Displaying 20 results from an estimated 3000 matches similar to: "Hookflash woes"
2003 Apr 14
2
Weirdness on "hookflash call pickup"
I'm sure dumb when it comes to describing things that happen on my system.
I'm making an outbound call on my ATA186 when another call comes in. I
first get the nasty CID screech and then the periodic call-waiting tone.
So far, so good.
Then I hookflash, and just like it's supposed to, the waiting caller is
on the line.
But during the duration of that conversation, my console
2004 May 17
0
Zap callwaiting hookflash idiosyncracy/flaw?
Don't know what else to call this. Googling and some time on the IRC
channel haven't gotten me anywhere.
Here's the sitch, which is a bit complicated but is something my
customers are in fact encountering on an everyday basis:
1. Bob is on a Zap channel talking through the PSTN to Carol. Both have
the misfortune, like so many of us, of having LECs who do not offer
disconnect
2005 Sep 01
0
Micronet 5050s FXO gateway and hookflash transfers.
Hi,
Has anyone out there managed to do a hookflash transfer with a Micronet
5050s gateway ?
We have just tried out this gateway and it seems to do everything we need
except this
particular feature. Also if you have succeeded where is the hookflash time
specified in the
gateway. There does not appear to be any parameter for this. Perhaps it is
not supported at
all.
Any help appreciated.
2004 Dec 27
1
transfer: hookflash vs #
I think I've managed to figure out that there are two ways to transfer a Zap
call, using hookflash (defined in zapata.conf) or the # key (the t and T
options of the Dial command in the dialplan), but not why there are two ways
to do this, nor what the difference is between them. Is there something
that explains this?
thanks
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2004 Dec 27
0
Fw: Hookflash timing with TDM400P
Hi all,
Is there a way to change the hookflash timing with the TDM400P?
Allready been searching the mailing list/google etc but i can't find
anything ;-(
I tried flash= in zapata.conf, but that only works with the T1/E1 cards.
Greetz,
Caspar
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2006 Jan 06
2
SPA-3000 is translating vocal sounds into DTMF
I'm sure there must be a setting I'm missing somewhere, so I thought I
might was well ask here.
Conversations are punctuated by sudden replacement of a given syllable
or so of conversation with a DTMF tone.
I would hope perhaps there's some kind of setting that has to do with
the way it detects inband DTMF? I'm pretty sure it's an artifact of
this particular ATA; my
2010 Oct 30
2
Exceptionally long queue length queuing . . . .
I wonder if anyone out there has a perspective on this. There are a
welter of tickets out there on the matter, most of them closed.
This problem began for me over a year ago, and continues up to the
latest versions I've installed (1.6.2.13).
It happens randomly, and the suggestion on one of the bug tracker
tickets that it is instigated by a small network leg looks to be on
point to me,
2003 Jul 27
3
Nortel 350
Wondering, since they appear to be plentiful on eBay, whether I could
get a Nortel 350 to use to learn my way around ADSI.
The vendor claims that these are "generic," and looking through the
archives I wonder if that means that they might be unlocked in the sense
that the word is meaningful to asterisk.
Of course I am green as could be on this topic, so this question may
even be a
2005 Feb 21
1
Problems with the FXS module in a TDMxxx card (no sound when receiving a call)
Hi all,
I have a brand new TDMxxx card with 3 FXO modules and one FXS.
It has replaced my old 3 X100P cards.
The FXO part work as before, after some adjustments in the rxgain/txgain
part.
The problem I have is with the FXS module.
I can place calls to SIP/IAX or PSTN destinations without any problems, but
the sound received by the other part is much to strong and a little bit
distorted.
I have
2005 Feb 21
0
Problems with the FXS module in a TDMxxx card (no sound when receiving a call
Hi all,
I have a brand new TDMxxx card with 3 FXO modules and one FXS.
It has replaced my old 3 X100P cards.
The FXO part work as before, after some adjustments in the rxgain/txgain
part.
The problem I have is with the FXS module.
