similar to: Any Method for capturing ISUP packets in DAHDI/ASTERISK

Is a normal functionality? when I do #dahdi_cfg -vvvvvv In my Asterisk console shows this.... [Jul 24 13:39:08] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 If I do this a lot of times...then [Jul 24 13:39:20] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jul 24

When I compile dahdi I see these warnings: WARNING: "kasprintf" [dahdi-linux-2.5.0.2/drivers/dahdi/wctdm24xxp/wctdm24xxp.ko] undefined! WARNING: "kasprintf" [dahdi-linux-2.5.0.2/drivers/dahdi/dahdi.ko] undefined! And "modinfo dahdi" shows that the driver was built for a 2.6.17 kernel, SMP mod_unload 586 4KSTACKS gcc-4.1 If I "modprobe -a dahdi", I get

hello , i try to install dahdi-linux/dahdi-linux-current.tar.gz ,dahdi-tools/dahdi-tools-current.tar.gz and libpri/libpri-1.4-current.tar.gz but got these error msg make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.6.1/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.6.1/drivers/dahdi/firmware' make -C

Hi all, I'm getting compilation error as trying to install latest version of dahdi on CentOS box 5.9 which I now updated from 5.6. I also installed the dependencies but still not getting the clue to get install the driver. Listing down the errors below; CC [M]

> > >On Thu, 2003-11-13 at 16:50, Freddi Hansen wrote: > > >>>> >Freddi Hansen wrote: >>> >>> >>>>> >> with boards from Aculab, we are replacing Aculab boards with Digium >>>>> >> boards BUT we would need more >>>>> >> Digium boards IF we could use both Digium and Aculab cards in

Hi all, I'm looking at my options for getting access to ISDN ISUP fields from DDI numbers, when connecting to a 3rd party Asterisk server. This is for a custom voicemail solution, and at this stage I want to avoid renting a PRI. The information I need to capture is: - Calling Number - Called Number (e.g. the DDI handling the call) - Redirecting Number (e.g. the device diverting to the

>Freddi Hansen wrote: >> with boards from Aculab, we are replacing Aculab boards with Digium >> boards BUT we would need more >> Digium boards IF we could use both Digium and Aculab cards in the same >> server. The reason being that >> TE410P doesn't support SS7-ISUP so we continue using only Aculab cards >> in the servers that must support >>

Hello, I can do simple, "yum install asterisk18-*" and it installs Asterisk and Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and smooth. However, if I want to compile dahdi-linux on the same openvz then I get the error, *"You do not appear to have the source for the 2.6.32-4-pve kernel installed".* * * 1- Based on above error and Google search I have

Hi All, Not sure if this is the right mailing list since the dahdi-dev seems not really active, so I'll try here. I'm developing a new DAHDI driver for a custom board. In this card I've implemented the reading of the TDM slots by 2 DMA channels, TX and RX. Each DMA channel has its own callback that decompose the each slot into a per-channel linear buffer useful for calling the

Hi, Is it possible to record say 30 seconds of audio and then have LumenVox convert to text ? or any available tool open source for speech to text . Regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100914/b56c3d9c/attachment.htm

Hello Group Members, I have one question regarding SIP-I/SIP-T support in any of Asterisk versions. We have client which send SIP-I/SIP-T request can asterisk handle it and serve as a normal SIP call. As per mine analysis SIP-I/SIP-T are variant of SIP protocol with adding of ISUP/SS7 packets to original SIP request. If we want to support it then how do we implement it and support it with

Hello, I just installed a Debian Jessie box from scratch which sports a Digium TE435 digital card. I installed the software, built and loaded the kernel modules: # dpkg -l|grep dahdi ii asterisk-dahdi 1:11.13.1~dfsg-2+b1 amd64 DAHDI devices support for the Asterisk PBX ii dahdi 1:2.10.0.1-1 amd64

hello all, i have 2 asterisk boxes on that 1 have public IP Address and another is only have local IP address now on public IP there are some 7 DID forwarded , now i want to forward 3 DID out of 7 DID to local machine we called server B , I know there are DIal , and Switch statement in asterisk , but is there any other convenient way to do this. because if call ratio is high then my call legs

hi all, any one know, about echo cancellation with digium card, is it actually needed or it okay if we dont purchase because it increase price which half of new card, regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090827/8d6c680a/attachment.htm

Hi all, I have n no. of extensions in my dialer. from 456 to 556 extensions. I was created 2 other extensions 667 and 668 I need to allow only STD calls to go from this extensions. These all extensions are same context . I need to define the STD dialplan for only this 2 extensions. how I can ? Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI |

Hello Friends, I am trying to Installl dahdi on amazon EC2 which have Open-SUSE-11.1 X86 version. and here is snap of uname- a command *Linux ip-10-160-86-41 2.6.32.19-0.3-ec2 #1 SMP 2010-09-17 20:28:21 +0200 x86_64 x86_64 x86_64 GNU/Linux* when I try to run DAHDI distribution dahdi-linux-2.1.0.4 I am getting following error *echo "You do not appear to have the sources for the

Dear All, How can we know the On board supports echo cancellation I have *Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02)*board all working fine but sometimes i got echo when user are calling a PRI. is there any way to know on board echo cancellation . regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, can anybody tell me how can i integrate asterisk and skype users so that skype users can dial my asterisk number or dial internal dialplan form skype regars Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090708/cccd4587/attachment.htm

hi all, can anybody help me to give Opensource SIP client information which can be modified as per our requirment regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090518/802cc3ac/attachment.htm

Hi, I need to implement trouble tracking system, we have 250 users in one premise & 3 desktop support technicians. I need to implement trouble ticket system, where user will enter their application / other issues. Mail will be sent to technician available on duty. trouble ticket will be provided to user & will be given close stat once resolved. Kindly suggest me one such application