Re: [Asterisk-Users] No busy-tone
Hi, I think you want this to be 102 since a busy returns n+101 n being the priority your Dial function was called. exten = _X.,101,Busy should be exten = _X.,102,Busy HTH -b Quoting Eric Wieling [EMAIL PROTECTED]: Nicklas Bondesson wrote: Just like this? It doesn't seem to work though. [wx3trunk-outgoing] include = internal-sip-callers exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,T) exten = _X.,101,Busy ___ 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 -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIPURA does not register with Asterisk
Hi, The first thing is I dont see a username=username in your config.I recommend keeping the username the same as what you have in the brackets, in this case [201] If you are behind a NAT device make sure to add nat=yes And if you dont have it configured in the [general] context be sure to handle your codecs. eg: disallow=all allow=ulaw ;or whatever Other than this, are you sure the 2000 is getting an IP address assigned to it. Look in your docs, off hand I think on the touch tone pad of the phone you have plugged into the 2000 you do a then 110 but you'd better check into this incase Im wrong. So, have at it like this: [201] username=201 type = friend host = dynamic secret=201 dtmfmode = rfc2833 context = default [EMAIL PROTECTED] nat=yes canreinvite=no callerid = 201 201 disallow=all allow=ulaw HTH Ciao, -b Quoting Naren Koka [EMAIL PROTECTED]: I have a SIPURA 2000 which is supposed to register with the Asterisk server. However it does not register at all. I have two lines registered in the SIP.conf [201] type = friend host = dynamic secret=201 dtmfmode = rfc2833 context = default [EMAIL PROTECTED] canreinvite=no callerid = 201 201 -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk call forwarding / remote dial-in/out?
Yes you can, It's called DISA. Realize that using DISA has it's potential security concerns. From the asterisk console type show application DISA for more information. DISA = Direct Inward System Access Ciao, -bh Quoting Angus Berry [EMAIL PROTECTED]: I haven't found this in any docs or faqs yet, so I'm wondering if I can achieve what I would like to do. On an Asterisk PBX with multiple PSTN lines, I'd like to call in from one PSTN line, probably via cellphone and access the PBX as if I were local to it. From here I'd like to get a dial tone and call back out. I know this isn't exactly call forwarding per se, but I'm wondering if this can be done. thanks ___ 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 -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't block CallerID outbound
Curious if anyone else has run into this. I am testing this with a Sipura-2000 with firmware rev 1.0.33. On FC1 with Asterisk 1.0 Stable branch 4/1/2004 CVS (no April fools jokes please :) ) The Sipura has the ability so when you dial *67 it turns ON CID block and *68 turns it back off. (This is for outbound calls) When I *67 (activate CID Block) dial, and look at the SIP INVITE I see that it added Anonymous in the From: Field When I *68 and look at the SIP INVITE Anonymous is no longer in the From: field, instead it has the SIP UserName. So I think the Sipura is doing it's job. However even though the From: field is set to Anonymous my callerID still gets sent through to the called number. I realize that Asterisk gets my callerID from the sip.conf file callerid= Name 555-555- but isn't Asterisk supposed to Block the callerID if it sees Anonymous in the SIP INVITE ? Oh, FWIW - the other *XX features like call forwarding work fine hopefully ruleing out dtmf and dial plan issues etc.. Thanks -bh -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G726 not working ?
