similar to: H.323 trunk between MD110 and Asterisk

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

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

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, 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

I have a call coming in from Asterisk-A going to Asterisk-B where it's determined that the called party is in fact yet another number in Asterisk-A so a new call is created from B to A and the two calls bridged (by Asterisk) at Asterisk-B. Originating Caller ==> Asterisk-A ==> Asterisk-B ==> Asterisk-A Now, what happens is that in my case both A and B are on the same network

Good question. I'm coming to the conclusion that using plain UDP and "home-grown" packet construction for transmitting the speex data (with timestamp/sequence counter) and implementing jitter control on the receiver end is an adequate implementation for a VoIP application. Assuming of course that I don't care about any interoperability issues with other applications etc. I was

Hi. I have installed B410P in Europe and the cards works more or less ok. My question is what color should the LED's on the back of the card be when connected to the PSTN NT box? Is there anywhere some information on the expected LED color in any given state (idle, call active, cord unplugged etc.)? On my card the lights are shining Red(orange-ish) but flashing to green every now and

Greetings. I wonder what is the best way in your opinion to share real-time extension state with applications outside of asterisk? What I'm after is the best way to have Asterisk update a central repository with the state of each extension configured in the local Asterisk setup. To try and explain what I am trying to achieve, Imagine for example if asterisk would call a url like this:

Hi. I'd like to get access to the CDR's generated by Asterisk (1.4) in real-time from a remote connection coming in on TCP. Basically what I have is a Windows application that is used to process incoming, outgoing and missed call records putting them into a database for some analysing etc. This app can connect to a TCP server and read from this connection the CDR's as they are

Hi, i managed to connect to Ericsson MD110 with PRI at last. And made a successful call thru asterisk to ericsson. But when i try to call from ericsson to asterisk i got an error on asterisk side. And i couldnt figure out why. Here's my extensions.conf about incoming calls. [DID_span_1] include = DID_span_1_timeinterval_all,${timeinterval_all} DID_span_1_timeinterval_all] exten =

Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to

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 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

Hi all I have a rather odd problem that I hope somebody can shed some light on. I have a Asterisk server (1.0.7) that is connected to a Cisco 2600 router (c2600-is-mz.122-28a) fitted with an E1 card. The E1 is configured for QSig and is conencted to our local Ericsson MD110 which goes out to the PSTN. We have a DID range assigned to the E1 and can make and receive calls from/to SIP attached

Hello, I'm trying to create a H.323 trunk between Asterisk 11 and Avaya. I have done this before between Asterisk 1.6 and Avaya but had some issues placing external calls from the Asterisk to the Public network which is connected to Avaya. I'm trying to create that trunk on Asterisk 11 because the 1.6 is outdated and has no support. On the Asterisk side I have Aastra 6731i SIP phones

I've looked around and not found details on the expected frequency range the Speex codec can be expected to carry. Is there any documentation available or a table of some sort that has been compiled which would give an indication of the frequency range based on the various compression options in speex? Best regards, Baldvin Hansson Reykjavik, Iceland baldvin@baldvin.com -------------- next

Hi, I'm trying to configure Asterisk as SIP-H.323 Gateway via ooh323, but I have an error relatively to the GK Confirmation message. >From the log: "H323 RAS channel creation - succesful Sent GRQ message Gatekeeper Confirmed (GCF) message received ERROR:No Gatekeeper ID present in received GKconfirmed message Ignoring message and will retransmit GRQ after timeout Error: Failed to

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

Hi. Using Asterisk 1.4.13 running on Ubuntu 7.04 with Intel CPU: 1) Is it possible/supported to install two or more B410P Digium cards in one computer (single Asterisk installation)? 2) Do they need to be hard-wired together with a PCM cable like I've seen explained in some beronet manuals (although that was specifically geared towards their cards, I must say)? Thank you for your time and