similar to: Cannot dial on E1 cards

Hi All, i'm going crazy trying to make asterisk work with the following hardware: 02:05.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) Subsystem: Cologne Chip Designs GmbH ISDN Board Flags: bus master, medium devsel, latency 16, IRQ 17 I/O ports at d000 [disabled] [size=8] Memory at ec000000 (32-bit, non-prefetchable)

Hello List, I am stuck with this problem for several days... anybody can give me a hint on this?? I know many of you dealt with problems similar to this, how did you address this?? Thanks in advance!!! -lars ---------- Forwarded message ---------- From: Lars Knopf <lars.knopf@gmail.com> Date: Jan 11, 2007 1:12 PM Subject: realtime sipusers and rtcachefriends... big headache!! To:

hi all i am using CVS with Realtime mysql on backend. Dial application is invoked again and again what is the reason. I have tested it by printing some message to debug. this application is invoked again and again here is debug you can see lot of messages from app_dial.c at the end. Any one tell me what is the reason. Is this a bug or what Kamran Ahmad

Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work

Hello, i added a existing MAILDIR as a Public Namespace and i can subscribe to this Namespace without a problem. I can access every folder except the INBOX. Thunderbird just shows a GREY "INBOX". Is there any way to access the INBOX of a maildir? Thanks Dieter Knopf

After updating to 1.4.15, I have the following issue: When I try to use the "M" macro option in the Dial() option, I get the following in the console: -- Executing Dial("Zap/1-1", "Zap/g2/w5051234|60|M(set-userfield^local)KT") -- Called g2/w5051234 -- Zap/3-1 answered Zap/1-1 [Dec 6 12:10:58] ERROR[19496]: app_dial.c:1541 dial_exec_full: Unable to start

[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer '3516533812' is now UNREACHABLE! Last qualify: 86 [Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke: Peer '3516533812' is now Reachable. (98ms / 2000ms) [Apr 26 12:08:49] WARNING[32273]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Hi. I'm looking at ordering a 30-channel ISDN connection from telia (a swedish operator) and then using a Wildcard TE110P card with that and asterisk to do IVR. Can I be certain that the TE110P card will work with that ISDN connection? A 30 channel ISDN certainly sounds like an E1 connection, but I couldn't get any clear answers from the operator if it is. Has anyone used the TE110P

Hi, I'm having a problem with the queue behaviour in my place: I have two ISDN channels to the outside (Zap/1) and two channels two a Siemens Gigaset (Zap/4). I also use a SIP gateway to call outside and have a couple of IP phones around as well (SIP). The Gigaset has about 5 phones connected to it (+base station). Whenever two people are using those, I always am blocking two internal

Hello, I am trying to do : Exten =>_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:10000)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:10000)' Why? I forgot something ? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que

Hi there I have a TE110P card fitted in my linux box running : Linux version 2.6.9-5.ELsmp (bhcompile@decompose.build.redhat.com) (gcc version 3.4.3 20041212 (Red Hat 3.4.3-9.EL4)) #1 SMP Wed Jan 5 19:30:39 EST 2005 I followed the installation steps on digium website...no errors reported. The modules seem to have loaded...here's what lsmod shows: Module Size Used by

Hi All After a Dial() I get: WARNING[7964][C-000075a8]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) if the subscriber is not registered. Is there a way from dialplan to know, *before* Dial(), if a destination Subscriber is a) not registered or b) busy ? I need to redirect a call to some other Subscriber if (s)he is not there

Hi, This error I got it just when I gonfigure zaptel support isdneuro 31 channels. But if I configure zaptel to support T1 and just 24 channels I have no problem. ######################################################################## #### # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # loadzone = it span=1,1,0,ccs,hdb3,crc4 bchan=1-23 dchan=16

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

I have an Asterisk system with a TE110P installed and connected to an ISDN E1 PRI that is delivered via a 2mb SHDSL connection. I am experiencing delays (the type of delay you would get on an international call) during calls. I am wondering if anyone could advise, would the problem be with any part of the Asterisk system or is the problem with the fact that the ISDN is delivered over the internet?

Hi, During tests with a IAX2/PSTN gateway I've been getting strange results for processor idle time and load. I (re)search(ed) this issue for a while, but I didn't get any good explainations. Can somebody help me? I have several sites that rely on a central server for connection to the PSTN. Calls to the PSTN are routed over the Internet to this PSTN gateway using IAX2 in trunk mode. To

All This is a small issue that I stumbled onto that has to do with the channel numbering on an E1 connection into an Asterisk Zaptel/DAHDI system. As most of us already know an E1 has 32 channels of which 30(1-15 17-31) are B-channels and 1 (16) is a D-Channel. The 32nd channel is not presented in Asterisk Zaptel/DAHDI. There are other configurations but this is the most common. ***

Hi, I have an Asterisk 1.2 (can`t upgrade to 1.4 because of some makefile error on my particular system, bug report opened). That being said, I doubt my particular issue is a bug, I think it's me not understanding something. Let`s take a simple dialplan command, i.e. make the phone ring for 15 seconds: Dial(SIP/some_sip_registration|15) It works well (most of the time). If I disconnect

Hi, when I'm trying to call the following extension exten => 6002,1,Verbose(1|Extension 6002) exten => 6002,n,Dial(Agent/6002) exten => 6002,n,Hangup() the call is terminated and I get the following warning from asterisk: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) When calling the agent with Dial(SIP/6002) no problem

Has anyone been able to successfully solve the following issue: WARNING[21725]: channel.c:3024 ast_request: No channel type registered for 'Zap' [Mar 11 01:26:53] WARNING[21725]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) Since we updated asterisk from 1.2.13 to asterisk 1.2.16 the module went away so we updated