similar to: Connecting a PSTN gateway to Asterisk using PRI

Hi, I would like to realize a voip testbed that should simulate the scenario in which two companies have an asterisk PBX connected through a PRI-ISDN to the Telco operator. I have no experience of T1/E1 connection but I think that the above could be relized with 3 asterisk boxes equipped with Digium TE405P cards. One of the box will represent the Telco, the other two, the two companies PBX. I

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

Hi guys, Can I use a Sangoma a101 to interface a legacy pbx to an Asterisk server? The pbx doesn't have sip and I want to come in off of a sip trunk and interface with the older system. I know I can use a pri card to hook in to the phone network, but can I use this same card to feed back the signaling as if I were the phone company to the older system? Thanks, Tom

Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890

hi, i'm having problems getting asterisk spliced between an E1 PRI (german Telco Arcor) and an Ericsson Business Phone 250 digital PBX. The Asterisk Server has a TE405P with it's port 1 connected to the E1 PRI provided by our telecommunications provider Arcor and port 2 connected to the E1 PRI of our Ericsson BP250. the setup before: Arcor TelCo PRI(E1)

Is it possible to send CLID NAME on a PRI? The numbers we send out are being received by telco and propagated, but the names we send out are not showing up. Is this a feature in PRI? Do we need to set PRI_NET instead of PRI_CPE? Is this just not possible? Is this a telco config issue? Thanks for your help... I've read voip-info, and various other sources, and search engines, and google...

Hi All, I'm having problems trying to get a TE120P operational in Canada. I keep getting a congestion error when I try to make a call. I'm not sure if my switching, parity, etc is correct. I'm hoping that someone will be able to verify my config. The Telco is SaskTel, with a 10 channel 50 DDI service. Zap show channels show and ztcfg -vv looks ok and the zttool show

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",

I have been getting quite a bit of PRI Resets using my Quad PRI Digium card. Prior to the resets I am getting similar notices to the following chan_zap.c:7482 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 Telco claims the PRI's are fine on their end and that it is my unit. Is this timing? (google somewhat leads to this) I am running 1.08 asterisk zaptel

Hello Group I have a Asterisk server running with 2 Digium T1 cards installed. 1 card connects to Telco via a PRI. The 2nd card is connected to a fax server via Digi DataFire RAS 24 PT1 Adapter (Digi0001). The idea is to have Asterisk route the calls based on DID or FAX tones. Everything is working great so far. The only problem is the Fax server does not see the DID. How can I tell if Asterisk

Hi, If I setup an IP PAX gateway to handle VoIP calls to a traditional phone line, I am wondering how each VoIP call to the PSTN connection get charged by a local Telecom. Thanks Sam

I've been going back and forth with my telco for several days, trying different configurations to get a new PRI to come up. The bchannels are all up and the T1 is not in alarm status. The dchannel refuses to come up however. We've tried ni2, qsig, and now dms100 for the switchtype. The telco tech I've been working with says that he's been sending "reset all channels"

So we've got a TE410P configured as E-1. The PRI is showing up as normal, I have green lights, but d channel doesnt seem to come up and i keep getting this error if i do a "pri intense debug" The carrier swears up and down that there are no issues on their end. Any thoughts? localhost*CLI> > Unnumbered frame: > SAPI: 00 C/R: 0 EA: 0 > TEI: 000 EA: 1 > M3:

Hi there Just a quick question, I will be building some servers in a lab utilizing Digium E1 cards. I would like if possible to avoid the expense of installing an e1/ISDN30 in my lab. I have two questions really, first does anybody know of an effective simulation tool I can use to replicate a real world PRI but without the telco line being installed. And secondly, can I have a scenario with 2

I have my peers registered to SER.asterisk seems to be sending mwi for the peers seen in the sip show peers CLI command. i have my ser server registered with asterisk as a type=friend and all clients register to ser.how do i get mwi to work for these clients registered to SER. Thank you, -AA

I need help on my zaptel.conf and Zapata.conf for a TE207P I'd like Span 1 to receive a PRI from the phone company(US PRI). I'd like Span 2 to interface with a Nortel Phone system as a PRI(acting as the phone company) Essentially my asterisk box is a man in the middle intercepting calls from the PRI passing certain DID to the Nortel, also intercepting calls from the Nortel passing them

Hi I have problem with TE121 Digium card. I connected it to modem keymile Music 200 (provided by telco) but I can see 2 red lights on modem (both bellow words rx) and my card is red too. I tried to make experiment and made loopback (pins 1 4 , 2 5) and put it in card and card become green (I hope that is sign card is ok) but on CLI i can see following error message WARNING: We think we are cpe but

Does anyone know of a way to simulate the signaling of a PRI trunk for testing/setup purposes? I realize this may be a rather naive question, but I was wondering if you could take a TE110, for example, and using a crossover cable (or not?) and some means of emulating the NI2 signaling protocol connect it to another TE110 on another machine to test and verify an installation before the telco

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 everyone: using chan_capi 0.7 and asterisk 1.4 Quick question: How can I detect the number of "voice" channels (B channels) in use at a given time. I'd like to call "Busy" if two B channels are used on my BRI to give the calling customer a Busy signal. Long question: On my single-line BRI (two channels) I'd like to give the 3rd simultaneous incoming call an