[Asterisk-Users] Busy, differences between SIP and Zaptel(bristuff)

Hi all, a lot of my snoms are being called with this macro: [macro-ohne-AB] exten = s,1,DBget(temp=UML/${ARG1}) exten = s,2,Goto(default|${temp}|1) exten = s,3,Dial(${ARG2},600,g) exten = s,4,SetVar(PRI_CAUSE=17) exten = s,5,Hangup [default] ... exten = 77,1, Macro(ohne-AB,77,SIP/snom8556)

Re: [Asterisk-Users] Attended call transfer

Does any one know if attended call transfer has been added into the STABLE release of asterisk yet? Any news? I am also looking for #-Transfers for asterisk-stable. Thomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Asterisk crashes on special Transfer with MGCP/ATA 186

Hi all, i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco ATA-186 3.1.1 atamgcp We are used to make an special ;) blind transfer like (Flash)Number(Hangup before anyone answers or ring). Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp If one

[Asterisk-Users] ZapRas problems

Hi I try to use zapras. I am using zaptel-bri-0.0.2 I compiled and patched pppd from ftp://ftp.digium.com/pub/zaptel/misc/ pppd is /usr/sbin/pppd Any idea whats going wrong here? Thomas -- Accepting call from '95' to '8526' on channel 1, span 1 -- AGI Script nuller.agi completed, returning

Re: [Asterisk-Users] Best Budgetone firmware?

Philipp von Klitzing wrote: Hi! I'm still running 1.0.3.81 because I read that once you move up to 1.0.4.x you can't go back again, and my experience isn't *that* crappy. You probably want to start with 1.0.4.26 although also 1.0.4.17 seems to have been relatively stable. Then there is

Re: [Asterisk-Users] Problems with ATA's locking up..

CW_ASN - Gus wrote: You must register with cisco in order to get ata image. I tried, but Cisco (Germany) has no idea how to do this... BTW, my ATAs sometimes cannot make calls. I first have to make a call to one ATA-Extension, wait for the Phone to ring, then i can make calls again. I am

Re: [Asterisk-Users] MWI message not seen on SNOM200

This has been covered before. I think the reason is that asterisk sends notifications from [EMAIL PROTECTED] and pressing the button dials that address. It's not fixed, yet. For this, i have an voicemail-extension for my snoms: exten = asterisk,1,GoTo(8518,1) exten = 8518,1,VoicemailMain2

Re: [Asterisk-Users] 3x AVM Fritz!Card PCI for a EuroISDN PBX.

David M. Wilson schrieb: Hi there! I'm currently considering various PBX solutions for our office telephone network, and would very much like to use Asterisk. Currently, my research is incomplete. I have been recommended to use the above cards, but it is unclear from my Googling whether my

Re: [Asterisk-Users] Call pickup (*8) on SIP devices.

WipeOut wrote: Ing. Angel Gomez Garcia wrote: WipeOut wrote: Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved ,

Re: [Asterisk-Users] [Asterisk-Dev] Cisco ATA 186 chan_mgcp Transfer problem

Florian Overkamp schrieb: Hey, if I press Flash asterisk gets the 'hf' event but does nothing. What gives ? :-) We can compare our ATA-configs, because transfering works fine with MGCP (SIP doesnt). By the way, I'd think maybe it's not actually transferring but rather 'bridging' through the

Re: [Asterisk-Users] Call transfer on ATA186

Hi all ATA-Users, after a lot of tests, i found the best (not complete working solution). If you use an an MGCP-Image then 1. CLIP-CallerID works fine (with one Phone Callername-transmission works too) 2. Blind transfer with # works fine 3. Attended transfer (Transfer with consultation?)

Re: [Asterisk-Users] mgcp problems

Pavel Zheltouhov wrote: When I connected over two mgcp channels and sending numerical indication to cisco ata it seems hangup one channel (receving ) and generate 'fast busy' tone. I hack chan_mgcp and my threewaycalling works ok! But why indications are sent after I press hookflash on

Re: [Asterisk-Users] three way calling and cisco ata 186

Pavel Zheltouhov wrote: Ok, if this is not working with sip or h.323, maybe it does with mgcp ? I tried to get ATA and Asterisk working with MGCP, but nothing worked! Any Howtos available about MGCP/ATA186/Asterisk? Thomas ___ Asterisk-Users mailing