[asterisk-users] SIP trunks going to the wrong context
Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes limitonpeers=yes callcounter=yes vmexten=5199 nat=no ; SE registrations register => user1:passwo...@sipgate.co.uk:5060/se2489 register => user2:passwo...@sipgate.co.uk:5060/se1268 register => user3:passwo...@sipgate.co.uk:5060/se0845 register => user4:passwo...@callcentric.com:5060/se1777 register => user5:passwo...@sipgate.co.uk:5060/se4130 register => user9:passwo...@sip.vohippo.com:5060/se1413 ; SJ registrations register => user6:passwo...@sipgate.co.uk:5060/sj0151 register => user7:passwo...@callcentric.com:5060/sj1777 register => user8:passwo...@sipgate.co.uk:5060/sj0203 I then have a selection of #included files. The first defines se2489: [se2489] type=friend insecure=port,invite secret=password1 defaultuser=user1 fromuser=user1 fromdomain=sipgate.co.uk host=sipgate.co.uk port=5060 outboundproxy=proxy.live.sipgate.co.uk disallow=all allow=ulaw context=external-se qualify=yes canreinvite=no dtmfmode=rfc2833 The second defines sj0151: [sj0151] type=friend insecure=port,invite secret=password6 defaultuser=user6 fromuser=user6 fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=proxy.live.sipgate.co.uk disallow=all allow=ulaw context=sj-main regcontext=sj-main ; Added to try to fix wrong context on IB calls subscribecontext=sj-main ; Added to try to fix wrong context on IB calls qualify=yes canreinvite=no dtmfmode=rfc2833 When an inbound call comes in to sj0151, I get the following error: NOTICE[10777][C-]: chan_sip.c:26407 handle_request_invite: Call from 'user1' (217.10.79.23:5060) to extension 'sj0151' rejected because extension not found in context 'external-se'. Surely it should have looked in sj-main, not external-se? Also, the "Call from 'user1' is always 'user1' no matter which sipgate account originated the call. The Callcentric numbers can't receive inbound calls, the vohippo number shows "Call from 'user9'" as one would expect. ALL of them look in context 'external-se', but the SJ registrations should all be looking in 'sj-main'. What's more, it seems to be struggling with pattern matching... The extension is passed correctly (albeit to the wrong context, for 3 of the numbers), so the following dialplan should pick them all up, surely?: [external-se] ; Transfer any call from any SE external trunk to the IVR @ the office. ; If the office is unavailable (no internet, for example), then go to voicemail) exten => _se.,1,Dial(IAX2/cloud/1000,30,r) same => n,Voicemail(5000) same => n,Hangup() However, it simply doesn't work. If I replace _se. with _se2489. (or just se2489), it works fine (for calls arriving on the se2489 extension; obviously the others bork). Can anyone tell me what I'm doing wrong, based on the above? FWIW; this seems to have occurred because I've been attempting to prune my dialplan; I used to have them all going into a single context, and I picked them out & routed them individually. I am _trying_ to simplify the structure/mess that is extensions.conf... but as a result I ran into this little conundrum. The main problem is to resolve the "wrong context"; I have a suspicion I could fix the "can't find extension" problem by getting rid of the letters & using a purely numeric extension. Many thanks, Ade. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and fwbuilder
Hi List, Until recently, I've been running an Asterisk server behind an MS ISA 2004 firewall. In general, this has worked fine - I've been able to connect to my SIP provider to make/receive calls (sipgate.co.uk in the UK and callcentric.com in the US), and DHADI runs the one traditional analogue line I have here. Then, the ISA server went pop, for the umpteenth time. Rather than replace it with yet another flaky second hand Dell server, I've put a spangly new 64-bit HP server in, which needs a 64-bit OS, hence Linux Mint. And, because I'm not entirely sure how well (if at all) ISA Server would work in a VMPlayer, I decided to use Linux's approach to firewalls, aka IPtables, using the GUI program fwbuilder. I finally got most of my network going through iptables/fwbuilder, but I cannot for the life of me make Asterisk talk SIP to the outside world. All attempts to register with sipgate fail. Callcentric appears to register OK, but attempting to make a call and it throws a critical packet not received error aborts. I have (I think) port forwarded 5060 UDP, 5060 TCP and 1-2 UDP to the Asterisk box, as well as IAX2. IAX2 works just fine, I have an external phone connected in using it. Has anyone used fwbuilder to create the rules required to let an Asterisk server make receive calls via SIP? Thanks in advance, Ade. -- _ -- 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] Half-height PCIe analog FXO card
