Re: [asterisk-users] ZAP device reference in Zaptel 1.4 - SIMILAR
I used the packages that were mentioned and have a link on the main website and www.asterisk.org/download I thought these were ok for production/use... I just compiled 1.4.1 (./configure and make, no make install) and copied the chan_zap.so module into /usr/lib/asterisk/modules, restarted asterisk (the 'old' 1.4.2) and it seems to work now. zap show channels gives me one channel, and that's correct, and when dialing in, the cli gives me status on the channel and the incoming calls... I just wanted to let you guys know how I resolved it (for now), and see if anyone knows when they are/were going to fix this issue? (1.4.3 or 1.4.2.x??) Thanks for the help!! Bram On Wed, Apr 04, 2007 at 11:51:21PM +0200, bram kortleven wrote: Well, I'm experiencing a similar problem with my setup... debian etch, asterisk 1.4.2, zaptel 1.4.1, ... I cannot find the chan_zap.so module file anywhere, tried recompiling with zaptel 1.4.0... no change... I tried 'make menuselect', and going to the channels-part, chan_zap is marked XXX - dependencies missing: and this is the message for it, as an explanation. Zapata Telephony Depends on: zaptel_vldtmf(E), zaptel(E), tonezone(E) Can use: pri Anyone any idea how to resolve Do you use the packages from Experimental? If so: indeed some of them were lacking chan_zap.so . Hwever that shoul have been fixed on the latest version (through a build-dependency on zaptel = 1.4.1 ) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP device reference in Zaptel 1.4 - SIMILAR
Well, I'm experiencing a similar problem with my setup... debian etch, asterisk 1.4.2, zaptel 1.4.1, ... I cannot find the chan_zap.so module file anywhere, tried recompiling with zaptel 1.4.0... no change... I tried 'make menuselect', and going to the channels-part, chan_zap is marked XXX - dependencies missing: and this is the message for it, as an explanation. Zapata Telephony Depends on: zaptel_vldtmf(E), zaptel(E), tonezone(E) Can use: pri Anyone any idea how to resolve this?? Thanks On Wed, Apr 04, 2007 at 10:11:01AM +0300, Tzafrir Cohen wrote: Hi On Wed, Apr 04, 2007 at 04:59:41PM +1000, Devraj Mukherjee wrote: monk*CLI zap show channels No such command 'zap show' (type 'help' for help) Does that mean I dont have ZAP support in Asterisk? Maybe. ls -l /usr/lib/asterisk/modules/chan_zap.so I also repeat my second question: What is the contents of /etc/asterisk/zapata.conf ? Follow-up: The issue seems to be an issue with the atrpms package: http://bugzilla.atrpms.net/show_bug.cgi?id=1165 Asterisk 1.4.2 is missing chan_zap.so ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE: [asterisk-users] ZAP device reference in Zaptel 1.4 - SIMILAR
Yes I did, I even started all over again using the original source tar.gz packages... The menuselect thing bothers me the most... something is wrong or at least, missing. When compiling, I see all chan_... things passing, but no chan_zap... Hate it ... Thanks for the help - Original Message: After recompiling zaptel, did you recompile Asterisk? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bram kortleven Sent: Wednesday, April 04, 2007 14:51 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ZAP device reference in Zaptel 1.4 - SIMILAR Well, I'm experiencing a similar problem with my setup... debian etch, asterisk 1.4.2, zaptel 1.4.1, ... I cannot find the chan_zap.so module file anywhere, tried recompiling with zaptel 1.4.0... no change... I tried 'make menuselect', and going to the channels-part, chan_zap is marked XXX - dependencies missing: and this is the message for it, as an explanation. Zapata Telephony Depends on: zaptel_vldtmf(E), zaptel(E), tonezone(E) Can use: pri Anyone any idea how to resolve this?? Thanks On Wed, Apr 04, 2007 at 10:11:01AM +0300, Tzafrir Cohen wrote: Hi On Wed, Apr 04, 2007 at 04:59:41PM +1000, Devraj Mukherjee wrote: monk*CLI zap show channels No such command 'zap show' (type 'help' for help) Does that mean I dont have ZAP support in Asterisk? Maybe. ls -l /usr/lib/asterisk/modules/chan_zap.so I also repeat my second question: What is the contents of /etc/asterisk/zapata.conf ? Follow-up: The issue seems to be an issue with the atrpms package: http://bugzilla.atrpms.net/show_bug.cgi?id=1165 Asterisk 1.4.2 is missing chan_zap.so -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070404/de4d4c46/attachment-0001.htm -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] x100p not showing in core show channels