Hi, Thanks for the reply! I am still having troubles I did try: disallow=all allow=g726 And still get: Mar 30 15:49:29 WARNING[-1137677392]: chan_sip.c:2128 process_sdp: No compatible codecs! -b Quoting Michael Manousos [EMAIL PROTECTED]: Hi, Bill Hamel wrote: Hi, I am running FC1 with latest patches of 3/30/04, and I have the latest CVS as of this morning 3/30/04 of asterisk, zap and libpri. The SIP device I am using is a Sipura SPA-2000 with G726-32 Forced. When I 'make clean and recompiled zaptel, libpri, asterisk and start asterisk I can see: [format_g726.so] [format_g726.so] = (Raw G.726 (16/24/32/40kbps) data) == Registered file format g726-40, extension(s) g726-40 == Registered file format g726-32, extension(s) g726-32 == Registered file format g726-24, extension(s) g726-24 == Registered file format g726-16, extension(s) g726-16 This is the support for reading/writing raw G.726 (all rates) data from/to files. It is not the G.726 codec. The codec is the codec_g726.so module (32 Kbps only). I 'Ass'ume this indicates that g726 is installed.. So in my sip.conf I put many variations of what I thought should go in there, finally includeing (to no avail): disallow=all allow=g726-40 allow=g726-32 allow=g726-24 allow=g726-16 allow=g726 allow=ima-adpcm You should allow g726 only. (Also tried G.726-xx etc... ) And none seem to work because when I dial out I get Mar 30 15:49:29 WARNING[-1137677392]: chan_sip.c:2128 process_sdp: No compatible codecs! I must not be putting the correct allow= value in sip.conf or possibly missing something. Can anyone point me in the right direction ? Thanks -bh Michael. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] C7960 busy notification
Perhaps the 7960 has Call Waiting set to YES - this being the case, you're really not hitting a busy but the 15 sec timeout instead. Try setting CallWaiting=No in the 7960 you should then get a 'busy'. Oh, another thing. If you have multiple line appearences configured as the same SIP phone on the 7960, you will have the same problem if the call rolls to the next available line. HTH, -b Quoting Rich Adamson [EMAIL PROTECTED]: Using the following defnitions with a C7960: exten = 3001,1,Dial(SIP/3001,15,r) exten = 3001,2,Voicemail2(u3001) exten = 3001,102,Voicemail2(b3001) exten = 3001,103,Hangup If someone is on this phone (real conversation) and another call comes in, the second call goes through the 15 second timeout and is dropped into the 2-priority as unavailable (not the 102 busy as expected). Do I need to do something different to hit the 102 busy priority? Rich ___ 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 -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Watchguard Firebox 1000 and Asterisk
The firebox has the UDP timeout set pretty low by default, this is a good thing to help prevent DOS attacks, but isn't a really good thing for a SIP device. There is no option in the GUI to set this. However you can go into the config file itself and modify the following: options.masquerade.udp.timeout: 30 options.services.dynamic.timeout.udp: 25 Set them higher than your register timeout on your 7960. Then save the config file and upload to the firebox. HTH -bh Quoting Glenn Dalgliesh [EMAIL PROTECTED]: Has any had any experiences with Watchguard Firebox 1000 and Asterisk. I have asterisk on public side and phones on the private side. I am able to get the phones to register and make outbound calls but the inbound calls are intermittent. I have NAT enable in asterisk and on the Cisco 7960. Any insight would be appreciated. Thanks ___ 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 -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G726 not working ?
Hi, I am running FC1 with latest patches of 3/30/04, and I have the latest CVS as of this morning 3/30/04 of asterisk, zap and libpri. The SIP device I am using is a Sipura SPA-2000 with G726-32 Forced. When I 'make clean and recompiled zaptel, libpri, asterisk and start asterisk I can see: [format_g726.so] [format_g726.so] = (Raw G.726 (16/24/32/40kbps) data) == Registered file format g726-40, extension(s) g726-40 == Registered file format g726-32, extension(s) g726-32 == Registered file format g726-24, extension(s) g726-24 == Registered file format g726-16, extension(s) g726-16 I 'Ass'ume this indicates that g726 is installed.. So in my sip.conf I put many variations of what I thought should go in there, finally includeing (to no avail): disallow=all allow=g726-40 allow=g726-32 allow=g726-24 allow=g726-16 allow=g726 allow=ima-adpcm (Also tried G.726-xx etc... ) And none seem to work because when I dial out I get Mar 30 15:49:29 WARNING[-1137677392]: chan_sip.c:2128 process_sdp: No compatible codecs! I must not be putting the correct allow= value in sip.conf or possibly missing something. Can anyone point me in the right direction ? Thanks -bh -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent / Queue help
Hi, First let me apologize if I sent this to the list twice. Is it possible to kick a caller out of a queue after 5 minutes and goto the next priority in the context where they were assigned to the queue ? My desired result is that even though one agent is dynamically logged into the queue and is on a call, I would like the 2nd caller to stay in the queue for 5 minutes and then timeout to the next priority if the agent is still busy and can't get to the call. Some observations: I have tried the n option with queue (if I don't the 2nd caller will stay in the queue infefinately) eg: exten = 401,1,Queue(support1|n) The problem with using n is that with one agent logged into the queue and he is busy on a call, when the 2nd call is placed in the queue it immediately timesout and goes to the next priority in the context even if timeout=300 is set in queue.conf. Any help appreciated. -bh Here are the configs: extensions.conf [supportq] exten = 401,1, Queue(support1|t) agents.conf [agents] autologoff=15 ackcall=no ;wrapuptime=5000 musiconhold = default queues.conf general] [support1] music = default strategy = leastrecent ;context = leavemessage timeout = 300 retry = 2 maxlen = 0 -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is there a MaxQueueTime for Queues ?
