similar to: Channel Bank ? Simple Switch Hangup?

Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT <-> Swyx The above setup works fine... what i'm trying to achieve is BT & SIP Trunks <-> Asterisk <-> Swyx I have connected to our BT (2 x ISDN30 UK) with

Hi, I'm using Sangoma's wanpipe together with dahdi, all software downloaded today at most recent version. Hardware is Sangoma A104, a 4xE1 card. Installation went well. Anyway, wanrouter status shows a different result than dahdi_tool or dahdi_scan. I've just put a hardware loop on port 1. All the other ports are open. wanrouter status shows the expected result: Device name |

I'm trying to send CallerID info to a MetaSwitch system over a PRI. The MetaSwitch gets the info exactly as it is sent by Asterisk, but I think it might be having trouble with the Hexadecimal b1 that is being sent just before the first character of the CallerID Name. Does anyone know what the significance is of the b1 being sent here? Or, is there a way to make Asterisk not send the b1

Hi, Do CDMA phones use the gsm codec? Rgds. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080530/2fe5b269/attachment.htm

Hi, Does anybody have an idea as to why dahdi_test results drop to unacceptable levels after doing a wanrouter stop/start using a Sangoma card? See below that it drops from 99.99% to 98.55%: [root at bin]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.999512% 99.992874% --- Results after 2 passes --- Best: 100.000 -- Worst: 99.993 -- Average: 99.996193, Difference:

I am connecting to a siemens hipath 4000 with dahdi 2.1.0.4 and asterisk 1.4.23 using a Te210P card. the phone guy is saying that the lines are reporting always BUSY. however on my end the status shows OK. Anyone seen this? Is there something different about connecting PRI to siemens hipath? system.conf shows: loadzone=us defaultzone=us span=1,1,6,esf,b8zs bchan=1-5 dchan=24

Hello, We are having issues with a NEW Sangoma A108D: -- Executing [691918892 at pbx1:1] Dial("SIP/xtravoip200-009d24b0", "DAHDI/g0/691918892|30|m") in new stack [Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 [Oct 1 10:04:43] WARNING[14171]: app_dial.c:1237 dial_exec_full: Unable to create

Anybody know a reliable fax solution for 1.4.30 branch? I am using PikaFax on another server and works very well (about 3000 faxes a week), but it appears they no longer offer their product to open source asterisk, only for there "WARP" appliance. NOT really looking to migrate from 1.4.x to 1.6.x -------------- next part -------------- An HTML attachment was

Hello, I am using Queue application with multiple agents in each queue. I want to set the CDR(userfield) for each cdr based on the agent answering the call. Is it possible to do this? Thanks

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know

I have several remote phones that experience a slight "call" delay when answering phones, ie, they will answer, speak a few words, and then the remote caller will hear them, and the first half is cutoff? Any idea what could be causing this? Thanks, Bill. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anybody notice log delays in this list, and very small amount of traffic? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110109/576a9b0e/attachment.htm>

I have asked a GSM operator in my country if he can route a number or a short code to my asterisk server via SIP (since they dont give DIDs in my country) the operator said they do not support SIP, they have no way of converting GSM calls to SIP to then send them to me. I would like to know what is needed from the operator side to do this, what kind of material is needed, or what can be done from

For years we've had the following simple context for outgoing calls: [outtrunk] ; match any NANP, and strip leading 1 off exten => _1XXXXXXXXXX,1,StripMSD,1 ; dial outbound on trunk group 1 exten => _XXXXXXXXXX,2,Dial,Zap/g1/${EXTEN} But when I upgraded on Friday to the latest CVSHEAD, this no longer works. If I send 13115552368 to this context, I get a message like pbx.c: Channel

Hi, Having problems with getting either RxFax or FaxReceive to compile. Running Asterisk 1.4 on CentOS 5. Does anyone have the free/open source executables that you could send me? Thanks for your help! P. S.: TxFax and FaxSend would also be appreciated.

Does anybody know of a Similar list for SwitchVoX? And would like to post to proper list if one is available. I had posted on digium forum, but have not received any responses yet. http://forums.digium.com/viewtopic.php?f=38 <http://forums.digium.com/viewtopic.php?f=38&t=77031&sid=4adb81c464701e0039d e21a300aa273f> &t=77031&sid=4adb81c464701e0039de21a300aa273f

I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel bank (12fxs/12fxo). I have the setup partially working thanks to some help from IRC. However I still have the following issues I can't seem to resolve 1. When calling into the system from the PSTN call hangup is not detected. * leaves line in use until it is shutdown. 2. When calling an analog phone connected to

I have a system that I am trying to get a port on a Sangoma A104d card connected to an Adtran 850 with 5 FXS modules and 1 FXO module. A problem I am having is figuring out what cable should be used from the port on the Sangoma to the JP2 port on the Adtran. Tried was a cross-over T1 (1->4, 2->5, 4->1, 5->2) as well as a straight T1 (1->1, 2->2, 4->4, 5->5). Neither one

Hi, I know you can access various sip variables via 'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status of the sip user - but what about variables? I have a user that has setvar=123456 in their users.conf (sip.conf if you prefer). I can read it with a 'sip show peer 201' - but that gives everything and parsing that isn't really an option. Anyone know how

Hello, We succeed to send faxes using FFA, when the files are converted to tif from PDF using gs, but it doesn't work with tif files we copy/upload directly from our PCs. We saw in the manual that the size is important, since we got the error "FAX handle 0: failed to queue document 'filename.tif'", so we set it to 1680x2285, but it's still rejected. Is there a way