similar to: transfer capabilities

Greetings, I have a quad port Tormenta 2 PCI card (uses the tor2 driver), which I have been successfully using for some time on an older, power-hungry Intel machine. I recently moved it to a newer machine with an Intel Atom D510 CPU in an effort to save on energy costs, but I cannot seem to get it to work there. Both machines are running Ubuntu 13.10 x64, with DAHDI support coming from the

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

Previously incorrectly sent to asterisk-dev list, sorry. I tried today while connected to a Jtec QSIG E1 card, with DAHDISendCallreroutingFacility with the following test dialplan: Extension 4888 is on the Fujitsu [incoming] exten => 8688,1,Answer() exten => 8688,n,Playback(connecting) exten => 8688,n,DAHDISendCallreroutingFacility(4888,8688) exten => 8688,n,Playback(goodbye)

the Telco, I can make calls in. What I am trying to get though is how to pass through the DID range. The E1 that I am connecting to the Telco with, used to connect direct to the NEC system and already has hunt group calling enabled for all 30 channels. Also, I was given a 100 number indial range (from 00 -> 99). If the E1 is connected to the NEC directly, I can call 5555 7320 and the NEC

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

Hello all. I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. Ie, pick up NEC handset, dial

Hi, sorry for the newbie hardware questions but here it goes scenario - our telco is feeding us e1 thru coax connection (unbalanced) - so the coax feed rx-tx goes to our old pabx using ericsson bp250 - what we wanted to do is to install asterisk in between hence telco<-->asterisk<-->bp250 using asterisk to power up the voip portion the problem is the we are getting crackling sound

Hello all, It 's dificult to explain; The system I need is an box option (based on *), that I would add to an existing PABX (ie: Nortel with 600 ext). I need two E1/T2 card to plug the system between Telco (FT) and PABX (Nortel)! One card for France Telecom Side (E1a) and one other to Nortel Side (E1b). --------- -------- ----------- Telco FT

Hi all I've got a setup where I use a Sangoma PRI card driven via Sangoma WanPipe to connect to a legacy Samsung PABX (I'm not sure which model) form Asterisk 1.8.11.0. The reason is the customer has a large installed base of Samsung phones physically connected to it and on each users desk. They did not want to spring for a complete replace of all their Samsung phones with generic, and

Hi, We've got a dedicated T1 with two trunks running into our ILECs selective router for 911. Split out of the T1 into two MF CAMA trunks on ILEC side. I'm trying to use asterisks e911 signaling, but I'm having trouble with the dial command. (== Everyone is busy/congested at this time (1:1/0/0)) I'm missing something and I'm thinking it has to do with the hookstate

Hi list, I'm having trouble with DTFM tones detection. Usually, some tones are being received duplicated in Asterisk, some not. As you can imagine, that's a very big problem involving IVR menu options, Meetme conferences protected with passwords, and so on. We are currently using DAHDi 2.2.0.2, module wct4xxp, which is managing a Digium TE220B card, with a hardware echo canceller

Hi list, I'm having trouble with DTFM tones detection. Usually, some tones are being received duplicated in Asterisk, some not. As you can imagine, that's a very big problem involving IVR menu options, Meetme conferences protected with passwords, and so on. We are currently using DAHDi 2.2.0.2, module wct4xxp, which is managing a Digium TE220B card, with a hardware echo canceller

I have a pretty large setup on one of my customers. Digium TE420B (with echo cancelling module), 3 Xorcom Astribanks with 32 FXS each and 1 Xorcom Astribank with 16 FXO. These FXO ports are NOT used for fax/data transmission, as they are connected to cell phones. Not really related to the issue, but there are also 250 SIP phones. The problem is that fax and dial-up connections are really

On Tue, Jun 30, 2015 at 3:34 AM, Tony Kasule <timotsmith at gmail.com> wrote: > Hello, > > Anyone to help me with this issue? It has never worked :( > > On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> > wrote: > >> Hello users, >> >> I have a Digium Te235 and asterisk 13 which have worked well with 1 >> carrier but

Hello all, I have the following working (somewhat) setup: TELCO | | E1 (30 Chan -- TE210 SPAN 2) | | Asterisk box 1.6 with DAHDI drivers loaded Digium TE210p | | E1 (30 Chan -- TE210 SPAN 1) | | NEC PBX

Hi Asterisk Users, We've been experiencing some tough time regarding a new Asterisk installation connected to the PSTN via an ISDN PRI with a Digium TE121 with the optional VPMADT032 echo cancellation module. For now, I'll focus on something very specific which is summarized on this email's subject. However, here are some general facts for the context: - System pbxfri went

Hi All, I am installing a Zaptel Quad E1 card with ISDN PRI in Brazil and I?m with this problem: Asterisk CLI : Mar 27 01:49:57 WARNING[2612]: chan_zap.c:7143 zt_pri_error: PRI: !! No channel map, no channel, and no ds1? W hat am I supposed to identify? Mar 27 01:49:57 WARNING[2612]: chan_zap.c:7143 zt_pri_error: PRI: !! Unable to add IE 'Channel Identification' == Restart

Hello everyone. I have server with 2E1 PCI card, asterisk 1.4.35, dahdi 2.4.0, libpri 1.4.12-beta2. One PRI trunk looks to PSTN and take a clocksource from telco. Another trunk looks to PBX with DECT system. Some outgoing calls from asterisk to PSTN drops. The last message that exists before hanging up process is: DEBUG[28467] channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI/... This

Hi, I'm having problems with a new ISDN PRI in a new server. The cable is connected and the E1 modem seems to have issues with syncing (blinking light on the modem). versions: CentOS 6, asterisk 1.6.2.20, dahdi 2.5.0.1, libpri 1.4.12 ------------------------------------------------------------------------------------ dahdi show status T4XXP (PCI) Card 0 Span 1 RED 0

Hello, I'm having issues connecting throu PRI with the following error "Requested transfer capability: 0x00 - SPEECH" Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003", "CALLERID(num)=xxxxxxxxx") in new stack -- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",