search for: callcentric

Greetings Asterisk users! I'm trying to setup Asterisk system to act as a callback system together with callcentric (http://callcentric.com) but it appears that I hit common DTMF issue and I want to workaround this problem. Basically my current setup is the following: 1) I have dedicated Asterisk server that it is linked to my callcentric account 2) I have US phone number (DID) from callcentric attached to my a...

On 11.9, trying to set up a callcentric peer: sip debug: > <--- SIP read from UDP:204.11.192.161:5060 ---> > INVITE sip:1777<myccid>@10.10.11.180:5060 SIP/2.0 > v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46aaaaef4249814d16a2ffb990d > f: <sip:<calling number>@66.193.176.35>;tag=3606475083-...

...he same time ;Eric on extension 105 exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)         same => n,VoiceMail(105 at default,u) Where problem comes in - if person not at the desk - his cell phone shows call from OFFICE number and there is no way to tell who is really calling. We use Callcentric as a trunk if it makes any difference. I'd like to add info about caller when passing to cell phone if possible. Is there any way to do that? Thank you,Ivan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attac...

Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get "handle_request_invite: Failed to authenticate user <sip:PSTNnumber" This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. B...

Hello everyone, I want to connect a fax to an FXS port (TDM420P). For testing purposes, I connected an analogue phone to it first. However, when I pick it up, I cannot hear anything at all. The power cable is plugged into the card, the port is configured to use fxo-signalling. Also, immediate=no. Here's the files: /etc/zaptel.conf: # Autogenerated by /usr/sbin/genzaptelconf -- do not hand

Antony, It is correct. Noone connects to Asterisk box/server from outside.Callcentric SIP trunk configured and Asterisk maintains connection to it itself. No special ports opened, nothing. Connection happens from us to Callcentric and all calls routed in from CallcentricI don't know exactly how it's doing it by it works. Again, keep in mind it is working for many years for m...

To deal with google dropping xmpp for voice, I've gotten a callcentric number. The cc number connects to asterisk, and all works fine. Then I set up the cc number as the gvoice forwarding number. If I'm on the gvoice site, I can make a call and it will ring my cc number and then the outside number. That also works fine. BUT, when an outside call comes into gv...

Hi Can anybody help me on this ? I am using Asterisknow 1.5.0-Beta(Freepbx) I am having a problem getting the sip trunks to register. It makes no different which provider one is using. Trunk name: callcentric Peer Details: context=from-pstn fromdomain=callcentric.com fromuser=1777xxxxxxx host=callcentric.com insecure=very secret=pasword type=peer username=1777xxxxxxx Register String: 1777xxxxxxx:password at callcentric.com/1777xxxxxxx I believe that the above configuration is correct. I have tried d...

Asterisk Users, I am looking for Sip Providers for my Asterisk 1.2.13, running Debian Etch system with McLeodUSA's T1 service. Has anybody ever used Callcentric for their Sip Provider? Any service issues with Callcentric? Best Regards, John _________________________________________________________________ Messenger Caf? ? open for fun 24/7. Hot games, cool activities served daily. Visit now. http://cafemessenger.com?ocid=TXT_TAGHM_AugHMtagline

...at yahoo.com> wrote: > John, > > FF1565AABB2D-SLS is probably invalid because it's not registered/lost > registration. This client is connected via VPN to our network, it usually > works when it's "warm". Not concerned about it too much. > > 15555555555 at callcentric OTOH is an actual cell phone that should be > dialed out via callcentric trunk. > Maybe I'm smoking something thinking it was working before. I know it > works from > > extensions.conf > ------------------------- > [globals] > ERIC_CELL=SIP/15555555555 at callcentric &gt...

Hello, I have queues.conf setup with a group like so: [Sales](StandardQueue) announce = first member => SIP/FF4C119EEBF8-SLS member => SIP/FF9EF375CCFC-SLS member => SIP/13145555555 at callcentric ;Eric's cell member => SIP/FF1565AABB2D-SLS ;Eric's Yealink So, my idea here that it should ring all 4 phones at the same time. And it does work but randomly.I did trace a call and this is what I see. Only 2 phones (internal) called. External SIP at callcentric is not being called. Any i...

