similar to: Call quality issue across VPN-> POTS vs SIP

Hi All, I've looked through the archives and tried several variations in Google, and I haven't found anything on-point... So I'm hoping someone here may be able to help this relative Asterisk neophyte shed some light on an issue: I have a box running Asterisk 1.4.22 in our lab with several Cisco 7961G phones and an AEX804E card (4 FXO, hardware echo cancellation). The server and all

Hi All, We've been running Asterisk for years in our offices but just recently replaced an Asterisk Appliance* in our smaller office with an actual server, upgraded the server in hardware in our HQ location and upgrading both ends to 11.5.0 with Gareth's patch for Cisco phones. 99.99% of our endpoints are Cisco 7961Gs. Each office is more-or-less standalone for ease of management and

Hi All, In the midst of trying to pilot a deployment of Microsoft Lync (mainly for non-voice collaboration, specifically IM) and integrate it with our Asterisk (11.6.0 if it matters) deployment and a "everything in one place" tool when people are out of the office. I have everything on the voice side playing nice from the Lync side (Lync->Lync, Lync->Asterisk,

Hi All, Thanks for all of the help I've been given in the past and info I've picked up from this list over the years. I have an "official" Asterisk appliance (the AA50) running my PBX at home (we previously also had an AA50 in a satellite office-that one was recently retired and replaced with Asterisk running on commodity server hardware). Anyway - the AA50

I checked the app store and haven't found anything promising, but I figured I'd ask here. Does anyone know of a SIP or IAX client for a non-jailbroken iPhone that will communicate directly with a machine running Asterisk? I know that there's at least one offering that seems like it's essentially a proxy (App runs on iPhone, iPhone talks to 3rd party server, 3rd party server talks

Hi All, I have a ticket open with Digium, but based on their previous lack of support for the Asterisk Appliance, I'm not really holding my breath - and, honestly, I'm not 100% convinced it's a Digium issue in the first place (but I don't know where else to point fingers). We have an AEX-804E (PCI Express, 4 FXO ports, Hardware Echo Cancellation) in a Dell PowerEdge 1950 with

Hi All, Is there any way to provide the user receiving an attended transfer with a tone or other audible indication that the transfer is completed (i.e. Party A calls Party B, Party B announces the call while transferring to Party C, Party C hears tone when Party B completes the transfer so that they know that they are now talking to Party A instead of Party B)? I know this is possible when

Hi All, We're using Asterisk 1.4, and Cisco phones exclusively (mostly the 7961G, but a few 7911Gs and one 7912G for the time being-all running the SIP firmware image, plus a few analog extensions until the next capital funding cycle). Each user has a phone at his or her desk, but there are also a growing number of "common area" phones (hallway, kitchen, conference rooms, data

Hi,

Hello. I would need some help trying to setup Asterisk 1.6.2.9-2+squeeze1 on a Debian 6.0 system. I'd like to use the Debian packages, hence the "strange" version number? Since I'm new to Asterisk, I'm trying to follow "The Asterisk Book" at http://www.the-asterisk-book.com/unstable/minimale-telefonanlage.html and created a VERY basic sip.conf; see

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like

Hi All, I posted this a couple weeks ago with no response, I'm hoping that someone will see it this time around and be so kind as to offer advice for resolving this issue (or point me in the direction of a better place to ask) "Some" (but not all) calls on one of our Asterisk boxes are being dropped in Voicemail -- only in voicemail -- after about 20 seconds with the error logged

Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the "domain" field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk

i am looking at iax to see if it is applicable to my needs. i would appreciate any corrections of what i think i have understood but probably have not. iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. trunking will save some bytes in flight iff one has four or more streams moving between two

"Why Siphon doesn't allow to receive a call when it doesn't run Apple doesn't accept (for the moment) an application runs in the background= . So, when Siphon doesn't run, the SIP server of your provider doesn't know= your iPhone." --_000_EC80F07C30CE3E46B2AD6B4407BE086F0C2AAD0248cworksmailcwo_ Content-Type: text/html; charset="us-ascii"

Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the

"Why Siphon doesn't allow to receive a call when it doesn't run Apple doesn't accept (for the moment) an application runs in the background= . So, when Siphon doesn't run, the SIP server of your provider doesn't know your iPhone." --0015174c3c60a73ef5046656ca27 Content-Type: text/html; charset=windows-1252 Content-Transfer-Encoding: quoted-printable

Hello, I have connected asterisk box with legacy PBX Alcatel OmniPCX 4400 (and also another * box connected to A4200). These PBXes have function to assign name to extensions and display it on phone. Asterisk box is connected via PRI with euroISDN signalling (also I have tried QSIG). Is it possible to set callerid with name and display it on alcatel digital phones? With command SetCALLERID

Does anyone have a working example of how to use the "switch" directive to peer two Asterisk PBXes? -- - Ian C. Blenke <icblenke@nks.net> (This message bound by the following: http://www.nks.net/email_disclaimer.html)

Hi Well it is well some time that my last DUNDI peer has become unreachable. I guess too many issues with spoofed numbers etc. But I am wondering, do people, especially larger entities like telcos, still use DUNDI? I know that in some Hamradio communities, DUNDI is used to interconnect PBXes, but that is with private phone number ranges, not connected to the public. Want some DUNDI peering?