Re: [asterisk-users] enable eyeBeam to accept only one call
Joao Pereira wrote: Hello I'm using eyeBeam, and Asterisk keeps sending my clients a second call, when they are still in one call (because eyeBeam has lots of channels). I was using X-Lite (with 3 channels) and Asterisk never sent the client a second call. How can I force Asterisk (or eyeBeam) just to send one call at each time. Is this a configuration I need to do in eyeBeam or Asterisk? Thanks Regards Joao Pereira try set limit=1 in the sip.conf file for the eyebeam extension ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] over 200 queues, anyone?
Lenz wrote: You are correct, this is more or less the scenario involved - the problem is that people want to call a personalized line AND speak to the same subset of agents preferably. I have never seen such a setup myself - I have seen CCs with 30 or 40 queues, never 200 - so I was wondering if anybody ever trued something on these lines; or if there are better solutions to the same problem. Best regards l. Hi Lenz We are doing this here is RSA. What i have done is use the DID trigger number to play a specific greeting for the customer. Then get the customer to route to their preferred language. We have close on 300 customers and 5 Q's with +/- 100 agents. Regards Terry ___ --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 + door opener
Dovid B wrote: can u get me the info on the part ? Hi Guys I have found this. Have not tested as yet, but have asked them for some more info. Might be of some help. www.its-tel.com Cheers Terry ___ --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] snom 360: how to make record button working ?
Another way would be to set the dtmf option to speed dial and then add a speed dial number 1: *1 ___ --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] Call to a queue killing Asterisk?
Avi Miller wrote: Hey guys, Last week I changed my queues from using proper agents and AgentCallbackLogin() to using the the FreePBX default with fixed agents (which uses the Local/[EMAIL PROTECTED] style for the member= field). I've also upgraded to Asterisk 1.2.10 and FreePBX 2.2.0 Beta 1. Since then, I noticed that my FOP would sometimes get stuck when a call hit the queue (showing all the agents being busy forever, until a op_server.pl reload). I started to track it this morning and actually saw Asterisk shutdown as the call got answered (and get restarted by safe_asterisk, of course). This accounts for the stuck FOP, but now I have the joy of working out why Asterisk is shutting down. I don't see anything in /var/log/asterisk/full -- I see the mysql CDR being recorded and then 4 seconds later, I see the Asterisk startup sequence happening. Anyone have any suggestions on where to start debugging this? Thanks, Avi Check http://bugs.digium.com/view.php?id=6626 cheers Terry ___ --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 version: 1.2.9.1 or older?
Don Pobanz wrote: I am running 1.2.9.1 and did not have any problems until setting up queues. Within a day of doing queue logins/logouts our T1 DID trunks (not PRI) stopped accepting calls from the local telco. Internal calls though channel banks continued to function properly. A restart would clear the situation. This happened on three separate occasions. After that, I got smarter and stopped doing anything with queues. ;) We will implement our queues at a later date! We too had problems with the queues. Seems to be more and AgentCallBackLogin issue. Especially when you have different agent code to extensions. Had to restart * up to 8 times a day. Solution was to make the agents Dynamic. Have not restarted the server in 4 days. Dont know what to do with all this spare time i have now. In Short, Queues do work in 1.2.9.1. You cant do CallBackLogin. Hope this helps Terry ___ --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 + nite affiliates
Hi Guys I have a client call center that has after hours agents. Once the call center closes they forward calls to the night affiliates. These nite operators are not constant and tend to swop with each other and then let the person in charge know who is on when. I have the mammoth (Mannie) task of getting a gui solution for asterisk. I would like to open a web browser and have a little drop down box where i can select the after hours workers number (be in home or mobile numbers) and then dial the numbers when incoming calls start hitting the system. They are currently using the Avaya Definity for this, but we are phasing it out. I am running on Suse 10.0 and asterisk 1.2.9.1 Is there already a software out there that can do this or should i have already started coding. Any help will be greatly appreciated. Cheers Terry ___ --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] Cant seem to send cidname to snom 320
Hi Guys
I am trying to send cidname info through to my Agents phones. I can get
the info out of th DB, but when i call an agent i only get the extension
number coming through.
here is my test
exten = 213,1,NoOp(*${DNID}*)
exten = 213,2,Set(CALLERID(name)=${DB(cidname/${DNID})})
exten = 213,3,NoOp(-${CALLERID(name)-ENG-)
;exten = 213,4,LookupCIDName(${DB(cidname/${DNID})})
exten = 213,4,Dial(SIP/78,30,tro)
exten = 213,5,Hangup()
I have added the o to the dial statement just to test. Still no luck. I
am testing this on an AAH 2.5 box, but will be moving it to * 1.2.9.1.
once it is working.
Any pointers will help. I have been googling.
