halalabu
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Posts posted by halalabu
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yes, snom 8xx, 3xx (Except snom 300) have a hard button.
snom 7xx you can use a context/(one of four below screen) buttons.
Hmmm, what specifically is the button labelled on an 821? I am not seeing anything resembling a conference button.
Thanks,
Hal
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Ahhh ok. Our snom phones have no hard button for it. I think i'll use the star code. Thanks!
Hal
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Is it possible to invite someone into a call with you? The same way a call-barge in works, but in reverse. E.g. if an employee is on a call with a client, and they want to get their supervisor on the line as well, they could enter a star code followed by an extension number and the extension would ring. When the extension is picked up, they are joined into the call.
The only other way I can think to do this is to have everyone hang up and call into the ad-hoc conference line, which is obviously not ideal.
Is there a way to do this? I see a "conference" entry on the star codes page, but I don't see any documentation for it.
Thanks so much,
Hal
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Based on the result, I believe the call is not even parked. How is this analog extension connected to the PBX? Maybe an ATA? If the ATA sends proper SIP messages, then the operation would work.
It is connected via a Linksys PAP2T Gateway. It's been working for other things like transferring calls. Any idea where to start looking?
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Based on the result, I believe the call is not even parked.
I'd agree. Are those the correct steps to park a call on an analog extensions?
Thanks,
Hal
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Hmmm, I tried making the park and retrieve codes different. When i try the call park code from an analog extension, it's just dead air. This is my sequence of events:
1) answer a call on the extension
2) press the flash/hook button (I now hear a dial tone)
3) dial my call park code *85, followed by the park orbit number 98
4) hear dead air
If i try the call retrieve code, *86, it just says there is no call to pick up.
Call parking and retrieving does work from the Snom IP phones.
Any ideas? I'm stumped.
Thanks so much,
Hal
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Hard to say, the log shows nothing that would explain it. I assume the file system is not full?
Correct. It's a brand new server. Uptime was 8 days, should the PBX be restarted on a schedule? I was in a bit of a panic so I didn't get to see the log file before I restarted. If it happens again I'll be sure to capture a copy of the log file before restarting.
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So, this just happened in the middle of the day. Nothing out of the ordinary. Every phone dropped it's registration and couldn't connect to the PBX. Calling into the PBX just rang continuously and then presented a busy signal. I was able to restore service by stopping the PBX and starting it again. Here is my log file (the important stuff, I think). Any ideas?
[0] 2012/06/14 11:57:17: Database sanity check: Deleting account 1 (type extensions)
[2] 2012/06/14 11:57:18: Deleting message 29 because the associated file recordings/20120523/111503-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 30 because the associated file recordings/20120523/111807-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 31 because the associated file recordings/20120523/111825-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 32 because the associated file recordings/20120523/111833-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 33 because the associated file recordings/20120523/111846-o-8885097627-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 34 because the associated file recordings/20120523/111939-o-8885097627-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 35 because the associated file recordings/20120523/111947-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 36 because the associated file recordings/20120523/112011-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 37 because the associated file recordings/20120523/120125-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 38 because the associated file recordings/20120523/120248-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 39 because the associated file recordings/20120523/120600-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 40 because the associated file recordings/20120523/120616-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 41 because the associated file recordings/20120523/125423-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 42 because the associated file recordings/20120523/125443-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 43 because the associated file recordings/20120523/125459-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 44 because the associated file recordings/20120523/125605-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 45 because the associated file recordings/20120523/125617-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 46 because the associated file recordings/20120523/125952-o-5186650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 47 because the associated file recordings/20120523/131006-o-6650411-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 48 because the associated file recordings/20120523/131022-o-18885097627-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 49 because the associated file recordings/20120523/131049-o-8885097627-701.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 50 because the associated file recordings/20120523/131344-o-6650411-501.wav is not present
[2] 2012/06/14 11:57:18: Deleting message 56 because the associated file recordings/20120524/113823-o-18884732963-103.wav is not present
[1] 2012/06/14 11:57:19: Starting up version 4.5.0.1075 Delta Aurigids
[1] 2012/06/14 11:57:19: Adding DNS server 127.0.0.1 to the dns server list
[1] 2012/06/14 11:57:19: Adding DNS server 10.0.0.202 to the dns server list
[3] 2012/06/14 11:57:19: Call limit reached. Call ports 0, g729 0
[1] 2012/06/14 11:57:19: Working Directory is /Applications/snomONE/conf
[1] 2012/06/14 11:57:20: UDP: TOS could not be set
[0] 2012/06/14 11:57:31: Administrator logged in from IP address 10.0.0.1, session 5k38ymo51fcvns4mrujy
[5] 2012/06/14 11:57:46: SIP Rx tls:10.0.0.151:4811:
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Is there a way to park a call from an analog extension? I thought maybe pressing the flash/hook button and then putting in the correct star code, but the message I get is that there's no call to pick up. I assume it's trying to retrieve a call from the park orbit, not send one to the park orbit. My park and retrieve codes are both *85. Thanks so much.