Hi, Is it possible to kick a caller out of a queue after 5 minutes and goto the next priority in the context where they were assigned to the queue ? My desired result is that even though one agent is dynamically logged into the queue and is on a call, I would like the 2nd caller to stay in the queue for 5 minutes and then timeout to the next priority if the agent is still busy and can't get to the call. Some observations: I have tried the n option with queue (if I don't the 2nd caller will stay in the queue infefinately) eg: exten = 401,1,Queue(support1|n) The problem with using n is that with one agent logged into the queue and he is busy on a call, when the 2nd call is placed in the queue it immediately timesout and goes to the next priority in the context even if timeout=300 is set in queue.conf. Any help appreciated. -bh Here are the configs: extensions.conf [supportq] exten = 401,1, Queue(support1|t) agents.conf [agents] autologoff=15 ackcall=no ;wrapuptime=5000 musiconhold = default queues.conf general] [support1] music = default strategy = leastrecent ;context = leavemessage timeout = 300 retry = 2 maxlen = 0 This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Hi, Have you checked for IRQ conflicts ? -b Quoting Steve Foy [EMAIL PROTECTED]: Hi, On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote: Steve, this really is a FAQ. You need add to EACH (!) sip user something like disallow=all allow=ulaw allow=alaw allow=gsm I do have that in my sip.conf. I am using ulaw. Calls from the SIP phones through Asterisk and out one of my X100P cards are working 95% of the time and also, incoming calls through the X100P cards to the SIP phones are the same. The only problem is that every once in a while, without any odd circustances that I can see, the call just drops and the remote user is gone. The box running asterisk isn't under heavy load, so I can't see why this is happening. I am not using g.729 or 723, just plain old ulaw, which I have got enabled in sip.conf Cheers, Steve -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ 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 -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Shot in the dark here ... Do you have: canreinvite=no Set in sip.conf for the SIP phones in question ? Ciao, -b Quoting Steve Foy [EMAIL PROTECTED]: Hi, I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating. Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards. There is nothing in the logs and nothing on the console, the call just seems to 'go away'! Can anyone shed any light on this? This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Try adding it to the phones involved so it looks like this: ; Shirley [100] type=friend username=xxx secret=xxx host=dynamic dtmfmode=rfc2833 callerid=Shirley O'Neill 100 context=internal [EMAIL PROTECTED] qualify=yes canreinvite=no -b Quoting Steve Foy [EMAIL PROTECTED]: Bill, On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote: Shot in the dark here ... Do you have: canreinvite=no Set in sip.conf for the SIP phones in question ? No, I don't. All I have in sip.conf is the general stuff like: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=all allow=GSM allow=G729 allow=iLBC allow=SpeeX; Allow all codecs allow=ulaw and then about 10 friends like this: ; Shirley [100] type=friend username=xxx secret=xxx host=dynamic dtmfmode=rfc2833 callerid=Shirley O'Neill 100 context=internal [EMAIL PROTECTED] qualify=yes -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ 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 -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do you turn on the 7960 msg waiting light?
