Re: [asterisk-users] Call Center Scenario -- take 2
Very possible in a number of ways - one that springs to mind if you are using FreePBX is the day/night mode settings - you could hit a code to send the incoming trunk to an IVR tree that includes a front end message and then options to queue or leave a message etc. We use this to divert out of hours calls to our branches over to 'on call' veterinary clinics. NK -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore Sent: Thursday, December 06, 2007 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call Center Scenario -- take 2 Not sure if my original message made it through. Going to try this again. :) --- Greetings, List. I would like to implement a procedure in my call center but am not sure the best way to implement it. I'm hoping I can describe it here and that I'll receive some feedback and/or suggestions on how to proceed. Here's my situation: My call center fields calls regarding internet access issues for local apartment complexes and businesses. Most of the time, we get a few calls here and there from new tenants unsure how to set up their connection. Every so often, however, there will be some sort of issue (ISP going down, router crashing, etc...) that will leave all users without internet access. When this happens, we get a flood of calls and the girls in my call center can quickly become overwhelmed. What I'd like to do is set up a system whereby incoming calls during a known outage are instead redirected to a recording explaining the issue and the option to have the caller leave a message (a la voicemail). All calls come down our T1 and we are able to identify the incoming account based on its DID. We would need to do this on a per-account basis. My girls would also need to have the ability to toggle on/off the redirection as well as record a message for the caller to hear -- at a moment's notice. Since my girls only field the calls and don't do any actual support (I do that), it'd be ideal if my VM indicator would also let me know if any callers left messages during a known outage. Again, this would be ideal, but most certainly not necessary. So, what say you list? Any suggestions on the most efficient way to do this? I am quite familiar with PHP and not adverse to writing a script to do this for me (I suspect I will have to anyway), but don't wish to reinvent the wheel if something like this already exists. Thanks in advance, Jay ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Possible mysql database corruption
Hi, I posted earlier about 'sip show registry' not showing any trunks and there seeming to be no attempt by one of my systems to register with our service provider. Following some notes elsewhere I ran 'myisamchk *.MYI' on the mysql database and it came up with an error in sip.MYI, which I repaired. Unfortunately, still no registration so I backed up the database, deleted the trunk details in the database using phpmyadmin and then re-made it. Still no joy. I am hoping I am close to solving this without a complete reinstall ad would appreciate any advice on sensible (and no so sensible!) things to try to fix SIP trunk registration. Thanks Nigel Kendrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip show registry shows nothing
Hi, In installed a new Trixbox-based system on Friday and initially had some problems with the system registering with our service provider (voip.co.uk), which I put down to the router's configuration. Anyway, the system was finally 'fixed' and worked well until Monday and then just stopped registering with voip.co.uk and I have spent the whole day trying to encourage it back online. Everything else works - inter-extension calls are OK, I have a Snom 360 at home registered via broadband and I have even put the server in the DMZ without any effect. I have two other servers with similar configurations successfully registered so I am a bit puzzled. I have deleted and remade the trunk to voip.co.uk several times. Thing is that 'sip show registry' shows nothing and I cannot see any evidence in the debug text to indicate that the server is even attempting to register (although I will admit to not being an asterisk guru). One of my checks was to configure a Snom 190 on site to register with voip.co.uk and this worked OK. The trunk does, however show in 'sip show peers': ELY/XX 80.249.108.21N 5060 OK (42 ms) 4851 (Unspecified)D N 0UNKNOWN 4850/4850 my ext at home D N 37731OK (127 ms) 4806/4806 192.168.113.207 D N 2054 OK (16 ms) 4805/4805 192.168.113.206 D N 5060 OK (7 ms) 4804/4804 192.168.113.205 D N 5060 OK (7 ms) 4803/4803 192.168.113.204 D N 5060 OK (7 ms) 4802/4802 192.168.113.203 D N 5060 OK (7 ms) 4801/4801 192.168.113.202 D N 5060 OK (7 ms) 4800/4800 192.168.113.201 D N 5060 OK (8 ms) The server can ping and traceroute to voip.co.uk and I am getting a bit lost for things to try. My final attempt has been to install the latest 1.2 from svn and so I am now running SVN-branch-1.2-r68732. Before I rush headlong into debug files, are there any specific modules or settings that I can check to see that the server really is trying to register the