[Asterisk-Users] Parial Hang with cvs-HEAD and queues/agentcallbacklogin
Title: Parial Hang with cvs-HEAD and queues/agentcallbacklogin Hi Last night I upgraded an asterisk install from cvs of early this year to current cvs head and all seemed to be working OK, but now Im having several problems which seem to be related to queues. First off queues dont work, theres no error message, the channel just seems to hang cli output as follows: -- Executing Answer(Local/[EMAIL PROTECTED],2, ) in new stack Jul 7 20:09:46 WARNING[27638]: channel.c:640 channel_find_locked: Avoided initial deadlock for '0x86e3948', 10 retries! -- Executing Playback(Local/[EMAIL PROTECTED],2, support-welcome) in new stack -- Local/[EMAIL PROTECTED],1 answered SIP/ed-1-fc54 -- Playing 'support-welcome' (language 'en') == Spawn extension (itg, 800, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2ZOMBIE' -- Executing Set(SIP/ed-1-fc54, CALLERID(name)=Support) in new stack -- Executing Queue(SIP/ed-1-fc54, support|t|||180) in new stack When I hang up the dialling phone there is no cli ouput and show channels shows the channel as still there: SIP/ed-1-93ce (macro-queueinbound s 4 ) Up Queue support|t|||180 Calling an agent produces the same result, and show agents on the CLI produces no output. Were using dynamic agents with agentcallbacklogin. Other calls seem to proceed OK, although it does seem to be rather slow for instance 4 gotos and a set callerid takes approx 6 seconds. This is a low load system using no more than 3-4% cpu normally and asterisk isnt using an abnormal amount of cpu or memory. Does anyone have any ideas whats causing this, or how to set about debugging it further? Many thanks Ed ___ 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] fxo connection in the UK
Most people get echo issues with x100p's in the UK due to mismatched impedance, the newer TDM400P is much better, and you could get this with 3 FXO modules (otherwise known as a TDM03B I believe). HTH Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Hoppe Sent: 30 November 2004 12:14 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] fxo connection in the UK Thank you very much for this hint. My apologies that I messed up a thread for my post - I had a message open and simply clicked on the link ... slap slap. Would anyone know of a better choice to multiplex three fxo lines into an asterisk box? I can still use three Digium X100P cards, but methinks, a seperate unit would be better. Thanks again, Peter I am located in the UK and am looking into connecting three analog BT lines to an astersik system which is replacing our current pbx. I could use three Digium wildcard x100p cards for that but I rather use a unit which is external to the computer to have a better separation of analogue/digital side. I would not like to go ISDN because the analog lines have so far sufficed in every repect and I tend not to fix what isn't broken. Today I found a unit on a supplier's website http://www.peripheralcorner.co.uk/product_info.php/cPath/113/products_id/544 which is a Micronet SP5054 VoIP Gateway 4 FXO Ports The website for this product is http://www.micronet.info/Products/voip/SP5054.asp and it appears to me that this unit would (similar to a channel bank) multiplex our three BT lines into one LAN port. If so, I could simply connect such a box to a LAN port in my asterisk server. I suspect the unit would appear to the asterisk box like three SIP-to-fxo converters (sorry for the horrible beginner-jargon). I basically would like to know whether I could use this unit instead of three x100P cards and it would functionally replace them. Questions * Would it be legal in the UK to connect such a unit to the PSTN ? In the specifications http://www.micronet.info/Products/voip/SP5054.asp#Specif there seems to be CE regulatory approval (see bottom row of table, Emission), but I don't know whether that is sufficient for use with a PSTN in the UK. * Could I operate the unit in a transparent fashion, i.e. it would look to the asterisk machine as if I had connected three SIP-to-fxo converters which I can control independently of each other from the asterisk machine? For example, could I initiate / receive a phone call while another phone conversation is running? For outgoing calls, could I specify which fxo port to use / for incoming calls, could I find out which port answered it? * Would the fxo ports match the UK PSTN specifications (impedance)? I am asking in this list just in case that someone has used / is using such voip gateways. I am still very much in the enquiry phase. Thank you very much for your consideration. Peter On 29 Nov 2004, at 17:30, Peter Hoppe wrote: Micronet SP5054 VoIP Gateway 4 FXO Ports We bought one of these units and had a lot of grief with them. The SIP firmware isn't great at all IMHO. For H.323 they work just fine. Stephan Wik -- There are 10 kinds of people in the world, those who understand binary, and those who don't. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sending broadcasts to all phones?
