similar to: Act! Integration

Hey Guys! We have two Asterisk with A102D Sangoma cards now i want to connect them back-to-back over PRI line via Cross-cable so what would be the configuration specially timing source and all? anybody did it before like this ? I want to make sure everything before putting in production.. (saving my downtime) -S -------------- next part -------------- An HTML attachment was

Hi! My customer want's to allow calls to landlines in EU and US and disallow calls to cells in EU. Rest of countries are blocked. Country blocking is easy... Is there a service that allows checking phone number? Maybe some specific Enum? I ask for number and server responds with info, for example: "Cell Phone, Belgium" or "Land Line, Germany". -- Piotr Gorski

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal :

I listened to your email using DriveCarefully and will respond as soon as I can. Download DriveCarefully for free at www.drivecarefully.com Sent via my BlackBerry from Vodacom - let your email find you!

I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from '<sip:22942 at 10.0.0.3>' failed for '10.0.0.228:5060' - No matching peer found my logger.conf

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. - -S - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -----BEGIN PGP

Hi folks, I repeat "as is" the title of a post someone did a few months ago, since I am facing the same problem and did not see one single answer to his post. Maybe I'll be a little bit more lucky. When I'm trying to issue an Atxfer AMI command, in the asterisk 1.8 branch, what happens is that some DTMF's are sent, like this : [Mar 22 15:46:27] DTMF[5910]: channel.c:3900

I am becoming frustrated with our current VOIP provider. Does anyone have any suggestions for a provider that supports asterisk well and provides solid service? Voip-info.org has a husge list of providers, but it is impossible to tell the fly-by-night operations from the reputable providers. --Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am having issues sending and receiving fax on my asterisk setup. Currently I have a server that has 2 x E1 TDM cards one is sangoma and the other one is openvox. Both support echo cancellation. One of the e1 is connected to our telco provider via mfcr2 where all our incoming calls originate. On the other end is a pri connection going to HICOM PABX where the local attached to a fax is

Hi, I am having a little problem and I hoped maybe I could get some help here. I deployed a Asterisk 1.8 server of my own to make SIP calls just between my friends. The server is configured with a public IP address. My friends and I are using "Acrobits Softphone for iPhone" as a client. I am using its push service which is hooked up to my Asterisk server. Now, the current situation is

Hello, I have Asterisk-1.8.3.2, dahdi-linux-complete-2.4.1+2.4.1, and libpri-1.4.11.5 installed and running on a Ubuntu 10.04 server all built from source. Everything is working nicely except one small issue. The CDR records are stored in the CSV file correctly and complete. The MySQL storage is working as it should and is automatically updating all the fields except the CLID field. I have

Hi Guys, We are getting a strange issue on certain polycom phones, sometimes when a call comes in it just "flashes" to show there is a call but does not play any sound. This problem is very intermittent and happens to maybe 2 out of 10 calls. Has any else experienced this before? -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Mon, Mar 9, 2015 at 9:40 AM, David Wessell <david at ringfree.biz> wrote: > Welcome to our hell. > > We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally > got Polycom to issue a hotfix firmware version. I'll be happy to share it > with you offlist, just email me. > > Officially Polycom will fix the issue in 5.3 in a few months.. > >

I have an asterisk 1.8.x box that intermittently returns a 401. Calls come through the same peer all the time, from the same carrier. However intermittently the asterisk box returns a 401. Below is the output of a failed call (1st) and a successful call (2nd). I can't see any difference until we get to these lines. Bad call: --- (17 headers 14 lines) --- Sending to carrierIP:5060 (no NAT)

Greetings- As many of your are Polycom "experienced", I was hoping some kind soul could provide direction on a specific issue. On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an instance where, using intercom/paging functionality of FreePBX, I need to override an end user's 'Do Not Disturb' selection on the handset. By default, DND simply

Matt, At some point you need to consider how much is too much... I run a call center with more then 125 commissioned phone sales reps and more than 60 customer service reps. We run dual servers, fiber from one provider and 6 bonded T1's from another provider. We purchase our so trunks from a wholesale company who is a major provider to resellers. Being so, their network is extremely

Package: xen Version: 4.0.1-1 Tags: patch Here is a patch to get rid off the obvious problem with string exceptions, and to change to rules to build with python2.6. -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: xen.debdiff URL: <http://lists.alioth.debian.org/pipermail/pkg-xen-devel/attachments/20100912/cb04d61b/attachment.txt>

Hey Guys, I knew this is stupid question but i just want to know what is the difference between Queue member logged out vs Pause ? -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110321/7646a66b/attachment.htm>

> I'd suggest try recording in ulaw first and then convert all to wav after. > It may have something to do with timing since you're using iax, what are you using as a timing source? Hardware or software? > Sent from my iPhone -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Normally daylight date change last sunday march for Turkey but this year daylight saving time will change for Turkey at 28 March 2011 at 03.00 AM so how can i change daylight save time setting for this year on CentOS servers ? I wrote rule file like below and compile it with zic but when I compile rule file,it has changed date immediately but it should be change at 28 March 2011 03.00 AM.