Thanks
Terry
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[Asterisk-Users] getting agentID and DNID help
Hi Guys
I have just installed a call center onto Suse 10. I have managed to do a
DBget (astdb) and extract the DNID numbers to play a DNID specific
greeting. We have installed Snom 320 and the customer would like us to
Send the DNID(nam) to the phone screens so that the agent will be able
to answer in the correct language and with the specific customer company
name (ie. Agent says Welcome to JP Morgan helpline as they will be
able to get the JP morgan helpline name on the screen). This is what my
incoming dial-plan looks like so far:
[Inbound}
exten = _X.,1,GotoIfTime(21:29-07:29|mon-thu|*|*?After-hours,s,1)
exten = _X.,2,GotoIfTime(17:59-07:59|fri-sat|*|*?After-hours,s,1)
exten = _X.,3,GotoIfTime(17:59-07:29|sun|*|*?After-hours,s,1)
exten = _X.,4,Answer
exten = _X.,5,Wait(2)
exten = _X.,6,NoOp(-ID ${CALLERID(num)}-)
exten = _X.,7,DBget(COMPDNID=checked/${DNID})
exten = _X.,8,GotoIf($[${DNID} = 4966]?clientX,4966,1:9)
exten = _X.,9,GotoIf($[${COMPDNID} = 1]?10:102)
exten = _X.,10,ResponseTimeout(7)
exten = _X.,11,Background(custom/${DNID}1)
exten = 1,1,Goto(english-q,1,1)
exten = 2,1,Goto(lang-2-q,1,1)
exten = 3,1,Goto ...etc
I have tried to do a:
exten = _X.,1,Set(CALLERID(name)=${DB(CIDNAMES/${DNID})})
but this has not seemed to help.
I am sure there is someone out there that will be able to point me in
the right direction. Please bear in mind that i am still fairly new to
asterisk and their dial plans.
The second question i would like to ask is as follows:
I have roaming agents. They have no fixed seating positions as the work
in shifts for 24 hour days. There are extensions on the desks and they
have unique agent login ID's.
This creats a small problem if i am trying to pause the agents from the
multiple queues that are in. 5 of them to be correct.
This is what i was thinking:
exten = PauseQueueMember(|Agent/${agentid}) ;
but i cant seem to manage to get the agentID from anywhere, I see that
once an agent has logged in there is a global variable
AGENTBYCALLERID_{EXTEN} set. Is there some way of using this to pause or
unpause the agent?
TIA
Terry
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Re: [Asterisk-Users] QuadBri card
Olivier Saulnier wrote: It's mark on some documentations... Where do i laucnh qozap ?? Best regards, Olivier S. Tzafrir Cohen a écrit : On Wed, Jun 07, 2006 at 05:44:16PM +0200, Olivier Saulnier wrote: Hello, I install the latest release of Asterisk, QuadBri driver. I compile al; that, copy zaptel.conf file, modify /etc/rc.d/rc.local for launch qozap... Bad place. rc.local is just about the last place in the init sequence to be run. After Asterisk is started. what i have done at some clients sites, is actually putting an entry into the /etc/init.d/zaptel file. search for the modprobe command and put your qozap line in at the bottom. ___ --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] Re: amportal doesn't start with brestuff(ISDN)HFC-PCI
Shenen Shenen wrote: On 5/27/06, *Shenen Shenen* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi!I've installed [EMAIL PROTECTED] and I have a ISDN card,(Cologne Chip Design GmbH ISDN network controller [HFC-PCI](rev 0.2) This is how I installed bristuff: how to install hfc card after unload asterisk and amportal whit amportal stop type setup unselect zaptel in system service... and set the lan ---reboot--- cd /usr/src wget http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1l.tar.gz tar -zxvf bristuff-0.3.0-PRE-1l.tar.gz cd bristuff-0.3.0-PRE-1l ./download.sh ln -s /usr/src/2.6.9-22.EL /usr/src/linux-2.6 cd zaptel-1.2.3 make clean make make install cd .. cd libpri-1.2.2 make clean make make install cd .. cd zaphfc make clean make cp zaptel.conf /etc/zaptel.conf --yes--- nano /etc/rc.d/rc.local at first add this line: modprobe zaptel insmod /usr/src/bristuff-0.3.0-PRE-1l/zaphfc/zaphfc.ko sleep 10 ztcfg -vv nano /etc/asterisk/zapata.conf this is my zapata: [channels] language=it switchtype=euroisdn ;If you connect to a hicom PBX set your ISDN Numbering Plan Identifier to unknown. pridialplan=local prilocaldialplan=local signalling = bri_cpe_ptmp ;signalling = fxs_ks rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 nationalprefix = 0 internationalprefix = 00 faxdetect=incoming group=0 callgroup=1 pickupgroup=1 immediate=yes context=from-pstn channel = 1-2 now you can recompile asterisk cd /usr/src/bristuff-0.3.0-PRE-1l/asterisk-1.2.4 make clean make make install reboot... when I reboot amportal doesn't start and I've the following error: [EMAIL PROTECTED] ~]# tail /var/log/asterisk/full May 26 08:45:59 WARNING[15240] pbx.c: Context 'ext-local' tries includes nonexistent context 'ext-local-custom' May 26 08:45:59 VERBOSE[15240] logger.c: [skipping chan_oss.so] May 26 08:45:59 VERBOSE[15240] logger.c: [chan_zap.so]May 26 08:45:59 VERBOSE[15240] logger.c: [chan_zap.so] = (Zapata Telephony w/PRI) May 26 08:45:59 VERBOSE[15240] logger.c: == Parsing '/etc/asterisk/zapata.conf': May 26 08:45:59 VERBOSE[15240] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found May 26 08:45:59 WARNING[15240] chan_zap.c: Unable to specify channel 1: Device or resource busy May 26 08:45:59 ERROR[15240] chan_zap.c: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 May 26 08:45:59 ERROR[15240] chan_zap.c: Unable to register channel '1-2' May 26 08:45:59 WARNING[15240] loader.c: chan_zap.so: load_module failed, returning -1 May 26 08:45:59 WARNING[15240] loader.c: Loading module chan_zap.so failed! and amportal start doesn't startand when I rebott before that coming the loggin and password answer this is the message on the monitor: zaphfc:no version for zt-receive found:kernel tainted Zaptel configuration - - - - - - - - - - - - - - - - - - - - - - - - SPAN 1:CCS/ AMI Build.out: 339 -533 feet (DSX-1) (FXO and FXS are ok) 10 thanks! Ciao -- This mail was scanned by AntiVir MailGate. This product is not licensed. See http://www.antivir.de/ for details. ___ --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 on the AAh forums, there is a INSTALL-bristuff scriptthat will download the correct files for your version number and do the install. Might be a nice idea to check that out first. Cant seem to find the link to the INSTALL-bristuff script. ___ --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] Netherlands zaptel.conf
