similar to: TELASIP DOWN?

I seem to be getting a connection from telasip but instead of dialing my default extension, nothing happens. I listen to dead air. I have a fxo card configured and working on both inbound and outbound calls. Telasip is working outbound. I put in the recommended (by telasip) changes to the trunk for incoming, e.g. host=gw4.telasip.com insecure=very qualify=yes type=user context=from-pstn Then

I've had the same excellent responsiveness from telasip, on the rare occasion that I've had issues. YMMV -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Vile Sent: Thursday, April 06, 2006 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Telasip

II noticed that Caller ID presentation is not making it to my cell phone through outound Telasip calls and I don't know why. It may very well have been this way for awhile (or always, not sure I called my cell phone during telasip testing). Does Telasip expect a different format than SetCIDNum(NXXNXXXXXX) ? It has always worked for the Teliax lines. BUT--- It doesn't have a problem

I can't seem to get consistant outbound caller ID working correctly. I have set the fromuser and callerid field in my sip.conf for my TelaSIP peer, but half the time it shows up as "No Caller ID" on my cell phone, other times it shows it correctly. Using asterisk CVS. Any ideas? Doug

I am running Asterisk@Home Voicemail works fine but does not email out the voicemail attachments. Any suggestion? ----------------------------------- Voicemail.conf [general] #include vm_general.inc #include vm_email.inc [default] 201 => {password},Jeff G Laptop,jrglass@columbus.rr.com,,attach=yes --------------------------------------------------------------------- Sip.Conf [201]

It said 'include zapata-channels.conf', where this line wasn't commented bij the ';'... Could you post me a working example of such a config (or a part of it, for the X100P cards...? Thanks guys! Message: 9 Date: Sat, 19 Mar 2005 18:04:26 -0500 From: "Jeff Glassman" <jrglass@columbus.rr.com> Subject: [Asterisk-Users] newbie question To:

I have installed asterisk @ home 2.6. I am using a Telasip VOIP account. When I make outbound or inbound calls the calls seem to connect and then get hung up. I was wondering if there was something that I am misisng. I have tried several different sip.conf configurations. Here is what they are currently. telasip-gw canreinvite=yes context=telasip-in dtmfmode=rfc2833 fromuser=jrasxxx

Hi all, I've just setup a sip line with Telasip and when they route the calls to my asterisk box, they include an extension along with the context that is defined in sip.conf for that DID. At first, I couldn't figure why they were getting 404 error from my asterisk box, but then figured out that they are sending the call to an extension that matches my number with them, in the

Hi, I'm seeking to change my service provider (after ten months, I've had it with broadvoice), but I would like to keep my 310 number. I've been digging through the lists of other providers and am considering telasip (good plans and support number transfers). My concern is what precisely happens when a number is transferred from one service provider to another. After the transfer is

I have the following appended to gluster logs at around 100kB of logs per second, on all 10 gluster servers: [2012-06-11 15:08:15.729429] I [dht-layout.c:682:dht_layout_dir_mismatch] 0-sites-dht: subvol: sites-client-41; inode layout - 966367638 - 1002159031; disk layout - 930576244 - 966367637 [2012-06-11 15:08:15.729465] I [dht-common.c:525:dht_revalidate_cbk] 0-sites-dht: mismatching layouts

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service. Is there a way to be able to transfer calls to other extensions of my asterisk server FROM the cell phone/ This isn't a

Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service. Is there a way to be able to transfer calls to other extensions of my asterisk server FROM the cell phone/ This isn't a

Am new to Asterisk - have it up and running & connected to a couple service providers (telasip & teliax). Nice! Am running our Asterisk PBX on a static DSL connection (~600 kbps up/~4mbps down), and would like to extend VoIP service to 10 non-profits we're working with. Am I correct in assuming that all calls from each organization would route through our Asterisk server & be

Probably just a simple syntax issue, but does anyone know how to dial a number and the once phone has been answered, play DTMF tones and then disconnect. I am trying to use this for page notification. Ive been trying the following string with out luck: exten => s,2,Dial(SIP/TelaSip-gw4/5198881212|D(12345678) Any help would be greatly appreciated! -------------- next part -------------- An

Hi folks, I have here a database node running # rpm -qa | grep mysql-server mysql55-mysql-server-5.5.52-1.el6.x86_64 on # virt-what vmware that seems to have a connection problem: # dmesg |grep SYN |tail -5 possible SYN flooding on port 3306. Sending cookies. possible SYN flooding on port 3306. Sending cookies. possible SYN flooding on port 3306. Sending cookies. possible SYN flooding on

I wish to have my Grandstream GXP-2000 phones make a different distinctive ring for internal calls ( Internal ) or if the incoming call has no caller id 'NOCID'. The Grandstream phones calls allow 3 distinctive rings depending on the caller id. I have one set up and working for 'Internal' calls but unfortunately the same tone will ring if caller id is absent on a call. My

> Am 20.07.2018 um 18:52 schrieb Nataraj <incoming-centos at rjl.com>: > > On 07/20/2018 03:56 AM, Leon Fauster via CentOS wrote: >> Hi folks, >> >> I have here a database node running >> >> # rpm -qa | grep mysql-server >> mysql55-mysql-server-5.5.52-1.el6.x86_64 >> >> on >> >> # virt-what >> vmware >>

Hello, after some days running without problems, my riello idialog 800 ups get disconnected, and usbfs started flooding syslog with messages like: usb 1-1.2: usbfs: process 57342 (riello_usb) did not claim interface 0 before use Version is 2.7.1-1 debian packages, which I backported from sid to wheezy. Any hint? rob

Hi guys, Sorry if this is a little offtopic, but I was wandering what can one do to prevent/stop arp flooding ? Thanks, Alex -- This message has been scanned for viruses and dangerous content by LG-Network(http://www.lgnet.ro), and is believed to be clean.