similar to: Broadvoice/MWI?

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

I'm experiencing a "Failed to authenticate on INVITE" error (see output below) whenever I try to MAKE a call through the Broadvoice account. I noticed some others had the same problem but it went away when they rebuilt Asteris w/ a new version. N such luck for me! I'd be grateful for any assitance. Here's what I've done so far: 1) I downloaded the latest stable

Hello all, I have an Asterisk installation with Kamailio using realtime integration. I have gotten my clients to register, but there is something odd about the sip message flow with some of my clients. My clients are Zoiper and Asterisk is 11.10.2. When I set 'Subscribe to MWI' value to 'both', after a normal, successful registration Asterisk begins to send REGISTER messages to

My SIP seems to be configured correctly as I can dial out and my minutes show up on my broadvoice bill, but whenever anyone calls my broadvoice # inbound they just get a busy signal. I dont get anything in the logs saying anything came in from broadvoice at all. Has anyone had this/simmilar problem with inbound from Broadvoice? Any suggestions? Thanks Neri -------------- next part

Hi all, We seem to be having an interesting issue maintaining our registrations with Broadvoice. It seems to be related to the fact that Asterisk does not normally include authentication with its first REGISTER attempt. Normally, Broadvoice doesn't care, responds with a 401 Proxy Auth Required, and asterisk answers accordingly. However, if the account is already registered, Broadvoice

Hi all - sorry if what I'm asking is FAQ by now - I only have 2789 digest messages that I've not read yet... ? The local phone company (Bell South) has gotten completely out of hand with their rates, and with them suing anyone who wants to compete against them...? So, I'm thinking very hard about going ALL VOIP here at home. ? Hardware I have: Old 586 chassis Old Pentium II laptop

Hi - We've got multiple offices with their own asterisk boxes (CVS HEAD 11/03/04-14:59:37) connecting to each other using IAX forwards. All users are on SIP phones. Voicemail is centralized to one location. Everything is hunky dory except that the users in the remote offices don't get MWI on their phones. I've seen the other posts to this list regarding this, and

I'm haveing trouble connecting an Ericsson Webswitch 100 to asterisk. Has anyone gotten a Webswitch running? When I try to connect asterisk thinks everything works fine, while the webswitch just rings. I belive chan_h323 is picking the wrong port to talk at the webswitch on, however I'm not sure, nor am I sure how to fix it. Any clues/hints? A tcpdump is attached to show the session.

Hi, I'm using a Cisco 7905 (with SIP image CP7905010300SIP050414A) on Asterisk 1.07. Whenever the user gets voicemail, the MWI goes on, and when the user listens to the voicemail, the MWI goes off. However, the phone is configured (as in the default config by Cisco) to re-read its configuration from tftp every hour. This 'warm reboot' also turns off the MWI, and it won't

I've gotten nothing from http://bbs.atcom.cn on this so far. Perhaps someone on the list has experience with this. Is there a way to get MWI support for PA168V-based ATAs? Apparently some IP phones based on the PA168V chip has this support already (Atcom AT-320 for example) by configuring Asterisk with 'mailboxdetails=yes' in iax.conf. On my ATA, however, it does nothing. Any

Greetings All, Regarding this archived post. http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html Did anyone ever find an solution to this? I've got a new box running 13.3.0 with the exact same issue. For those that don't read the link. I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, These are loaded into asterisk without

I have a situation that I need help with. I have 2 phone systems, 1 Asterisk and 1 Avaya. All voicemail is kept on the Avaya system. Whenever a call comes into an extension that the Asterisk server owns, I re-direct it to a different number that is owned by the Avaya System. If that Avaya extension does not answer it, I send it to the voicemail on the Avaya Messaging system for the extension that

BroadVoice sells a wireless SIP phone for $149. Does this phone as sold by BroadVoice work with Asterisk or is it a locked down device like the Vonages ATA186? Adi

Hello - I've been using Broadvoice with Asterisk for a couple of months with no issues. Today, it has stopped registering. The Sip Debug from CLI is below. It tries to register five times and then gives up. Any suggestions? As you might suspect, I have not been able to get Broadvoice on the phone and usually get cut off after being on hold about 5 minutes. Strategic portions of IP

Hi, i'm having problems with broadvoice incoming calls. I can perfectly place calls but my Asterisk Box is having problems when registering with the SIP Proxy. Sometimes it register and the call gets into asterisk, but without sound (seems to be NAT problems) and sometimes its not possible for asterisk to receive the calls. Everything was working great exactly for a month, but a week ago it

I have searched the list and cannot find a sip 400 solution posted that solves my problem. If anyone has any thoughts or suggestions on the following I would greatly appreciate it. I didn't have this error before Broadvoice made their changes this weekend. Now when I make a call it connects but, I cannot hear anything on the other end... The full message I have is: 8 headers, 0 lines

> > I've tried setting up my sip.conf in two ways: > > > ------------------------------------------------------ > register => [240xxxxxxx]:[my_password]@sip.broadvoice.com > > > [Broadvoice] > type=peer > username=[240xxxxxxx] > fromuser=[240xxxxxxx] > secret=[my_password] > host=sip.broadvoice.com > context=incoming >

I've been wondering for a long time why my reinvite option is not working with Broadvoice anymore. It happend during the time Broadvoice was having a lot of issues, so I decided to wait. Recently I decided to test the same sip.conf with another VSP (SIPphone) and it worked fine! No issues on the reinvite. Note: clients and server using ULAW (only), no NAT or firewalls, public ip address and

In a 1.2 release of asterisk, I've had no problem connecting to a Broadvoice SIP peer, to allow routing outgoing calls from Asterisk to Broadvoice. Now, with the same SIP configuration, I cannot establish the peer. I've enclosed a SIP log in the hope that someone can help me analyze this failure. I'd guess the issue is NAT related and wondering if someone can spot a problem in the

Hello, I'm using Asterisk@Home. I'm still new to Asterisk, and trying to grasp it all. I'm wanting to do a simple setup of One SIP provider (Broadvoice) and 3 SIP Software Phones. I'm able to call the VoIP provided line fine and get dropped to the digital receptionist (or mailbox). However, when I try to send outbound calls I get "Error 503 Service Unavailable" and