Michiel van Baak wrote: I thought the wcfxs module is used for fxo cards? Anyhow, if I load the wcfxo module, then I get errors with ztcfg (below). ZT_CHANCONFIG failed on channel 1: No such device or address (6) Normally, if I load the wcfxs module and the zaptel module, then this is what I get (listed below). It all looks reasonable and the only issue seems to be that the card doens't answer the line. what port is the module in. If it is on port 3 or 4 then the zaptel and zapata needs to be the same. Why not just run the genzaptelconf file, or is that specific to AAH? Cheers Terry ___ --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 and Cisco Callmanager
Rob Lith wrote: Hi Terry Could you outline more what existing set-up they have already - do they want to use existing PBX's , if so which kind etc, and these Asterisk or Ciscos as the voicemail application? you could also add on call recording as a carrot. Do they have analogue or ISDN, maybe PRI? Regards Rob On 10/04/06, Terry Wade [EMAIL PROTECTED] wrote: Hi Guys I have just come from a customer that is looking to install 13 Cisco CallManagers into all their branches, (i tried to convince them to go *). They are looking for a voicemail solution. Now as Kinesis and Unity are way too expensive (apparently cisco is launching a cheap voicemail system too) I was thinking of installing * as the voicemail solution. Lots of goggling i have found plenty information telling me that this is possible. I just wanted to know if there are any success stories out there and whether or not i need any additional hardware, other than a PC. I know that one could use ztdummy as a timing source. Is this the best way, or is there some wise words out there one should be heeding. TIA Cheers Terry ___ --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 -- This mail was scanned by AntiVir MailGate. This product is not licensed. See http://www.antivir.de/ for details. Hi Rob They currently have Philips IS3000 in place, but will be tossing them out for the Cisco Solution. The main Office in PTA will be running the callmanager and the branches will all be Cisco 82XX's. They will be running 4.1 on the callmanager. They syustems will be connected to PRI's. This all goes live at the end of June 2006. Cheers ___ --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 and Cisco Callmanager
Hi Guys I have just come from a customer that is looking to install 13 Cisco CallManagers into all their branches, (i tried to convince them to go *). They are looking for a voicemail solution. Now as Kinesis and Unity are way too expensive (apparently cisco is launching a cheap voicemail system too) I was thinking of installing * as the voicemail solution. Lots of goggling i have found plenty information telling me that this is possible. I just wanted to know if there are any success stories out there and whether or not i need any additional hardware, other than a PC. I know that one could use ztdummy as a timing source. Is this the best way, or is there some wise words out there one should be heeding. TIA Cheers Terry ___ --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] open source queue analyzer
Michiel van Baak wrote: On 13:11, Thu 16 Mar 06, nik600 wrote: browsing the web i don't find any opensource (and free of charge ) software for the web statistic about queues... i've tries queue_stats made from asteriskguru, it is a good tool, and it is free of charge, but it's not open-source :-( i'm considering to develop myself a web application, before that i would ask you if you are interested of this, i would like to activate a sourceforge project the main requirements of the project are: /// realtime - possibility to login/logout from the queue via web interface - monitor the state of the queue (logged in agent/extension, queued calls, ecc) ///statistics - average wait time - average call time - average calls per agent/extension - average calls per hour - average calls per day - average calls per week - average calls per month ///supervisors - define new users that can access to the software - set for each user the operation to do in the queue (login/logout/real time monitor/statistics) now i've realized the firse section, realtime, and i'm using it in my callcenter sice 2 weeks the software use php and mysql o postresql as database (i would add some ajax module for refreshing some data without reloading the page) so, would you like to contribute? what do you think of that? Sounds like a nice project. If it's on sf.net I will for sure checkout the source and see if I can contribute :) I'm a fulltime php/ajax/mysql/postgres developer. Michiel ___ --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 if this becomes a reality, i am happy to contibute financially. This is eactly what i am looking for here, but dont have the skills to complete it. Terry ___ --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] Zaphfc.ko module error
