Vodia PBX
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Posts posted by Vodia PBX
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This home office user wants to appear as responsive and available as possible for his clients without having a receptionist.
When he is not on a call he wants his extension to ring for incoming calls. No Auto Attendant.
When he is in an active call and a second call comes in he will see the caller ID and he wants to be able to choose on the fly whether he answers the call immediately, allows the call to go to voicemail, or sends the call to the Agent Group and gives the client the opportunity to wait for him to finish his current call. The client that gets sent to the Agent Group would realize that he knows that they are on the phone waiting on hold. Almost like a receptionist took the call and placed them on hold for him.
Make sense?
If I sent all calls to the Agent Group then it would be a different experience for the caller and he wouldn't have the ability to choose whether or not the call should just go to voicemail -or- get picked up immediately. Particularly if it is a call that he knows is unimportant or if it is a call he knows is urgent.
Okay, so he wants a kind of glorified park orbit. Yes that makes sense.
The first step would be trying that without OCS - just a deflection into the agent group. If that works (can't try it right now) then we have an OCS issue.
Could be a good feature also without OCS. Maybe worth a button on a phone.
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We have a problem on getting the correct IVR
When a caller A make a call to caller B and caller B is engaged or no answer the call goes to Exchange IVR but ask for the A's pin number for the mailbox and not the message to leave a message for B.
Is there any setting on the pbxnsip or Exchange to be done. So that the prompt would be to leave a message for B
Very strange. Does it work properly if someone from outside (PSTN) calls and gets into the mailbox?
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I find that the Presence / Availability indicators are not updated regularly (or at all) on some of my extensions.
For presence, the extension itself is responsible for updating the state.
What does "Poll time" do?What does "Refresh interval" do?
I believe they have no effect when using the agent mode.
Furthermore the page also states "Presence is also not supported yet by the PBX")Well, the PBX is a "stupid" presence agent. We were thinking about making it a smart agent, which calculates the presence state based on registration and calls.
However, the whole presence story so far must be seen as a kind of sales hype; practically nobody used it in the PBX context (AFAIK). In the PBX environment, people look more for BLF and SLA. Maybe Microsoft OCS will change that.
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This solved the problem. Thank You!
Great! The next 3.3.1 version will have it in the GA build.
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I am trying to implement an Agent Group creating a personal call queue that an active extension can use to selectively put inbound callers into using the 'Redirect' capability of OCS Communicator. When the user is done with their current call the next holding inbound call that they redirected to the Agent Group should ring back to them.
Why don't you send the call to the agent group in the first place? This agent group would have only one agent (small group), which may use a static registration that points to the OCS.
I believe the redirect will get very tricky. Maybe we need to set something up in our environment as well, to see how we can deal with this.
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i am having a weird issue in an enviornment that has 20 Aatra 480i's and three Polycoms, the polycoms work fine but the users using the Aastras have an issue where they will hear beeping like someone is pressing buttons on their phone even though the person on the other end of the call cannot hear it.
i tried upgrading firmware to latest version but it didnt help!
any help would be greatly appreciated.
Is the hearing limited to the Aastra handsets? That would be very strange. If you can get a wireshark trace it should be possible to see what is going on on the media level. Filter for the IP address during the capture, then we don't have to screen MB of data.
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"http://pbx.pbxnsip.com:2345/provisioning/snom820.htm
is it necessary to use a Port?
same error,
Snom 370
[5]24/12/2001 00:00:34: Fetching URL: http://xxx.xxx.xxx.xxx/provisioning/settin...00413266111.xml
[1]24/12/2001 00:00:34: Conf setup: code: 404, host: xxx.xxx.xxx.xxx:80, file: /provisioning/settings-000413266111.xml
tail -f /opt/pbxnsip/pbxnsip.log
Web Server: File snom370/snom370-firmware.htm not found
If you are running the HTTP port on port 80, then you don't have to put the port there. If you are using a snom 370, then the URL should look like this: http://xxx.xxx.xxx.xxx/provisioning/snom370.htm.
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Ok .. disabled PNP trust MAC and now the internal phone will not provision over HTTP using a proper user/pass that tests good. Seems something is broken here?
* Replaced actual private IP with LANIP *
0326163952|copy |3|00|'http://111%40rcp.local:****@LANIP/0004f20457d6.cfg' from LANIP
0326163959|copy |4|00|Download of '0004f20457d6.cfg' FAILED on attempt 1 (addr 1 of 1)
0326163959|copy |3|00|transport res: 22 respCode 401
0326163959|copy |3|00|transport error: Curl Error strings have been compiled out.
