[asterisk-users] surge protector?
I lost one channel on an FXO module on a Sangoma A200 card due to a lightening zap in the area (well - it died the same night as a major thunder storm came through)Is there a recommended/standard surge protector for phone lines I should be using? My server has 2 POTS lines. thanks Todd ___ --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] open source screen pop software for asterisk
I like ADM as it has a URL popup feature (open a URL with a DID or CallerID in URL). The problem is that for each call, I tend to get 4 or 5 popups... But as the other author said, there are many programs to choose from... Todd On Jul 13, 2007, at 11:54 PM, RENZZO SOTOMAYOR wrote: > Hi! I am new here. Well I'm doing a call center using asterisk and > I'm looking for an open source screen pop software to pop the > caller's information, its call history and others things. i was > looking around and find the U-rang2 the problem is that it isn't > open source. if someone knows about an open source screen pop > please tell me. > > thanks in advance > > renzzo ___ --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
[asterisk-users] queue beep
Hi - I have a queue setup with 4 agents. The people working the queue tell me that when a new call goes into the queue, both the agent and the caller hear a tone. These are static agents - could it be ringing the line again, even though the agent is on a call? We are using GXP2000's with only 1 line programmed in. thanks Todd queue.conf > [601] > announce-frequency=0 > monitor-format=wav > monitor-join=yes > music=default > queue-callswaiting= > queue-thankyou=queue-thankyou > queue-thereare= > queue-youarenext= > retry=5 > rtone=0 > strategy=ringall > timeout=15 > wrapuptime=1 > leavewhenempty=no > eventwhencalled=no > joinempty=Yes > context= > maxlen=0 > announce-holdtime=no > eventmemberstatus=no > member=Local/[EMAIL PROTECTED]/n,0 > member=Local/[EMAIL PROTECTED]/n,0 > member=Local/[EMAIL PROTECTED]/n,0 > member=Local/[EMAIL PROTECTED]/n,0 > member=Local/[EMAIL PROTECTED]/n,0 > ___ --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
[asterisk-users] transfer/conference
Hi All- I have an asterisk server and GXP2000. If I want to send a call to someone else (external), I can transfer the call where I can announce it, and then send it over. But what I'd like is to start a 3-way conference, and then drop out. But if I do a conference button on the phone, and then drop out, the other two are not left to finish their conversation (the call is ended). Should I be using the MeetMe application instead or is there a different way to fix this problem? thanks Todd ___ --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
[asterisk-users] BLF for Queue
Hi - I guess it's not possible to use Asterisk BLF function for queues... Can someone confirm that? I'm looking for that type of function with calling queues. I have Grandstream gp2000 phones. thanks Todd ___ --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] connecting asterisk (trixbox) to traditional phone lines?
To go nice and cheaply, you could just get a free number from IPKALL.com or Stanaphone.com.. And do it all over IP... -t- On Jan 2, 2007, at 10:15 PM, blackwater dev wrote: Ok, I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. 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] Sangoma Remora A202
Hi - I just got a Sangoma A200 card with a single 2FXO module and what appears to be an empty module. I put the card in my Dell GX260, but the power light on the front of the box just blinks and won't power up. I did take the power cable from the CDROM to put on the card - I don't need the CDROM right now.. I'm looking for direction in getting this card working - I currently have a new Trixbox, hoping it'll have the software for this card already. If not, I'll be back asking what drivers I need. Sangoma seems to have a lack of documentation, but it may just be me thanks Todd ___ --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] Sangoma Remora A202
Thanks - that turned out to be the problem. Well- one of those solutions. I removed the blank and swapped the FXO module to the other port. I don't know if it was a bad port on the A200, but since I don't plan on using it, I won't worry about it- just regret it in a year when I get a second FXO module ;) As for documentation, I did find the info on WanPipe, but am not sure what Wanpipe is.. I'll do some more reading tonight. Thanks for the info. Todd On Jan 3, 2007, at 11:40 AM, Bruce Reeves wrote: Try switching the order of the blank module and the FXO or remove the blank, I had a similar Dell do the same and after some experimenting found that the removing the blank solved the problem. On 1/3/07, Rob Schall <[EMAIL PROTECTED]> wrote: If the light on the dell is blinking amber... that typically means you have a power issue. Rob Time Bandit wrote: >> Hi - I just got a Sangoma A200 card with a single 2FXO module and >> what appears to be an empty module. I put the card in my Dell GX260, >> but the power light on the front of the box just blinks and won't >> power up. > Maybe your card is not properly seated. > >> seems to have a lack of documentation, but it may just be me > It is just you ;) > > http://wiki.sangoma.com/ > > If you still have problems with the card, contact Sangoma, they have > very good customer support : http://www.sangoma.com/main/contact > > hth > ___ > --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 -- Bruce Nortex Networks ___ --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