Hi Remco Thanks for the response. I am running Suse 9.3 kernel 2.6.11.4-20a-default. Will check on the auto update, but I don't think so. Cheers Terry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: 18 August 2005 08:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaphfc.ko module error Hi! You didn't state what distro you are running but my guess is that you have autoupdate / up2date running. Before the powerfailure the kernel was updated and after the powerfailure the box booted the new kernel for which you need to recompile the module. Cheers! Remco On Thu, 18 Aug 2005, Terry Wade wrote: Hi Guys I have been running a test server for a few days now with * 1.0.9 bristuff RC8n. I had a power failure and the test machine was not on the ups. When power was restored I found the following error: FATAL: Error inserting zaphfc (/lib/modules/2.6.11.4-20a-default/misc/zaphfc.ko): Unknown symbol in module, or unknown parameter (see dmesg) My dmesg output: zaphfc: unsupported module, tainting kernel. ^^ that makes me believe you are now running a newer kernel ___ 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] Zaphfc.ko module error
Hi Guys I have been running a test server for a few days now with * 1.0.9 bristuff RC8n. I had a power failure and the test machine was not on the ups. When power was restored I found the following error: FATAL: Error inserting zaphfc (/lib/modules/2.6.11.4-20a-default/misc/zaphfc.ko): Unknown symbol in module, or unknown parameter (see dmesg) My dmesg output: zaphfc: unsupported module, tainting kernel. zaphfc: disagrees about version of symbol zt_receive zaphfc: Unknown symbol zt_receive, st_info == 0x1 zaphfc: disagrees about version of symbol zt_ec_chunk zaphfc: Unknown symbol zt_ec_chunk, st_info == 0x1 zaphfc: disagrees about version of symbol zt_transmit zaphfc: Unknown symbol zt_transmit, st_info == 0x1 zaphfc: disagrees about version of symbol zt_unregister zaphfc: Unknown symbol zt_unregister, st_info == 0x1 zaphfc: disagrees about version of symbol zt_register zaphfc: Unknown symbol zt_register, st_info == 0x1 load_module: err 0xfffe (dont worry) I have tried to google and voip-info this problem, but to no avail. Modprobe zaptel works just not the zaphfc. I have recompiled zaptel and zaphfc, but still get the error. Any pointers would be appreciated. TIA Kind Regards Terry Wade Mobile: +27 82 802-5750 Office: +27 11 784-7642 Fax: +27 11 388-0855 Linux is like a Wigwam - No gates, no windows, Apache inside Disclaimer and Confidentiality Warning This message is intended for the addressee only. If you are not the intended recipient of this message, you are notified that any distribution, use of or copying of this communication is strictly prohibited. If you have received the communication in error, please notify the sender immediately. The views and opinions expressed in this message are those of the individual sender of this message and do not necessarily represent the views and opinions of ActiCom. Consequently, ActiCom does not accept responsibility for such views and opinions and this message should not be read as representing the views and opinions of ActiCom without subsequent written confirmation. Each page attached hereto must also be read in conjunction with this disclaimer. ___ 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] Database querie
Hi Guys Just a quick question. Does * write directly into PGSQL database like MySQL? Kind Regards Terry Wade Mobile: +27 82 802-5750 Office: +27 11 784-7642 Fax: +27 11 388-0855 Linux is like a Wigwam - No gates, no windows, Apache inside Disclaimer and Confidentiality Warning This message is intended for the addressee only. If you are not the intended recipient of this message, you are notified that any distribution, use of or copying of this communication is strictly prohibited. If you have received the communication in error, please notify the sender immediately. The views and opinions expressed in this message are those of the individual sender of this message and do not necessarily represent the views and opinions of ActiCom. Consequently, ActiCom does not accept responsibility for such views and opinions and this message should not be read as representing the views and opinions of ActiCom without subsequent written confirmation. Each page attached hereto must also be read in conjunction with this disclaimer. ___ 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] HFC BRIstuff woes
Add another span= line and then extra chans .. look in the zaphfc build directory and you will find some examples -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: 18 July 2005 12:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HFC BRIstuff woes Tzafrir Cohen wrote: Hmmm... didn't you say you had a x100p card somewhere on your system? Nope - only the 2 pci HFC ISDN cards. OK - I hope I'm not buggering up the thread, but I need to answer myself as there now seems to be only 2 remaining concerns - but then I haven't yet had a chance to plug this box into an ISDN wall-socket yet - I'll get that joy tomorrow (today) afternoon!!! I'm still concerned about the auto-generated context being demo. I keep getting out-is-busy messages (remember I'm not plugged into the ISDN socket, though) Chan Extension Context Language MusicOnHold pseudodemoen 1demoen 2demoen Tzafrir, should I worry about these messages that keep coming up on the console? zaphfc: bchan rx fifo not enough bytes to receive! (z1=3015, z2=3008, wanted 8 got 7), probably a buffer overrun. (the z1= and z2= numbers keep changing - message repeats circa every minute or so) (I've fixed the registration of z1 and z2 - iax_additional.conf had the wrong phone names in it) Thanks yet again, good night, Zoltan. -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ 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] Re: hfc-s card, brii-stuff.0.1.0-RC4a, zaphfc:sync lost, pci performance too low. you might have some cpu throtteling enabled.