I can only speak for the SIP IOS load on the 7960's (We're running 6.1 ) but if you add: [EMAIL PROTECTED] It should work Note: 7188 being the mail box number and ContextInVoicemailConf being the context in the voicemail.conf file where the mail box 7188 exists. Example: [7188] type=friend username=7188 secret=7188 host=dynamic nat=no dtmfmode=inband context=mycontext callerid=Bubba (555)-555-1212 [EMAIL PROTECTED] canreinvite=no amaflags=default disallow=all allow=ulaw allow=alaw ;End HTH -b Quoting Paul Mahler [EMAIL PROTECTED]: Does anyone in Asterisk land know how to turn on the message light on the back of the earpiece of a cisco 7960 when a message is waiting? -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: help with voicepulse connect IAX2
Curious what your iax.conf looks like. Also FWIW - if you are connecting directly to VoicePulse with a SIP phone, wouldn't that mean that you have a SIP account and not an IAX account ? -b Quoting yair hakak [EMAIL PROTECTED]: hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am pretty convinced the SIP setup is OK. This is the error message: Jan 29 12:21:54 NOTICE[262161]: app_dial.c:527 dial_exec: Unable to create channel of type 'IAX2' when i try to call the PSTN from the SIP device. i've tried the wiki, the handbook, the voicepulse site, and all sorts of other sites, and nothing helps. i also downloaded and compiled the code today (jan 29) and that didn't help either. if anyone has ideas i would be eternally grateful - this is driving me crazy. thanks- yair p.s. i am using the right login and password; not the ones from the website, and i know my account at voicepulse works because i can connect direct through a SIP client. it seems to be a specifically IAX2 problem. here are my files sip.conf ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=gsm [yairphone] type=friend insecure=no username=yairphone secret=yairphone host=dynamic dtmfmode=inband callerID = Yair Hakak nat=true extensions.conf [general] ; static=yes writeprotect=no [default] exten = _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],20) exten = 8665,1,Dial(SIP/yairphone,20) iax.conf [general] port=5036 disallow=all allow=ulaw jitterbuffer=no [voicepulse] context = VPWS secret=mypassword auth=md5 type=friend host=gw5.voicepulse.com _ The new MSN 8: smart spam protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ 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 -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is there a way to # of agents logged into a queue ?
Hi, Looking around I can't seem to find a way to show the number of agents currently logged into a queue and if possible who they are. Is there a way to do this ? Thanks -b -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent timeout then Dial() ?
Hello, I have agents / queues working to the extent that agents can login, logout and I can send a caller into the queue and the logged in agent's phones will ring. Maybe I've spent to much time googleing and reading and my eyes are crossing now, but what I am trying to do is this but cannot find any reference to it. 1. Xfer the caller into the Queue... If Noone is logged into the queue, the caller will be directed to a PSTN number instead (or extension, same thing) 2. Xfer the caller into the Queue... Agents are logged in, but the call times out for whatever reason, I would then like to have it go to an extension as in above 3. When say 6PM rolls around and all agents are gone I would like to automagically log them out just incase they forgot to. I will be happy with an answer for 1 and 2 - I can always use a big stick for #3 :) I did find a reference to adding a member local in queues.conf eg: member = local/[EMAIL PROTECTED],10 And have a context in extensions.conf like this [timeout] exten = s,1,Wait,1 exten = s,2,Answer exten = s,3,Playback(transferring_you_offsite) exten = s,4,Dial,IAX2/office/[EMAIL PROTECTED] Even with the metric of '10' to try and give the local member less preference it will give logged in agents like half a ring and then xfer to the timeout context right away. Any help, pointers would be greatly appreciated. Many thanks -bh -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 Phone disconnects when dialing using speaker