Re: [asterisk-users] queues - limiting ringing calls to queue members
My GXP2000 does what you are talking about. I solved the problem by assigning lines 2-4 to other extensions which are not queue agents. Then those lines don't ring. hth -t- On Jan 2, 2007, at 5:03 PM, Nikola Ciprich wrote: Hello, I'm using asterisk queues, for reception phone, and I have small problem: I have only one phone as queue member, and the problem is, that ALL channels waiting in queue are ringing on it. And if there are too many people ringing on it, it's not possible to use attended transfer then... Is it possible to limit maximum ringing calls from queue? or some other tip? thanks a lot in advance! best regards Nik ___ --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] Sangoma Remora A202
For those following this discussion, below is my original message to the list and a reply from Sangoma Tech Support. -t- On Jan 3, 2007, at 8:43 AM, Todd H wrote: Hi - I just got a Sangoma A200 card with a single 2FXO module and what appears to be an empty module. I put the card in my Dell GX260, but the power light on the front of the box just blinks and won't power up. I did take the power cable from the CDROM to put on the card - I don't need the CDROM right now.. I'm looking for direction in getting this card working - I currently have a new Trixbox, hoping it'll have the software for this card already. If not, I'll be back asking what drivers I need. Sangoma seems to have a lack of documentation, but it may just be me thanks Todd On Jan 4, 2007, at 3:03 PM, David Yat Sin wrote: Hi Todd, We just looked at the blank modules that we usually ship to insert in the empty slots of the A200, and there is a small trace that connects two pins. These two pins have been switched to a different function in our newer A200 builds and this is causing a short. Please do not use the blank module. We will either make new blank modules or stop shipping the blank modules. I guess this was not an issue with the FXO module in 2nd slot, but rather an issue with the blank module. Regards, David Yat Sin tel: 905.474.1990 x119 tel: 800.388.2475 x119 fax: 905.474.9223 msn: [EMAIL PROTECTED] wiki: wiki.sangoma.com ___ --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] Nat Question
To Read about: http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+externip To fix it: Edit the file /etc/asterisk/sip.conf Add in the following lines: externip=xx.xx.xx.xx (your real internet address here. go to http://www.myipaddress.com/ to find out) localnet=192.168.2.0/255.255.255.0 (your internal network information here). Then reload asterisk and you should be set. Hope this helps! Todd On Jan 12, 2007, at 3:55 PM, [EMAIL PROTECTED] wrote: Hello all, iam setting up an asterisk box behind NAT to get SIP calls from outside or internet. In that eschema i can setup SIP calls but, while from the outside nat people can hear me, Im unable to listen anything behind NAT. Out of firewalls settings( I checked this to port fowarding) what can i do to get this working fine?. Thanks G. ___ --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] Suppliers in Canada
http://www.canadianvoipstore.com/home.php VOIPSupply has a candian store-front. I bought from VoipSupply and products shipped from Canada... Todd On Feb 8, 2007, at 1:47 PM, [EMAIL PROTECTED] wrote: I am looking for some Linksys and GrandStream ATAs in Canada. I am looking for places that ship from Canada so I don't have to deal with the clearing of customs and tax remittance. Any suggestion? -- 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] Accessible documentation vor blind users
If you are a Mac user, Apple's speech synthesis works well. Open the PDF in Preview, not Acrobat. Then choose 'Start Speaking' from the Services->Speech menu. I'm not sure what tools Windows has for reading PDF files though I expect there's something... Might want to ask on a Windows mailing list. Todd On Feb 24, 2007, at 1:53 AM, [EMAIL PROTECTED] wrote: Hi Hi Is there any accessible ocumentation, ie plain text or html, how to configure Asterisk. The book 'Asterisk: The Future of Telephony'' is availablly only as and pdf document and is thus unreadable for a blind user. Any pointers welcome. ___ --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] YAACID and manager.conf security
Hi - I am going to open port 5038 on my firewall so that I can use YAACID to spawn browser popups on an incoming call. My question is, under manager.conf, what are the suggested settings so that I can get the browser popups only? I'll be at different IPs so I can't lock it down that way.. I guess I don't need any write access? [managername] secret=secretword read=system,call,log,verbose,command,agent,user write=system,call,log,verbose,command,agent,user thanks Todd ___ --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] GXP-2000 DST Change
