Re: [asterisk-users] MeetMe - ConfBridge: hint not working
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com wrote:
I'm trying to migrate from MeetMe to ConfBridge:
[conferences]
exten=_8[1-9],1,Answer()
;;exten=_8[1-9],1,Meetme(${EXTEN},1Ms,1234)
exten=_8[1-9],2,ConfBridge(${EXTEN},1Ms)
exten=_8[1-9],n,Hangup
And that works.
Also changed the hints:
;;exten = 81,hint,MeetMe:81
exten = 81,hint,ConfBridge:81
;;exten = 82,hint,MeetMe:82
exten = 82,hint,ConfBridge:82
;;exten = 83,hint,MeetMe:83
exten = 83,hint,ConfBridge:83
;;exten = 84,hint,MeetMe:84
exten = 84,hint,ConfBridge:84
And that does not work. The blf does not go on when a party is in
ConfBridge. Is there some new syntax for hints with ConfBridge?
sean
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Re: [asterisk-users] Wireless Desktop VoIP Phone?
Cisco also make a wireless adapter for the 500 series phones. On Fri, Dec 17, 2010 at 7:40 AM, Matt mhop...@gmail.com wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with HUD Lite
Forget about HUDlite you want iSymphony, http://www.getisymphony.com/ On Sun, Nov 7, 2010 at 5:02 PM, Rupert Utteridge rupe...@dtasia.com.auwrote: Has anyone used HUDlite recently and got it operating with Open Source Asterisk 1.6 or 1.8? I have read the instructions on HUDLite but it appears that it is only suited to Fonality versions like Trixbox. I would like to test HUDLite as a presence panel. If there are other options we are open to this? Rupert Utteridge -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CEL Documentation
Anybody know where to find some good information on the new CEL in asterisk 1.8? I'm very anxious to check out the new logging features but can't find anything but the cel.conf.sample file in the source package. I'd like to get this setup with ODBC. Thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CEL Documentation
Thanks guys, I think I've got enough to go on, I've setup the ODBC stuff before so I think I should be able to figure this out. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon
Make sure to set the voicemail_odbc.so ans voicemail_imap.so modules to noload in module.conf On Wed, Aug 11, 2010 at 4:52 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Wed, Aug 11, 2010 at 06:14:52AM -0400, unsero...@aol.com wrote: Hi all, using Asterisk 1.8 beta3 installed from scratch I am not able to stop/start/restart Asterisk deamon with /etc/init.d/asterisk stop|start|restart It just happens nothing, no warnings, errors etc. Next step: start tracing. sh -x /etc/init.d/asterisk start -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk compatible cards?
I have always had very bad experiences with the x100p cards, they always have very bad echo. If you need decent call quality I would wait until you can afford a Digium card. On Tue, Aug 10, 2010 at 2:49 AM, Gordon Henderson gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote: On Tue, 10 Aug 2010, Daniel Petre wrote: hello all, for my home purpose, i found nothing in my budget in romania, very limited options, so i turned to ebay where i found two cards in the 25 usd price range: a voxzone x100p and a infomatrix x100p, their descriptions from ebay: Never used them myself - always used Digium or openvox cards. I'd be surprised if there was any difference between them though. Google for x100p images and compare the chips... Gordon voxzone x100p: * Universal Voxzone X100P FXO PCI for DigiumTM Asterisk * Works with Official Asterisk Zaptel Driver * Connects Asterisk Box to PSTN * No Echo issues compared to MD3200 * Support via www.voxzone.com/forum/ * Original DAA Chipsets with Caller ID * Low profile bracket is available on request infomatrix x100p (a2 and b2) : # InfoMatrix ? X100P(A2), Intel Chipset, 2nd Generation, PCI, Single FXO port , # Guarantied Caller ID/Redirection, Call transfer, Ring and Remote hang up detected # Support and work with Mutiple Protocols: SIP, IAX, H.323, MGCP, Skinny/SCCP... # Support Global codecs for all countries to use: G711, G726, G723.1, G729A, GSM # Fully tested to support the Asterisk and