As explained on the wiki page http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config you don't just do a dial(phone1phone2) you put all the phones through to a conference, the one drawback of this is that you have to set one of the cisco's lines to autoanswer, which you probably won't want most of the time. If you've got a 7960 this probably won't worry you too much (who needs 6 lines anyway?), but on a 7940 it could be a pain. I can't see any way around this without modifying the cisco firmware (unless you could do something cunning with the cisco phone's telnet interface?). HTH Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: 17 October 2004 22:30 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Sending broadcasts to all phones? I am in the process of writing an app to do this with Cisco phones7940/60. The feature on most PBX's is Page Groups, This allows paging through the speaker phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Baker Sent: Saturday, October 16, 2004 5:36 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sending broadcasts to all phones? The Polycom phones will do this. Use the meetme feature. It's well documented on the Wiki. John David J Carter wrote: I have a Panasonic switch here and it a paging system on the switch. It will output the page message to all phones and also to an RCA (Phono) socket on the side of the switch to a PA amplifier if required to drive a 100Volt line system around a building. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kevin Walsh Sent: 16 October 2004 22:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Sending broadcasts to all phones? Kristian Kielhofner [EMAIL PROTECTED] wrote: Stan Brinkerhoff wrote: A friend of mine has a real panasonic PBX setup at his house, and is able to pick up the phone, dial an extension, and it broadcasts what he says over every phone in his house without the phones having to be picked up. What is this feature called? Would it be possible to set this up with Asterisk given the appropriate phones? (Cisco?) This can be done with Cisco phones and 6.x or 7.x firmware. It is on the wiki. Well, actually, it's not on the WIKI. The WIKI would help you set up a Cisco phone to auto-answer, but that's not all he needs here. The problem is that if you dial phone1phone2 then the first phone to auto-answer will receive the broadcasted call. The other phones in the list will not hear anything. Well, that'd be what I'd expect to happen with Dial(), anyway. Stan seems to be asking for a system where the caller hears a ring tone until all phones (auto)answer, and is then able to speak to them all at once. It'd be kind of like an enforced conference call, but with one speaker and multiple listeners, and with all audio received from the called phones thrown away rather than distributed. It could be done, but would need a new Dial()-based application to do it, I think. Perhaps there's an existing facility that can be used to to do this. If there is then I can't think of it. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] spandsp
I think the port.h in this distribution may have been created from tiffv3.5.7 while you have tiffv3.6.0 - (or maybe something else), anyway I had this problem, and installing tiffv3.5.7 and copying the port.h from that distribution to /usr/local/include fixed it HTH Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maurizio Marini Sent: 17 September 2004 11:31 To: [EMAIL PROTECTED] Cc: administrator tootai Subject: Re: [Asterisk-Users] spandsp On Thursday 19 August 2004 23:29, administrator tootai wrote: I made one. Can be found at http://ftp2.tootai.net/spandsp-0.0.1k-whole.tar.gz The 3 headers files are included, made a short readme file for installation and modify the Makefile.patch (remove the dtmftotext). Comments welcome. debian sid with littiff3-dev libtiff4-dev installed; compiling spandsp i get this error: gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -c t4.c -MT t4.lo -MD -MP -MF .deps/t4.TPlo -fPIC -DPIC -o .libs/t4.lo In file included from /usr/include/tiffiop.h:45, from t4.c:38: /usr/include/tif_dir.h:240: error: conflicting types for `TIFFFieldInfo' /usr/include/tiffio.h:448: error: previous declaration of `TIFFFieldInfo' make[2]: *** [t4.lo] Error 1 `TIFFFieldInfo' is defined in tif_dir.h and in my tiffio.h: /usr/include# grep TIFFFieldInfo * tif_dir.h:} TIFFFieldInfo; tif_dir.h:externvoid _TIFFMergeFieldInfo(TIFF*, const TIFFFieldInfo[], int); tif_dir.h:externconst TIFFFieldInfo* _TIFFFindFieldInfo(TIFF*, ttag_t, TIFFDataType); tif_dir.h:externconst TIFFFieldInfo* _TIFFFieldWithTag(TIFF*, ttag_t); tiffio.h:} TIFFFieldInfo; tiffio.h:const TIFFFieldInfo *info; tiffio.h:extern void TIFFMergeFieldInfo(TIFF*, const TIFFFieldInfo[], int); tiffio.h:extern const TIFFFieldInfo* TIFFFindFieldInfo(TIFF*, ttag_t, TIFFDataType); tiffio.h:extern const TIFFFieldInfo* TIFFFieldWithTag(TIFF*, ttag_t); tiffiop.h: TIFFFieldInfo** tif_fieldinfo; /* sorted table of registered tags */ what do u suggest me? -- Maurizio Marini ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Patching UK Caller ID