Hal
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Maybe you have to terminate your entry with a # (thats a question for the ATA when it detects the end of the dialled number).
Ahhh yep that does it. Thanks!
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I meant set the conference account number as part of the conference name itself. There is no variable for that. Instead of setting the name as Alabu Conference Room, set it as Alabu Conference Room x555
Ahhh got ya. Thank you.
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Currently, there is no option for the body. But the number 5555551212 should be your access number for the conference. If you still want to send alias for the conf room, then you can probably make it part of the description. Say, The conference room name is "AlabuConference Room x555"..
Ok, thank you. Can you tell me what the variable name is for the conference room number?
Hal
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Hi Folks,
Just have a couple of questions on conferences.
1) Is there a way to add to the body of the conference email? I see the email_conference.txt, but that appears to only affect the calendar invite.
2) On the calendar invite, how do I add the extension number of the conference room? I have our email_conference.txt set to default. It give the main phone number for the domain and the pin, but not the extension of the conference room.
Thanks so much,
Hal
Conference email:
SUBJECT:{lng subject1}{ssi variable name}{lng subject2}
SUMMARY:{lng summary1}{ssi variable name}{lng summary2}
DESCRIPTION:{lng description1}{ssi variable pin}{lng description2}{ssi variable room}{lng description3}
LOCATION:{lng location1}{ssi variable number}{lng location2}
Contents of the calendar invite:
BEGIN:VCALENDAR
VERSION:2.0
METHOD:REQUEST
PRODID:-//PBX/4.5.0.1075 Delta Aurigids//EN
BEGIN:VEVENT
ORGANIZER:MAILTO:mail@example.com
ATTENDEE;ROLE=CHAIR;RSVP=FALSE:MAILTO:mail@example.com
ATTENDEE;ROLE=CHAIR;RSVP=FALSE:MAILTO:mail@example.com
DTSTART:20120606T143000Z
DTEND:20120606T153000Z
DTSTAMP:20120606T145926Z
SUMMARY:Telephone conference "Test"
DESCRIPTION:The access code is 2763. The conference room name is "Alabu
Conference Room".
LOCATION:Conference Server 5555551212.
UID:F5DBB4876EB84E325BAF042FCB72AC65
STATUS:CONFIRMED
SEQUENCE:1
TRANSP:OPAQUE
CLASS:PUBLIC
PRIORITY:5
END:VEVENT
BEGIN:VALARM
TRIGGER:-PT15M
ACTION:DISPLAY
DESCRIPTION:Reminder
END:VALARM
END:VCALENDAR
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Well, there is a transfer star code (*77 for blind transfer). However, the ATA must be able to put the call on hold. Usually that can be done with a flash hook (tap the on hook key for a short time). Then you should get secondary dial tone where you can dial *77123 to transfer the call to 123.
Ahhh that works. There is a very long timeout after dialing before the PBX tries to transfer the call. I'm assuming there is a setting for how long it waits (or is it a setting in the ATA?) I'll look around. Thanks!
Hal
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Is there any way to transfer a call from a phone on an analog extension? I can't get the star code to work (pressing hold does not product a second dial tone), and using the star code while on the line does nothing. Any ideas?
Thanks so much.
Hal
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It's a bug.
Read through this thread - also a temp fix explained.
http://forum.snomone.com/index.php?/topic/5967-delta-aurigids/
Ahhh thanks. Will set to unlimited for now.
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One of the extensions on our snom one gives the message that the mailbox is full when trying to leave a message. It even sends an email saying there are 159 messages in the mailbox and that it is full. However when checking the mailbox both from the web interface and the phone, it shows completely empty. Also, this is a new system, there's no way there has ever been 159 messages in that mailbox. Maybe 10 or 15 tops. Any ideas?