Thanks for the info, Ken. I was about to research this tonight. Todd On Mar 12, 2007, at 12:53 PM, Ken Williams wrote: In case it hasn't been posted before, here's instructions to get the correct time to show up on your Grandstream GXP-2000's: 1. Login to phone 2. Go to Basic Settings tab 3. Change Daylight Savings Time to yes 4. Change Optional Rule to 3,2,7,2,0;11,1,7,2,0;60 (this means change clocks the second sunday of March and back again the first sunday of November - i.e., the new savings times). -snip-___ --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] A200 card problem
Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http:// sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or compiling. I also downloaded trixbox 2.0 with sangoma drivers included directly from sangoma (http:// wiki.sangoma.com/Trixbox-1xx). I know how this list feels about trixbox, but still, the card/configs are the same, no? Any advice is appreciated. thanks Todd +++ /var/log/asterisk/full Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP)) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so] => (Local Proxy Channel) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so] => (Linux Telephony API Support) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap.so] => (Zapata Telephony w/PRI) Mar 15 16:12:37 WARNING[31964] chan_zap.c: Unable to specify channel 1: No such device or address Mar 15 16:12:37 ERROR[31964] chan_zap.c: Unable to open channel 1: No such device or address here = 0, tmp->channel = 1, channel = 1 Mar 15 16:12:37 ERROR[31964] chan_zap.c: Unable to register channel '1' Mar 15 16:12:37 WARNING[31964] loader.c: chan_zap.so: load_module failed, returning -1 Mar 15 16:12:37 WARNING[31964] loader.c: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# +++ END /var/log/asterisk/full +++ /var/log/messages [EMAIL PROTECTED] ~]# tail -20 /var/log/messages Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_config.so] => (Text Extension Configuration) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_functions.so] => (Builtin dialplan functions) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_ael.so] => (Asterisk Extension Language Compiler) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_dundi.so] => (Distributed Universal Number Discovery (DUNDi)) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_loopback.so] => (Loopback Switch) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_spool.so] => (Outgoing Spool Support) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_agent.so] => (Agent Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_h323.so] => (Objective Systems H323 Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_sip.so] => (Session Initiation Protocol (SIP)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_features.so] => (Feature Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [skipping chan_oss.so] Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_local.so] => (Local Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_phone.so] => (Linux Telephony API Support) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_zap.so] => (Zapata Telephony w/PRI) Mar 15 16:24:08 asterisk1 safe_asterisk: Asterisk died with code 1. Mar 15 16:24:08 asterisk1 safe_asterisk: Automatically restarting Asterisk. Mar 15 16:24:08 asterisk1 asterisk: safe_asterisk startup succeeded [EMAIL PROTECTED] ~]# +++ END /var/log/messages ++/etc/zaptel.conf+++ [EMAIL PROTECTED] ~]# more /etc/zaptel.conf # Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us # Sangoma A200 [slot:9 bus:1 span: 1] fxsks=1 fxsks=2 fxoks=3 fxoks=4 [EMAIL PROTECTED] ~]# ++END /etc/zaptel.conf+++ ++/etc/asterisk/zapata.conf+++ [EMAIL PROTECTED] asterisk]# more zapata.conf ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing
Re: [asterisk-users] A200 card problem
FYI, here's the result from Sangoma... -t- Hi Todd, It’s look like issue with your power supply. It’s not providing enough power to the FXS module. Make sure that power cable is connected properly to the A200 main board. You can try to swap the power cable from CD/DVD ROM for temporary testing. You can find this information at http://wiki.sangoma.com/sangoma-hardware#A200 . I have also added signaling method for channels in your /etc/ asterisk/zapata.conf . I received following error in your /var/log/messages Mar 16 19:43:39 asterisk1 kernel: wanpipe1: Module 2: TIP/RING is too low on FXS 0! Mar 16 19:44:11 asterisk1 last message repeated 2 times Mar 16 19:44:44 asterisk1 last message repeated 2 times Mar 16 19:44:44 asterisk1 kernel: wanpipe1: Module 2: FXS failed! Mar 16 19:45:13 asterisk1 kernel: wanpipe1: Module 3: TIP/RING is too low on FXS 0! Best regards, On Mar 15, 2007, at 4:38 PM, Todd H wrote: Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http:// sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or