[asterisk-users] SIP ATA Device Problems
I have an MG3 SIP ATA. This sip phone is registered and I am able to call the phone from another softphone. However, I am unable to place a call from the phone. In addition, after calling the SIP ATA phone the sip phone does not see to hang up the call in complete. Can anyone shed some light on this problem. 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] FXO PCI Master abort
So I'm all excited, ready to install Trixbox at home. Purchased my X100p card installed in a computer. I run Trixbox setup and boom I get this error message FXO PCI Master abort It repaets across the screen and I have to reboot. When I reboot the system hangs at adding hardware. Loading wcfxo and the system will not go any further. If anyone has an idea of something to try let me know. I have tried on two different computes. One compaq and One Dell system. I think the card is defective ir is a WildCard X100P(A) 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
RE: [Asterisk-Users] Asterisk 1.2.6, VMWare, Playback/Background GSM prompts
For what it is worth I had the same experience with VMWare server. I got better sound from my 5 year old workstation. From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED] Sent: Tue 3/28/2006 12:40 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk 1.2.6, VMWare, Playback/Background GSM prompts I've spent the past week experimenting with [EMAIL PROTECTED] 2.6, and then Asterisk 1.2.6 individually, on VMWare Workstation 5.5. I have an entirely IP (hard soft)phone setup (IAX and SIP) so I have no requirements to support any Digium PCI cards, etc. All in Asterisk works extremely well except for one thing: Playback of sounds (GSM format) such as an ivr greetings, sound terrible. Choppy, uneven, broken audio, etc to the caller. I have a fairly fast system as the host: dual core AMD X2, 2GB mem, running SMP 64-bit centos-4.2. Asterisk guest OS is centos-4.2 32bit. No other guests are running. Just for grins, I also installed Asterisk on a lowly 3-year old AMD Duron system (no VMWare). Using the same Asterisk config files and sound files as above, but in this case the audio playback (using the Asterisk 'background' command) sounds perfect. Anyone else notice this when using Asterisk with VMWare Ideas? 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] Exchange 12 Unified Messaging
Exchange 12 will support "OVA" Outlook Voice Access. It will do this using VOIP. I was wondering if anyone has given any thought as to how Asterisk might interface with Exchange 12. It will do this using a VOIP gateway. If this could happen it seems to me this could be a really big win for Asterisk. I am going to setup a test server as soon as I can and start digging into this. Thanks Jerry___ --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] Asterisk @ Home 2.6 Call hangs up
I have installed asterisk @
home 2.6. I am using a Telasip VOIP account. When I make outbound or
inbound calls the calls seem to connect and then get hung up. I was
wondering if there was something that I am misisng. I have tried several
different sip.conf configurations. Here is what they are
currently.
telasip-gw
context=telasip-indtmfmode=rfc2833fromuser=jrasxxxhost=gw4.telasip.cominsecure=verynat=yessecret=xyztype=peerusername=jrasxxx
551212
context=from-pstndtmfmode=rfc2833host=gw4.telasip.cominsecure=verynat=yesqualify=yessecret=xyztype=peerusername=jrasxxx
The odd thing is it worked once or twice then stopped. If anyone
could shed some light it would be greatly apperciated.
