similar to: OC3/STM-1 Line Card

Thanks James, What i need is to make the fax machines connected to the audiocodes mediant 1000 be able to send and receive fax throught Asterisk (connected to a pri) I know it's not reliable, but it should work at leaste, what should i do on Asterisk and Mediant to make this work? Im quite confuse with all these fax issues :S Thanks in advance > > Message: 11 > Date: Fri, 9 Apr

Hello my friends, I want to make fax work in the following scenario: My versions are: Asterisk 1.4.21.2 WANPIPE Release: 3.4.7 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P The E1 pri is connected to our Sangoma A102DE, we also have a SIP Mediant Audiocodes 1000 where we have some fax machines connected to fxs ports, what we need is to make fax machines through mediant

hello,all there is a AudioCodes Mediant 2000 out there. i want to realise ip to PSTN and PSTN to ip connection. after some configuration on AudioCodes Mediant 2000, PSTN to ip connecttion works. next ,i want to dial from asterisk to PSTN now. i have see the sample in the extensions.conf relevent to PSTN as follow: ; If you are freely delivering calls to the PSTN, list them here ; ;exten =>

Hi, I am configuring an audiocodes Medant1000 to talk to my asterisk box. So far I have successfull in landing a single call from mediant to my *box. my sip conf is as follows: [general] context=sip bindport=5060 bindaddr=0.0.0.0 srvlookup=yes [3911700] type=friend host=dynamic dtmfmode=info secret=blah context=sip where 3911700 is my E1 telephone no. in my extensions.conf I have exten =>

Does anyone have any information for me? Welinghton. Citando Welinghton Magno Guimaraes <welinghton.guimaraes at ufvjm.edu.br>: > Hello! > ? > I am setting up an Asterisk server with a Mediant 1000 (Audiocodes) > to make external links. Does anyone have any manual or instructions on > how to proceed? > ? > Asterisk ?=>? Mediant 1000 (AudioCodes) ?=>?

Greetings... We've been having some interoperability issues between Asterisk and an AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000 somewhere. So, I've been pondering using iptel.org's SIP server (SIP Express Router) as a "front end" for PSTN calls going out to the Mediant, while using Asterisk for everything else. Has anyone done something similar, or

I sent this yesterday, but saw zero traffic, so I think it got lost in the ether, so I'm sending again. I'm having an issue using Asterisk 1.2.14 and an Audiocodes Mediant 1000 ISDN gateway. For the most part, everything is working except for attended transfers. When I do an attended transfer, and complete the transfer before the 3rd party answers, the PSTN side hears dead air until the

Hello Everyone, Unlike going through DIDx, DIDLogic etc.., we have an option of getting DIDs directly from local telco Bell Canada. Currently our SIP Trunk provider assigned a DID to us, and as you know, they just redirect requests it to our PBX. However, when dealing directly with a telco, what equipment will we need? Basically giving us the same capability as a DID provider. If someone can

The callback struct drm_driver.get_scanout_position() is deprecated in favor of struct drm_crtc_helper_funcs.get_scanout_position(). Convert stm over. Signed-off-by: Thomas Zimmermann <tzimmermann at suse.de> --- drivers/gpu/drm/stm/drv.c | 1 - drivers/gpu/drm/stm/ltdc.c | 65 ++++++++++++++++++++------------------ drivers/gpu/drm/stm/ltdc.h | 5 --- 3 files changed, 34 insertions(+),

Hi FOlks, I am trying to setup remotely an "AudioCodes Mediant 2000" MG Module 2 to work with Asterisk through SIP or H323. But since I don't the product manual, it's being a little hard. Anybody would the manual in PDF(file or URL) to indicate to me? Thanks a lot, Isamar

I just put in a Audiocodes Mediant 1000, which seems to be working well except for one annoyance. I am using Polycom 501's and 601',s and if I do a supervised transfer of a PSTN call where I complete the transfer before the 3rd party has answered, the PSTN party hears dead air until the call is answered or goes to voicemail. I'm not really sure where to start my troubleshooting. Any

The callback struct drm_driver.get_scanout_position() is deprecated in favor of struct drm_crtc_helper_funcs.get_scanout_position(). Convert stm over. Signed-off-by: Thomas Zimmermann <tzimmermann at suse.de> Tested-by: Yannick Fertr? <yannick.fertre at st.com> --- drivers/gpu/drm/stm/drv.c | 1 - drivers/gpu/drm/stm/ltdc.c | 65 ++++++++++++++++++++------------------

VBLANK callbacks in struct drm_driver are deprecated in favor of their equivalents in struct drm_crtc_funcs. Convert stm over. Signed-off-by: Thomas Zimmermann <tzimmermann at suse.de> --- drivers/gpu/drm/stm/drv.c | 1 - drivers/gpu/drm/stm/ltdc.c | 1 + 2 files changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/stm/drv.c b/drivers/gpu/drm/stm/drv.c index

VBLANK callbacks in struct drm_driver are deprecated in favor of their equivalents in struct drm_crtc_funcs. Convert stm over. Signed-off-by: Thomas Zimmermann <tzimmermann at suse.de> --- drivers/gpu/drm/stm/drv.c | 1 - drivers/gpu/drm/stm/ltdc.c | 1 + 2 files changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/stm/drv.c b/drivers/gpu/drm/stm/drv.c index

VBLANK callbacks in struct drm_driver are deprecated in favor of their equivalents in struct drm_crtc_funcs. Convert stm over. Signed-off-by: Thomas Zimmermann <tzimmermann at suse.de> Tested-by: Yannick Fertr? <yannick.fertre at st.com> --- drivers/gpu/drm/stm/drv.c | 1 - drivers/gpu/drm/stm/ltdc.c | 1 + 2 files changed, 1 insertion(+), 1 deletion(-) diff --git

The redundant fb helper .load_lut is no longer used, and can not work right without also providing the fb helpers .gamma_set and .gamma_get thus rendering the code in this driver suspect. Just remove the dead code. Signed-off-by: Peter Rosin <peda at axentia.se> --- drivers/gpu/drm/stm/ltdc.c | 12 ------------ drivers/gpu/drm/stm/ltdc.h | 1 - 2 files changed, 13 deletions(-) diff

Greetings... I'm working on getting an AudioCodes Mediant 2000 big box o' PRI's going with Asterisk, and am running into a problem with DTMF handling. The box is sending DTMF packets to Asterisk as INFO packets, and they are apparently being seen by Asterisk. However, the DTMF knowledge doesn't seem to actually do anything -- the VM system doesn't recognize the digits,

On Thu, Jun 22, 2017 at 11:49:34AM +0000, Philippe CORNU wrote: > > > On 06/22/2017 08:06 AM, Peter Rosin wrote: > > The redundant fb helper .load_lut is no longer used, and can not > > work right without also providing the fb helpers .gamma_set and > > .gamma_get thus rendering the code in this driver suspect. > > > > Hi Peter, > STM32 chipsets

On 06/22/2017 08:06 AM, Peter Rosin wrote: > The redundant fb helper .load_lut is no longer used, and can not > work right without also providing the fb helpers .gamma_set and > .gamma_get thus rendering the code in this driver suspect. > Hi Peter, STM32 chipsets supports 8-bit CLUT mode but this driver version does not support it "yet" (final patch has not been upstreamed

I'm currently looking to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 However they are