its appliance, all PC systems(old, new) # 100% compatible with Digium WildCard X100P, X101P card and more features. anyone has any idea which one should i buy? i intend to use it in a SFF dell computer with one pci port (its pretty tight to the PSU wall but i hope it will fit.. any interference because of that?) thanks! On Mon, 2 Aug 2010, Daniel Petre wrote: hello, i just subscribed to this list, i discovered asterisk and i would like to try it at home on my personal pc. the computer is a p4 at 3 ghz with 2 gb ram and 80 gb hdd, a 1 Mbit guarranted connection and runs a gentoo linux. i search about digium products but i can't find them in my area on any shops, i was wondering if good people here could recommend some PCI or PCIex cards for a beginner to play with one telefonic line (which i will install it soon via provider) If you really can't get digium cards, then look on ebay for x100p cards - you might get lucky... Failing that, OpenVox have some compatable cards - you might find an importer locally who deals in them. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.0-beta1 Connectedline
I'm pretty sure only hardphones (Aastra, polycom) support this feature. At least I do not know of any softphones that do. On Sat, Jul 24, 2010 at 5:21 AM, unsero...@aol.com wrote: Hi, i just tried to use the CONNECTEDLINE() feature but it does not work, at least with my softphones (zoiper, 3CX, Xlite) in sip.conf under general I have: trustrpid = yes sendrpid = rpid,pai rpid_update = yes in extensions.conf I have: exten = 2000,1,Set(CONNECTEDLINE(number,i)=98) exten = 2000,n,Set(CONNECTEDLINE(name,i)=test) exten = 2000,n,Set(CONNECTEDLINE(pres)=allowed) exten = 2000,n,Dial(SIP/2000,20) It seems to be executed correctly -- Executing [2...@default:1] Set(SIP/1000-002e, CONNECTEDLINE(number,i)=98) in new stack -- Executing [2...@default:2] Set(SIP/1000-002e, CONNECTEDLINE(name,i)=test) in new stack -- Executing [2...@default:3] Set(SIP/1000-002e, CONNECTEDLINE(pres)=allowed) in new stack -- Executing [2...@default:4] Dial(SIP/1000-002e, SIP/2000,20) in new stack -- Called 2000 -- SIP/2000-002f is ringing -- SIP/2000-002f answered SIP/1000-002e -- Remotely bridging SIP/1000-002e and SIP/2000-002f == Spawn extension (default, 2000, 4) exited non-zero on 'SIP/1000-002e' but neither the number is changed on the calling softphone nor the name is displayed. Did anyone successfully test this, maybe with a hardphone? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX endpoints not Registering after upgrage from Asterisk ver 1.4.26.1
requirecalltoken does not work in the [general] section it must be defined per peer. On Fri, Jul 16, 2010 at 5:43 AM, Steve Edwards asterisk@sedwards.comwrote: On Fri, 16 Jul 2010, Vidura Senadeera wrote: I am experiance a issue with my IAX clients. I have upgradeed Asterisk to 1.4.28 After then IAX clients are not working and It's not registering even. On Fri, 16 Jul 2010, Gordon Henderson wrote: Put requirecalltoken=no in iax.conf for each account. I have: calltokenoptional = 0.0.0.0/0.0.0.0 requirecalltoken= no which I only needed to add to [general]. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi T1 CRC4 errors?
I have a system setup with two T1 circuits, the customer complains of call quality issues. I've checked logs and don't see anything too strange, but when I run dahdi show status I see: Description Alarms IRQbpviol CRC4 Fra Codi Options LBO T2XXP (PCI) Card 0 Span 1OK 0 0 19 ESF B8ZS YEL 0 db (CSU)/0-133 feet (DSX-1) T2XXP (PCI) Card 0 Span 2OK 0 0 774 ESF B8ZS YEL 0 db (CSU)/0-133 feet (DSX-1) The value in the CRC4 column seems to be growing with time. From what I understand CRC4 has to do with E1 circuits not T1. I am I missing something? Is there any further debugging I can do? What could the CRC4 errors mean? I have check the SIP devices latency times and they are all under 40ms so I'm pretty sure it's not an issue with the local network or SIP device. Thanks in advance. Asterisk Version: 1.6.2.7 Dahdi: 2.3.0 Kernel: 2.6.18_164.15.1.el5 /etc/dahdi/system.conf span=1,1,0,esf,b8zs