Re: [asterisk-users] problem starting asterisk, unable to load chan_zap

Hi, Two things come to mind, (1) being that you don't have the TE110P card jumped for an E1. (2) UDEV isn't creating the devices fast enough for the driver load. My guess is it's UDEV. You can test this theory by creating a startup script that loads the modules, put a sleep statement

Re: [asterisk-users] problem starting asterisk, unable to load chan_zap

Hi, Easy to check if the problem is udev: ls -l /sys/class/zaptel there are lots of subdirectories in there, at least after a rmmod modprobe. I can reboot later to see how it looks like before that. If there are files there and not under /dev/zap, udev is to blame. in /dev/zap there

[asterisk-users] problem starting asterisk, unable to load chan_zap

Hi all, I am running asterisk 1.2.18, zaptel 1.2.18, libpri 1.2.4. on a suse 10.2, running kernel 2.6.18.2-34-default. The zaptel drivers are loaded on boot via /etc/init.d/zaptel, but Asterisk is unable to start. It ends with the following message: [chan_zap.so] = (Zapata Telephony w/PRI)

Re: [asterisk-users] problem starting asterisk, unable to load chan_zap

Hi, When you type ztcfg -vvv, what does it display? After reboot it shows the following: asterisk1:~ # ztcfg -vvv Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01)

Re: [asterisk-users] problem starting asterisk, unable to load chan_zap

Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com wrote: Sebastian Reitenbach wrote: Hi, When you type ztcfg -vvv, what does it display? How about your zaptel.conf, zapata.conf and the snip of your dial plan

[asterisk-users] iax trunking on OpenBSD

Hi, do I have a chance to use iax trunking on OpenBSD where there is no zaptel driver or ztdummy available? Do I can use sth. else as timing source? kind regards Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

Re: [asterisk-users] iax trunking on OpenBSD

Hi Arun, Arun Kumar [EMAIL PROTECTED] wrote: you can use FreeBSD 6.1 its working fine for me with ztdummy and I'm able to use IAX2 trunk. thanks for the suggestion, but the OS is not really exchangeable. Before I am going to do this, I'd exchange asterisk with sth. else. kind regards

[asterisk-users] alive check for HA constellation

Hi, I have setup two asterisks with ucarp, to build a HA cluster. Everything works fine, if one of the machines is going to die completely. But if the asterisk software is running, but behaving not correctly, this cannot be detected by the ucarp software. I think I need a script that

Re: [asterisk-users] alive check for HA constellation

portugese. Sebastian Leonardo Silva [EMAIL PROTECTED] wrote: Hi Sebastian, This url http://underlinux.com.br/content/view/6330/70/ have some thinks that you need. Leonardo Silva 2006/10/10, Sebastian Reitenbach [EMAIL PROTECTED]: Hi, I have setup two asterisks

Re: [asterisk-users] overlapdial and DID not always working

Hi, I am wrong... sorry :-) pridialplan not switchtype! Ok... this is: the number on the PRI is always: XXXYYY where is the radical and YYY is the local number this is wath is working for me: switchtype = national pridialplan = unknown prilocaldialplan = unknown

[asterisk-users] overlapdial and DID not always working

Hi, another try with a hopefully better subject. I am here in Germany connected to the telephone system with a PRI interface: 00:0b.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface. I am using asterisk 1.2.7.1 and zaptel 1.2.5. To let DID work, I have set the options

Re: [asterisk-users] overlapdial and DID not always working

Hi, This is normal, some parameter cannot be changed with reload, the only way to change them is stop asterisk and restart (stop now, restart asterisk, you found all parameters correct). This message is like ouch you are reloading the configuration, but i cannot change this parameter by

Re: [asterisk-users] overlapdial and DID not always working

Hi, we have a problem, there is always the last digit missing. This is a 1-800 service in the US, forwarding the call to our asterisk. As a workaround I configured it to call to X580 and have an inbound route set for X58 to the number I want to reach. any idea what I can do?