Hi, I recently decided to change my setup from AsteriskNow to plain-asterisk 1.4, which I wanted to set up and configure myself on a server running Debian Etch 64bit version. Hardware: Asrock motherboard, model 775Dual880-Pro, with a Celeron D running at 2.8GHz, 1GB memory, standard Nvidia GF4MX videocard, and one X100P clone card. Running the AsteriskNow, everything worked fine, except for incoming calls, not being routed right, but they entered the system, and mostly they ended up into the voicemail. To change that behaviour and to have more control of what I'm doing, I reinstalled the same machine with Debian Etch, the 64bit version, as the CPU (and the replacing one in a few weeks) runs EM64T nicely... The setup ran OK, compilation etc too, except for zttool which I still cannot compile. When configging the server, I used several HowTo's and guides, but no solution: hereby the config parts that I did/changed: /etc/zaptel.conf: # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # This file is parsed by the Zaptel Configurator, ztcfg # It must be in the module loading order # Span 1: WCFXO/0 Generic Clone Board 1 fxsks=1 # Global data loadzone= be defaultzone = be /etc/asterisk/zapata.conf: [channels] signalling=fxs_ks group=1 context=incoming channel=1 ;X100P /etc/asterisk/extensions.conf: [incoming] exten = s,1,Echo ;for testing the connection ;exten = s,1,Playback,demo-thanks ;for playing a file Nothing happens when dialing in. BUT: ztmonitor 01 -vv gives levels, and when dialing in, the levels change according to dialtone in my phone I use for calling the server. AND: core show channels gives me this: asterisk*CLI core show channels verbose Channel Context ExtensionPrio State Application Data CallerIDDuration Accountcode BridgedTo 0 active channels 0 active calls What am I doing wrong??? Anyone that can give a hand? Thanks! Just email me! bram_at_antwerpen_dot_be ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskNow - H323 support for trunks
First of all, I was wondering if this is the right place to post questions and issues around AsteriskNow. If so, here's my question: I've been using Iphonecom.com's services for abour 3-4years now, to connect and call my brother who lives in the US and my parents, sister and me (all located in Belgium). We're OK with their services, but we wanted to be more free in call-numbers, options, ... So I decided to give Asterisk a go, and try to host these things myself (dedicated dsl line, dedicated server, ...) That way, we could also use my landline here to let my brother use the belgian phone system to call other relatives and friends, after-working-hours, which is free overhere. (for me that is). Now, Iphonecom.com only uses H323 as they mentioned 'they have too many problems with SIP thus sticking with H323 for the moment to guarantee everyone's service'... I was wondering if I can use this account in my Service Providers tab, and connect through my account, and receive my brothers call, before migrating him over to my Asterisk system. Just as a test-scenario... Anyone tried or did this before, pref'd with AsteriskNow. My knowledge about asterisk (especially configging) is rather basic, but I'm willing to learn ;) Thanks people! Great product, great user-base support! Love it!!! Bram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] menu system - configurator
We are currently looking for a way to easily configure a 'auto attendant' system on our asterisk pbx. More in detail, I'm looking for a webbased (or something similar) configuration generator, that has a feature like asking me how many 'menu levels' I want, what text to play, and in the first, how many items I want, and then per item what text and description it has to set, etc ... Is there something out there that does something similar? Or does anyone know how to make such a script? If possible, we prefer mysql-driven menu's... as all other stuff is in mysql already... Thanks guys!!! Bram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call limit function on sip channel to external pop
Hi, We've been using asterisk as our main telephone-communications platform for years now, and we wrote several extra scripts and features for it. Now we 're looking for a solution to limit the number of channels going to an external SIP provider. We recently upgraded our system from asterisk 1.0 to 1.2(.9 now) to be able to use such features, but nothing helped... When we configure a new channel, it seems to work, but putting the call_limit on an existing sip channel going out, it doesn't do anything. Anyone already had such an issue, or anyone knowing the best config for limiting outgoing sip channels to external sip providers? It's kind of urgent... Thanks in advance, and keep up the good work! Bram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Huawei EP201S