Quoting Brian West [EMAIL PROTECTED]: Works fine here.. got two of em. bkw Hmpf! I donno whats wrong then, both phones do the same thing. So you can keep the headset in the cradle, hit the 'speaker' button, dial a call and it doesn't disconect ? I wonder, are you using an xml dial plan or anything on you phones ? Thanks -bh On Fri, 16 Jan 2004, Bill Hamel wrote: Hi, Just got some CISCO 7960 phones. They seem to work great except if I make any SIP call using the speaker phone (leaving the hand set in the cradle)the call will disconnect in about 5 or so seconds. If I pick up the hand set and make a call, it's fine. Has anyone else run into this ? Any solution ? The phone is on SIP v6.1 - it did the same thing on 4.4 5.0 and 6.0. Thank you in advance, -bh -- This message was sent using IMP, the Internet Messaging Program. This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voicepulse
Atually with the root servers dropping their domain name announcement nothing would have helped. Well, except for hard codeing the IP rather than using fqdn in the config. Or making a static entry in the local hosts file ( both have it's issues) I prefer to use IP rather than fqdns when possible. But that can introduce other problems if the host system decides to move you to another host machine by just changing the DNS name. Using fqdns in mission critical applications is not a good idea IMHO, it just adds another layer of something that can go wrong. Just my $.02 worth ;) -b Quoting Chris Albertson [EMAIL PROTECTED]: --- Steve Sobol [EMAIL PROTECTED] wrote: Matt Lawson wrote: I was just about to write the same thing. It says busy. Is is REALLY busy or is something else wrong? This on the heels of switch-1.nufone.net being missing out of DNS. We have customers that expect their VOIP to work. Is there anybody that's reliable? I've been doing some testing and so far I'm not 100% impressed by the VOIP services I've seen. They provide a good service but my local phone company and ATT longdistance service is more reliable. But this is not to say _you_ can't built a reliable VOIP based system. Get _two_ providers and set up your dial plan in extensions.conf to fail over if one service fails to connect to dial via the next one and finally if both fail use pstn. your users will see a system the just works. About Nufone's problem. I bet they'll start thinking about getting a backup DNS service and maybe geographic deversity. A company should be able to even stay on the air if there is a server room fire using techniques like round robin DNS and West cost and East coast servers run by different, unrelated hosting companies. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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 -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7960 Phone disconnects when dialing using speaker
Hi, Just got some CISCO 7960 phones. They seem to work great except if I make any SIP call using the speaker phone (leaving the hand set in the cradle)the call will disconnect in about 5 or so seconds. If I pick up the hand set and make a call, it's fine. Has anyone else run into this ? Any solution ? The phone is on SIP v6.1 - it did the same thing on 4.4 5.0 and 6.0. Thank you in advance, -bh -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZT_CHANCONFIG failed on channel 2
Hi, FWIW This issue had been resolved. The fix is nothing to speak of except that maybe this post may be informative for someone out there. It turned out to be a hardware issue in the PC, after swapping the Zaptel cards to another PC, it has been up and running with no ZT_CHANCONFIG failed on channel 2: No such device or address (6) [FAILED] errors anymore. Unfortunately I don't have the cycles to research exactly what on the MB or NIC may have been causing this, it could be anything, it's easier just to leave the Zaptels in the new box and find something clever to do with the old one like tie it to my bumper and drag it around for a bit :) I can say this though, the box would loose network connectivity ~60 seconds after the error. An ifconfig -a showed the NIC Up and the routing table on the box was correct as well. Link lights looked fine and the CISCO catalyst port showed up/up as well. The Zaptel cards did not need to be powercycled because a shutdown -r worked. Regards, -bh Quoting [EMAIL PROTECTED]: Hi, Thank you for the reply, actually the cards installed are a TDM400P (Single port) and an X100P. I don't need to power down the PCI cards by turning of the PC, a simple shutdown -r now does it. -bh Quoting Tilghman Lesher [EMAIL PROTECTED]: On Thursday 11 December 2003 17:42, [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] asterisk]# service zaptel start Loading zaptel framework: [ OK ] Loading zaptel hardware modules: wcfxo wcusb Running ztcfg: ZT_CHANCONFIG failed on channel 2: No such device or address (6) [FAILED] Typically this means that the driver cannot detect the device. Since it is the S100U that seems to be undetected, try unplugging it, waiting 20 seconds, and plugging it back in. -Tilghman This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