compiling. I also downloaded trixbox 2.0 with sangoma drivers included directly from sangoma (http:// wiki.sangoma.com/Trixbox-1xx). I know how this list feels about trixbox, but still, the card/configs are the same, no? Any advice is appreciated. thanks Todd +++ /var/log/asterisk/full Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP)) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so] => (Local Proxy Channel) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so] => (Linux Telephony API Support) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap.so] => (Zapata Telephony w/PRI) Mar 15 16:12:37 WARNING[31964] chan_zap.c: Unable to specify channel 1: No such device or address Mar 15 16:12:37 ERROR[31964] chan_zap.c: Unable to open channel 1: No such device or address here = 0, tmp->channel = 1, channel = 1 Mar 15 16:12:37 ERROR[31964] chan_zap.c: Unable to register channel '1' Mar 15 16:12:37 WARNING[31964] loader.c: chan_zap.so: load_module failed, returning -1 Mar 15 16:12:37 WARNING[31964] loader.c: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# +++ END /var/log/asterisk/full +++ /var/log/messages [EMAIL PROTECTED] ~]# tail -20 /var/log/messages Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_config.so] => (Text Extension Configuration) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_functions.so] => (Builtin dialplan functions) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_ael.so] => (Asterisk Extension Language Compiler) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_dundi.so] => (Distributed Universal Number Discovery (DUNDi)) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_loopback.so] => (Loopback Switch) Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_spool.so] => (Outgoing Spool Support) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_agent.so] => (Agent Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_h323.so] => (Objective Systems H323 Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_sip.so] => (Session Initiation Protocol (SIP)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_features.so] => (Feature Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [skipping chan_oss.so] Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP)) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_local.so] => (Local Proxy Channel) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_phone.so] => (Linux Telephony API Support) Mar 15 16:24:08 asterisk1 safe_asterisk: [chan_zap.so] => (Zapata Telephony w/PRI) Mar 15 16:24:08 asterisk1 safe_asterisk: Asterisk died with code 1. Mar 15 16:24:08 asterisk1 safe_asterisk: Automatically restarting Asterisk. Mar 15 16:24:08 asterisk1 asterisk: safe_asterisk startup succeeded [EMAIL PROTECTED] ~]# +++ END /var/log/messages ++/etc/zaptel.conf+++ [EMAIL PROTECTED] ~]# more /etc/zaptel.conf # Autogen
[asterisk-users] managers
Hi - Am I allowed to have multiple managers logged in with the same manager username at the same time? I'm referring to the id names in manager.conf. I expect so, but just want to check to help in troubleshooting a problem. thanks -todd- ___ --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] wireless desktop phones
Any comments on an ATA and an analog wireless? I've been doing it that way and it works well... Todd On Mar 28, 2007, at 8:31 AM, Dean Collins wrote: Yeh Jordan, my suggestion is don’t. If you read this list you’ll find plenty of people complaining about wireless functionality, the hardware/technology just isn’t there yet. Stick with wired phones and one or two wireless for particular people for now, maybe in 12-18 month things might change. ___ --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] Linksys SPA 3102 causing me problems
Is it behind a router? -t- On Mar 29, 2007, at 6:26 PM, Alan Chandler wrote: I have a linksys SPA 3102 with a DECT phone connected into its Telephone port. It has been working, but something I've done (and I don't know what) means that now everytime asterisk tries to dial it, it says it is busy. I can make calls from it through asterisk I am at a complete loss to know what to try next to fix it. Any ideas? ___ --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] SPA3102 PSTN fallback
Hi - I got a SPA3102. I've set it up without to many problems. If the unit looses power, calls to the PSTN are bridged which is nice. However, if the Asterisk server is unavailable (I turned it off to test), calls out are not bridged to the PSTN. I've rebooted the SPA3102 with the asterisk server off, but still it gives me no dial- tone. Under the configuration, Auto PSTN Fallback is set to Yes. Is there anything else I need to do? thanks Todd ___ --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] Stanaphone business ok?
I see that stanaphone is not accepting new customers. Does anyone know if they are doing ok? I have a number with them and would like to start redirection people before it gets canceled on me if they are having trouble thanks Todd ___ --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