Spoke to Klaus-Peter about this PCI performance issue. He says it has to do with the CPU not supporting cpufreq stepping. I had to get a quad card to get the issue resolved. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Szlezak Sent: 17 July 2005 11:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: hfc-s card, brii-stuff.0.1.0-RC4a, zaphfc:sync lost,pci performance too low. you might have some cpu throtteling enabled. HI Hartmut, I do have the same problem as you decribed earlier. The Billion HFC Cards (two of them) work flawlessly in my old Pentium II, but in my more powerful Athlon XP 2400+ (Via KT400 Chipset), I allways get the pci performance too low message and syslog kills the system. I'll try it now with commenting out the message. Do you know anything more in the meantime. Did Mr. Junghanns have any statement about it? I wonder if chan_mISDN is a better choice by now? Thanks for any advice you might have! yours, Alexander Hartmut Wahl wrote: Hello, I have investigated the issue a bit further, I was not able to find the root cause, maybe it is the KT133 Chipset of my ASUS A7V. However I found a bad hack to make it work under some circumstances. I commented out the line: printk(KERN_CRIT zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled.\n); in zaphfc/zaphfc.c. Since when this situation happened once, the syslog started and this caused the situation again - endless loop with high sysload. Now the sync lost happens probably every now and then but it does not go into an endless loop. I also recognized that I must not run setiathome since then the audio quality of connections via the hfc-card will suffer (crackling). Amazingly cpuburn (takes every cpu-Time it gets as well) does not have this effect. I have no idea what strange things setiathome does to cause this but it reminds me of a sound card problem on this board. My SB-Live had crackling when I ran setiathome but I think with a newer driver (and ALSA I think) this problem was gone. Things that did not help: - Trying to change the latency (is fixed to 16 and cannot be changed) - Trying to change the latency of my other pci-devices (much higher and much lower). - Playing around with BIOS Options like delayed transaction, etc. - Changing PCI-slots - Making sure that the ISDN-card did not share the IRQ - Having only the ISDN-card and the Video-card in the System Hope that helps others who encounter this problem as well Hartmut On Sat, Sep 11, 2004 at 12:55:21PM +0200, Hartmut Wahl wrote: Hi! On Sat, Sep 11, 2004 at 10:38:48AM +0200, Maurizio Marini wrote: my machine did hangup as growing logs fullfilled partition hmm I see, mine is 8G, but it has gronwn from 0.5G to 2.0G it does apply to asterisk, not to zaphfc :( it was a misleading suggestion, so i solved it installing in an other more powerful machine: processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Pentium III (Coppermine) stepping: 10 cpu MHz : 999.556 cache size : 256 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 mmx fxsr sse bogomips: 1957.88 with this hw i've no issues at all; even strange messages i complained about in my previous posts like: ok we are getting closer, although speed shouldn't be a problem: vendor_id : AuthenticAMD model name : AMD Athlon(tm) Processor stepping: 4 cpu MHz : 1059.618 cache size : 256 KB bogomips: 2097.15 it is a 1,4GHz underclocked, since my ASUS A7V with KT133 does only support 100MHz FSB, but it requires only slow and quite fans. I think I'll try different kernels maybe something in the Debian k7-kernel-image interferes. Greetings Hartmut ___ 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] calling shell scripts from within *
Hi Guys Just a stooped question, if I may. I have a bash script that I would like to run from the acd. Would I use the System command, the API command or is there a better way of doing this. I want to create a smb connection to a microsnot machine and pull a .wav file off to playback to the caller. TIA Kind Regards Terry Wade Mobile: +27 82 802-5750 Office: +27 11 784-7642 Fax: +27 11 388-0855 Linux is like a Wigwam - No gates, no windows, Apache inside Disclaimer and Confidentiality Warning This message is intended for the addressee only. If you are not the intended recipient of this message, you are notified that any distribution, use of or copying of this communication is strictly prohibited. If you have received the communication in error, please notify the sender immediately. The views and opinions expressed in this message are those of the individual sender of this message and do not necessarily represent the views and opinions of ActiCom. Consequently, ActiCom does not accept responsibility for such views and opinions and this message should not be read as representing the views and opinions of ActiCom without subsequent written confirmation. Each page attached hereto must also be read in conjunction with this disclaimer. ___ 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] calling shell scripts from within *