0326163959|copy |3|00|transport error buffer: The requested URL returned error: 401.
401 means that the authentication did not work. Are you sure that you have chosen the right password (the web password)?
It does not matter if you disable the MAC trust or not. If you are not in the LAN than that flag has no effect.
I tried it again from here. It worked if the phone uses a standard port (80). Using a non-standard port did not work.
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If i dial the auto attendent and dial the ext for the callback the dtmf works, but if i point a number via hunt group to the callback extension it does not work here are the logs. The inbound access number i am dialing is 973 528 7903 logs are attached
Oh you want a callback when a hunt group becomes available? Call back only works if you dial a specific extension. (It is hard to imagine how you would find out if a hunt group becomes available again.)
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yes, i need the download link for this build!
What OS?
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I tested this further and it appears to provision properly on the INSIDE, so I would guess this is related to the sip IP rewrite rule change in this version?
No, there is also a change regarding the identification of the device that is being provisioned.
If the ARP cache contains the IP address of the device and it also matches a known MAC address, then the PBX trusts that MAC address (can be disabled).
Otherwise, it will always use HTTP authentication.
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We removed the pnp.xml file and restarted the service, still no luck, the phone cannot get the MAC config file from the service? This is from a real-world IP to our DMZ real-world IP which should provision using the SIP rewrite, correct?
When you provision outside of the LAN, then the MAC plays no role any more; you better remove it. The PBX identifies the account only by the authentication info that you provide.
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If you are using callback, and when call comes back to the cellphone or landline, after that it asked to dial the destination number, can it be setup to call an extension number on the pbx.
Also I am setting up so it does not use the auto attendent, but when it goes directly to the callback or i use the hunt group i do not get dtmf tones, but when i use the auto attendent is works great
Currently the callback is really just for the callback. You cannot do something else than dialling the person that became available.
Did you try to press "*"? I am not sure, but it could be that this is a escape code that brings you back to the auto attendant.
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i have 3.2.0.3144
Any chance to try 3.3.1 out? What OS?
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Attached is a copy of the sip trace.
Hmm. The PBX sends the T.38-reinvite to the gateway, but there is no response in the log. Does the gateway reply at all? If there is no reply, it is understandable that the FAX does not work.
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it doesen't work, now i get some error in the Log from my Snom 370 (Firmware 7.3.14)
[5]24/12/2001 00:00:55: Fetching URL: tftp://xxx.xxx.xxx.xxx/snom370.htm
[5]24/12/2001 00:00:55: Conf setup: return code 500; requeueing >tftp://xxx.xxx.xxx.xxx/snom370.htm<; attempt: 8, state: 32
i need to provisioning by "HTTP" not "tftp"!
I dont see errors in the PBXnSIP Log!
The Snom Phones and the PBX are not in the same network!
If you are not in the same LAN; you must set the username and password now.
I used a string like "http://pbx.pbxnsip.com:2345/provisioning/snom820.htm", and that worked without problems. Using 3.3.1.3171, though. If you need a build to try this out, let us know.
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We are on 3.1.2.3120
Better don't use that version. Move to 3.2.
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My Snom 370 is configured to Update from http://<IP-Adress of pbxnsip>/prvisioning/settings-000413266111.xml
Is the "prvisioning" a typo (should be "provisioning")?
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Would it be possible to get the polycom_phone.xml template that is built into pbxnsip 3.3.0.3165? I noticed that it's based on Polycom SIP 3.1.1 and I wanted to create a version based on the phone1.cfg that comes with Polycom SIP 3.1.2RevB.
Didn't we have this topic yesterday already?!
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When i try to update my firmware from 6.x to 7.x or 7.x to 7.x the phone gets itself in a loop and updates the firmware then boots up fine then restarts itself and downgrades itself to the last firmware on the phone.
Is there a setting im missing when i start to do firmware updates? the one setting i do change is for it to ask wether or not it should update but that setting dissapears after the firmware update and just gets in a update loop then
What version of the PBX?
I had the same problem myself today
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I DO NOT know!!! but Time Warner did a call trap for me and sent me this info:
We made a test version that disables the detection of the caller-ID when the line is not in the ringing state. Can you give that version a try? After upgrading and testing, you can downgrade back to the version that you are running right now if the situation should not improve.
