similar to: Problem in receiving calls from E1

I've done a decent size update of the OpenSSH portable from the upstream tree. There is an AES upgrade that needs to take place, but I need people to test and tell me what endedness changes need to be applied (if any). Attached is the patch. It is geared towards the latter snapshots..Unsure how it will patch (if at all) against 2.9pX series. Thanks. - Ben "This is dark day in

Hello, We are having issues with a NEW Sangoma A108D: -- Executing [691918892 at pbx1:1] Dial("SIP/xtravoip200-009d24b0", "DAHDI/g0/691918892|30|m") in new stack [Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 [Oct 1 10:04:43] WARNING[14171]: app_dial.c:1237 dial_exec_full: Unable to create

Hi, I'm using Sangoma's wanpipe together with dahdi, all software downloaded today at most recent version. Hardware is Sangoma A104, a 4xE1 card. Installation went well. Anyway, wanrouter status shows a different result than dahdi_tool or dahdi_scan. I've just put a hardware loop on port 1. All the other ports are open. wanrouter status shows the expected result: Device name |

Dear Rxperts! A simple example where "write.csv" does not seem to accept user specified arguments.. Why? write.csv(t(1:10),"./te1.csv",quo=F,col.names=F) Warning message: In write.csv(t(1:10), "./te1.csv", quo = F, col.names = F) : attempt to set 'col.names' ignored However, write.table does fine..

Hi everyone, I am reading through Sangoma Wiki right now. But someone may already and quickly notice this. I have a system that is down since the morning (maybe power intruptions). All seems fine except for "wanrouter status" shows disconnected. Following are the alarms raised. Should I call telco (they have long wait times) or should I just keep searching online for troubleshooting

Has anyone seen something like this before. Randomly, on longish calls, the local side of the call audio goes dead. Meaning remote caller can hear us but we cannot hear the remote person? Linux voip 2.6.18-128.1.6.el5 #1 SMP Wed Apr 1 09:10:25 EDT 2009 x86_64 x86_64 x86_64 GNU/Linux Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. WANPIPE Release: 3.4.1 Wanpipe Config:

I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start: Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:5976 mkintf: Unable to get parameters Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:9478 setup_zap: Unable to register channel '1-15' Aug 5 10:47:29 WARNING[1076842624]: loader.c:328 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered

Hi list, I have search for a clear explanation about this mensage " WARNING T.30 ECM carrier not found", but until now I dont succed on it.Anybody know how can I handle with this problem? I have Asterisk 1.6.2.13 with TDM 410p and the Fax is connected to Dlink FXO dvg 2032s. Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda

Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5 -- Executing [21341400 at local:1] Dial("SIP/4804-00000000", "DAHDI/g11/21341400,,t") in

Hi , Is it possible to have two meetme room in asterisk 1.6 which each one have a different language? I mean, one room the annoucement is in Portuguese an another in english? Today I can change over the sip.conf and it is valid for all room. regards! Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part -------------- An

Hey there, I am trying to setup a2billing on asterisk 1.6 , but ,when I try to access its web page I see the a2billing directories:Index of /a2billingNameLast modifiedSizeDescriptionParent Directory -admin/15-Sep-2010 19:19-agent/15-Sep-2010 19:21-common/15-Sep-2010 19:18-customer/15-Sep-2010 19:20-Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny8 with Suhosin-Patch Server at Att, Flavio Roberto

Hi all, I am trying to set up Web-meetme in Asterisk 1.6. After some attemps, I am receiving the message:DB Error: connect failed What could be ? Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, What is the easier way to make call using a password? I have A2billing but its authentication is too big, I would like four digits long. Something like that: In any extensons, the user dial the password and make call. Thanks in advanced! Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part -------------- An HTML

Hi all, What hell hapen here? asterisk:/etc/asterisk# /etc/init.d/dahdi startLoading DAHDI hardware modules:FATAL: Error inserting dahdi (/lib/modules/2.6.26-2-686/dahdi/dahdi.ko): Device or resource busy wct4xxp: done wcte12xp: done wct1xxp: done wcte11xp: done wctdm24xxp: done wcfxo: done wctdm: done wcb4xxp: error wctc4xxp: done xpp_usb: doneError: missing /dev/dahdi! When

Hi all, After a lot of trouble with a TE110p working with mfcr2 , brazil variant, everything looks great,but I can not go out of my calls. When I try I receive the following log: == Using SIP RTP CoS mark 5 -- Executing [33220567 at local:1] Dial("SIP/4804-0000001a", "DAHDI/g11/33220567,,T") in new stack == Everyone is busy/congested at this time (1:0/1/0) -- Auto

Hello everybody, Anybody here knows about Astmail ? I have set up in a server but something is going wrong! i can open its web interface but when I put the extension number and its password I receive:Invalid mailbox or password My asterisk is 1.6 and my S.O is debian lenny. I know this is not asterisk but as the project has not a forum or mail list, I am trying help here! Thanks in advanced!

Hi there! Anybody knows why I am receiving this output from CLI: No such command 'dialplan reload' (type 'core show help dialplan reload' for other possible commands) Look like asterisk dont see dialplan? Is it possible to restart it ? Thansk Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part

Hi all, There is a feature very common in PBX called PADLOCK , and I'd like to set up it on Asterisk 1.6. I have seen it in the internet but such scripts never work to me. I am trying to do something like that:Create a password and associate it with the callerid: exten => _*11*xxxx,1,Set(DB(CADASTRA/${CALLERID(num)})=${EXTEN:4}) Create a flag in order to verify later if PADLOCK is ON

Hi there! I am trying to configure Web-meetme on Asterisk 1.6. I have followed the README and everything looks ok,therefore, when I try to open the webpage appear the folowing messages:DB Error: connect failed Testing with a php script, the message Connected successfully is shown. Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda

Hi all, I know Hylafax is an application and not Asterisk but I'd like to post a problem found in configuring such application and Asterisk. I am able to reveive fax,but , I can't receive it in e-mail. Although I put my e-mail in /etc/hylifax/Dispatch I can't receive. Anybody know where I must to add something else in order to make it works! Thanks in advanced!! Att,