fxsls=1-22 echocanceller=oslec,1-22 unused=23-24 echocanceller=oslec,23-24 span=2,1,0,esf,b8zs em=25-48 echocanceller=oslec,25-48 /etc/asterisk/chan_dahdi.conf [channels] language=en echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0 txgain=0 group=0 callgroup=0 pickupgroup=1 faxdetect=no callreturn=yes cancallforward=yes transfer=yes context=from-zaptel signalling=fxs_ks rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=no immediate=no signalling=fxs_ls language=en context=from-zaptel echocancel=yes echocancelwhenbridged=no rxgain=0 txgain=0 faxdetect=no immediate=no group=0,1 callgroup=1 channel = 1-22 signalling=em_w callgroup=1 group=0,2 immediate=no faxdetect=no txgain=0 rxgain=0 echocancelwhenbridged=no echocancel=yes context=from-pstn language=en channel = 25-48 sip show peers: Name/username HostDyn Nat ACL Port Status 3200/3200 10.10.2.63 D N A 5060 OK (32 ms) 3202/3202 10.10.2.31 D N 5060 OK (12 ms) 3204/3204 10.10.2.88 D N A 5060 OK (13 ms) 3205/3205 10.10.2.85 D N 5060 OK (14 ms) 3206/3206 10.10.2.65 D N 5060 OK (12 ms) 3208/3208 10.10.2.68 D N 5060 OK (12 ms) 3210/3210 10.10.2.59 D N 5060 OK (9 ms) 3211/3211 10.10.2.75 D N A 5060 OK (13 ms) 3213/3213 10.10.2.72 D N A 5060 OK (8 ms) 3214/3214 10.10.2.10 D N A 5060 OK (13 ms) 3216/3216 10.10.2.91 D N A 5060 OK (13 ms) 3217/3217 10.10.2.23 D N A 5060 OK (8 ms) 3219/3219 10.10.2.66 D N 5060 OK (8 ms) 3221/3221 10.10.2.71 D N A 5060 OK (14 ms) 3223/3223 10.10.2.80 D N 5060 OK (12 ms) 3224/3224 10.10.2.37 D N A 5060 OK (14 ms) 3226/3226 10.10.2.83 D N A 5060 OK (14 ms) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Soft-phone on Black Berry
Blackberry has a very high dollar proprietary solution for what you are trying to achieve, I don't think they ever allow SIP soft-phones on their devices. -- Jeremy Betts (714) 388 6015 Ext. 304 Freevoice -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Locking Up?
I am running a server with a Digium TE410P, about 40 grandstream gxp-2000's, 10 Polycom 500's, running svn branch 1.2 rev 44144,and FreePBX. The server only has one PRI at the moment, and at times all 23 channels are full, the phones are all onthe local networkwith the server. Every day or two all SIP devices seem to stop responding, calls still come in over the zap channel are handle appropriately by the queue app and seem to be attempting to ring the SIP device, but the device never returns the ringing status. The call then times out and logs out the dynamic agent as it should.When I try to dial out using a sip device, it seems to hang afterI press send and I see nothing on the Asterisk CLI. Although if I try to dial out using an IAX device, all is well.If I do a sip show peers all the sip devices show as registered and qualify. Can someone help me? Is this a bug? How can I trouble shoot this further? The log file is here Thanks, Jeremy ___ --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] Queue Monitoring Broken?
I have my queues set for monitoring but all I get is a small
corrupted sound file. Any ideas?
Queues.conf
[200]
wrapuptime=10
timeout=15
strategy=ringall
retry=5
monitor-format=wav49
member=Local/[EMAIL PROTECTED],0
member=Local/[EMAIL PROTECTED],0
extension.conf
exten =
200,n,Set(MONITOR_FILENAME=/var/spool/asterisk/monitor/Queue-${EXTEN}-${TIMESTAMP}-${CALLERIDNUM})
exten = 200,n,Playback(PleaseHold)
exten = 200,n,Queue(200|t|||180)
Jeremy Betts,
Direct: 714-388-6013
Toll Free: 866-526-9729
Main: 714-871-9388
Fax: 714-871-6154
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[Asterisk-Users] Audiocodes MP-108
Does anyone have any experience configuring the Audiocodes MP-108 for use with asterisk? Im trying to achieve an easy setup, with 4 POTS lines that will be used for both inbound and outbound calling thru the asterisk server. Im confident I can figure out how to set up asterisk, but the Audiocodes config pages are a bit confusing (to me at least). Any help with the Audiocodes config would be very appreciated. Thanks in advance! Jeremy Betts ___ --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