Hello, This is not technical post, just looking for suggestions on what to check.I have asterisk for long time, no updates, just maintain OS updates. I use SPA504G phones Very rarely and randomly when we pickup a phone - other side does not hear us. Call them back and all works. Now I have couple people I'm talking to and it seems like very call like this. Someone can't hear someone.

I have installed several call centers in the netherlands with the eyebeam softphone (from the counterpath guys) It is not free, but very stable, and pretty easy to use. It works great with asterisk (specially the presence option, so agents can see whether somebody is actually ready to take a call). In combination with sennheiser headset CC series, I have had no complaints. We also use a tapi

Thanks all, I did contact Callcentric about it and their tech support helped meget those headers established. They even helped to troubleshoot Asterisk dialplan. A the end all works as it should. Thank you,Ivan Message: 2 Date: Mon, 15 Oct 2018 23:39:31 +0200 From: Daniel Tryba <daniel at tryba.nl> To: Asterisk Users Maili...

Hello All! I asked week a so ago about how to call multiple phones alltogether (home, office, cell) Dial app looks simple, this is kind of what I have now: --------------------- [globals] IVAN_HOME_OFFICE=SIP/BF8 IVAN_OFFICE=SIP/CFC IVAN_CELL=SIP/83 at callcentric [internal] exten => 101,1,Dial(${IVAN_HOME_OFFICE}&${IVAN_OFFICE}&${IVAN_CELL},60) same => n,VoiceMail(101 at default,u) ?????????? Now, I have basic automated attendant and I have Queues setup ??????????? [general] autofill=yes shared_lastcall=yes [StandardQueue](!) musicclas...

Hi, I've been trying to configure my extensions.conf and sip.conf for two days now and I'm pretty sure it's just a small typo or anything I can't find by myself. My setup: - Asterisk connected via Fritz! PCI Card to a HiPath 3500 (2 channels) - Callcentric.com SIP channel to dial out to foreign countries - Cisco 7912 attached to asterisk using SIP (in another city) When I dial extension 85 my Cisco phone is ringing and I can talk and everything works fine. But when I try to dial an extension from the dialplan I never get a connection. I've post...

...callcounter=yes vmexten=5199 nat=no ; SE registrations register => user1:password1 at sipgate.co.uk:5060/se2489 register => user2:password2 at sipgate.co.uk:5060/se1268 register => user3:password3 at sipgate.co.uk:5060/se0845 register => user4:password4 at callcentric.com:5060/se1777 register => user5:password5 at sipgate.co.uk:5060/se4130 register => user9:password9 at sip.vohippo.com:5060/se1413 ; SJ registrations register => user6:password6 at sipgate.co.uk:5060/sj0151 register => user7:password7 at callcentric.com:5060/sj1777...

...setting up asterisk for my company. I had 94 employee to set this up for ... I never heard of asterisk before to b honest, so after researching a bit.. I started with a digium card with ZAP though that didn't work out as the card were flawed.. so ended up setting up SIP for everyone using a SIP callcentric accounts as well as sipura for pstn lines.. now it's working at it's minimal state.. but as am out of the heat of pressure from management.. so now It's time to learn about asterisk the right way as I had lots of help from this mailing list as well as the IRC channel that I'm not s...

...assume that I can use same context, add ?fax? extension and if someone calls to send fax - it will autodetect. Right? Per book, I made following setup additions: 1. In sip.conf [general] I added: ;FAX stuff faxdetect=yes t38pt_udptl=yes 2. In extensions.conf I hade something like this: [from-callcentric] exten => s,1,Goto(automated_attendant,s,1) ; FAX handling stuff AS IN BOOK exten => fax,1,Verbose(3,Incoming Fax) same => n,Set(FAXDEST=/tmp) ; folder where faxes will be stored same => n,Set(tempfax=${STRFTIME(,,%C%y%m%d%H%M)}) same => n,ReceiveFax(${FA...

Hey Guys We are replacing a BM4 with a trixbox (asterisk) virtual numbers as the customer wants to move the callcentre. They are asking for an equiv to the ipview I gather HUD may be or the panel view The problem is that we need to see (a) total calls in the queue (b) calls for specific DID - How can you give 1 DID preference to another DID ie DID 61740410001 = Fred Electrical DID