You need to run: patch -p1 ast-UK-and-DTMF-pol-CID.diff You may need to change the -p1 to -p0 depending on the paths in the diff. I don't think this patch applies cleanly with current CVS head - I know for sure the 31-08 version does. HTH Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Gardiner Sent: 14 September 2004 20:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Patching UK Caller ID I'd be grateful for some help on this. I've been following the various e-mails on the UK CID issue, particularly the last posting in bug fix 9. It seems that all I need now is to apply ast-UK-and-DTMF-pol-CID.diff. I apologise in advance for what is probably a very simple question, but how do I apply this patch. I've copied the file in to the directory containing the source files for asterisk (which is /usr/src/asterisk) and then I've run: patch ast-UK-and-DTMF-pol-CID.diff Unfortunately nothing happens, just a blank line appears and no command line prompt. The other question I've got is the directory structure. In the diff file the location of the file is given as: RCS file: /usr/cvsroot/asterisk/callerid.c,v I don't have a directory called /usr/cvsroot/. I've installed Asterisk by following the standard CVS install procedure. Am I missing something blindingly obvious or should I be RTFM (in which case, could someone point me to the right manual!). Even if I replace cvsroot with src I get the same no-activity/blank line. I really would like to get UK caller ID working so I'd be grateful for any pointers from those on the list who know a hellava lot more than I do. Many thanks, George PS I have looked at three linux manuals I've got at the patch command and can't find anything obvious for me to follow. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK Disconnect supervision with TDM400P
Dan, I get a long single beep that continues for about 20secs and then hangs up. I use irc all the time (as Whisk), but don't always remember to connect to freenode/#asterisk, I'll jump on in the next few days and maybe we can have a chat about this in some more detail. Thanks Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Tucny Sent: 06 September 2004 18:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] UK Disconnect supervision with TDM400P On Wed, 2004-09-01 at 22:02, Edward Eastman wrote: Hi, thanks for the reply, only just got round to having a look at it again (annoying how real life gets in the way of the important stuff ;) I've had a go at ramping up the tx/rx gain but it doesn't seem to make any difference. FWIW it's the same with the module in normal fcc mode. Does anyone know if bt do normally provide disconnect supervision or whether it has to be done with e.g. busydetect (and can either be detected by the tdm400p in uk mode)? Thanks Ed Edward Eastman wrote: I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line, loading wcfxs with OPERMODE=UK. All's working well, except if I get an incoming call through my bt line, and the remote party hangs up, I get approx 20secs of the bt line hungup tone before asterisk hangs up, which leads (if nothing else) to the well documented 20secs of beep on vm problem :) I have tried: busydetect=yes / busycount=7 / other busycounts / callprogess=yes but none of these make any difference. I have loadzone/defaultzone=uk and country=uk in indications.conf and fxs_ks signalling. Try increasing your RX gain in 1db steps, until it reliably hangs up. I had a box with X100Ps which busydetected perfectly with default gain settings. When they were replaced with TDM FXOs, busydetect stopped working and I needed 3db of RX gain added to get it working again. Regards, Richard Ed, When someone does hang up on you with your BT line, what do you hear? Here I get a click/pop following by a 4 second unobtainable tone followed by a click/pop... The clicks are BT's 'k-break's... It obviously doesn't seem to be what * expects... Investigating this is something I'm hoping to have a look at soon, but, if you have time beforehand BTW have you used the IRC channel? Dan (dant) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK Callerid bug #1719 TDM400p