Did you check the Grandstream BudgeTone series? They're really cost-effective, and have good sound quality, and are perfectly supported by Asterisk. We used them for over 2 years now, and no problems to be mentioned. In belgium, they cost around EUR80, so that's about US$95-100, I think. (don't shoot me if I'm wrong!) This is the link: http://www.grandstream.com/y-bt100.htm Let me know what you're going to choose and how it ends up Bram Has anybody used Huawei EP201S IP phone? I need 9 SIP phones that cost around 100USD, and those phones are one of options. Can anybody suggest anything else that costs around 100USD? -- Tomislav Parhina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr http://www.lama.hr/ winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie question on 1.2 extension configs
Are there any example configs? Or does anybody have a default config for this setup: 1 analog digium clone card for an analogue line (my home line) Several sip phones (a few of them on the outside of my lan (NAT fw between) and 2 insde my lan) Or a simple way of configging through a frontend/script/management utility... I installed astlinux But it does not allow to install and use AMP... Anyone having another script? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie question on 1.2 extension configs
Are there any example configs? Or does anybody have a default config for this setup: 1 analog digium clone card for an analogue line (my home line) Several sip phones (a few of them on the outside of my lan (NAT fw between) and 2 insde my lan) Or a simple way of configging through a frontend/script/management utility... I installed astlinux But it does not allow to install and use AMP... Anyone having another script? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: Asterisk 1.0/1.2 on cobalt Raq2-4
Title: Asterisk 1.0/1.2 on cobalt Raq2-4 Anyone ever tried to install on a Cobalt Raq device? This is a 1U 19" rack computer, the Raq2 using a Mips processor, the Raq4 using a K6-2/3 processor.As I do have a few of these as spares, I was wondering if I could use them as my pbx system, because of their low power-system and dence system box.I simply need the pbx to serve 2 phones in my appartment, a SIP- connection for 4 external internet devices (my brother, living in the USA, my parents, living a few miles from here, and my nefew living in France and his mother, living here in Belgium too)Has anyone done this setup on a Raq2? Or do I need to use the extra power of the Raq4 (faster cpu and mem, bigger faster disk, ...)Anyone having a pkg-installer for the raq devices, as they are used for updates etc...?ThanksBram ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0/1.2 on cobalt Raq2-4
Anyone ever tried to install on a Cobalt Raq device? This is a 1U 19 rack computer, the Raq2 using a Mips processor, the Raq4 using a K6-2/3 processor. As I do have a few of these as spares, I was wondering if I could use them as my pbx system, because of their low power-system and dence system box. I simply need the pbx to serve 2 phones in my appartment, a SIP- connection for 4 external internet devices (my brother, living in the USA, my parents, living a few miles from here, and my nefew living in France and his mother, living here in Belgium too) Has anyone done this setup on a Raq2? Or do I need to use the extra power of the Raq4 (faster cpu and mem, bigger faster disk, ...) Anyone having a pkg-installer for the raq devices, as they are used for updates etc...? Thanks Bram ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie question
I guess the first time it didn't get through... I didn't see it appear in the list, that is... I installed an [EMAIL PROTECTED] machineand configured a few SIP accounts on it. They seem to run fine inside my network, so that's OK. Now, I want to start using a X100P to connect it to my phone line, to make call routing between internal SIP phones/softphones, my local phoneline and an external SIP server. How do I enable and configure the X100P? I ran the configuration tool locally on the machine (the genzaptelconf thing) and it added a line to the config. Now using the number it gave me, in the trunk config in AMP, I still cannot get an outside line (connected it to a simple analogue pbx system) and call outside the *-server.. Could anyone help me with this? Thanks guys ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
