similar to: DAHDI, PRI and callerid

I have a 'CONGESTION' Status with R2 protocol. While testing this scenario sip GW--?Asterisk ?Digium E1 R2 Protocol?Cisco E1 R2 protocol?sip Gw Find below my error and configuration ,where are the errors in my configuration ? ========================================================================= Connected to Asterisk SVN-branch-1.6.2-r235775 currently running on

Can anyone describe the asterisk implementation of this any better than the sample config files do? from zapata.conf ; Trunk groups are used for NFAS or GR-303 connections. ; ; Group: Defines a trunk group. ; group => <trunkgroup>,<dchannel>[,<backup1>...] ; ; trunkgroup is the numerical trunk group to create ; dchannel is the zap channel which will

Hi list, Ive got a server with 6 ports on it (4+2 port card) we have a DS3 delivering all voice DS1's to us. Carrier has a trunkgroup for the first 8 span (we only have the first 6 plugged in right now). Everything works fine until we fail the primary D channel (D's are on 24,48) the secondary then picks up and outbound calls do not work, if we reboot Asterisk the D on 48 comes up and it

We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get over 500 frame errors and over a 500 slip errors per hour. When the errors reach 1000 per hour the Mitel will take it's T1 card offline. At that point no calls can be routed from the Asterisk server to the Mitel and the TE110P reports a Yellow alarm. What can be causing all these Frame and Slip errors? We have been

Hi, I'm a lab manager / supervisor at our labs. We've had Asterisk in use for over a year directly hooked to the PSTN - a no brainer for configuration (although I had to fix some AT&T specific things in libpri). Right now I have two big challenges. One is to hook our box up lineside to a Lucent Definity G3R. Avaya is chasing this from their end, but we had the highest level of

I currently have two channel groups in my zapata.conf file. I would like one group to be immediate=yes and the other immediate=no Does not seem to matter which way I go, the first entry in overrides my explicit setting for the second group. I am running * 1.0.9 on FC1 [trunkgroups] ;trunkgroup => 1,24 trunkgroup => 1,48,72 ;spanmap => 1,1,0 spanmap => 2,1,0 spanmap => 3,1,1

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

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

Hi, I'm trying to set up an "old" PBX (that supports SIP) to go through our new Asterisk server, so that our old phones can be used still for some time. How can I set up Asterisk to deliver a trunk sip connection that our old PBX can connect to? Is it just to sett up a normal sip device in sip.conf? Or is there some other / extra magic for this to work? I can't test this with

Hi all, As i tested with Asterisk+dahdi+libpri with openvox BRI with NT mode, the ISDN phone does not work. There is the setting and error. ***********************Enviroment***************************** Asterisk-1.6.1.18 dahdi-linux-2.3.0.1 dahdi-tool-2.30. libpri-1.4.11.2 CentOS-5.5 OpenVox B400P ****************************************************************** In my case , I set prot 1 and

[18884732963 at from-fax-machine:... - your call is hitting the from-fax-machine context - yet your 'fax' exten is in the from-pstn-4 context. See the "[2011-03-17 13:40:29.6] NOTICE[8825] chan_dahdi.c: Fax detected, but no fax extension" line. When Asterisk detects an incoming fax tone - it tries to automagically route the call to the 'fax' extension in the SAME

We fairly recently switched service providers for our 4 PRI circuits. Since that time, we started to notice some failed inbound calls. These calls terminate with an ISDN cause code 47 'resource unavailble'. Most of the time I see this error on the first or second channel on the second span in a trunk group (This is the providers trunk group for hunting, not an Asterisk trunk group). All

Dear Customer Support, i connected the asterisk to a e1 interface of our hipath4000. outgoing calls from a sip peer of my asterisk to an up0 telephone which iss connected to the hipath4000 are working. If you want to dial from an up0 device to the e1 interface where asterisk is connected to, you have to use the prefix 83. But when you enter the 3rd cipher this error appears at the cli

I've got a system with 5 pri circuits configured into a CFAS group with a primary and secondary d channel. There are three TE410P cards in the system. The 5 circuit span are located as follows: circuit 1 on span 5 circuit 2 on span 1 circuit 3 on span 6 circuit 4 on span 2 circuit 5 on span 9 primary d chan is on chan 24 of span 5 (chan 120) secondary d chan is on chan 24 of span 1 (chan

I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS modules. I'm trying to set-up things to route analog fax calls from a FXO port to an analog fax machine on a FXS port on the same card. Outgoing faxes work just fine. But incoming faces are routed to the right DAHDI extension, but the call dropped right as the fax machine rings for the first time. The fax machine

Sir, i am using goautodial server , bri card is showing ok but when i try to call that showing below , This configuration is in doing in dubai , so kindly help me how can connet the call from this , what is my mistake is in this :::chan-dahdi.conf [channels] #include dahdi-channels.conf language=en context=default usecallerid=yes hidecallerid=yes callwaiting=yes usecallingpres=yes

Hi all, As i tested with Asterisk+dahdi+libpri with openvox BRI with NT mode, the ISDN phone does not work. There is the setting and error. ***********************Enviroment********************************** Asterisk-1.6.1.18 dahdi-linux-2.3.0.1 dahdi-tool-2.30. libpri-1.4.11.2 CentOS-5.5 OpenVox B400P ******************************************************************** In my case , I set prot 1

Hello- After configuring DAHDI and starting asterisk, I get the following message continuously on the Asterisk console: WARNING[2057]: chan_dahdi.c:4158 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! My card is a D410P configured for E1, only the first span is configured, and configuration snippets are as follows: From /etc/dahdi/system.conf:

Hi All, I've a BT Versatility PBX that I want to connect to my asterisk 11.9.0 box via BRI port in NT mode but actually I wasn't able to get it working. I've another standalone PBX, it's a Panasonic model, which works fine connected to the same port. The BT connected to a UK BT BRI line works quite fine after roughly a minute once connected. The PBX can be seen at

We've been working for the past 2 weeks to get a new V400P working with our PRIs from the telephone company. We're trying to get the Asterisk server setup as a VoIP gateway for SIP and AIX. We can make SIP-SIP calls, but all calls from or to the PRI fail. This is the applicable entries from the Asterisk log (configuration files follow) for a call coming from the PSTN on the PRI. I