similar to: asterisk & google contacts

Displaying 20 results from an estimated 10000 matches similar to: "asterisk & google contacts"

2015 Jun 11
2
asterisk & google contacts
2019 May 01
4
very high traffic without any load
Hi everyone,   I am new to using tinc and currently trying to set up a full IPv6 mesh between 4 servers of mine. Setting it up went smoothly and all of the tinc clients do connect properly. Routing through the network works fine as well. There is however a large amount of management traffic which I assume should not be the case.   Here is a quick snapshot using "tinc -n netname top"
2015 Jun 15
1
small homebrew pbx
James Cass wrote: > I picked up a cheap JS200-FX on ebay: http://x100p.com/products/js200fx.php for $30, and it works great for a home install. Very low power draw as well. > > James Cass <http://goog_987864563> > jcass78 at gmail.com <mailto:jcass78 at gmail.com> The JS-200 runs a very old ( 1.4 ) version of Asterisk I have set up more than 30 nodes using the HP thin
2019 May 02
4
Aw: Re: very high traffic without any load
2016 Mar 02
4
Dial your phone and contact phone from within outlook?
I am wondering what the best solution is for initiating a call from Outlook Contacts. I imagine something that would start the call from the outlook card (or similar) and then dial the user's extension and the contact's phone number and place them in a bridge. Anyone use something like this? Travis Ryan Director of Information Technologies Oscar Winski Company 2407 North Ninth Street
2019 Aug 25
2
tinc 1.1pre17 on fedora 30
2018 Jul 07
4
dsync panic
Hi, I just upgraded from dovecot 2.2.19 to 2.3.2. "doveadm backup" worked fine in v 2.2.19, but now panics (user with shared folder): /opt/dovecot/bin/doveadm backup -u testuser -1 sdbox:/tmp/testuser dsync(standl2): Panic: file mailbox-attribute.c: line 360 (mailbox_attribute_get_stream): assertion failed: (value_r->value != NULL || value_r->value_stream != NULL) dsync(standl2):
2015 Jun 25
2
asterisk email to fax
I hope his mother in law doesn't live with him. That's a support issue for sure. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin Larsen Sent: Thursday, June 25, 2015 2:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk email to fax > Since the O.P. said
2015 Jun 17
4
small pbx for the office [it was: small homebrew pbx]
Lukasz Sokol wrote: > but have you considered a web-managed config-builder such as FreePBX? > Instead of building your dialplan from scratch ? I've never used FreePBX, but, after having looked at its website, I think I have a general understanding of what it can do. What I don't understand is how FreePBX answers my question about the Linksys SPA3102 being good for a mission
2019 May 06
4
very high traffic without any load
Lars, interesting - do you have an example of what that might look like in the config file? Thanks! On Sun, May 5, 2019 at 6:00 PM Lars Kruse <lists at sumpfralle.de> wrote: > Hello Christoph, > > I am glad, that you discovered the source of the problem! > > > Am Sat, 4 May 2019 08:30:28 +0200 > schrieb "Christopher Klinge" <Christ.Klinge at web.de>:
2018 Jul 13
4
dsync panic
I think I get pretty much the same issue: dsync(support): Panic: file mailbox-attribute.c: line 360 (mailbox_attribute_get_stream): assertion failed: (value_r->value != NULL || value_r->value_stream != NULL) dsync(support): Error: Raw backtrace: /usr/lib64/dovecot/libdovecot.so.0(+0xc9e06) [0x7fba8a348e06] -> /usr/lib64/dovecot/libdovecot.so.0(default_fatal_handler+0x2a) [0x7fba8a348e4a]
2015 Jul 02
2
asterisk email to fax
2016 Apr 26
7
my dahdi dont'n start
<!DOCTYPE html> <html><head> <meta charset="UTF-8"> </head><body><p>Hello,</p><p><br></p><p>Having installed DAHDI to be able to use the meetme() application , when I start the dahdi service it generates me the following error:</p><p>-bash: /etc/init.d/dahdi: No such file or
2020 Jul 05
1
export of information about mail into csv
On 05 Jul 2020, at 09:10, Aki Tuomi <aki.tuomi at open-xchange.com> wrote: > doveadm -ftab fetch -u user "hdr.from hdr.to date hdr.subject" mailbox FooBox | sed -e "s/\t/;/g" > result.csv > > Might need bit of tuning hdr.date. Other than that, seems to work fine. -- "Are you pondering what I'm pondering?" "I think so, Brain, but should
2015 Jun 11
0
asterisk & google contacts
On Thursday 11 Jun 2015, tux john wrote: > Hello everyone. i am running an asterisk server and i would like to have > the contacts from google. so every inbound call with fetch the caller ID > from google contacts and present it to my screen. This is really three problems, as follows: (1) Accessing the Google Contacts API to retrieve someone's details based on their phone number.
2016 Feb 17
5
1000 analogue lines with asterisk
Hello all, Can someone recommend what hardware to use for a 1000 analogue line capacity asterisk PABX? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/2bcd322f/attachment.html>
2019 Apr 12
2
Mailbox INBOX: Opening INBOX failed: Mailbox doesn't exist: INBOX. Maybe master user + namespace problem?
2019 Jun 19
2
Instable graphics with GeForce GT 730M, especially on external monitor
2018 Jan 29
0
geo-replication command rsync returned with 3
Hi all, by downgrade of ii? libc6:amd64 2.23-0ubuntu10 to ii? libc6:amd64 2.23-0ubuntu3 the problem was solved. at least in gfs testenvironment running gfs 3.13.2 and 3.7.20 and on our productive environment with 3.7.18. possibly it helps someone... best regards Dietmar Am 25.01.2018 um 14:06 schrieb Dietmar Putz: > > Hi Kotresh, > > thanks for your response... > > i
2014 Jul 30
2
SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines
I'm having a problem with a new SIP trunk. Calls within the UK to fixed lines are fine, but calls to mobiles have noticeably poorer audio quality. I thought it might have been a codec issue; we have used G.726 for internal and external calls (over primary ISDN and GSM). So I tried allowing "alaw", (G.711 A-law) which is the native codec used within the PSTN in this country,