similar to: asterisk and Broadvoice Outgoing Again :(

Since Thursday evening my asterisk box has been failing to register with broadvoice. I haven't changed any of my config files in the last week. Can anyone suggest anything? Asterisk is reporting: *CLI> Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout: Registration for '703XXXXXXX@147.135.8.129' timed out, trying again -- Got SIP response 404 "Not found"

I finally got my incoming and outgoing to work on Broadvoice with a configuration file that is no where close to the one given by them. Here it Is (sip.conf). For others who have a working config could u please share so that I could compare. Thank You [broadvoice] type=friend username=[number] fromuser=[number] secret=[password] host=sip.broadvoice.com fromdomain=sip.broadvoice.com

Hi, I have two accounts with broadvoice. Now, I want to be able to distinguish between them. I though that this would be simple by adding "/EXTEN" at the end of the register statement. For example: register => num1:pass@sip.broadvoice.com/1000 Unfortunately, this is not working. When I call into my box I hear busy tone. My config looks like this: [root@voip asterisk]# cat sip.conf

I've tried everything with the * box after this weekend. I have read every document on the problems people are having with them after this weekend as well, but none of them address my problem. I checked my settings in my sips which I have below as well, I have changed the host file a few times, but this was new to me and I never had modified it before. I have and the same results

Is the Broadvoice service up? I just signed up with them and started receiving calls in no time but could not make calls. And after a few minutes I cannot even place calls. register => [number]:[password]@sip.broadvoice.com [broadvoice] type=peer fromuser=[number] host=proxy.lax.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice dtmfmode=inband any help would be

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on

Hi, I have installed Asterisk, Zaptel, Libpri, Addons, Sounds in my Linux system. I got registered with sipgate.co.uk and got the UK phone number i.e., 0207100xxxx. I configured my Asterisk server with 0207100xxxx. When I made a call to this number from outside phone, my XLite extension is not ringing. Its directly going to Voicemail or telling that "person is unavailable". When I

Currently, o2net_accept_one() is allowed to accept a connection from listening node itself, such a fake connection will not be successfully established due to no handshake detected afterwards, and later end up with triggering connecting worker in a loop. We're going to fix this by treating such connection request as 'invalid', since we've got no chance of requesting connection

Hello, Have a weird problem when using asterisk (1.0.6). There are certain numbers I cannot dial when using asterisk with my broadvoice account. No problems with inbound. With outbound calls, I can call some numbers (for example broadvoice customer support number) and unsuccessfully with some. However, when I configure my account directly on x-lite, I dont see these outbound problems. Here is a

All, I signed up for a broadvoice BYOD plan over the weekend (very excited about their offering) and after about an hour I had asterisk registered and was making in and out bound calls. However, the next day (without changing anything) I couldn't call in or out and haven't been able to get it going again. I can connect using a softphone (X-Lite) and make calls in and out

Hi! Useful advice, thanks! Actually I had been using foobar2000 to recode, because it just makes it so easy to convert multiple files while keeping the metadata (I confess, I'm a "tagger"). But it's easy to miss some encoder option when being presented some default suggestions in a dialog form... Apart form that I always had the impression that Opus could be quite smart

I have configured iax.conf and extensions.conf as instructed on FWD website (http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76) and I can successfully receive calls and make test calls to 612, 613, etc. The problem is that I can not make a call to another FWD user. Here is what asterisk says: -- Executing [393xxxxxx@default:1] Set("Zap/1-1",

Greetings, What is the recommended settings for using SPA-3000's FXO port for dialing out to PSTN in regard of the DTMF? The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports registered to the Asterisk box with unique username/passwords. The inbound PSTN DTMF works excellently, e.g. people calling from PSTN into the * box are able to pick IVR items with DTMF reliably. The

Hello, With an extensions.ael enabled system, I keep getting whatever I change into my "astup.call" file : [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/astup.call [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service:

Changes since v1: - Set probing source port to be nbd_local_port+1 instead of always using 50124 to deal with multi-disk scenario. - Set SO_REUSEADDR on client socket to avoid issues with old connections in TIME_WAIT. I've been running this for a few hours now using the updated multi-disk test and haven't seen any problems.

Hi All First time poster, long time reader. I just ran into something really bizarre. I've enabled videosupport and have been testing sip calls with Xten Eyebeam software, it works (mostly) However, when I have Videosupport=yes In the [general] section of my sip.conf, broadvoice calls fail w/ "We're sorry your call cannot be completed at this time" So... I've

- perform dns lookup on hostname, - randomize local tunnel port - bump rpm spec version updated to include danpb's feedback regarding getaddrinfo --- main.c | 36 ++++++++++++++++++------------------ ovirt-viewer.spec | 9 ++++++++- tunnel.c | 29 ++++++++++++++++++++--------- 3 files changed, 46 insertions(+), 28 deletions(-) diff --git a/main.c b/main.c index

Hi all, I am trying to configure my Asterisk PBX to connect to a broadvoice account. I want incoming calls to go to the "s" extension and to ultimately have outbound calls routed through my broadvoice account(s). I cannot for the life of me get this to register correctly. I am currently receiving this message on the console: Failed to authenticate on Register to

HELP! Ok, so I have the following SIP.CONF: [general] context=default port=5060 bindaddr=10.1.1.200 externip = XX.XXX.XX.XX localnet=10.0.0.0/255.0.0.0 disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw register => ##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234 [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromdomain=sip.broadvoice.com

- perform dns lookup on hostname, - randomize local tunnel port - simple autobuild script - bump rpm spec version --- autobuild.sh | 41 +++++++++++++++++++++++++++++++++++++++++ main.c | 14 +++++--------- ovirt-viewer.spec | 9 ++++++++- tunnel.c | 29 ++++++++++++++++++++--------- 4 files changed, 74 insertions(+), 19 deletions(-) create mode 100755