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Thanks a lot for posting this! It's very helpful. If you'd like, I can parameterize Polycom's one for SIP 3.1.2RevB and post it here. It would be great to have that be part of the new builds.
Sure!
I do see that several parameters were renamed such as {sip-port layer} to {sip-port polycom_layer}. That explains why my polycom_sip.xml wasn't working once I upgraded to 3.3 from 3.2.We consolidated thhe parameters, so that they appear only once. Especially for Polycom it was strange that you had to specify the transport layer twice.
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PBXNSIP .. anything .. it would be great to have this issue resolved ASAP?
Just tried it here... It worked... I even tried over the Internet. It also worked...
Both provisioning over the Internet and provisioning in the LAN. Using version 3.1.2 on the phone (bootloader 4.1.2.0037). All phones firmware files are in the tftp directory.
Do you have a pnp.xml file in your html directory? If that should be so, consider moving it away and restart the service.
I attached the log file from the phone for reference.
0324204419|log |*|02|Initial log entry. Current logging level 4
0324204419|so |*|02|Initial log entry. Current logging level 3
0324204419|so |*|02|---------- Initial log entry ----------
0324204419|so |*|02|Platform: Model=SoundPoint IP 550, Assembly=2345-12500-001 Rev=A
0324204419|so |*|02|Platform: MAC=0004f214058e, IP=192.168.178.21, Subnet Mask=255.255.255.0
0324204419|so |*|02|Platform: BootBlock=2.7.0 (12500_001) 18-Jan-07 13:27
0324204419|so |*|02|Platform: Bootrom=4.1.2.0037 11-Aug-08 16:16
0324204419|so |*|02|Application, main: Label=SIP, Version=3.1.2.0392 07-Jan-09 13:08
0324204419|so |*|02|Application, main: P/N=3150-11530-312
0324204419|so |3|02|Platform: Flash detected is 9 which is different than expected.
0324204419|wdog |*|02|Initial log entry. Current logging level 4
0324204419|ethf |*|02|Initial log entry. Current logging level 4
0324204419|so |5|02|utilCertificateInit failed.
0324204419|hw |*|02|Initial log entry. Current logging level 4
0324204419|ares |*|02|Initial log entry. Current logging level 4
0324204419|dns |*|02|Initial log entry. Current logging level 3
0324204419|cfg |*|02|Initial log entry. Current logging level 3
0324204419|cfg |3|02|RT|Checking DHCP option 160 type string
0324204419|cfg |3|02|RT|Runtime basic IP parameters updated.
0324204419|cfg |3|02|RT|Runtime SNTP parameters updated.
0324204419|dns |*|02|DNS resolver servers are '192.168.178.1' '0.0.0.0'
0324204419|dns |*|02|DNS resolver search domain is ''
0324204419|so |4|02|[soFontsC]: Font item (2)(2) is NULL.
0324204419|curl |*|02|Initial log entry. Current logging level 3
0324204419|utilm|*|02|Initial log entry. Current logging level 4
0324204419|copy |*|02|Initial log entry. Current logging level 3
0324204419|rtos |*|02|Initial log entry. Current logging level 4
0324204419|sec |*|02|Initial log entry. Current logging level 4
0324204419|cfg |3|02|Prm|Beginning to provision phone
0324204419|copy |3|02|'http://44%40localhost:****@192.168.178.20/2345-12500-001.bootrom.ld' from '192.168.178.20'
0324204419|cfg |3|02|Prm|Image 2345-12500-001.bootrom.ld has not changed
0324204419|copy |3|02|buffered_write: transfer Terminated on entry. Return 0
0324204419|copy |3|02|Download of '2345-12500-001.bootrom.ld' succeeded on attempt 1 (addr 1 of 1)
0324204419|cfg |3|02|Prm|Downloaded bootROM is identical to current version 4.1.2
