similar to: incoming call problem from pri

Hi, i know i am asking a lot of questions lately in this forum..sorry about that first of all. :) Ok, here is the deal.. I am trying to make a hybrid system with an ericsson MD110 and asterisk. Internally we have 4 digit phone extensions on ericsson.. and so in asterisk. So, what i want to do is to call pbx side without adding 9 or etc to the begining of the number from asterisk clients.. For

I've got an asterisk 1.4.24 box with dahdi complete 2.1.0.3+2.1.0.2 I've got a 2 port T2XXP card attached with on T1 currently plugged in. Inbound calls work fine, but outbound fail with the rather cryptic: [May 19 17:28:07] WARNING[11360]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) There are similar threads I've found on the

Hello, for a long time i am using asterisk 1.6 with astgui. but for production system i intend to use asterisk 1.4 which i think might be more robust. And for a more developed service options i preferd to install with freepbx. But still there are big plusses and minusses for both system. My complain about astgui+1.6 was.. For example there were no backup trunk config running on that version.Even

I was just wondering if someone has experiences to use Asterisk in an existing Ericsson MD110 environment. Particulary I'd like to know if it is possible to use the MD110's system phones directly connected to Asterisk. I'm not very familiar with it but would it be possible to use ADSI with these phones? Are they more like analog or more like digital phones or is the protocol even more

Hi all, I was using dahdi 1.6.2.0.9 version for a long time. We decided to upgrade to 1.6.2.22 a few days ago. After that we started to have some problems with dahdi channels. PS:DAHDI Version: 2.6.0 Echo Canceller: HWEC, MG2 We have 2 PRIs between Ericsson pbx and asterisk and a sip trunk for outside calls. At begining everything works fine but in a few hours, calls from asterisk to ericsson

Hi, Perhaps there's something wrong in my config... I did some tests connecting Asterisk to an Ericsson MD110 PBX by setting up an h323 trunk. When dialling into asterisk I got some problems when the entire number is not in the setup message, i.e. I'm dialling digit by digit on the ericsson phone. Lets say I have 4001 in my extensions, and dial that from the Ericsson PBX, then the

Hi. Calling from Ericsson MD110 via H.323 trunk to an asterisk 1.4.4 I get the call established but no sound heard on either end. What is the best/correct way to try and see what codecs Asterisk is using on each end of the call as it passes through Asterisk? And is there any way to see that voice is in fact being passed through Asterisk during the call (some counters etc.)? Thank you

Hello, I just tried to upgrade to 1.6.1.0 from 1.6.0.9 and i had problems in registering users. As i see from debug it successfully reads from users.conf but later,when a user tries to logon it say peer not found.... And there were an error msg about mysql about the username field..Smthing changed in mysql tables??? Now i downgraded to 1.6.0.9 again and everything is working..

Hello, I have a default installation of asterisk 1.6.1.9-2 When i create a user in users.conf via asterisk-gui, calls, voicemail etc works. But if i create a user realtime (and my realtime caching is available too) i can see the realtime user with sip show peers. But, my local dial rules does not work. I can call from realtime user to static users(the ones in users.conf) and if they are not

I have been asked by a potential customer whether we can connect an Asterisk box to an Ericsson MD110 that has an E1 port with ISDN-USR. They are unable or unwilling to upgrade their E1 port to QSIG. Has anyone here had experience of successfully making such a connection? I have found a couple of hits on Google that suggest it "should" work, but I'm after something a little more

Hello all, I was wondering if someone already wrote something to support a serial connection(ICU) on PABX's that's used for signaling. What I currently have is a connection between an Ericsson MD110 and * with analog trunks. Problem with this is, that all CallerID info is send over a serial link (ICU). Is there anyone who knows if there is support for this on * or to find the

--- span_1 = DAHDI/g1 1,1,dial(${span_1}/${EXTEN:0}) --- I can only presume some form of precedence overrides the group configuration in the recent asterisk installs and not for the servers set up earlier. On 26/5/09 4:01 PM, "Kal Feher" <kalman.feher at melbourneit.com.au> wrote: > Ok I've solved the problem. I do not think it was as switchtype issue after > all as

Hi guys! We have a call centre that has been moved across from an old Ericsson MD110 PABX to an Asterisk server with those in the call centre using X-Lite as their softphone. I'm trying to get Agent presence configured so that X-Lite gives the operators a visual indicator of their status - logged on, off and on "pause". I'm using chan_agent for the agents, so agents are

Hello all, as good? It would like to make a question, asterisk supports the protocol qsig, for interconnections in ISDN with equipment Siemens HiPath 4000 or same Ericsson MD110, so that it could identify to the name and the number of hosts and also to use some features of asterisk in the Siemens/Ericsson equipment. Greetings Josu? -------------- next part -------------- An HTML attachment was

Hi, I am having problems with an Asterisk with a Digium TE205P card. The issue is that the Ring Back Tone is noisy. I am making modem's calls and this noise influences on the initial negotiation protocol, so modems have to recall. My configuration is: Asterisk version: Asterisk 1.4.21.2 Linux version: CentOS release 5.2 (Final) Card: Digium TE205P

--- span_1 = DAHDI/g11 1,1,dial(${span_1}/${EXTEN:0}) --- The configuration was rsync'd from a working pair of asterisk servers in another office. The only difference was the version 1.4.22 for the original servers that were operating as expected, 1.4.24 and 1.4.24.1 for the new servers. Included in both working and non working servers is the following configuration:

Hi. Anyhone have any experience with trunking between Ericsson MD110 and Asterisk using H.323? I've tried both ooh323 and the /channels/h323 one in version 1.4.4 and 1.4.0 of Asterisk. ooh323 does not manage to establish the call (starts to ring but then disconnection when answering the call on the Asterisk end) but using the channels/h323 driver I can get the call established from

Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I already have t38pt_udptl=yes in sip.conf Thank you.

Hello, I am trying to create a simple IVR for testing.. What i did is to create a voice file from asterisk-gui. And i saw it created that under /var/lib/asterisk/sounds/record/ as deneme.gsm Then i tried to make a IVR menu and play that file. I tried exten=s,4,Playback(/record/deneme.gsm) exten=s,4,Playback(record/deneme.gsm) exten=s,4,Playback(deneme.gsm)

Hi, Previously i was using asterisk 1.4 with freepbx installation. To try the 1.6 version i installd anc configured everything.. Just one thing didnt work so far.. I am using grandstream 2000 and it has a line busy indicator for chef secretary phones. But now, this feature does not work. I can see the line is online..with a green steady light.. But when the line is busy or DND, it wont change to