Hi Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the best/only way to get callerid working in the UK with a tdm400p? I thought Id seen a patch thatd gone into cvs, but maybe I was just imagining things ;) Should this patch work against current cvs? Of the 3 files 2 are .patch and one is .diff whats the difference between them, and how should I apply the diff (at the moment Im doing patch p1 patchname.patch for the others which seems to work, but Im doing this slightly blind and Im not quite sure if this is correct. Thanks Ed ___ 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] UK Callerid bug #1719 TDM400p
Brilliant - thanks, took me half an hour but it's working now. Just for the record, settings as follows: The patch on http://bugs.digium.com/bug_view_page.php?bug_id=009 (ast-UK-and-DTMF-pol-CID.diff) doesn't seem to work for current cvs, but I backed up to cvs as of 31/08/04 and that worked fine. Zapata.conf: usecallerid=yes cidsignalling=v23 cidstart=polarity usecallerid=uk doesn't work, has this changed somewhere along the way, or is this something else? Caller ID detects fine, although I get this logged to asterisk console: Sep 6 13:56:22 WARNING[393238]: chan_zap.c:3369 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. I'll try and add this to the wiki when I get time Thanks Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Soren Rathje Sent: 06 September 2004 13:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK Callerid bug #1719 TDM400p Edward Eastman wrote: Hi Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the best/only way to get callerid working in the UK with a tdm400p? I thought I'd seen a patch that'd gone into cvs, but maybe I was just imagining things ;) Check the bug tracker for id=9, there has been some development here. UK BT CLIP and DTMF CLIP for the TDM400 (will not work for the X100P) is now merged into one patch. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call back on failed transfer or dial?
I think what you want is an attended or consultative transfer, this can be accomplished in different ways depending on your setup, for zap channels see http://www.voip-info.org/wiki-Asterisk+tips+zap+transfer with SIP this will normally be implemented on your hard/soft phone. Another alternative is to use call parking (not exactly what your after, but can achieve the same end): http://www.voip-info.org/wiki-Asterisk+call+parking. HTH Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shabanip Sent: 04 September 2004 13:43 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] call back on failed transfer or dial? hi, i'm under the impression that this feature is not available in asterisk, consider this scenario: - you are the operator. you answer a call from outside and you want to transfer it to one of the extensions. after you transfer, if the person you transferred the call to, doesn't pick up or if his line is busy, the call is transfered back to you, you can speak to the caller and tell him, for example, that the person you want to talk to is not in, and ask if he would like to talk to leave a message or talk to another person instead. now in asterisk, it seems to me that after you transfer a call to an extension, there's no way to have the caller transfered back to yourself if the called extension doesn't answer or if it's busy. is this correct? thanks, - shabanip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK Disconnect supervision with TDM400P
Hi, thanks for the reply, only just got round to having a look at it again (annoying how real life gets in the way of the important stuff ;) I've had a go at ramping up the tx/rx gain but it doesn't seem to make any difference. FWIW it's the same with the module in normal fcc mode. Does anyone know if bt do normally provide disconnect supervision or whether it has to be done with e.g. busydetect (and can either be detected by the tdm400p in uk mode)? Thanks Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Scobie Sent: 28 August 2004 21:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK Disconnect supervision with TDM400P Edward Eastman wrote: I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line, loading wcfxs with OPERMODE=UK. All's working well, except if I get an incoming call through my bt line, and the remote party hangs up, I get approx 20secs of the bt line hungup tone before asterisk hangs up, which leads (if nothing else) to the well documented 20secs of beep on vm problem :) I have tried: busydetect=yes / busycount=7 / other busycounts / callprogess=yes but none of these make any difference. I have loadzone/defaultzone=uk and country=uk in indications.conf and fxs_ks signalling. Try increasing your RX gain in 1db steps, until it reliably hangs up. I had a box with X100Ps which busydetected perfectly with default gain settings. When they were replaced with TDM FXOs, busydetect stopped working and I needed 3db of RX gain added to get it working again. Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Headset for Cisco 7960?