Thanks guys -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 05 July 2005 09:58 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] calling shell scripts from within * On Tue, Jul 05, 2005 at 09:44:50AM +0200, Giorgio Incantalupo wrote: Hi, we are using AGI command to call external python script but it can be a good solution if you have a small PBX . There are res_perl and res_python that promise a better performance but we are still working on it. I suggest you to try them if your call flow is very intensive. A fork/exec is involved anyway because of calling an external program. And \ bash is lighter than than python/perl . This is exactly what System is for, as it does not need to interact with the dialplan. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] make error for zaptel
Ln -s /lib/modules/'uname -r'/build /usr/src/linux-2.6 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: 01 July 2005 01:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] make error for zaptel Hi, I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the box (in a hope to sort out the uname -r issue mentioned below). I'm using the Asterisk Doc Proj vol 1 to guide me through the initial setup. I have no special HW and intend to use asterisk on an internal network just to get some experience. I have downloaded what I think I need and placed it in /usr/src (see listing below). I run make clean ; make linux26 (what about the usual make with no parameters?) and I get a crash. Note that uname -r returns a *different* version of what the linux is linked to (thanks to YOU??) I have tried make clean ; make (no params) and it still crashes. Can anyone offer me some suggestions? - or do I go first to the SuSE list to sort out the uname -r usr/src/linux issue? TIA, Zoltan. gl0:/usr/src # ls -la total 499 drwxr-xr-x 17 root root680 2005-06-30 14:46 . drwxr-xr-x 13 root root368 2005-06-30 09:21 .. drwxr-xr-x 3 root root320 2005-05-19 06:53 astcc -rw-r--r-- 1 root root 130943 2005-06-25 19:09 astcc.tar drwxr-xr-x 22 root root 2336 2005-06-23 04:16 asterisk-1.0.8 drwxr-xr-x 7 root root432 2005-06-23 04:21 asterisk-addons-1.0.8 drwxr-xr-x 3 root root184 2005-06-23 04:23 asterisk-sounds-1.0.8 drwxr-xr-x 2 root root440 2005-05-23 06:47 btp -rw-r--r-- 1 root root 32975 2005-06-25 19:08 btp.tar drwxr-xr-x 23 root root776 2005-06-12 17:24 dicts drwxr-xr-x 5 root root416 2005-04-01 18:50 gastman -rw-r--r-- 1 root root 332857 2005-06-25 19:08 gastman.tar drwxr-xr-x 25 root root736 2005-06-12 17:29 kernel-modules drwxr-xr-x 2 root root520 2005-06-23 04:11 libpri-1.0.8 lrwxrwxrwx 1 root root 19 2005-06-25 22:01 linux - linux-2.6.11.4-21.7 drwxr-xr-x 3 root root 72 2005-06-25 22:01 linux-2.6.11.4-20a drwxr-xr-x 3 root root 72 2005-03-24 02:03 linux-2.6.11.4-20a-obj drwxr-xr-x 19 root root720 2005-06-25 22:01 linux-2.6.11.4-21.7 drwxr-xr-x 3 root root 72 2005-06-02 16:54 linux-2.6.11.4-21.7-obj lrwxrwxrwx 1 root root 23 2005-06-25 22:01 linux-obj - linux-2.6.11.4-21.7-obj drwxr-xr-x 7 root root168 2005-06-12 17:43 packages drwxr-xr-x 2 root root 2720 2005-07-01 12:43 zaptel-1.0.8 gl0:/usr/src # uname -r 2.6.11.4-20a-smp gl0:/usr/src # cd zaptel-1.0.8/ gl0:/usr/src/zaptel-1.0.8 # make clean rm -f torisatool makefw tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f *.ko *.mod.c .*o.cmd rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core gl0:/usr/src/zaptel-1.0.8 # make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o -lm -L. libtonezone.a cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -c ztspeed.c cc -o ztspeed ztspeed.o make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/zaptel-1.0.8 modules make[1]: Entering directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.11.4-20a-obj/i386/smp' make: *** [linux26] Error 2 gl0:/usr/src/zaptel-1.0.8 # ___ 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:
[Asterisk-Users] Echo Problems
Hi Guys I have installed an * system and we seem to have loads of echo problems. Sometimes worst than others. I have googled and voip-info ed my little mind out. I am running 3 x zaphfc cards in the machine. Not sharing irqs, other than themselves. It is on a PIII 1Gig machine with 1Gb ram. My question is this. Does the 2.6 kernel affect (or can) the echo? Could a busy network cause this problem? Vmstat shows cpu usage spike to about 48% Starting to pull my hair out. Any suggestions would be muchly appreciated. Kind Regards Terry Wade Mobile: +27 82 802-5750 Office: +27 11 784-7642 Fax: +27 11 388-0855 Linux is like a Wigwam - No gates, no windows, Apache inside Disclaimer and Confidentiality Warning This message is intended for the addressee only. If you are not the intended recipient of this message, you are notified that any distribution, use of or copying of this communication is strictly prohibited. If you have received the communication in error, please notify the sender immediately. The views and opinions expressed in this message are those of the individual sender of this message and do not necessarily represent the views and opinions of ActiCom. Consequently, ActiCom does not accept responsibility for such views and opinions and this message should not be read as representing the views and opinions of ActiCom without subsequent written confirmation. Each page attached hereto must also be read in conjunction with this disclaimer. ___ 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] cisco 7960 firmware
