similar to: I give up!!

I have an asterisk installation at a client, it's quite simple. Basically it's an asterisk downloaded from CVS about a week ago, with 3 Zaptel FXO cards (the digium ones) and 10 Grandstream Budgettone SIP phones ... Every now and then, especially when a call is ringing and not picked up immediately, Asterisk quits with a segmentation fault error. IT seems quite inexplicable, my dialplan

hi there .. I asked sometime ago regarding getting a Budgettone working with Asterisk over G729. My system is quite simple, Asterisk server with 1 G 729 license installed, and 10 Grandstream phones. Only one of them needs G729, because it's on a remote link via an ADSL bridge. The rest run happily on G711 on a local network. I added the lines disallow=all allow=g729 to the sip.conf entry

well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my

just three "stupid" questions I need to ask .. 1. what's the sequence to press on a SIP phone to transfer a call to another extension. 2. what's the same thing if you want to hold an incoming call, speak to the other extension, then pass the call? 3. what's the extensions.conf syntax to dial two SIP extensions at once? many thanks Dave

hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

Dear all, I was testing the wonderful package APE. However upon testing a particular Newick's format tree - which I think to be a non-binary tree - it yields different result as expected. > library(ape) > tree.hiv <- read.tree(text="(rat,mouse,(human,chimp));") > is.binary.tree(tree.hiv) [1] TRUE Was that a bug in APE package? - Gundala Viswanath Jakarta - Indonesia

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

Hi, i have one firewall/gateway server with two interfaces and a routing problem (?). eth0: external interface eth1: internal interface. Both ip address are valid. Services like DNS, HTTP is configured to run using eth1 ip address. The problem is when i try to connect from internet to firewall, i can´t see eth1 ip address... only eth0 ip address. So, when i try to connect to web

hi, can someone who has used Budgettone phones tell me how to do the following: an incoming call comes in and is answered by the receptionist. she need to put the call on hold, speak to whoever the call is for, and either (after that) pass on the call, otherwise speak again to whoever was on the call and hang up .. so far i've got as far as a blind transfer by pressing transfer button and

Hi everybody, I would like to be awareabout what happened to me. Two weeks ago, on a Sunday morning a French guy called me. Ask me to fix some problems with his asterisk. After fixing his problem, he asked more and more, after 10 hours of work I ask him to pay me for the first milestone. However, lucky me that I did not finish, since he never paid me. Be afraid and take your action if some

Hi people, What are the pros and cons of Winbind, LDAP and Kerberos in a Samba 3.0.2 plus Active Directory environment ? What could be the best design for this scenario ? Estevam Henrique ========================================================= Esta mensagem pode conter informacao confidencial e/ou privilegiada. Se voce nao for o destinatario ou a pessoa autorizada a receber

hi, all asterisk 1.4.24 , zaptel 1.4.10.1 , E1 Manager API Action : Action: Originate Channel: ZAP/G1/8888888 Callerid: 12345678 Context: callout Exten: s Priority: 1 extensions.conf [callout] exten => s,1,Answer() exten => s,n,Wait(10) exten => s,n,Hangup() when the phone 8888888 pick up , it will come to callout context, after hangup, one cdr generate, but the

I upgraded from RC2 last night, but have a major call quality issue. Heres our setup: 1 FXS and 1 FXO card. Incoming/Outgoing calls via IAX trunking from our provider. G729 running between us and the VoIP provider. Two handsets, one BudgetTone 102 and a Cisco 7940G running the 7.2 SIP firmware. Both these phones are using ULAW to the server, and we have plenty of G729 licenses on the server.

Hi, It's unfortunate that with recent revisions of R 2.13 (this appeared in revision 54640, March 2), 'R CMD build' now removes empty dirs in the package. People might have good reasons for having empty dirs in their packages. For example, in Bioconductor, we have some tools to automatically generate annotation packages and those tools are implemented in software packages that use

Hello, I have read the information on echo and SIP in the FAQ and I have scoured the mailing list for possible solutions, but as yet I have not been able to get rid of this echo. I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed into an asterisk server. If I call between the Sip Phone (Budgettone-100) and the 4 FXS ports everything sounds great. If I call out to the PSTN

You should use the ntlm_auth module provided by samba. -----Mensagem original----- De: samba-bounces+ecarvalho=bmf.com.br@lists.samba.org [mailto:samba-bounces+ecarvalho=bmf.com.br@lists.samba.org] Em nome de Riccardo Baldanzi Enviada em: quinta-feira, 22 de abril de 2004 16:51 Para: samba@lists.samba.org Assunto: [Samba] Problems with ntlm_auth --helper-protocol=squid-2.5-ntlmssp Hi Guys,

On Tue, 2003-12-23 at 19:22, Brian West wrote: > I'm just double checking.. I was told it wasn't possible but i'm going to > ask just in case. > > Can you set outbound callerid on a channelized T1? > >I think there is a way to do something like DID with the 4 digits of >DTMF passed before the call. It is unlikely though that you will find >someone interested

Hi, i am recording a message, but it is always cut off at 40 secs. There are no time out configured. Gabriel -- The educated person is not the person who can answer the questions but the person who can question the answer.

Hi All, I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan: [callback-outbound] exten => _00.,1,Macro(callout|${EXTEN}) [macro-callout] exten => s,1,AGI(getchannel.php|${ARG1}) exten => s,2,Dial(Local/${OUTBOUND}@from-internal/nj||tr) exten => s,3,Hangup() but for some reason i am not receiving the argument: Executing [s at macro-callout:2]