similar to: PRI unknown signalling on TE405P

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)

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

Hi all, Does somebody know how I have to setup my TE405P ? Is it correct my configuration below ? Otherwise, can somebody help me ? Thanks, Angel. zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,1,ccs,hdb3 span=3,0,1,ccs,hdb3 span=4,0,1,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31

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

Hello all, I really checked voip-info.org but it still seems to be not very easy and I just hope that there is anybody with a simular config. We have one PRI (euroisdn with 30 channels) coming from the DTAG. The second PRI should be connected to a Siemens Hicom 150E Pro Office PBX (was cheaper than a channel bank :-) Carrier ----S2M------ * -----S2M------Siemens | |

We originally purchased a TE405P so we could connect our Asterisk server directly to the T1 PRI from our provider, however due to all the problems reported with T1 PRI's interacting directly with Asterisk, we scrapped that idea and decided to stick with our Cisco router making the conversion to SIP. Anyways, we found a new potential use for it. We have an old location with an existing PBX,

Im having a problem getting the 2 cards working together. I added the TDM with 3 FXO and 1 FXS board. The TE405p is working fine. THe TDM registers and identifies the FXO's and FXS but cant get them to go off hook. I hope someone can help. Asterisk seems to THINK its dialing it sends manager status as dialing, and it just sits there with no tones, on the phone dialing out. If I dial,

I'm installing TE405P card. This is my zaptel.conf. -------------------------------------- span=1,0,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 loadzone = us defaultzone= us -------------------------------------- When i modprobe wct4xxp, -------------------------------------- PCI: Found IRQ 11 for device 02:0b.0 PCI: Sharing IRQ 11 with

I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of the office calls go through the Definity. Here's the issue: Calls to internal SIP extensions, Definity extensions, other offices within our private network (through the

Hi! I need your advice. My problem is that I have very bad sound quality calling to cellular phone via asterisk router. There are some kind of noises and echo effects when you try to speak louder. I have the following components in my call routing schema: - PBX with E1 port. - asterisk router with TE405P card(32bit/4 E1 ports). - Teles server with PRI interface card(3 E1 ports) and VTM

Hi, We've exhausted our internal capabilities as well as Sangoma tech support and were hoping someone with some expertise could help us with a pointer. Briefly, our issue is as follows. Periodically (several times an hour), we get either of the following error messages in our asterisk messages log. These correspond with dropped outbound calls on a one-to-one basis when the second error

Hello, I have a busy problem with Asterisk when I try to transfer a call from PRI directly to IVR. This problem appear sometime after 2 hours or 2 minutes. The log file contain : Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) When this problem appear I must restart Asterisk to solve it. Another thing, I don't know why the alarm is set to NOP on SPAN

Any ideas? 51] logger.c: [chan_zap.so] => (Zapata Telephony) Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata-auto.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing

Did you compile and install libpri *before* Asterisk? I had same problem (among others) b/c I didn't install in the correct order. Try the awesome asterisk_update.sh shell script. Are you trying to emulate CPE or NET? Try signalling=pri_cpe Check for whitespace behind the statement, zapata.conf seems bitchy about whitespace. hth -----Original Message----- From: Steve Totaro

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

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Hello everyone, We have recently turned up a new T1 from McLeod (Midwestern CLEC). It is configured like so: /etc/zaptel.conf: loadzone=us defaultzone=us span=1,0,0,esf,b8zs ;(also tried 1,1,0,esf,b8zs) bchan=13-23 nethdlc=1-12 dchan=24 /etc/zapata.conf: switchtype=national context=pri-in signalling=pri_cpe group=1 channel => 13-23 I can get hdlc0 (and pvc0) up just fine after the

Good Day All, I am experiencing some weirdness using the E&M channel and hope you can offer a little assistance with the problem I am having. 1) call comes into channel 25 (Second Span first channel of a Digium Quad PRI from SBC-PRI) 2) Call is sent to channel 1 (First Span first channel on the Digium Quad PRI connecting an ADTRAN via E&M Feature Group D) 3) Between rings one and two

Hello, I have a problem with Asterisk, on 10% of incoming calls the IP Phone ring but I don't hear the caller and the caller doesn't hear me (all IP Phones have the same problem). This problem appear also if the call is directly send to the second E1 of the digium card who is connected to an IVR. It does not depand on the charge of the server (I have the problem with only one call).

I have an NXP i.MX6-based armv7l-dey-linux-gnueabihf system in which I am seeing some as-yet-unaccountable behavior in sshd when compiled with Arm/GCC 10/11/12. That is, when attempting to scp/slogin/ssh to 'root@<host>', where <host> is either a name or IPv4 or IPv6 address, the connection is quickly closed by the server without prompting for a password. The variable I can