Cisco headset pinout is different from normal ones (grr) If it's just for you, (ie nothing too professional ;) you can snip the lead of an existing plantronics type headset and do some reordering - this will give you the necessary info (sorry - can't remember exactly how I did it): http://www.mml.uni-hannover.de/einhorn/headset/index_e.html If you're after something more professional then obviously one of the leads/adapters will be a better approach. HTH Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nate Carlson Sent: 31 August 2004 21:05 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT: Headset for Cisco 7960? Sorry, I know it's OT, but does anyone know of a relatively inexpensive headset that is compatible with the Cisco 7960? I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queue Announcement not until after # accept callpressed
This is something I'm after as well, what I have found is the following: http://bugs.digium.com/bug_view_page.php?bug_id=0001082 http://lists.digium.com/pipermail/asterisk-dev/2004-February/003201.html which pretty much does what I(you) want, the one problem with it is that while the agent is listening to the pre # announcement, MOH for the queued party stops. Other than this I can confirm the patch works well with CVS Head 08/03/04. Does anyone else have anything better, or any status on the above patch? Thanks Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Brown Sent: 27 August 2004 15:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Queue Announcement not until after # accept callpressed When using the callback feature on agents I notice that when the queue calls one of the agents and the agent picks up the call they hear nothing until pressing the # to accept the call. Only then does my announcement play back to the agent after which the call is immediately connected. Is there a way to have the announcement played to the agent before they press # to accept. I have ackcall=yes in agent.conf Can't find anything on the wiki. Thanks Andrew [exten.conf] exten = s,1,Answer exten = s,2,background(custom/100) ; Sales exten = 1,1,ringing(2) exten = 1,2,playback(custom/101) exten = 1,3,queue(sales) [queue.conf] [default] ; ; Default settings for queues (currently unused) ; [sales] music = default announce = sales_queue; This not played until after # pressed .. How can i get announce to play as soon as call answered? announce-frequency = 20 strategy = roundrobin timeout = 15 retry = 5 maxlen = 0 member = Agent/7001 member = Agent/7005 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK Disconnect supervision with TDM400P
Hi I know this gets covered fairly regularly, but I've had a search through the archives and can't find anything dealing with this specifically - apologies if I've missed it though. I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line, loading wcfxs with OPERMODE=UK. All's working well, except if I get an incoming call through my bt line, and the remote party hangs up, I get approx 20secs of the bt line hungup tone before asterisk hangs up, which leads (if nothing else) to the well documented 20secs of beep on vm problem :) I have tried: busydetect=yes / busycount=7 / other busycounts / callprogess=yes but none of these make any difference. I have loadzone/defaultzone=uk and country=uk in indications.conf and fxs_ks signalling. I've seen mention of the different BUSYDETECT flags in the * Makefile, but I can't seem to find exactly what they do, or whether they're likely to improve anything. Does anyone have disconnect supervision with a TDM400P working well in the UK? Can anyone provide some pointers to getting this working? Thanks Ed ___ 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] randomize Dial() target
Look at call queues http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20queues if you don't want to mess around with MOH holding positions etc use the 'r' argument for the queue app. HTH Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marcin Mazurek Sent: 16 August 2004 14:22 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] randomize Dial() target Hi, is it possible to randomize extension which would be choosed by Dial()? I would like to forward phone calls to one of sales rep in randomized way (not to harm anyone;) ). tia mazek -- http://www.marcinmazurek.com/ ::: nic-hdl: MM3380-RIPE GnuPG 6687 E661 98B0 AEE6 DA8B 7F48 AEE4 776F 5688 DC89 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?
