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hi, i'm having problems getting asterisk spliced between an E1 PRI (german Telco Arcor) and an Ericsson Business Phone 250 digital PBX. The Asterisk Server has a TE405P with it's port 1 connected to the E1 PRI provided by our telecommunications provider Arcor and port 2 connected to the E1 PRI of our Ericsson BP250. the setup before: Arcor TelCo PRI(E1) <------------------------> Ericsso...

Hello, What is the set of configuration that should be done in the Asterisk 1.0.8 using FreePBX that can allow a simple video call between two extensions? Thanks in advance. [http://www.ericsson.com/shared/images/Email_line.gif] JULIO ARAUJO TE ENGINEER MS Ericsson ITTE & Test Environment S?o Jose dos Campos, Brazil Phone +551239084121 SMS/MMS +551281150089 julio.araujo at ericsson.com<mailto:julio.araujo at ericsson.com> www.ericsson.com<http://www.ericsson.com> [http:/...

Hi all, I've got an Asterisk box with 1 Sangoma A102 and 1 Ericsson PBX. I need to use Asterisk as E1 line for the Ericsson PBX. How do I have to connect them? I'm trying to connect the Sangoma to the Ericsson, but RED alarms remain. Any suggestions? Thanks -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://l...

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 proto...

Hello, I am trying to install my E1 card to make a conection with an Ericsson MD-110 PBX. I installed dahdi drivers as: dahdi_hardware pci:0000:04:08.0 wcte12xp- d161:8000 Wildcard TE121 ran dahdi_genconf and it created all my e1 ports. On the other side i also configured the pbx to communicate with TE121. On ericsson side, i have no error messages. On asterisk side,...

...ZE 16750 INTERNALDATE "22-May-2020 05:02:34 +0000" ENVELOPE ("Fri, 22 May 2020 03:46:54 +0000" "RE: Http2 tomact server taking time in responding when 1st StreamId is a large integer value like 2147483641" (("Prateek Kohli" NIL "prateek.kohli" "ericsson.com.INVALID")) (("Prateek Kohli" NIL "prateek.kohli" "ericsson.com.INVALID")) (("Tomcat Users List" NIL "users" "tomcat.apache.org")) (("Tomcat Users List" NIL "users" "tomcat.apache.org")) NIL NIL "...

Hi. I want to create a full-virtualization xen VM with autoyast installation. how can i do that? and how does the configuration file to be configured? I use suse linux 11 SP2. looking forward your replay. Jing. -- View this message in context: http://xen.1045712.n5.nabble.com/how-to-create-a-full-virtualization-xen-VM-with-autoyast-tp5714901.html Sent from the Xen - User mailing

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 phones through this gateway setup. The problem i have is with transfering calls. I can do supervised and unsupervised transfers from/to any of the IP phones behind Asteri...

Hello, Our asterisk is connected to an ericsson pbx by PRI. What i want is the asterisk clients should call operator numbers by dialing 0 But, when a call is made to ericsson via number 0, it assumes that the call is made from outside, so it doesnt allow to be dialed. There are 3 real operator extensions which is grouped by ericsson for operato...

I am trying to use TE110p card for Euro ISDN with Ericsson AMS switch. I consistently get one of the following errors: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 or PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 My zaptel.conf file: -------------------------------- span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadz...

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 Ericsson switc...

Hi List, This is somewhat off-topic since the problem itself isn?t asterisk but the Ericsson BP250 I want to connect to. But since there have been a couple of posts relating in part to that system I am hoping someone can help me out. What we want to do: PRI <---> BP250 <---> Asterisk Currently the BP250 handles extension 000 - 899. We now would like to forward a chunk of...

Hi, How does cpu and memory allocation works in KVM, If your total vm's on a certain hyp are alloted more RAM and CPU then total physically available CPU-Cores and RAM -- Ericsson Signature line *Mohit Kumar* * Back Office* Ericsson India Global Services Pvt. Ltd BUGS-GSC, Back Office A-8, Tower- A, 3rd Floor, Knowledge Boulevard, Sector-62, Noida, Uttar Pradesh-201301, India mohit.m.kumar at ericsson.com www.ericsson.com http://www.ericsson.com/current_campaign &l...

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 stops with app_dial.c:1780 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) My call rules are as DAHDI/g1/ Even i restart...

...atches replacing EVT with MVT at a number of places in TargetLowering. The last two patches are related cleanups in SelectionDAGBuilder. /Patrik Hägglund > git log --stat --reverse origin/master.. commit 8dabe3eb005360347eabb86a2e88c3b6e9098ed5 Author: Patrik Hägglund <patrik.h.hagglund at ericsson.com> Date: Tue Dec 4 10:37:37 2012 +0100 Change TargetLowering::getRegClassFor to take an MVT, instead of EVT. Accordingly, added helper funtions getSimpleValueType (in parallel to getValueType) in SDValue, SDNode, and TargetLowering. include/llvm/CodeGen/FunctionLoweringInfo.h...

Hello All I am wondering is someone knows how to configure the * to work with an Ericsson MD-110 with SL60 signaling?? through a TLU76 card. What is the right configuration in the zaptel.conf ? I currently have it configured as span=3.0.0,ccs,hdb3,crc4 but it doesn't detect anything when I connect it to the PBX and no activity can be seen either in the logs or in the asterisk c...

I'm haveing trouble connecting an Ericsson Webswitch 100 to asterisk. Has anyone gotten a Webswitch running? When I try to connect asterisk thinks everything works fine, while the webswitch just rings. I belive chan_h323 is picking the wrong port to talk at the webswitch on, however I'm not sure, nor am I sure how to fix it. Any clue...

...ry to do that next week. If you do not hear back from me by end of next, do not hesitate to ping me! Cheers, -Quentin > On Apr 22, 2015, at 9:32 AM, Quentin Colombet <qcolombet at apple.com> wrote: > >> >> On Apr 21, 2015, at 11:49 PM, Jonas Paulsson <jonas.paulsson at ericsson.com <mailto:jonas.paulsson at ericsson.com>> wrote: >> >> I looked at SplitKit, but I am not sure how to best do it, so it would be great if you could take a look. > > Sure. > I will try to do that by the end of the week. > > Cheers, > Q. > >> &g...

Does anyone have info on using the Ericsson Webswitch 100 Model G4 with *? The documentation and brochures make it sound like it's some kind of specialized FXO device purely to provide legacy PBX with access to VOIP, and not a general purpose FXO-VOIP gateway. Is it possible to use it to route incoming PSTN calls into * with full ca...

My asterisk server has an external IP. Also my Ericsson DRG22 has an external IP. Tried setting up the Ericsson DRG22 device. *SIP Server IP (primary): MYIPNUMBER User name: drg22-2 password: password Caller ID name: drg22 Telephone number: 12345 *Telephone domain name: ?? (I only have stighost.someotherdomain.com ) Do I have to run my own dns on...