0324204419|copy |3|02|'http://44%40localhost:****@192.168.178.20/0004f214058e.cfg' from '192.168.178.20'
0324204419|copy |3|02|Download of '0004f214058e.cfg' succeeded on attempt 1 (addr 1 of 1)
0324204419|copy |3|02|'http://44%40localhost:****@192.168.178.20/2345-12500-001.sip.ld' from '192.168.178.20'
0324204419|copy |4|02|Download of '2345-12500-001.sip.ld' FAILED on attempt 1 (addr 1 of 1)
0324204419|copy |4|02|Server '192.168.178.20' said '2345-12500-001.sip.ld' is not present
0324204419|cfg |4|02|Prm|Could not get all 512 bytes of the header
0324204419|copy |3|02|'http://44%40localhost:****@192.168.178.20/sip.ld' from '192.168.178.20'
0324204419|cfg |3|02|Prm|Image sip.ld has not changed
0324204419|copy |3|02|buffered_write: transfer Terminated on entry. Return 0
0324204419|copy |3|02|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1)
0324204419|copy |3|02|'http://44%40localhost:****@192.168.178.20:80/provisioning/polycom_phone.cfg' from '192.168.178.20'
0324204419|copy |3|02|Download of 'provisioning/polycom_phone.cfg' succeeded on attempt 1 (addr 1 of 1)
0324204419|cfg |3|02|Prm|Updated http://192.168.178.20:80/provisioning/polycom_phone.cfg
0324204419|copy |3|02|'http://44%40localhost:****@192.168.178.20:80/provisioning/polycom_sip.cfg' from '192.168.178.20'
0324204419|copy |3|02|Download of 'provisioning/polycom_sip.cfg' succeeded on attempt 1 (addr 1 of 1)
0324204419|cfg |3|02|Prm|Updated http://192.168.178.20:80/provisioning/polycom_sip.cfg
0324204419|cfg |3|02|Prm|Check of configuration files suceeded
0324204419|cfg |3|02|Prm|Phone successfully provisioned
0324204419|cfg |*|02|Prm|Configuration file "polycom_phone.cfg" is from template Unknown, revision Unknown
0324204419|cfg |*|02|Prm|Configuration file "polycom_phone.cfg" SHA1 digest: 87F30DB8EBF7457481EE3F197F6EB5BD0F5F1CCA
0324204419|cfg |*|02|Prm|Configuration file "polycom_sip.cfg" is from template Unknown, revision Unknown
0324204419|cfg |*|02|Prm|Configuration file "polycom_sip.cfg" SHA1 digest: CE500E55FED94E2E0EDC4E34221C147AB86290A0
0324204419|so |3|02|Success provisioning.
0324204419|so |*|02|Configuration files: http://192.168.178.20:80/provisioning/polycom_phone.cfg, http://192.168.178.20:80/provisioning/polycom_sip.cfg
0324204419|copy |3|02|'http://44%40localhost:****@192.168.178.20/0004f214058e-phone.cfg' from '192.168.178.20'
0324204419|copy |4|02|Download of '0004f214058e-phone.cfg' FAILED on attempt 1 (addr 1 of 1)
0324204419|copy |4|02|Server '192.168.178.20' said '0004f214058e-phone.cfg' is not present
0324204419|utilm|4|02|uBLFCompressed: File /ffs0/local/0004f214058e-phone_cfg.zzz does not exist or is empty
0324204419|sec |4|02|utilCryptoConfigFileEncrypted: Could not read file /ffs0/local/0004f214058e-phone_cfg.zzz.
0324204419|cfg |3|02|Edit|Local cfg not found: /ffs0/local/0004f214058e-phone_cfg.zzz
0324204419|cfg |3|02|Edit|Loaded local file: /ffs0/local/0004f214058e-phone_cfg.zzz
0324204419|cfg |4|02|Edit|Error 0x380003 attempting stat of /ffs0/local/0004f214058e-phone_cfg.zzz
0324204419|utilm|4|02|uBLFCompressed: File /ffs0/local/0004f214058e-phone_cfg.zzz does not exist or is empty
0324204419|cfg |5|02|Prm|Error when decompressing config file /ffs0/local/0004f214058e-phone_cfg.zzz
0324204419|cfg |3|02|RT|Checking DHCP option 160 type string
0324204419|cfg |3|02|RT|Runtime SNTP parameters updated.
0324204419|so |3|02|Using time offset 3600 seconds.
0324204419|so |3|02|SNTP: sync with server '192.168.178.1' 1 of 1 OK on try 1 of 2 timeout 500 of total 0 ticks (0 sec)
0324204419|so |3|02|Setting daylight savings string to "TIMEZONE=UTC::-60:::".