Here is what the asterisk output looks like:
-- AGI Script fixlocalprefix completed, returning
0 -- Executing SetVar("IAX2/100-2", "OUTNUM=770555")
in new stack -- Executing Cut("IAX2/100-2",
"custom=OUT_2|:|1") in new stack -- Executing
GotoIf("IAX2/100-2", "0?16") in new stack -- Executing
Dial("IAX2/100-2", "SIP/telasip-gw/770555") in new
stack -- Called
telasip-gw/770555 -- SIP/telasip-gw-3091 is
ringing -- SIP/telasip-gw-3091 answered
IAX2/100-2 == Spawn extension (macro-dialout-trunk, s, 14) exited
non-zero on 'IAX2/100-2' in macro 'dialout-trunk' == Spawn extension
(from-internal, 770555, 1) exited non-zero on
'IAX2/100-2' -- Executing Macro("IAX2/100-2",
"hangupcall") in new stack -- Executing
ResetCDR("IAX2/100-2", "w") in new stack -- Executing
NoCDR("IAX2/100-2", "") in new stack -- Executing
Wait("IAX2/100-2", "5") in new stack -- Executing
Hangup("IAX2/100-2", "") in new stack == Spawn extension
(macro-hangupcall, s, 4) exited non-zero on 'IAX2/100-2' in macro
'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero
on 'IAX2/100-2' -- Hungup 'IAX2/100-2'
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RE: [Asterisk-Users] Asterisk @ Home 2.6 Call hangs up
Thanks for the response Joseph. It ended up that Telasip needed to make a change on there end. They needed to disable re-invites. BTW, I wanted to give a big plug for Telasip. I thought when I called they would simply tell me it was my problem and they did not support asterisk. This was not the case at all. I recieved promt friendly curtious service. These guys had my problem fixed within 15 min of sending them my log file. I cannot say enought good things about them right now. From: [EMAIL PROTECTED] on behalf of Joseph Tanner Sent: Thu 3/9/2006 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk @ Home 2.6 Call hangs up My guess, is nat problems. Just for fun, try dialing your inbound number from something not connected to that asterisk box, say a cellphone. I know you're using IAX and SIP, so you'd think you wouldn't run into a double-nat problem (nat going out, nat coming in), but you never know. I have odd issues pop up sometimes when I try calling from my asterisk box right back into it, and I don't even have any nat in the way. Do outgoing calls generally work fine? How do incoming calls work when dialing from an outside line? For the heck of it, try calling out normally, and use a cellphone (or whatever) to dial into the asterisk box. Can it handle an outgoing AND incoming call at the same time, as long as it's not calling itself? If incoming calls still fail, then look into nat issues. Perhaps you can permanently forward port 5060 or 5061 (whichever you use, probably 5060) to your asterisk box, see if that helps any. May need to forward ports 1000-2000 as well. Joseph Tanner On 3/9/06, Jerry Rasmussen [EMAIL PROTECTED] wrote: I have installed asterisk @ home 2.6. I am using a Telasip VOIP account. When I make outbound or inbound calls the calls seem to connect and then get hung up. I was wondering if there was something that I am misisng. I have tried several different sip.conf configurations. Here is what they are currently. telasip-gw context=telasip-in dtmfmode=rfc2833 fromuser=jrasxxx host=gw4.telasip.com insecure=very nat=yes secret=xyz type=peer username=jrasxxx 551212 context=from-pstn dtmfmode=rfc2833 host=gw4.telasip.com insecure=very nat=yes qualify=yes secret=xyz type=peer username=jrasxxx The odd thing is it worked once or twice then stopped. If anyone could shed some light it would be greatly apperciated. Here is what the asterisk output looks like: -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar(IAX2/100-2, OUTNUM=770555) in new stack -- Executing Cut(IAX2/100-2, custom=OUT_2|:|1) in new stack -- Executing GotoIf(IAX2/100-2, 0?16) in new stack -- Executing Dial(IAX2/100-2, SIP/telasip-gw/770555) in new stack -- Called telasip-gw/770555 -- SIP/telasip-gw-3091 is ringing -- SIP/telasip-gw-3091 answered IAX2/100-2 == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'IAX2/100-2' in macro 'dialout-trunk' == Spawn extension (from-internal, 770555, 1) exited non-zero on 'IAX2/100-2' -- Executing Macro(IAX2/100-2, hangupcall) in new stack -- Executing ResetCDR(IAX2/100-2, w) in new stack -- Executing NoCDR(IAX2/100-2, ) in new stack -- Executing Wait(IAX2/100-2, 5) in new stack -- Executing Hangup(IAX2/100-2, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'IAX2/100-2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/100-2' -- Hungup 'IAX2/100-2' ___ --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 ___ --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 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
RE: [Asterisk-Users] Oneway voice