trunk? Happy to post logs etc. here if someone lets me know what is best to show. Thanks Nigel Kendrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] A couple of questions for the Mitelgurus(phone-related - not systems)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Porch Sent: Friday, May 11, 2007 8:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] A couple of questions for the Mitelgurus(phone-related - not systems) Nigel, You cannot upgrade a non-dual mode 5220 to SIP. If you are referring to the cable that connects the 5310 to a 5235, that is a standard CAT5 straight-through cable. Barry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nigel Kendrick Sent: Friday, May 11, 2007 7:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] A couple of questions for the Mitel gurus(phone-related - not systems) Hi Folks, Just in case there are any Mitel gurus here: 1) Is it possible to convince a non-dual mode 5220 phone to 'upgrade' to the SIP firmware? I have inherited one that's Minet only. 2) I have a 5310 conference unit and 5235 phone in SIP mode, but someone's lost the connecting lead. Can anyone recommend anywhere in the UK for a replacement lead or confirm the pin-out so I can check whether a generic RJ-RJ lead will work without frying anything. Thanks Nigel Kendrick Thanks Barry, I managed to find some specs for the 5310 and when it mentioned that it was PoE powered I took the plunge and tried a cat 5 patch lead and everything's working fine. Shame about the 5220 - but no big loss. Nigel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A couple of questions for the Mitel gurus (phone-related - not systems)
Hi Folks, Just in case there are any Mitel gurus here: 1) Is it possible to convince a non-dual mode 5220 phone to 'upgrade' to the SIP firmware? I have inherited one that's Minet only. 2) I have a 5310 conference unit and 5235 phone in SIP mode, but someone's lost the connecting lead. Can anyone recommend anywhere in the UK for a replacement lead or confirm the pin-out so I can check whether a generic RJ-RJ lead will work without frying anything. Thanks Nigel Kendrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Feedback on Linksys SPA-921 and GrandStreamGXP-2000
Feedback on the GXP2000 - we have around 10 of them: 1) Great if the firmware's recent (but not too recent - see GS info over at http://www.voip-info.org/wiki/view/GXP-2000) 2) Good caller ID 3) Speakerphone OK 4) Good features - Asterisk friendly and they support paging/announcements 5) BLF works fairly well but has the occasional hiccup 6) Power plug/sockets are a loose fit and moving a phone will often 'glitch' it so it reboots - this is the biggest PITA we have found - go with PoE where possible 7) LCD backlight LEDs (white) fade within a month or so if they are left on permanently, which can make the display hard to read in some conditions. Aiming to take a look at how easy these are to replace. 8) We have 4 phones connected back to base via 512K ADSL and NAT/STUN works well, plus the phones do not tend to disconnect randomly and fail to re-register (like our test Safecom phones) Overall, the GXP-2000 seems to be good for the money. It's our phone of choice for the spec/price. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] WIFI SIP- The Best phone
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mitcheloc Sent: Sunday, December 31, 2006 8:52 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] WIFI SIP- The Best phone Those wifi phones are neat but I'd rather not carry around two devices, does anyone know of any good dual-mode GSM/SIP phones? I'm using a T-Mobile MDA right now and it is way too slow. Apparently the Nokia e61 has a built in SIP client, but there might be a new model around the corner (worth the wait?) Suggestions? I've had a play with a Nokia E70 - the 'bar' version of the E61 and gave up! Menu navigation is dire - I went through hoops trying to get SIP working - I know from others it can be done, but I bailed out when I realised that to put these phones in the hands of inexperienced users would be a recipe for a lot of frustration and support calls. I'd expect the 'PDA style' E61 might be easier to use. I have an HTC Hermes phone (Vodafone V1605 in the UK) running Windows Mobile 5. I have fired up the beta version of SJPhone on it and it was just about useable, but not 'production ready'. I hope that there will be some decent WM5 software in the near future but am wondering what sort of battery life can be expected. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Selecting outbound trunks
Hi Folks, Can you point me towards some info on how to specify that certain extensions use specific outbound trunks - we can only set outbound caller ID against the SIP accounts managed by our service provider (we cannot pass CID info to them at the moment although they are promising this facility) and so I'd like to do the following: Extn A + Extn B + Extn C - Outbound via SIP account 1 only Extn D - Outbound via SIP account 2 only I presume there's some grouping arrangement that will do this? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
