similar to: PRI channel hangs after BUSY

I have a Toshiba DK424 connected via T1 E&M to a TE110P card. Intermittently when a user dials a number I am getting 'getdtmf' errors on the Ast server and the calls do not go through. If they dial the number once or twice more, it works fine and I receive no DTMF problems. On another note, end users are complaining about intermittent disconnects. T1 is ESF/B8ZS - 24 chan. Other

Was'nt sure if this mail got through earlier: I have been having a weird issue with my telco's ISDN PRI occasionally resetting on a incoming call, i suspect it to possibly be a timing issue since some of the incoming call work. This problem happens very frequently. I am using asterisk-1.6.0.1 with libpri-1.4.9, the asterisk server is connected viw TDMoE to a Redfone Fonebridge into

Hi, I've connected Asterisk with 4 PRI to a Siemens HiPath 4000. For CALLERID(name) feature I wanna use Q.SIG as switchtype. Cause Siemens PBX orders Channels logical I need the parameter qsigchannelmapping=logical. Here is my chan_dahdi.conf trunkgroups] [channels] language=de context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes

Hi Srs. we have to integrate a Nortel MATRA M6501-L with Asterisk with a TE410P. All call from outside get into asterisk and asterisk send to Nortel in a correct way. My problem is when a call is made from Nortel to Asterisk. If we digit a national Number in Spain([98]ZXXXXXXX or 6XXXXXXXX) all work find. But if we digit an international number call doesn't progress. I Have seen in

Hi, I am having issues sending and receiving fax on my asterisk setup. Currently I have a server that has 2 x E1 TDM cards one is sangoma and the other one is openvox. Both support echo cancellation. One of the e1 is connected to our telco provider via mfcr2 where all our incoming calls originate. On the other end is a pri connection going to HICOM PABX where the local attached to a fax is

Hello List! We have problems setting the right caller id on outgoing calls. The Asterisk Pbx is located in Bucarest(Romania), our Telco provider is rcs-rds.ro. We have the local telefon number 40787 00-99, associated to our PRI E1 Line. Where 00-99 are the DID numbers available. The telco is aspecting a 3 digit long Callerid from us, for example like "710", for the extension 10.

Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local

I have a client who is using a Shoretel PBX. This PBX apparently does not send DTMF information OOB, but instead sends this inband via the B-channel. This is traversing an Asterisk box via a PRI. The user calls the IVR (1-800-CALL-DHL), receives audio, but is not able to present DTMF to engage the IVR. With some light research it appears that the DSP is not activating until the call is

I'm having (I think) timing issues in relation to bridged T1-T1 calls via dynamic spans. Fax calls are intermittently working, but voice is fine. My box has a Sangoma A400 inside it as the primary Zaptel timing source. My T1 PRIs that are hooked to the box come in via a foneBRIDGE2 (dynamic TDMoE spans). PRI #1 is the telco and PRI #2 is an existing Comdial FX-II. For some reason, bridged TDM

Hello All, I apologize for the slightly off-topic question, but I'm sure that the people best acquainted with the issue would be hanging around here. We recently deployed several Linksys POE switches for some smaller customers (10-24 station) and appear to be suffering from intermittent lock-ups of the SNOM phones attached. Obviously we are running Asterisk for the gateway, but I was

Hi there, we have a problem with dahdi and overlapdial. We are running an E1 in Germany and are in need of overlapdial. The E1 is connected to a Sangoma A101. As soon as overlapdial is set to "yes" we have problems with incoming audio on the dahdi channels. When set to "no" all audio is fine. Basically we can choose between being able to receive calls or to place calls

Has anyone fed a Nortel BCM from Asterisk? I'm interested in switching our company over, but don't want to replace all the handsets in one fell swoop. I imagine some of the PRI cards can "emulate" a switch? I'd still like to pass CallerID into the Nortel, etc but all the external traffic would be VOIP, not TDM.

Hi I have this escenario: |SIP or H323 phone|---->|Cisco2600|----E1-pri---->|Asterisk|------>IVR, A2Billing, etc... The problem is that I can not hear any audio when call from 'sip or H323 phone' and configure something like: exten => _01XXXXXXX,1,Playback(thank-you-for-calling|noanswer) ... It works if I remove the 'noanswer' parameter but in this case it connects

Hi list, Attempting to get an ISDN-BRI line connected using an HFC-S PCI card together with Asterisk v1.4.14 and Zaptel 1.4.7 on a Debian etch system, I find that I can't access the card's resources because the channels are always be busy. An attempt to call out results in the following CLI output: == Primary D-Channel on span 1 down == Primary D-Channel on span 2 down

Hi I have recently upgraded my Asterisk system (from 1.2 to 1.4) and I have started to notice the following messages when I recieve a call on my Zap channel :- [Mar 11 09:20:17] WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 bytes LLC I have a single PRI ISDN 30 link to a Siemens Realitis DX. Here is my zapata.conf :- [channels] echocancel=no echocancelwhenbridged=no rxgain=-5.0

Hi, I have just set up Asterisk to use an E1 line with a Digium card. And I can call both in and out, but my outgoing line is all ways identifying itself as the same number, and i can't even change it to another number in the same number series. Do anyone have some clue on how to fix this. I'm using Asterisk 1.6.2.13, libpri 1.4.11.4 and DAHDI 2.4.0. /etc/dahdi/system.conf:

Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2 I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf. The detection is not working with call file, manager originate and not with the dial command to the mobile. I have no ideas left. I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...) But with the same vaules on a second call there

SYNOPSIS Erratic red alarm T1 PRI on asterisk, but zttool running concurrently during alarm shows no errors, irq misses, or alarms, on any span. Using asterisk and quad Digium T405P, configured as follows: Span 1 connects to ISDN PRI (fractional 8 B channels, D channel 24). Span 2 connects to T1 Mux and analog stations. Span 3 connects to ISDN PRI Nortel BCM hybrid key system digital trunk.

Hello All, ?I have a Dell Vostro 1550 laptop. i3 core, 64 bit, 8gb ram etc etc. I had a nic interface, but some how the wired nic is not working. So I am trying to use the BCM 4313 wireless chip, of this system, to connect to the internet. ?I have Cent os 6.5 (64 bit) installed with the required libraries and kernel sources, headers etc. I got the source code of the driver,

Hi, I've an PRI coming to my asterisk ,calls are coming fine and my agents are able to answer no prob. but I've an agreement with my telco with some incoming no if the no of calls on these no are more then 3 then send to another no. they use busy signal to divert call on another number so I'm sending the call to Congestion() if no of calls in this group are more then 3. But my