If your connection to the internet is being nated you may need to add this entry to your sip.conf externip=210.x.x.x From: [EMAIL PROTECTED] on behalf of ram Sent: Thu 3/9/2006 12:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Oneway voice Hi all I have installed AAH 2.6 created extension, and created Trunk created outbound routing iam able to make calls out and configured incoming, also working fine with the extension I have problem here I ahve extension sitting in same network where the AAH installed My provider support canreinvite=yes when iam making calls, its not consuming any b/w and voice quality is good in sip_additional.conf i have made in extension also canreinvite=yes another extension sitting another Country and he is behind nat here also made extension caninvite=yes i get one way Voice, later i have made the extension config( out side country extension) canreinvite=no the voice quality is good, but its taking 128Kb b/w how can i resolve this problem using g729 codec and save b/w thanks any suggestions ram 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] Asterisk @ Home 2.6 Call hangs up
I have installed asterisk @ home 2.6. I am using a Telasip VOIP account. When I make outbound or inbound calls the calls seem to connect and then get hung up. I was wondering if there was something that I am misisng. I have tried several different sip.conf configurations. Here is what they are currently. telasip-gw canreinvite=yescontext=telasip-indtmfmode=rfc2833fromuser=jrasxxxhost=gw4.telasip.cominsecure=verynat=yessecret=xyztype=peerusername=jrasxxx 551212 canreinvite=yescontext=from-pstndtmfmode=rfc2833host=gw4.telasip.cominsecure=verynat=yesqualify=yessecret=xyztype=peerusername=jrasxxx The odd thing is it worked once or twice then stopped. If anyone could shed some light it would be greatly apperciated. ___ --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] Advice on Asterisk-based home voicemail+fax+datasystem
Asterisk can do all this and more. I would suggest starting by using this project http://asteriskathome.sourceforge.net/. You can also check out this site http://www.asteriskdocs.org they just published a book and can download for free that is full of great information. BTW when I set up my server I only used a X100P card. This was more than enough to do all I wanted. This card is much cheaper than a TDM 400P. Good luck -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, November 13, 2005 7:00 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Advice on Asterisk-based home voicemail+fax+datasystem I took a look at vgetty as a solution for my home telephony needs, but the lack of documentation (at least beginner-level documentation) led me to give up. I have a couple of GNU/Linux (gentoo Debian) and FreeBSD boxes at home. I have a POTS line and a digital cable modem. I am not interested in dropping my local tone dial service for VOIP. I want to do the following: 1. Voice mail answering system. Message by person is a must if I am going to forward by email. 2. Voice mail is recorded as a .wav file (or similar) and either forwarded to email accounts or at least put in folder where it can be forwarded to the right user based on either the folder it is in or the message's file name. (So, I can write a script to email the message.) 3. Receive faxes and save them as a file in the same folder as the voice mail (so that I can forward it.) 4. not interfere with outgoing voice calls. 5. Act as a fax server and dial out faxes for home LAN pcs (linux, unix, windows) clients. 6. Not interfere with outgoing voice calls. *** less important *** 7. Be able to dial out data calls. 8. Act as PPP Internet connection for backup to cable Internet. 9. Be able to use Skype through my digital cable-based Internet connection. So, with these goals in mind, I started looking at Asterisk. It looks like I need some sort of hardware card like the Wildcard TDM 400P. I guess I am looking for some idea of whether I am barking up the wrong tree. Asterisk seems very alive and well while vgetty docs are much older, but it is a real PBX, not a home modem solution. Any suggestions on how I can piece together a reasonably priced solution to do the above? I am flexible on what hardware goes in. Any ideas/comments appreciated. Regards, Bud Roth ___ --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 ___ --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
RE: [Asterisk-Users] modprobe wcfxo fails.
Can you help me remember the details. Its been a while since I have touched the system. From: [EMAIL PROTECTED] on behalf of Tim KingSent: Sun 7/17/2005 3:09 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] modprobe wcfxo fails. I was reading a thread where you were helping someone out and noticed it ended without resolve. Was this issue ever taken care of?I seem to be having the exact same problem. Thanks Tim King Network Engineer Computer Network Solutions LLC 1331 Plainfield Ave Grand Rapids MI 49505 Phone: 800-669-3290 ___ 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
RE: [Asterisk-Users] X-Lite Softphone