If memory serves correctly. You need to go from firmware version 3 to 4 to 5 and then you can jump to 7. or something like that. Terry From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Kenna Sent: 09 May 2005 08:14 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] cisco 7960 firmware do some searching in google, just the other day i came across a site that had v7.3 SIP availiable. pretty good, since v7.4 is the newest. Chris [EMAIL PROTECTED] 5/9/2005 1:43 PM Hi List! As a child did you ever receive a present that required batteries only to find all the shops were shut? That's the way I feel at the moment with my cisco 7960 IP phone. Recently purchased off ebay was looking forward to connecting up to my asterisk pbx ( installed a few days ago ). The firmware the phone came with ( 3.1 I think ) is not the SIP firmware I need. So I though I'd have a look on voip-info.org, did some light reading and noticed I needed a contract from cisco to download the firmware. Thought this a little strange so called cisco and they asked me to call a re-seller. Now I have to wait for them to get me a price, then order it ( said it could be a few days ) then add it to my cisco registration ( another few days ) Is there anywhere I could get the files I need to upgrade my phone just so I can get the thing working while I wait for my contract? Any help would be appreciated. Phil ___ 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] OctoBRI and 2.6kernel
Hi Guys I am trying to get the Junghanns card to load on Suse 9.3 and tried to get it running on Fedora Core 3 (latest kernels). I have heard from a source here in South Africa that this is about as hard as pulling teeth. Could someone please confirm this for me and if they do have it working properly is it possible to get a guide. I can get the zaptel and qozap to load the card and all the ports and inside asterisk I see the zap channels. But I cannot get a line out or make any incoming calls. Are there some 2.6 tweaks that I need to do in the kernel. Kind Regards Terry Wade Mobile: +27 82 802-5750 Office: +27 11 784-7642 Fax: +27 11 388-0855 Linux is like a Wigwam - No gates, no windows, Apache inside Disclaimer and Confidentiality Warning This message is intended for the addressee only. If you are not the intended recipient of this message, you are notified that any distribution, use of or copying of this communication is strictly prohibited. If you have received the communication in error, please notify the sender immediately. The views and opinions expressed in this message are those of the individual sender of this message and do not necessarily represent the views and opinions of ActiCom. Consequently, ActiCom does not accept responsibility for such views and opinions and this message should not be read as representing the views and opinions of ActiCom without subsequent written confirmation. Each page attached hereto must also be read in conjunction with this disclaimer. ___ 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] BRI channels not answering
Hi Guys I am trying to install a HFC-S isdn card into asterisk. Bri-stuff installs fine and I have managed to get *-v-1.0.7 installed. All seems to run without a problem, well except for the fact that the card does not answer any incoming calls or make any outgoing calls. My zaptel.conf looks like this: more /etc/zaptel.conf # hfc-s pci a span definition # most of the values should be bogus because we are not really zaptel loadzone=us defaultzone=us span=1,1,3,ccs,ami bchan=1-2 dchan=3 I have changed the default zones to a few different countries, nothing has made any difference. more /etc/asterisk/zapata.conf ; ; Zapata telephony interface ; ; Configuration file [channels] language=en ;switchtype = euroisdn ; p2mp TE mode ;signalling = bri_cpe_ptmp ; p2p TE mode ;signalling = bri_cpe ; p2mp NT mode signalling = bri_net_ptmp ; p2p NT mode ;signalling = bri_net pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 09 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context=demo channel = 1-2 [dmesg] zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xd89a1000 fifo 0xd1ed8000(0x11ed8000) IRQ 3 HZ 1000 zaphfc: Card 0 configured for TE mode zaphfc: 1 hfc-pci card(s) in this box. Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) [EMAIL PROTECTED] zaphfc]# ztcfg -vv Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. I have plugged the system into an ISDN NT1 box, and still cannot manage to get something good. I have googled to my wits end and any help would be appreciated. If anyone has some documentation on all the different options one has to ones disposal, would also be appreciated. All my SIP phones work internally, I can send my extensions.conf as well as sip.conf if needed. TIA Kind Regards Terry Wade Mobile: +27 82 802-5750 Office: +27 11 784-7642 Fax: +27 11 388-0855 Linux is like a Wigwam - No gates, no windows, Apache inside Disclaimer and Confidentiality Warning This message is intended for the addressee only. If you are not the intended recipient of this message, you are notified that any distribution, use of or copying of this communication is strictly prohibited. If you have received the communication in error, please notify the sender immediately. The views and opinions expressed in this message are those of the individual sender of this message and do not necessarily represent the views and opinions of ActiCom. Consequently, ActiCom does not accept responsibility for such views and opinions and this message should not be read as representing the views and opinions of ActiCom without subsequent written confirmation. Each page attached hereto must also be read in conjunction with this disclaimer. ___ 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] OctoBRI - unable to specify channel 1