Thanks so much!
Hal
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Nevermind. My fault. Somehow I had a ringtones.xml file in the html directory that was overriding the files in the web interface. All fixed now.
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I'm trying to change the default external ringtone to match the default internal ringtone. I tried updating my ringtones.xml file via the web interface, but it does not seem to have changed anything. Here is my default ringtones.xml file
<?xml version="1.0"?> <ringtones> <tone name="custom1"> <vendor ua="snom.*" type="alert-info">Custom 1</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="custom2"> <vendor ua="snom*" type="alert-info">Custom 2</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="custom3"> <vendor ua="snom.*" type="alert-info">Custom 3</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="custom4"> <vendor ua="snom*" type="alert-info">Custom 4</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="internal" type="internal"> <vendor ua="snom.*" type="alert-info">Internal</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr3></vendor> </tone> <tone name="external" type="external"> <vendor ua="snom*" type="alert-info">External</vendor> <vendor><http://127.0.0.1/Bellcore-dr2></vendor> </tone> <tone name="intercom" type="intercom"> <vendor ua="snom*" type="alert-info">Auto Answer</vendor> <vendor type="call-info"><{from-uri}>;answer-after=0</vendor> </tone> </ringtones>
And this is what I changed it to: (note the last number in the Bellcore- line for external ring
<?xml version="1.0"?> <ringtones> <tone name="custom1"> <vendor ua="snom.*" type="alert-info">Custom 1</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="custom2"> <vendor ua="snom*" type="alert-info">Custom 2</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="custom3"> <vendor ua="snom.*" type="alert-info">Custom 3</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="custom4"> <vendor ua="snom*" type="alert-info">Custom 4</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="internal" type="internal"> <vendor ua="snom.*" type="alert-info">Internal</vendor> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr3></vendor> </tone> <tone name="external" type="external"> <vendor ua="snom*" type="alert-info">External</vendor> <vendor><http://127.0.0.1/Bellcore-dr3></vendor> </tone> <tone name="intercom" type="intercom"> <vendor ua="snom*" type="alert-info">Auto Answer</vendor> <vendor type="call-info"><{from-uri}>;answer-after=0</vendor> </tone> </ringtones>
Should this not change the external ring to be the same as the internal one? If not, what am I doing wrong?
Thanks so much.
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Did you check out http://wiki.snomone.com/index.php?title=How_to_implement_fax_on_snomONE%3F ?
Thanks for the link. Yes I did set up my auto attendants per the documentation. As I mentioned internet fax services are routed correctly, it seems just that a physical fax machine on an analog line is not working. Any ideas? Thanks so much.
Hal
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Hi Folks,
We're trying to set up the AA to redirect faxes to a fax extension. This is the Setup: PSTN feeds to a Patton 4114, which is a functioning trunk on the Snom One. The fax machine is plugged into an ATA gateway (Linksys PAP2T), which is properly registered with the PBX. After some finagling with the PAP2T, I can fax directly from the PSTN to the fax extension (I had a friend send a fax, and I routed his number to the fax extension based on his CID). This works fine. However, when dialing in to the main number where the auto attendant picks up, the auto attendant does not transfer the fax to the extension. For what it's worth, when I'm sending a fax from the PSTN, I believe I hear the CNG tone while the line is ringing, but not after the AA picks up (not sure about this though). Additionally, if I send a test fax from an internet fax service like freefaxbutton or similar, the AA routes that to the fax extension and it comes through fine. Any ideas?
Thanks,
Hal
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Right now thats hard coded. We can change it or make it a setting, this would be a safe path for you to get this problem solved finally. However, it would be good to know why other programs can use PLAIN and we cant. Can we give you a test build where try one more time to use PLAIN, but with only one ID? (What OS are you using?)
Hmm alright. Yeah I don't have a problem using a test build. I'm using Mac OS X 10.7.4 Lion Server. Thanks for your help figuring this out.
Hal
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Hmm.
Actually the PBX does not support CRAM MD5; it supports only PLAIN and LOGIN. Is there something in the middle?!
Ahhh yep my bad. It was reverting to LOGIN not CRAM MD5. In any event, is there a way to force it to use LOGIN instead of PLAIN?
Thanks,
Hal
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Ahhh ok thank you. No idea how those got customized.
Call Invite
in Call Treatment
Posted
Ahh look at that! Thank you!
Hal