Just to add my 2 cents I have had the best luck with Sjphone as well. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dana Olson Sent: Tuesday, February 15, 2005 3:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] X-Lite Softphone I've had the best luck with SJphone. Using a USB headset really helps as well, opposed to the sound card or onboard audio with a standard headset. I don't like the skins that come with most softphones, and XLite is no exception. SJphone lets me disable it, and the profiles are nice, since you can copy them from computer to computer. I haven't tried XPro or eyeBeam. -- Dana On Tue, 15 Feb 2005 08:56:48 -0800, Richard J. Sears [EMAIL PROTECTED] wrote: Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk box and had the same luck. My laptop is the Dell XPS, so power, ram, etc are not problems, and loading it onto my desktop system revealed the same results. There was also no difference between a NAT implementation and a regular (live IP) implementation of the software. I am getting stuttering speech, cutouts, etc all the time. Running my Cisco 7960 at the same locations and it works fantastic with no issues at all. Is anyone else using this softphone or does anyone know of a better softphone or some hints on configuration that may make X-Lite work better..? TIA ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ 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 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 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
RE: [Asterisk-Users] On Hold music
This may sound kind of crazy and I maybe missing something. But are you placing the call on hold so you can hear the hold music. This may not be the case but you may have to place the call on hold to here the music. Also you mentioned sound, you do not need a sound card in the asterisk box to use this hold music feature. Hope this helps. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Computer Onsite Support Sent: Monday, January 17, 2005 10:51 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] On Hold music Please be more specific regarding symbolic link of mpg321 so I can troubleshoot it myself. The strength thing is that I tried this in three other different computers and can't get it to work using same installation guide was able to get it to run on a PIII 500 which I want to get rid of it now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matt Riddell Sent: Monday, January 17, 2005 10:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] On Hold music Computer Onsite Support wrote: Thanks you but that didn't work. Any other solutions? Make sure you are using the correct version of mpg123... I think from memory the correct version is 0.59r Also, I think Redhat has simply made a symlink to mpg321 (which is *not* the same). -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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 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 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] zapata.conf not being parsed by *
I am running * 1.0.3 for some reason when I start * is does not appear to be parsing my zapata.conf file. I do not see any errors * just does not seem to know to look for zapata.conf. I am unable to use my FXO card to make calls or receive calls. I have been able to configure SIP to work correctly. Any help would be greatly appreciated, I spent most of last night searching for an answer. Thanks Jerry ___ 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
RE: [Asterisk-Users] zapata.conf not being parsed by *
Also when I try to dial outbound I get the following errors channel.c:1920 ast_request: No channel type registered for 'Zap' and Unable to create channel of type 'Zap' (cause 66). My assumption is I am getting these errors because Zapata.conf is not being parsed From: [EMAIL PROTECTED] on behalf of Leif Madsen Sent: Wed 12/29/2004 2:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] zapata.conf not being parsed by * On Wed, 29 Dec 2004 13:59:19 -0500, Jerry Rasmussen [EMAIL PROTECTED] wrote: I am running * 1.0.3 for some reason when I start * is does not appear to be parsing my zapata.conf file. I do not see any errors * just does not seem to know to look for zapata.conf. I am unable to use my FXO card to make calls or receive calls. I have been able to configure SIP to work correctly. I've not seen or heard of that before... but the first thing that comes to mind would be some module not being loaded in modules.conf? Since no one has responded yet, thought I'd throw out a shot in the dark :) Leif Madsen. http://www.leifmadsen.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 winmail.dat___ 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
RE: [Asterisk-Users] zapata.conf not being parsed by *
You know I think the I compiled them in the wrong order. I bet you that is it. I will give it a try and let you know. From: [EMAIL PROTECTED] on behalf of Eric Wieling aka ManxPower Sent: Wed 12/29/2004 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] zapata.conf not being parsed by * Jerry Rasmussen wrote: Also when I try to dial outbound I get the following errors channel.c:1920 ast_request: No channel type registered for 'Zap' and Unable to create channel of type 'Zap' (cause 66). My assumption is I am getting these errors because Zapata.conf is not being parsed Or you have a noload = chan_zap.so in /etc/asterisk/modules.conf or you installed Asterisk before you installed zaptel. Install zaptel before you install Asterisk or the chan_zap modules won't be built. You should also confirm that ztcfg -vvv shows your card and the correct ports. --Eric ___ 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 winmail.dat___ 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