Hi Guys I have installed * with an OctoBRI card. The card laods fine without and errors, ut when I start * I get: == Parsing '/etc/asterisk/zapata.conf': Found Apr 15 09:41:59 WARNING[9893]: chan_zap.c:924 zt_open: Unable to specify channel 1: No such device or address Apr 15 09:41:59 ERROR[9893]: chan_zap.c:6460 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Apr 15 09:41:59 ERROR[9893]: chan_zap.c:10247 setup_zap: Unable to register channel '1-2' Apr 15 09:41:59 WARNING[9893]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Apr 15 09:41:59 WARNING[9893]: loader.c:440 load_modules: Loading module chan_zap.so failed! I have googled for an answer but dont find anything specific. Please can someone just ,let me know where I am going wrong. Thanks Kind Regards Terry Wade Mobile: +27 82 802-5750 Office: +27 11 784-7642 Fax: +27 11 388-0855 Linux is like a Wigwam - No gates, no windows, Apache inside Disclaimer and Confidentiality Warning This message is intended for the addressee only. If you are not the intended recipient of this message, you are notified that any distribution, use of or copying of this communication is strictly prohibited. If you have received the communication in error, please notify the sender immediately. The views and opinions expressed in this message are those of the individual sender of this message and do not necessarily represent the views and opinions of ActiCom. Consequently, ActiCom does not accept responsibility for such views and opinions and this message should not be read as representing the views and opinions of ActiCom without subsequent written confirmation. Each page attached hereto must also be read in conjunction with this disclaimer. ___ 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] zaptel not starting issues
Hi Guys I have been running a test system of asterisk on a PIII 550 machine and now I want to implement it. I have a digium TDM02B card. I am running on Mandrake 10.1. I have now moved it to a PIII 1.2 backplane server and now I cant get the wcfxs module to load. I have also tried wctdm. I did all the udev mods and the install didnt have any errors. I am currently running * v1.0.7 stable. I have googled and have not come up with anything concrete. Error is as follows: ZT_CHANCONFIG failed on channel 4: No such device or address (6). I only have 1 FXO module in at the moment, and have moved it through all the positions. Does the card need the power in the back to correctly start it? Kind Regards Terry Wade Mobile: +27 82 802-5750 Office: +27 11 784-7642 Fax: +27 11 388-0855 Disclaimer and Confidentiality Warning This message is intended for the addressee only. If you are not the intended recipient of this message, you are notified that any distribution, use of or copying of this communication is strictly prohibited. If you have received the communication in error, please notify the sender immediately. The views and opinions expressed in this message are those of the individual sender of this message and do not necessarily represent the views and opinions of ActiCom. Consequently, ActiCom does not accept responsibility for such views and opinions and this message should not be read as representing the views and opinions of ActiCom without subsequent written confirmation. Each page attached hereto must also be read in conjunction with this disclaimer. ___ 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] Unable to create channel of type 'SIP'
Hi I get the following error when i dial a sip extension, please help NOTICE[1681]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time The SIP extension you are trying to dial has not registered with *. T ___ 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] * and Philips IS3090 PBX
Hi I have been playing with * for the last couple of weeks now. I am also speaking to one of my customers about installing a * server in addition to their Philips IS3090 switch. They are busy building a new office block and I have convinced them to go VoIP. Currently the client is thinking about a Philips IS2000 Voip switch. I am thinking * solution. 1) Would I use one the T1 cards to interconnect the two, does anybody have any experience with the Philips switches? 2) Would I be able to make calls to the local PSTN via the IS3090? 3) Do I create the extensions on the IS3090 or the *? 4) Would the IP phones be able to use the Kinesis voicemail on the IS3090? 5) Would the TMS still log all the activity from the IP phones? 6) Will the cell/mobile routers still make their allocated calls? 7) They are also running dect phones, so this would this be able to work as well? Sorry for all the questions, but this could be a big deal for me. Starting with 150 new exts and can replace upto 750 Exts. Thanks for advice in advanced Cheers Terry ___ 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
