Re: [Asterisk-Users] Panasonic PBX -to- Sirrix BRI: Numbers getting echoed/duplicated

Any ideas anyone ? Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent

Re: [Asterisk-Users] Panasonic PBX -to- Sirrix BRI: Numbers gettingechoed/duplicated

Thanks Peter. Any other takers on the list on this one ? Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion

Re: [Asterisk-Users] Panasonic PBX -to- Sirrix BRI: Numbers gettingechoed/duplicated

Hi Peter, Thanks for your reply. Would srx show ccmsgs 1 help ? Regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion

[Asterisk-Users] Panasonic PBX -to- Sirrix BRI: Numbers getting echoed/duplicated

Hi guys, How's things going ? Got a bit of a weird one here that I've been unable to solve. I have a Panasonic PBX linked to a Sirrix Quad BRI card that is running in TE (ptp) mode in a Asterisk box -this then links through Internet to another Asterisk box via IAX2. When a user on the

Re: [Asterisk-Users] Panasonic PBX -to- Sirrix BRI: Numbers gettingechoed/duplicated

to disable it ? Thank you for your help so far - greatly appreciated. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion

Re: [Asterisk-Users] Weird ring back

Hi Yair, Thanks for your email. Unfortunately no reply or response from anyone yet. Please let me know if you hear anything - I'm also battling to resolve the problem. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82

Re: [Asterisk-Users] Monitoring Sirrix quad BRI channels

with debug set to 1 and the channels definitely shows up as Srx/gout--01 etc. Any ideas ? Many thanks. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux

Re: [Asterisk-Users] Monitoring Sirrix quad BRI channels

Ah thanks Nicolás. I will give it a try and let you know how it goes. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion

Re: [Asterisk-Users] Monitoring Sirrix quad BRI channels

Thanks Nicolás, Thanks for your reply. Sounds interesting ! Any ideas on how to do that ? :) Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven

[Asterisk-Users] Monitoring Sirrix quad BRI channels

Hi all, How are things going ? Is there a way for me to individually identify each BRI channel on the Sirrix quad BRI board. The reason I ask is because our client uses the "Asterisk Flash Operator Panel" to monitor its external lines and transfer calls from the lines to the various SIP

[Asterisk-Users] Weird ring back

Hi guys, I have a weird thing happening sometimes with users calling from a GrandStream phone through Asterisk onto a PSTN. Sometimes after a user hangs up a call on a GrandStream phone the phone starts ringing after a couple seconds. When the call is answered there is no one there.

[Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation

Hi guys, I'm running a Sirrix PCI4S0 quad BRI card in a box with Asterisk CVS-HEAD (20050614). Something I've come across is that with 'echocancel = yes' in /etc/asterisk/sirrix.conf the echo is unbearably loud - so loud in fact that the echo distorts. To remedy this I've set 'echocancel =

Re: [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation

either that or I'm doing something wrong. :) Did you have to modify the patch in any way or did you just apply it 'as is' ? I will keep in touch to let you know the outcome. Many thanks. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33

Re: [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation

that there's a slight echo, much softer than the echo I get when I have 'echocancel = yes' with or without the patch applied. Hows JHB today ? :) Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za

Fw: [Asterisk-Users] Analogue phone transfering

Hi guys, Any other ideas on this one ? Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers

[Asterisk-Users] Analogue phone transfering

Hi guys, How are you keeping ? I have an analogue phone plugged into a Digium FXS Zap module on my TDM card. The phone works well except that I cannot seem to transfer calls using the "flash" key. I don't seem to get another dialtone as indicated in:

Re: [Asterisk-Users] Analogue phone transfering

=no busydetect=yes busycount=8 adsi=no relaxdtmf=yes faxdetect=incoming channel=1-3 signalling=fxo_ks context=default relaxdtmf=yes ;threewaycalling=yes transfer=yes adsi=no usecallerid=no channel=4 ;rxgain=70.0 ;txgain=50.0 Kindest regards David Wilson ___ D c D a t a Tel

Re: [Asterisk-Users] how do i get rid of this blasted echo !!!

Is MMX support not enabled by default in the Zap drivers ? So this is something we need to do if using any PII, PIII, P4 AMDK6/Duron/Athlon and Celeron CPU ? Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http

Re: [Asterisk-Users] OT: Active channels bridging with SNOM190

Hi Matt, Thanks for all your help. Things have gone well today. No bridged Zap channels so far ! Thank you so much for all your help. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL

[Asterisk-Users] OT: Zap channels intermittently bridging with SNOM190

Hi guys/girls, We are running a TDM04B card with Asterisk in a Linux box that has 15 GrandStream102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly into our telecoms provider's analogue lines. Something I've picked

Re: [Asterisk-Users] OT: Zap channels intermittently bridging withSNOM190

right - thank you for your assistance so far. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers

Re: [Asterisk-Users] OT: Zap channels intermittently bridging withSNOM190

a difference in firmware ? Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent

Re: [Asterisk-Users] OT: Zap channels intermittently bridging withSNOM190

and thought it could be something in Asterisk that was causing it. Thanks so much for all your help. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux

Re: [Asterisk-Users] OT: Active channels bridging with SNOM190

Yea, True. No sweat. Should be better now ? :-) Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion

Re: [Asterisk-Users] OT: Active channels bridging with SNOM190

Hi Matt, Cool. The upgrade went through well and the Call join on Xfer option appeared. I've now turned the option off and so far things are working nicely. Thank you so much for your help. It looks like this issue has been sorted ! Keep well. Kindest regards David Wilson

[Asterisk-Users] Zap channels intermittently bridging with SNOM190

Hi guys/girls, We are running a TDM04B card with Asterisk in a Linux box that has 15 GS102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly into our telecoms provider's analogue lines. Something I've picked up with

[Asterisk-Users] Zap channels intermittently bridging with SNOM190

Hi guys, Sorry to bug you on this. Any ideas ? Really stuck with this. Hi guys/girls, We are running a TDM04B card with Asterisk in a Linux box that has 15 GS102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly

[Asterisk-Users] Manager redirect action does not appear to work in some cases.

Hi there, I am currently developing the ability to have a unified system/telephone login, with SIP phones paired to a computer. When a user logs into a computer, a notification is sent to an external service program which connects to Asterisk through the manager API. Besides that, the service

[Asterisk-Users] Running AGI script on answer.

Hi there, Is there an accepted way of running an AGI script on answering of a channel? Is it even possible? I don't need to execute AGI commands, I just need to know a channel has been answered. Thanks, David. -- One world, one web, one program -- Microsoft promotional advert. Ein Volk, ein

Re: [Asterisk-Users] Grandstream Budgetone 101 channels don't disappear on hangup.

. In the sip.conf file, I put bindaddr=209.43.121.215. After that, calls come up normal and end completely normally. Like I said, this is probably not the problem in your situation.. but hopefully it'll lead you in the right direction? Tom On Jul 23, 2004, at 5:43 AM, David Wilson wrote: Hi

[Asterisk-Users] Grandstream Budgetone 101 channels don't disappear on hangup.

Hi there, I'm having problems with the Grandstream Budgetone 101 on hangup - show channels/show channels concise output is still showing the call's channels as active. The problem does not exist when I use SJPhone, so I'm assuming it isn't an Asterisk configuration issue. Has anyone seen this,