RE: [Asterisk-Users] zapata.conf not being parsed by *
That was it. I compiled them in the wrong order. You know sometimes it really pays to read the instructions From: [EMAIL PROTECTED] on behalf of Jerry Rasmussen Sent: Wed 12/29/2004 3:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] zapata.conf not being parsed by * You know I think the I compiled them in the wrong order. I bet you that is it. I will give it a try and let you know. From: [EMAIL PROTECTED] on behalf of Eric Wieling aka ManxPower Sent: Wed 12/29/2004 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] zapata.conf not being parsed by * Jerry Rasmussen wrote: Also when I try to dial outbound I get the following errors channel.c:1920 ast_request: No channel type registered for 'Zap' and Unable to create channel of type 'Zap' (cause 66). My assumption is I am getting these errors because Zapata.conf is not being parsed Or you have a noload = chan_zap.so in /etc/asterisk/modules.conf or you installed Asterisk before you installed zaptel. Install zaptel before you install Asterisk or the chan_zap modules won't be built. You should also confirm that ztcfg -vvv shows your card and the correct ports. --Eric ___ 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 winmail.dat___ 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] Digium TDM11B
Does anyone have a Digium TDM11B they would like to part with? I would be interested in purchasing such a card. I am new to * and would like to have this card to practice with. If you are interested in selling this card please contact me off list. P.S. I would be interested in any card in the TDM400P series. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FC2 zaptel compile failure
I would suggest two things. (Note I am not an expert) 1. Upgrade the Kernel. I had to do this to get mine to complile correctly 2. Have only one symlink like this one only pointing to the new kernel /lib/modules/linux-2.6 - /lib/modules/2.6.7-1.494.2.2 And of course good luck -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff Borders Sent: Friday, September 17, 2004 3:41 PM To: Asterick Users Subject: [Asterisk-Users] FC2 zaptel compile failure I've got a fresh FC2 install and I'm trying to get the symlinks right according to the /usr/src/zaptel/README.Linux26 instructions. I've created two symlinks: /usr/src/linux-2.6 - /usr/src/linux-2.6.5-1.358 /lib/modules/linux-2.6 - /lib/modules/2.6.7-1.494.2.2 When I do a make linux26, I get a million warnings and errors with the result being: make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory '/usr/src/linux-2.6.5-1.358' make: *** [linux26] Error 2 Am I looking at the right thing? Or do I have another problem? Jeff Borders [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RC2 zaptel compile problem
What I believe you want is this.
ln -s /lib/modules/2.6.5-1.358/build linux-2.6 Only pointing to your
kernel. Run this in the /usr/src directory
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
Borders
Sent: Wednesday, September 15, 2004 7:46 AM
To: Asterick Users
Subject: [Asterisk-Users] RC2 zaptel compile problem
I'm a newbie with a TDM11B. I've read the FAQs about linking
/usr/src/linux-2.6 to /usr/src/linux-2.6.8-1.521 and /lib/linux-2.6 to
/lib/linux-2.6.8-1.521 but still get a million errors and eventual abort
during compile.
Could someone point me in the right direction? I do a yum update every
day and I'm using the CVS from 9/14/04. I'm also using an ASUS CUSL2
system board. Here's the end of the error, although I don't think it's
useful.-Jeff Borders (jeffATjeffbordersDOTcom)
/usr/src/zaptel/zaptel.c:6203: warning: excess elements in struct
initializer
/usr/src/zaptel/zaptel.c:6203: warning: (near initialization for
`zt_fops')
/usr/src/zaptel/zaptel.c:6204: error: unknown field `flush' specified in
initializer
/usr/src/zaptel/zaptel.c:6204: warning: excess elements in struct
initializer
/usr/src/zaptel/zaptel.c:6204: warning: (near initialization for
`zt_fops')
/usr/src/zaptel/zaptel.c:6205: error: unknown field `fsync' specified in
initializer
/usr/src/zaptel/zaptel.c:6205: warning: excess elements in struct
initializer
/usr/src/zaptel/zaptel.c:6205: warning: (near initialization for
`zt_fops')
/usr/src/zaptel/zaptel.c:6206: error: unknown field `fasync' specified
in initializer
/usr/src/zaptel/zaptel.c:6206: warning: excess elements in struct
initializer
/usr/src/zaptel/zaptel.c:6206: warning: (near initialization for
`zt_fops')
/usr/src/zaptel/zaptel.c: In function `zt_prechan_ioctl':
/usr/src/zaptel/zaptel.c:4188: warning: unreachable code at beginning of
switch statement
/usr/src/zaptel/zaptel.c: At top level:
include/linux/elf.h:433: warning: array `_DYNAMIC' assumed to have one
element
include/linux/sched.h:227: error: storage size of `mmap_sem' isn't known
include/linux/sched.h:261: error: storage size of `default_kioctx' isn't
known
include/linux/sched.h:286: error: storage size of `shared_pending' isn't
known
include/linux/sched.h:378: error: storage size of `wall_to_prev' isn't
known
include/linux/sched.h:502: error: storage size of `rlim' isn't known
include/linux/sched.h:510: error: storage size of `thread' isn't known
include/linux/sched.h:522: error: storage size of `pending' isn't known
include/linux/stat.h:68: error: storage size of `atime' isn't known
include/linux/stat.h:69: error: storage size of `mtime' isn't known
include/linux/stat.h:70: error: storage size of `ctime' isn't known
include/linux/fs.h:276: error: storage size of `ia_atime' isn't known
include/linux/fs.h:277: error: storage size of `ia_mtime' isn't known
include/linux/fs.h:278: error: storage size of `ia_ctime' isn't known
include/linux/quota.h:224: error: storage size of `dq_dqb' isn't known
include/linux/fs.h:356: error: storage size of `bd_sem' isn't known
include/linux/fs.h:357: error: storage size of `bd_mount_sem' isn't
known
include/linux/fs.h:431: error: storage size of `i_atime' isn't known
include/linux/fs.h:432: error: storage size of `i_mtime' isn't known
include/linux/fs.h:433: error: storage size of `i_ctime' isn't known
include/linux/fs.h:440: error: storage size of `i_sem' isn't known
include/linux/fs.h:441: error: storage size of `i_alloc_sem' isn't known
include/linux/fs.h:447: error: storage size of `i_data' isn't known
include/linux/fs.h:574: error: storage size of `f_owner' isn't known
include/linux/fs.h:745: error: storage size of `s_umount' isn't known
include/linux/fs.h:746: error: storage size of `s_lock' isn't known
include/linux/fs.h:773: error: storage size of `s_vfs_rename_sem' isn't
known {standard input}: Assembler messages:
{standard input}:747: Error: symbol `rv' is already defined {standard
input}:761: Error: symbol `rv' is already defined {standard input}:768:
Error: symbol `c' is already defined {standard input}:774: Error: symbol
`rv' is already defined
include/linux/pci.h:515: error: storage size of `dev' isn't known
/usr/src/zaptel/zaptel.c:6194: error: storage size of `zt_fops' isn't
known
include/linux/proc_fs.h:210: warning: `create_proc_read_entry' declared
`static' but never defined
/usr/src/zaptel/zaptel.c:3671: warning: `recalc_maxlinks' defined but
not used
/usr/src/zaptel/zaptel.c:589: warning: `zt_first_empty_conference'
defined but not used
/usr/src/zaptel/zaptel.c:600: warning: `zt_get_conf_alias' defined but
not used
/usr/src/zaptel/zaptel.c:982: warning: `free_tone_zone' defined but not
used
/usr/src/zaptel/zaptel.c:2338: warning: `ioctl_load_zone' defined but
not used
/usr/src/zaptel/zaptel.c:2690: warning: `zt_common_ioctl' defined but
not used
/usr/src/zaptel/zaptel.c:2949: warning: `recalc_slaves' defined but not
used {standard input}:5367: Error: symbol `default_zone' is already
defined
RE: [Asterisk-Users] Final Help on setting up x100p
I am new to this as well and I want to be 100 % sure. I can, using only a x100p card, have an advanced Voicemail server that will answer my PSTN line and use it as a SIP server. I am trying to learn * in hopes that I can convince my company to use it one day. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Sunday, September 12, 2004 2:19 PM To: Rodolfo Grave Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Final Help on setting up x100p The Phone port wired to the Line port so you can still use a phone plugged into the card when the server is down or powered off. Let me repeat this: You cannot plug a phone into the X100P and expect it to work with Asterisk. On Sun, 2004-09-12 at 13:09, Rodolfo Grave wrote: But I see there is a plug for LINE and another for PHONE. and also I'm almost sure I read you could connect a normal phone to the x100p... Eric Wieling wrote: The X100P is for connecting to phone LINES (telco lines) not for connecting to phones. FXO = expects to RECEIVE dialtone and ring voltage FXS = expects to PROVIDE dialtone and ring voltage. On Sun, 2004-09-12 at 12:59, Rodolfo Grave wrote: Hi. I have installed a x100p (THE x100p for those who have seen my former post). Now I just want to connect a normal phone (not an IP phone) to the card and use it as a sip extension (I have a FWD account)... more clearly: I want to be able to pick up the phone and call any FWD user using my FWD account... receive the FWD calls in that phone, and also to be able to make normal PSTN calls... I suppose that's possible (I find it very simple for the things you can do with asterisk), but I'm a bit lost in the configuration options and I want to use this simple settings to go in there and understand all that better I do have read the documentation, but I'm still a bit lost. Thanks to everybody, you're a helpful community. RODOLFO --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 0437-1, 09/09/2004 Tested on: 12/09/2004 19:59:19 avast! - copyright (c) 2000-2004 ALWIL Software. http://www.avast.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 0437-1, 09/09/2004 Tested on: 12/09/2004 20:09:51 avast! - copyright (c) 2000-2004 ALWIL Software. http://www.avast.com -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