IAX2 uses udp port 4569, so youll probably have to open that up on your
firewall/router.
http://www.voip-info.org/ is a good starting place for any asterisk problems
- specifically:
http://www.voip-info.org/wiki-Asterisk+firewall+rules
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
HTH
Ed
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Blunt
Sent: 15 August 2004 23:06
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Inbound Free World Dialup - extension not
ringing?
Hi Lyle,
Thank you so much for your help, I think your information points to using
IAX2 rather than registering with FWD from the sip.conf
I have made an attempt to understand this, added the appropriate information
into iax.conf, remove old info from sip.conf, gone to fwd and ticked the IAX
registration box, and I now get my local sip phone ringing when I dial in
from FWD! Hurrah, unfortunately I get no sound in either direction. Do
you have any experience of this or could it be due to me being inside a NAT
firewall? I have port 5060 forwarded to my * server, should I forward any
other ports? (I can only forward a maximum 20 single ports due to a
limitation on my home router).
As yet I am unable to make outgoing calls over FWD, I figured I would look
at this next.
Is there a NAT solution that could be used with sip.conf rather than the
IAX?
Again your help is most appreciated.
Best regards
Chris
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: 15 August 2004 15:14
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Inbound Free World Dialup - extension not
ringing?
You need a defination for the inbound FWD and what to do with that.
In my extensions.conf, I have:
[globals]
FWDNUMBER=123456 ; your actual fwd number
FWDCIDNAME='My Name'
FWDPASSWORD=myfwdpasswd
FWDRINGS=sip/office
FWDVMMBOX=1010
[fwd_out]
exten = _123.,1,SetCallerId,${FWDCIDNAME} ; replace 123 with the desired
access code to dial out via FWD
exten =
_123.,2,Dail(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN}:3},60
,r)
exten = _123.,3,Congestion
[local]
include = fwd_out :add to local context
[default]
;inbound dialing from FWD
exten = ${FWDNUMBER},1,Goto(housemenu,s,1) ; I have mine set to hit a
menu, no reason you cann't forward to an extension instead
- Original Message -
From: Chris Blunt
To: [EMAIL PROTECTED]
Sent: Sunday, August 15, 2004 3:29 AM
Subject: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?
Hi to all the * people out there,
Please kind to me as I am both new to Asterisk and to Linux But I am
learning fast.
My config is quite simple, Im just following examples and the Wiki: I have
two PCs running X-Lite phones, these work without problems between each
other, and I have a GS BudgeTone-100 registered to Free World Dial UP
(working no problem).
I have tried to set up Asterisk to accept calls from FWD on another number I
have registered, but I cant get my local X-Lite to ring on an inbound call
from FWD, and I get the busy tone on the BT100
When I sip debug, I can see that I am registered with FWD, and when I call
the number from the BT100 I can see all the incoming information but still
nothing on my X-Lite.
My extensions.conf:
[general]
static=yes
writeprotect=no
[globals]
[sip]
exten = 1,1,Dial(SIP/phone1,20,tr)
exten = 2,1,Dial(SIP/phone2,20,tr)
exten = 2,2,VoiceMail,u1234
exten = 2,102,VoiceMail,b1234
;exten = 1000,1,Dial(SIP/phone1SIP/phone2,20,tr)
exten = 1001,1,Ringing
exten = 1001,2,Wait(2)
exten = 1001,3,VoicemailMain,s1234
exten = 6601,1,WaitMusicOnHold(60)
exten = 232999,1,Dial(SIP/phone1,30,tr)
exten = 232999,2,Hangup
I am behind a NATed fire wall, but Im not sure that is related.
Any ideas or help (working simple confs) would be much appreciated.
Best regards
--
Chris Blunt
SIP: [EMAIL PROTECTED]
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