I can place calls to SIP/IAX or PSTN destinations without any problems, but
the sound received by the other part is much to strong and a little bit
distorted.
I have
2006 May 29
4
app_conference DTMFs?
We need to conference together a call center agent, a customer and a third
party IVR and send DTMF tones from the agent to the IVR.
MeetMe has been eating our DTMFs so we'd like to try app_conference.
Has anybody setup such a configuration in app_conference and how did you
configure it?
-HJC
2003 Jul 11
2
Hook Flash INFO messages
Here is a question that needs a few opinions...
Recently we installed a couple of FXS gateways into a site with a SIP Proxy/Softswitch other than Asterisk. One of the things that the users on this site need to do is receive calls on single line phones on the FXS gateways and then hookflash and transfer them to other SIP users.
We found that the FXS units, true to their nature as VoIP gateways,
2006 Jun 16
17
Voicemail with NFS
I have /var/spool/asterisk/voicemail NFS mounted from another server. Everything is fine, until I simulate an NFS server failure, by shutting down the NFS server process.
At this point, Asterisk becomes almost non-responsive. It won't even process a 'sip show peers' command correctly. It displays a few lines of text, pauses for several seconds, and then displays the rest. When a call
2008 Jun 17
1
looking for help / input with Blind transfer from asterisk to zap
List,
Having a little trouble with the following. Let me prefix by saying I
have blind transfers working from the following setup.
Inbound call [from-zap] (SIP/sv0071iv) answers.
Zaptel -> Asterisk -> SIP extension
SIP extension then blind transfers [from-sip]
---
SIP extension -> Asterisk -> Zaptel
During this whole process, the original channel off the trunk
(lineside T1) is
2010 Feb 20
2
Sending a hook flash to a DAHDI channel
I've got a piece of CPE equipment that has an FXS port that I have tied
to an FXO port on a TDM400 clone card. Normally, if I go off-hook with a
standard telephone connected to it, I get a dialtone. If I dial a digit,
and send a hookflash, the device will provide a dialtone back for the
next available channel on the device.
I'm trying to recreate this same behavior with Asterisk,
2003 Dec 03
2
Cisco IAD with MGCP
I repost a message I put a week ago:
I have a Cisco IAD 2431 which has MGCP protocol; I cannot make
it to work againts Asterisk; at least there is some MGCP conversation
between them but when I offhook a phone attached to IAD I get no tone at
all.
As anybody managed to get working Asterisk against an MGCP Cisco
gateway ?
Which MGCP version should I use ?
Also I recently
2005 Mar 02
2
Dual X100P cards
I know the X100P cards are not supported by Digium any more, but for home office use, are they still acceptable? I have two POTS lines, one residential and one business line comming into the house. I'd like to get both into my * server and $15 total compared to > $100 for the newer TDMxxx card sure is desirable. Having said that, will the sound quality, functionality, stability, etc. be as
2006 Jun 26
4
Oh oh. Micro$oft just noticed VoIP
It will be interesting to see how many standards get broken, and how
many proprietary hooks get thrown into the pot. The bean counters smell
some money, and their OS franchise is waning:
http://www.nytimes.com/2006/06/26/technology/26soft.html
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2005 Feb 11
0
Transfers to engaged extensions
Hi,
I'm using zaptel with three way calling and call transfers with a hookflash.
If I try transfering a call to an extension that is engaged I get an
engaged tone. This is fine, but how do I get back to the caller?
If I hookflash again I seem to put on a three-way call and the caller
can hear the beeping. I can press hookflash again but I'd prefer the
caller to hear only the hold
2004 May 18
0
problems with asterisk-oh323
Hello,
I've been trying to send traffic to a Cisco Call Manager 3.2, but with
no luck.
Here's whats happening:
* Call gets to CCM
* Call gets to the gateway
* Rings a couple times on destiny
* Call gets hungup.
On the CCM I get the following error: MediaManager - ERROR
wait_AuConnectErrorInd
On the Gw (Cisco AS5300) I get a disconnect cause of 2F (Resource not
available)
On asterisk: