similar to: '#' doesn't work for me

hi, Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call. ie, the operator answers the call, and presses hash key to transfer, and then enters the extension number, some times, it timeouts too quickly before the operator enters the whole extension number (may be bcos the operator is slow). I tried the following, but it doesn't seems to be helping

G'day all, I've been googling myself silly looking for help on this one but have come up blank. I have an AVM Fritz!Card PCI, and I'm using chan_capi v 0.3.1 with * from CVS-HEAD-05/08/04-22:48:00. I can start * and make and receive calls on ISDN fine but after a few hours of * uptime, on any ISDN call I make or receive from my SIP handsets (7960 or ATA-186) I get bad audio: on the

Hello List, I am an Asterisk newbie, and I got a question about Asterisk Background command's option "noanswer": What is required from the user agent, such as a SIP phone, to be able to hear the playback without Answer()? I'm asking this because when I used X-Lite, I could hear the the audio file but when I used a hardware phone (an ATA in fact) I couldn't hear it. The

Hello everyone, I have posted once a message that I had problem with Asterisk, ATA and X-Lite. The problem was: When I called from ATA to X-Lie it did not want to work. The connectuion apper in astersik but I could not hear anything. Right now I updated asterisk from CVS and I still have this problem but... When I call directly to X-Lite from ATA it doesn't work but when I call to X-lie

Hello, I have been trying to get my coders to work without a conversion. I have read all the available asterisk documentation and support groups without any luck. Here is my issue. (Please feel free to ask questions if you do not understand what I am talking about.) I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if sip-server request g711) I have 2 SIP-services to

I am experiencing a situation where I am getting intermittent choppy audio. Here is the network layout: Termination provider -> IAX2 over the Internet -> 20Mb fiber connection -> router -> Asterisk My ATA connection goes into the router between the fiber and the Asterisk server on another interface here is the layout from me to Asterisk: Sipura ATA (SPA1001 running

Hi, I've just setup a second asterisk system and am having a wierd problem. If I am on the same network as the system it works fine... if I am outside coming in from the internet through a NAT I get the following: I can place calls.... I can hear the asterisk system. The asterisk system does not seem to hear me. When I do a tcpdump I don't even see the ATA sending anything to the

My asterisk box doesn't recognize DTMF from my analog phone, plugged into my ATA(linksys pap2 version2). I can make/receive calls fine... it's just that, for example, I cannot login to my asterisk voicemail. Softphones (such as x-lite) are fine. I've turned up a few articles via google where some people have this trouble, but have not seen suggestions on how to fix. I presume

Hi guys, sorry to be iterating this on the list once more, but I'm not able to get this stuff to work as I'd expect. So far, I've always managed to keep it out of NAT environments :-> My home LAN is NATed by a simple Draytek router. In the home LAN is an ATA186 with SIP. On the internet (public) is an Asterisk server. I have nat=yes in the sip.conf and the connectmode is set

*This message was transferred with a trial version of CommuniGate(tm) Pro* Will asterisk actually convert between two different codecs????? ie, a SIP endpoint running GSM and another running G.711? Wouldn't that add quite some latency? I was always under the impression Asterisk did not recompress and was smart enough to negotiate the right codec at each end and just pass through the RTP

Hello, I have read the information on echo and SIP in the FAQ and I have scoured the mailing list for possible solutions, but as yet I have not been able to get rid of this echo. I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed into an asterisk server. If I call between the Sip Phone (Budgettone-100) and the 4 FXS ports everything sounds great. If I call out to the PSTN

Hello Michael, Here is the BackTrace of the program which i forgot to attach BACKTRACE OF Asterisk -vvc #0 0x42074d60 in _int_realloc () from /lib/tls/libc.so.6 #1 0x420738c4 in realloc () from /lib/tls/libc.so.6 #2 0x47c7da89 in PAbstractArray::SetSize(int) () from /home/sip/pwlib/lib/libpt_linux_x86_r.so.1.5 #3 0x47c7cf4d in PContainer::SetMinSize(int) () from

I am new to VOIP, Linux and Asterisk. Through a lot of reading (this list, voip-info.org, documentation, etc.), I successfully installed FC3 and * on a new Dell SC420 with two X100P connecting to two PSTN lines at my office. I've also installed AMP to help me configure IVRs, call groups, extensions, etc. I use a Handytone-286 ATA and x-lite clients on the internal network and all works

I have a Lite-On 24102B burner on an Epox 7KXA (VIA 686A southbridge) together with a few SCSI disks. I recently upgraded from FreeBSD 4.5 to 4.7 then almost immediately to 4.8. I've had trouble with the burner since. With FreeBSD 4.5, I used "sysctl hw.atamodes=---,---,dma,---" to set the burner to DMA. Everything then worked fine. As far as I know it was using UDMA33.

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users

I have this silly problem : When I place a call, being either to an extention or to an outside line, DTMF signals are ignored by Asterisk. This is serious because I can't even transfer calls (#) or park them (#70). When I receive a call there's no such problem. When I recover a call from parking (71) all goes OK too, and so goes call capturing with *8... I already tested dtmfmode=inband,

Hello, Does asterisk do transcoding when the call goes through the system, codecs are the same but signaling protocol is changed. example: SIP with GSM ---> IAX with GSM What quality destruction happen when I use transcoding? I know this is not a concrete/precise question, but I would like to know how is it in general. What CPU performance is needed for transcoding 30 channels e.g. from

Hi: We have got SIP clients connecting to our Asterisk fine with a DSL connection behind router (NAT), but when we bring the Sipura 2000 ATA to a Rogers Cable connection behind a Netgear router (NAT), the SIP clients aren't able to reach the Asterisk at all. We enabled the SIP debug in Asterisk, and it doesn't see any request coming from these SIP clients, and we also tried the to use a

Are there any other codecs that can be used with the 7960 and the ATA-186? I have been using the default gsm codec and wanted to see if I could make use of something a little less bandwidth intensive. Kim Callis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030723/34e950e0/attachment.htm