0324214508|log |*|02|Log render level set to 1
0324214508|lic |*|02|Initial log entry. Current logging level 4
0324214508|copy |3|02|'http://44%40localhost:****@192.168.178.20/000000000000-license.cfg' from '192.168.178.20'
0324214508|copy |4|02|Download of '000000000000-license.cfg' FAILED on attempt 1 (addr 1 of 1)
0324214508|copy |3|02|transport res: 28 respCode 0
0324214508|copy |3|02|transport error: a timeout was reached.
0324214508|copy |3|02|transport error buffer: connect() timed out!.
0324214508|copy |3|02|Making further download attempts for '000000000000-license.cfg'
0324214508|copy |3|02|'http://44%40localhost:****@192.168.178.20/000000000000-license.cfg' from '192.168.178.20'
0324214508|copy |4|02|Download of '000000000000-license.cfg' FAILED on attempt 2 (addr 1 of 1)
0324214508|copy |4|02|Server '192.168.178.20' said '000000000000-license.cfg' is not present
0324214508|copy |3|02|'http://44%40localhost:****@192.168.178.20/0004f214058e-license.cfg' from '192.168.178.20'
0324214508|copy |4|02|Download of '0004f214058e-license.cfg' FAILED on attempt 1 (addr 1 of 1)
0324214508|copy |4|02|Server '192.168.178.20' said '0004f214058e-license.cfg' is not present
0324214508|lic |4|02|cfgCfsFlashKeyScratchLoad: key is blank
0324214508|srtp |*|02|Initial log entry. Current logging level 4
0324214508|rdisk|*|02|Initial log entry. Current logging level 4
0324214508|rdisk|*|02|RAM disk created, size: 4,194,304 bytes
0324214508|res |*|02|Initial log entry. Current logging level 4
0324214508|httpa|*|02|Initial log entry. Current logging level 4
0324214508|cdp |*|02|Initial log entry. Current logging level 4
0324214508|cdp |*|02|CDP is DISABLED. CDP not detected at boot.
0324214508|sys |*|02|Initial log entry. Current logging level 4
0324214508|copy |3|02|'http://44%40localhost:****@192.168.178.20/0004f214058e-directory.xml' from '192.168.178.20'
0324214508|copy |3|02|Download of '0004f214058e-directory.xml' succeeded on attempt 1 (addr 1 of 1)
0324214508|cfg |4|02|Edit|Local file compressed: /ffs0/local/local-directory_xml.zzz
0324214508|ssps |*|02|Initial log entry. Current logging level 4
0324214508|dbuf |*|02|Initial log entry. Current logging level 4
0324214508|so |*|02|System Info Reports:
0324214508|so |*|02|.CPU is TNETV1055/C55x, rev 2 running at 125MHz with memory at 125MHz.
0324214508|so |*|02|.Board is identified as PolycomSoundPointIP-SPIP_550.
0324214508|so |*|02|.DRAM_LO: 0x94000000. DRAM_SIZE: 32 MB
0324214508|so |*|02|.Clocks are VBUSP: 125MHz, USB: 25MHz, LCD: 31MHz, DSP: 100MHz.
0324214508|key |*|02|Initial log entry. Current logging level 4
0324214508|ht |*|02|Initial log entry. Current logging level 4
0324214508|httpd|*|02|Initial log entry. Current logging level 4
0324214508|ssps |*|02|Application, comp. 1: Label=PolyDSP Titan Mem1 FS3 (G.729), Version=3.1.2.0022 08-Dec-08 10:52
0324214508|ssps |*|02|Application, comp. 1: P/N=3150-11580-312.
0324214508|pps |*|02|Initial log entry. Current logging level 4
0324214508|sip |*|02|Initial log entry. Current logging level 4
0324214508|copy |3|02|'http://44%40localhost:****@192.168.178.20/SoundPointIPWelcome.wav' from '192.168.178.20'
0324214508|cfg |4|02|Edit|Error 0x380003 attempting stat of /ffs0/local/0004f214058e-phone_cfg.zzz
0324214508|cfg |4|02|Edit|Error 0x380003 attempting stat of /ffs0/local/0004f214058e-phone_cfg.zzz
0324214508|lic |4|02|License: feature VoiceQualityMetrics is not available
0324214508|so |4|02|No license for voice quality monitoring.
0324214508|so |4|02|All voice quality monitoring features disabled.
Multiple Msg Waiting Lights
in Mailbox
Posted
If the secretary's phone is able to send the neccessary subscription for MWI then that is no problem.
When using the "buttons" feature it is also possible to put the MWI on a button.
PAC also supports this.