search for: evision

Im glad to let you know that finally I invested some time to make work Unicall in Asterisk 1.4, I must say not much testing could be done since I have no hardware available ( cards, servers ), however a friend was able to test it with a couple of calls with success, I need you to test this and report some feedback. The sources are available in: http://moy.ivsol.net/unicall/soft-switch/r1b1/

...ng (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.3.4.98:5060;branch=z9hG4bK102F7CD4855F4C4E927C3398E3C57BF4 From: Tobias <sip:tobias@10.3.1.6>;tag=2760968676 To: <sip:2@10.3.1.6>;tag=as396962de Call-ID: 79A5523F-AFCA-4DBE-9AA2-F51377E8B5AE@10.3.4.98 CSeq: 58303 INVITE User-Agent: evision PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:2@10.3.1.6> Content-Length: 0 Perhaps anyone can point me to the right direction ?? Tobias

I am trying to maximise a complicated loglikelihood function with the "optim" command. Is there some way to get to know the estiamtes at each iteration? When I put "control=list(trace=TRUE)" as an option in "optim", I just got the initial and final values of the loglikelihood, number of iterations and whether the routine has converged or not. I need to know the

I am looking for completly wireless desktop phones. Until I realized we needed wireless i was going to use polycom soundpoint 501's. Any suggestions on a comparable wireless phone? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070328/88e22671/attachment.htm

Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because TDMoE module in asterisk is not so stable and TDMoE is stale. And It seems that RedFone doesn't not support load

THE SCHEMA IN THE MODEL a small write up on ''putting the schema in the model'' This is a write up on an issue best covered in a mailing list thread of Januari 2006 (see the links in the text), I repost it because I think it deserves a place on the agenda. == Why? == I was switching back and forward between the model files and the schema.rb -- off course I have

Hi, i was successful in compiling app_conference and setting up an conference was quite easy. :-) Does anyone knows if it is possible to have an IVR accessable from inside the conference. So, if i dialed into an conference i want to be able to press '*' and then the actual discussion is muted for me and i and menu is read to me. Something like the ${MEETME_AGI_BACKGROUND} in MeetMe.

Hi, today i have taken a strong look at meetme.c what i am trying to accomplish is the following: it should be possible to access an menu from within the conference in order to perform special tasks, eg. to dial another number so that the called person is joined with the conderence. my first try was to use an agi-script for this, but as with agi enabled sip-channels (for which

Hello all, i am using * 1.0.9 and chan_capi-cm 0.5.4. So far everything works good. But if i want to initiate an call over an capi channel either from the manager api or from an call file * quits with an segfault. If i do the same thing over sip channels everything works perfectly. Had anyone the same problem? I am using Linux Kernel 2.4.27 and an Fritz!PCI v2.0 Card. Thx in advance :)

Hi, the following szenario leads to a problem: I connect an CAPI channel to an AGI-Script per Manager API. This Agi script starts the MeetMe-Application. The Person on the Capi Channel is now able to speak with the other conferess in the MeetMe-Room. But if the CAPI channel hangs up, the busy tone is streamed into the MeetMe Room for several seconds, until the CAPI HANGUP-Signal is finally

Hello, i've got the following setup: exten => _X.,1,Dial(Zap/${EXTEN},15,T) exten => 9000,1,AGI(agi://localhost/myagi.agi); Now i want to do the follwing. With the catchall extension i make an outbound call to another person. This Person will then get transfered to extension 9000 and will be connected to the AGI-Script. So far this works fine. But the AGI Script has to know with

Hi, i've been trying to connect Asterisk with Google Talk such as some others have tried. Therefor i followed the instructions on http://www.voip-info.org/wiki/view/Asterisk+Google+Talk I took the latest version of asterisk from trunk. My Asterisk server is not NATed but the Google Talk Client is. Signalling a call is no problem, but after the call is set up, no audio is passed. I can see a

Hi all, just for learning purposes i made a little gui frontend that visualizes incoming and outgoing calls in realtime, using the events of asterisk. I experienced a strange behaviour for outgoing calls. The callerid for the *called* person got changed to one of my own numbers, after the channels git linked. After looking into the flow of events i saw that asterisk keeps sending an

Hello all, recently i stumbled upon the I-TDM standard, e.g. see here http://www.picmg.org/v2internal/news2005.htm "SFP.1, also known as I-TDM (Internal TDM), is a companion protocol specification to SFP.0 that is optimized for TDM traffic over high-speed fabrics such as 1 and 10 Gigabit Ethernet (PICMG 3.1), Advanced Switching (PICMG 3.4), Infiniband (PICMG 3.2), etc. SFP.1 and SFP.0

Confirmed in squeeze. Almost two years old. Thanks for the fix, works well. -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4371 bytes Desc: not available URL: <http://lists.alioth.debian.org/pipermail/pkg-xen-devel/attachments/20120321/ea6d5566/attachment.bin>

I had another look at this and found the source of the problem - the reason why XEN_SCRIPT_DIR is not set: The hotplugpath.sh file is missing in the package. In the "rules" file the "install" target includes this line: $(MAKE) -C $(BUILD_DIR)/tools/hotplug/Linux install-udev install-scripts UDEV_RULES_DIR=/lib/udev/rules.d The problem is that make is never run in the

Hi, I have some trouble with setting my CallerID if i make an international Call. No Problems with National Calls, i can set whatever I want. We pay for this service but our telephone provider was not able to state clear, wether the number we set on an international call should be shown on the other side. Actually only our base number shows up. If I understand it correctly, in every call the

Hi, i've been discovering app_sms and it states that it can act as an smsc for landline sms. Receiving SMS from my Gigaset Phones is no problem and the SMS are stored as files on by * box. So far so good. Let us assume that i have a couple of phones which should be able to receive SMS directly from my * box ( and not from an SMSC from BT or Deutsche Telekom ), So all these phones have

Hi, i've been trying to get fixed line sms working for some time now. Can anybody tell me, if he is actualy using this with asterisk in germany? I have followed the instructions found on voip-info. I was successful a couple of years ago with asterisk 1.0.7 and an normal telekom isdn line. Now i want fixed line sms over an Dokom PRI with Asterisk 1.2.9. Here in Germany the Materna

Hi, we are searching for wireless IP Phones (DECT preferred) with have an solution for an external telephone book. We don't want to enter all of our numbers into every telephone, but have one location for all the numbers and every phone looks them up there, e.g. an ldap server. We have tried Kirk but they are working on an solution without any information when it will be available. Does