polycom">FileTransport PolycomSoundStationIP-.*</ua> <ua vendor="Polycom">FileTransport PolycomSoundPointIP-.*</ua> <ua vendor="snom">snom-m3-SIP/.*</ua> </http> BUT - The system did not let me save the change. Its weird actually. It created a blank 'template' with no name and no content - Keeping the original php.xml without the new line. How do we make the change stick, and is there any documentation or dll.xml so we can know what is allowed by the system? Thanks again Sudo

Could you elaborate a little more? Many Thanks - Sudo

I tried this too. It does work in the respect that the redirection message does not play, but he wants it to hit his cell phones voicemail, not snomone. With other pbx's, I have been able to set the amount of time it rings the cell phone. I could set it for 60 seconds and the cellphone voicemail would pick up the call before the full minute had passed. The new system does not allow me to set the amount of time to ring the cell phone. So 1 of the following 2 options would be preferred: 1. Remove the redirect message and leave the cell number in the "Call forward on no answer to:" field. -or- 2. Increase the amount of time the cell phone rings so the cellphone voicemail answers the call before its kicked back to the office extension. Thanks again - Sudo

Awesome this worked - Except for one thing (Story of my life) The phone loaded the bootrom and registers with the pbx, but it boots up with the following error: "http://IP.ADD.OF.PBX:80/prov/poly..." (The small display screen does not show the full url nor give me the option of scrolling over) Looking at a pcap I see the phone requesting a 'prov/polycom_phone.cfg' and a 'prov/polycom_sip.cfg', but both of these requests are met with a 'Continuation or non HTTP Traffic" which tells me that the server is responding to the request with the needed file(s). The phone is registered and I can call out/in and internal extensions, but there is a "Caution" Icon in the top right of the display (a triangle with an ! inside.) I greatly appreciate your previous advice! Any suggestions with this error I'm getting? Sudo

The HTTP request is as follows: GET /3111-15600-001.bootrom.ld HTTP/1.1\\r\n HOST: pbx.domain.com\r\n Accept: */*\r\n User-Agent: FileTransport PolycomSoundStationIP-SSIP_6000-UA/4.2.1.0275\r\n \r\n [Full request URI: http://pbx.domain.com/3111-15600-001.bootrom.ld] Had a thought that it may be because I have an earlier firmware version that the phone did not like. So I downloaded 3111-15600-001.bootrom.ld ver 4.0.1 and replaced it in the /tftp directory, but no dice. Still sending the same requests...not getting anywhere. Thanks

I have an extensions set to forward to the cell phone after 10 seconds. When it does this, it plays the message, "Please hold the line while the system re-directs the call". Ive looked around and cannot find where I can turn this message off. Anyone have any thoughts on disabling this redirection message? Sudo

Im running into this same issue Just to confirm - There is no PnP support for the Polycom sound point IP 6000? Even with the pbxnsip license? I have programmed the 6000 with the SIP credentials and its still looking for the BootROM for the 6000. In my pcap the phone requests "3111-15600-001.bootrom.ld" (which IS located in /usr/local/snomONE/tftp) Server kicks back a '302 File moved temporarly" Then the phone asks for 'login.htm' over and over again, getting the same 302 error form the server (During which the phones display is stuck on "Updating initial configuration"). If there is no PnP support, does anyone know where to find instructions for configuring this phone manually to connect? I thought the SIP server, un, and passwd were enough. Evidentially not. Thanks Sudo PS - are there any other Polycom conference phones that are supported?

I have a snomONE box setup with a pbxnsip license to support 3rd party devices. In this case Polycom IP 331. For this particular domain, I needed the phones to have 2 lines, so under the account settings > registration > lines, I put '2'. The phones booted up with the additional line. Great. The problem is as follows: This customer is in UT and uses the 801 area code often. When they open a line and begin to dial "801" the phone stops it at "80" and tries to connect, getting a busy signal. I tested this on an other domain on the same pbx, and I was able to dial the full number (801.555.1234) before it connected. Initially, I thought this was a dial plan issue, but it is not. The dial plan looks good. Comparing the two domains, the only significant difference was the number of lines. The working one had the default value (a blank 'lines' field). I added 2 lines to the working extension, rebooted, and I got the same issue - the phone connects at '80'. Removing the line does remove the additional line from the phone, but does not fix the issue. This does not happen with any other 2 number string, only '80' (even strings starting with 8 like 81, 82, 83, etc, do not cause it, only specifically the string '80' I know that '8' is the suffix for an extensions voicemail so I wondered if the system is attempting to hit the voicemail of a nonexistent extension '0'. I changed the "Mailbox Direct Dial Prefix" from 8 to 9. Saved and rebooted the phone. Same issue. Connects after dialing '80' Suggestions are appreciated. Sudo

Solved. The issue was their firewall. It was set to - "SIP Deep Packet inspection". Similar to SIP ALG. Turned it off and the phones can connect to vm/AA. Silly mistake - thought Id post the solution to any for anyone else who might experience this issue. Thanks to those who posted!

Looking at the pcap: I get a "405: method not allowed" response from the client side. I would guess that it could be a firewall blocking this traffic, but i think that this all runs off port 5060 and 80. I had them try dialing the AA and it rang as busy. Again I can dial the AA from my VoIP lab connected to the same domain and get through. What could be blocking this? Any suggestions or info is greatly appreciated. Sudo