Re: [asterisk-users] overlapdial and DID not always working

I found the same indentical problem, the trouble was the switchtipe, i am using national and i switched to unknown. is unknown allowed for switchtype? when I take a look here: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf then there is no unknown switchtype? jut

[asterisk-users] did sometimes not working

Hi, I am here in Germany connected to the telephone system with a PRI interface: 00:0b.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface. To let it work, I have read that I have to set the options overlapdial=yes and immediate=no in zapata.conf. here is my zapata.conf:

Re: [asterisk-users] did sometimes not working

Hi, forgot to mention, I have asterisk 1.2.7.1 and zaptel-1.2.5 running. Sebastian Sebastian Reitenbach [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com wrote: Hi, I am here in Germany connected to the telephone system with a PRI

Re: [asterisk-users] problems to call brazil from germany

? :) 2. Try describing the Asterisk behaviour under every circumstance. Regards On 7/17/06, Sebastian Reitenbach [EMAIL PROTECTED] wrote: Hi, I have problems to call to brazil, frome here in germany. the asterisk is connected to the telephone system via a pri interface. I use a preselected

Re: [asterisk-users] problems to call brazil from germany

Hi, Johann Steinwendtner [EMAIL PROTECTED] wrote: Sebastian, This is possible and most likley the reason. To make sure, check the location code of the cause IE in your ISDN disconnect message. I have a PRI interface, here ISDN with 30 channels. I am a bit unsure what you mean with the

[asterisk-users] problems to call brazil from germany

carriers have different ways to route the call to brazil and the preselection provider has not so many lines for overseas? kind regards Sebastian -- Sebastian ReitenbachTel.: ++49-(0)3381-8904-451 RapidEye AG Fax: ++49-(0)3381-8904-101 Molkenmarkt 30

Re: [Asterisk-Users] unable to set outgoing callerid

Hi, just answering myself: I am not allowed to send the leading 0 for my prefix with the callid, then it works well. Sebastian Sebastian Reitenbach [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: Hi *, now for a long time

[Asterisk-Users] unable to set outgoing callerid

Hi *, now for a long time i am trying to set the outgoing callerid, without luck. I am here in Germany, my asterisk has a pri interface connected to a PMX installed by Telekom. All telephone calls are preselected to EcoVoice. I am using asterisk 1.2.7.1, zaptel 1.2.5 and libpri 1.2.2. A week

[Asterisk-Users] some problems with asterisk and E1

Hi, I am using asterisk 1.2.5 and have some problems with asterisk connected with an E1 card to our PRI. Dialling in and out generally works. When someone dials in from a mobile phone, all numbers are sent as a block, and the called extension rings as intended. when someone picks up his phone

Re: [Asterisk-Users] Asterisk on BSD?

Hi, The subject says it all I think. I'm looking at maybe needing to run it under BSD 5 It runs fine on OpenBSD 3.8 No zaptel though, but for FreeBSD there's a zaptel port. http://ezine.daemonnews.org/200409/asterisk.html http://www.voip-info.org/tiki-index.php?page=Asterisk+FreeBSD

Re: [Asterisk-Users] cannot set outgoing cid

provider? Do you have a PRI or is it a POTS line? On 3/31/06, Sebastian Reitenbach [EMAIL PROTECTED] wrote: Hi, sorry for the long debug output below. I configured Asterisk with AMP to send the whole number including the extensions of the callers to the called party. Whatever I

Re: [Asterisk-Users] cannot set outgoing cid

provider? Do you have a PRI or is it a POTS line? On 3/31/06, Sebastian Reitenbach [EMAIL PROTECTED] wrote: Hi, sorry for the long debug output below. I configured Asterisk with AMP to send the whole number including the extensions of the callers to the called party. Whatever I

[Asterisk-Users] cannot set outgoing cid

to Sebastian Reitenbach Mar 31 16:53:56 DEBUG[11747] pbx.c: Expression result is '0' Mar 31 16:53:56 DEBUG[11747] pbx.c: Launching 'GotoIf' Mar 31 16:53:56 VERBOSE[11747] logger.c: -- Executing GotoIf(SIP/451-0e31, 0?5) in new stack Mar 31 16:53:56 DEBUG[11747] pbx.c: Not taking any branch Mar 31 16

[Asterisk-Users] asterisk doesn't wait for whole extension

Hi, maybe a dumb question, but it seems that some calls are directed to our central dial in number despite the extensions the callers say they dialled. E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown extension, where it is right, and redirects the call to the

Re: [Asterisk-Users] asterisk doesn't wait for whole extension

that only 12343 arrives at the asterisk. kind regards Sebastian Aaron Daniel [EMAIL PROTECTED] wrote: Can you post your dialplan? We'd be much better at troubleshooting the problem if we could follow the path that calls take. Aaron On Thu, 30 Mar 2006, Sebastian Reitenbach wrote: Hi

Re: [Asterisk-Users] asterisk doesn't wait for whole extension

Hi, thanks for answering my question. I used AMP to setup the dial plan. I have attached the extensions*.conf files created by it. My message was too large for the list, therefore i omitted the extensions_custom.conf. let me know if you need it. thanks for looking. kind regards Sebastian