similar to: Forwarding issue.

I just tried to go from CAS to PRI on my T1 (Sangoma), and failed pretty badly. Seemingly everything worked -- Asterisk would see the incoming call (including CID and DID info), try to route it, and fail -- giving me a telco (not Asterisk) call failure message. My zapata.conf and zaptel.conf files are at http://pastebin.com/558349 Below's the log dump. Note that, because I was simply going

Hopefully I'm not just missing some little detail here. We're trying to set the HANGUPCAUSE on SIP channels to have our softswitch play the proper recording instead of answering the call on Asterisk to play the message. It appears that no matter what the HANGUPCAUSE is set to, Asterisk always just sends "603 Declined". I looked through the source code briefly and it appears

I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 320893f1-50c13ba3-78c26164 at 192.168.1.54 for seqno 2

I have a small system, server, client and 2 phones. Phones are polycom 501's. In general all is working fine. I can call the two phones, speak etc... I can have the server call each phone and play a wave file. However, when trying to setup a direct dial number of 1044 that calls another machine running asterisk - something ODD is happening. ; This is not working.... [smvoice-sip] exten

Is there a way to configure a friend in sip.conf that allows a station to register using a username other than the [name]? I want to have something like this in sip.conf: [1234] username=something_really_long_and_random secret=something_else_really_long_and_random ... Then allow a SIP REGISTER like so: REGISTER sip:10.0.0.200:5060 SIP/2.0 Via: SIP/2.0/UDP

Hi all, I just configures my PRI and incoming calls are working fine but outside calling giving error PRI is BUSY :( any idea ? I have same setup on other box and that boxes works perfect. -- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-00000002 -- DAHDI/i1/6463279153-2 is making progress passing it to SIP/7328-00000002 -- DAHDI/i1/6463279153-2 is busy -- Hungup

I have the following situation: Asterisk 1.2.1 25+ Polycom 501 telephones. Bootrom 2.6.2.0032 Application 1.6.2.0041 Some 501's local to my network, some across the great INTERNET divide. PRI connected to Sangoma card. Issue: horrible echo (and squeals, and "underwater-like" sound) on speaker phone when calling from extension to extension. echo not present when calling outbound

Hi, I've recently bought new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 2.0.1. I just noticed something, which I first blamed on Asterisk and NATs (a 2 second silence at the beginning of a call). Something I've noticed also on my old phone (which is having the same problem now, but its also been upgraded). My keys are sticky. Simple as that. Sometimes I press a number

Hi List... I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, and it also has the echo canceller... I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 2.6.9-34.0.2.EL I'm using Polycom's 501 with the SIP 1.6.2.0041 The problem is when someone dials to or from the PSTN through the TDM2400, the voice quality is crappy...Instead of hearing:

Hi, all. Every now and then, some of my users get "Error" on their phones. A reboot fixes it, but it's quite annoying/inconvenient. I'm running Asterisk 1.2.4, and have the following firmware, etc.: Bootrom: 2.6.2.0032 BootBlock: 2.5.0(11500_030) SIP application: 1.6.2.0041 Any ideas as to why this might be happening? Thanks! -Ken

While I know this is not a true asterisk problem, I figure someone where may know. When you click on Messages and it gives you the count of Urgent, New, etc. How can you make the phone gather that information? For example, my phone shows me there is an e-mail. It also sends an e-mail. Yet, when I click on message before I connect to the contact center, it doesn't have any counts. Here is

I have just upgraded my Polycom 501's from 1.6.2.0041 to 2.1.0.2708 to get the microbrowser. Almost everything is fine except when receiving calls from a BT200 (1.1.14 and earlier) the Polycom rings but when answered, drops out and the BT200 gets a busy tone. I have many PAP2T's and SPA3000's etc and they all cal call the Polycom without problem. Does anyone know what could be going

I'm stumped on this one and any help would be greatly appreciated. I'm just trying to get my Polycom 601 to have multiple extensions on it. For example, on line 1 I want extension 21, on line 2 I want extension 22, and on line 3 I want extension 23. Ideally I'd actually have each extension appear on 2 lines and therefore filling up all 6. I should be able to do that with the

Once upon a time, Stephan Bosch <stephan at rename-it.nl> said: > On 5/15/2015 5:56 PM, Chris Adams wrote: > > Once upon a time, Stephan Bosch <stephan at rename-it.nl> said: > >> You can check the handling of a particular message yourself using the > >> sieve-test tool (there is a man page for it). By specifying the `-t - > >> -Tlevel=matching`

Once upon a time, Chris Adams <cma at cmadams.net> said: > I can confirm that a message with multiple Subject: and multiple From: > headers does not get filed correctly into the Spam folder. The > sieve-test tools shows the correct action, but when the message comes in > via LMTP, it goes into INBOX. Okay, digging some more, it looks like something in sieve is overwriting the

On Thu, January 8, 2015 10:44 am, Les Mikesell wrote: > On Thu, Jan 8, 2015 at 9:48 AM, James B. Byrne <byrnejb at harte-lyne.ca> > wrote: >> >> A perusal of the contents of both the Fedora devel list and users list >> does not give one much hope that such a point of view would be >> tolerated, much less welcomed. > > Exactly. They don't care about

Estimada comunidad, para mi trabajo uso latex y R normalmente, ahora debo viajar sin mi portatil, pero tengo la opcion de llevar un pequeño tablet (con android) ... para suplir latex he estado usando www.overleaf.com y trabaja excelente, practicamente todos los paquetes que uso estan disponibles ahi ... pero no he encontrado algo similar para R. Saben ustedes si existe algun proyecto que