Thanks Eric Tim, I'll put Sangoma back on the shopping list then :) I also found a thing called a Jingletel A400EL which looks like an A400-module-compatible low-height PCIe card but the name suggests it's yet more Chinese knock-off cr*p. With postage to the UK, it's about the same price as a Sangoma, which in turn is about the same as I paid for the server (after cashback). Harumph. I wonder if I could hacksaw an A400E into fitting. If anyone in the UK has an old Sangoma A200 (I only need 1x FXO, if it comes with 2x FXS as well that's a brucie bonus) they'd be willing to sell for up to about 60 quid, please drop me a line. Cheers, Ade. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: 01 June 2012 15:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Half-height PCIe analog FXO card Last time I checked (a few years ago) Sangoma has half height brackets available. Contact their support or sales. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ade Vickers Sent: Friday, June 01, 2012 10:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Half-height PCIe analog FXO card Hi, Does anyone do a low profile PCIe FXO card? I just picked up an HP ProLiant microserver for $nuppence, which I'd hoped to migrate my Asterisk setup onto. I currently use an A400P analog card, but the ProLiant only has PCIe slots, and they're short ones too, so I can't use an A400E card. Even the Sangoma cards, which seem to be low profile, have full-height brackets on them - which, of course, won't fit in the box. Is it just me, or is this whole half-height PCIe thing a complete b***ocks? Any advice appreciated. I'd prefer not to have to spend mega$ on this, the server only cost $200, it seems silly to spend $1000 on a PCI to PCIe converter (Magma.com) to keep using a $100 card... Cheers, Ade. -- _ -- 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] Half-height PCIe analog FXO card
Hi, Does anyone do a low profile PCIe FXO card? I just picked up an HP ProLiant microserver for $nuppence, which I'd hoped to migrate my Asterisk setup onto. I currently use an A400P analog card, but the ProLiant only has PCIe slots, and they're short ones too, so I can't use an A400E card. Even the Sangoma cards, which seem to be low profile, have full-height brackets on them - which, of course, won't fit in the box. Is it just me, or is this whole half-height PCIe thing a complete b***ocks? Any advice appreciated. I'd prefer not to have to spend mega$ on this, the server only cost $200, it seems silly to spend $1000 on a PCI to PCIe converter (Magma.com) to keep using a $100 card... Cheers, Ade. -- _ -- 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] Wanted: UK-specific hardware recommendations (FXOand FXS)
Roger Burton West wrote: I want to hook one of them to the PSTN. Given that I am in the UK, what is a reasonably easily-available device to provide an FXO interface from a Linux box, with a minimum of faffing around with drivers? Just one line is needed, though in theory two might eventually be useful. My usual white-box hardware suppliers don't seem to play in this field. I've had good experiences with an OpenVox A400P, once you've done the Dahdi dance, it settles down to be very reliable. Reasonable price, too. I bought mine from Voipon, although I'm sure a bit of shopping around will find other vendors. Cheers, Ade. -- _ -- 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] UK CallerID -v- Wildcard W100P
Brian wrote: At the risk of being flamed Has anyone had any success get the 'El cheapo' Wildcard W100P clone's (£20 flavour) to work with UK Caller ID? I'm not sure what the status of Asterisk 1.6 is with respect to UK caller ID, being that we have an odd method of sending the FSK ahead of the ring, but I'm guessing I can't be the first to ask this? Nope, I asked some years ago :) I could never get my Wildcard clone to work with UK CLID, no matter what patches I applied. I gave up in the end implemented a very roundabout solution using a Pace modem, second computer, and a database... It worked, albeit a little slowly. Keeping in mind that cost is the most important factor, my searches I've found a couple of suggestions - the most promising of which was reading the CID from a serial modem. However, I've tried a couple - on of which is a BT Enabler that no amount of AT commands can get to give up the CID - and concluded that the chances of finding a compatible modem are probably slimer than getting the clone to work. There are a very small number of modems which work with CLID. Pace being the only ones that I know of, which worked reliably... and Pace went bust years ago. You can pick up 2nd hand Pace modems off eBay, but by the time you've done that, you may as well have bought an A400P card... which will do UK CLID out of the box. If you want to take the Modem route, send me a mail off-list, as I have an implementation that may work for you. Has anyone been able to get the cheap clone cards to offer CID in the UK? Only the A400P. But, TBH, at £55 with 1xFXO module, that's pretty cheap these days. I can heartily recommend them for being a) more reliable, and b) quicker at CLID detection than the modem option... HTH. Ade. -- _ -- 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] IAX2 help needed...
I run a phone in a remote office using the IAX2 protocol. It mostly works fine; except that every 5 mins it loses connection with Asterisk, before reconnecting 30 seconds later; rinse repeat. Using the IAX2 debugging, I'm seeing this a lot: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00018ms SCall: 04050 DCall: 0 [**.**.***.***:4673] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00018ms SCall: 16174 DCall: 04050 [**.**.***.***:4673] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00018ms SCall: 16174 DCall: 04050 [**.**.***.***:4673] RR_JITTER : 0 Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00018ms SCall: 04050 DCall: 16174 [**.**.***.***:4673] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00018ms SCall: 16174 DCall: 04050 [**.**.***.***:4673] RR_JITTER : 0 Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 04050 DCall: 16174 [**.**.***.***:4673] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 16175 DCall: 0 [**.**.***.***:4673] USERNAME: 5111 REFRESH : 60 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGACK Timestamp: 00019ms SCall: 08339 DCall: 16175 [**.**.***.***:4673] USERNAME: 5111 DATE TIME : 2009-06-30 15:27:40 REFRESH : 60 APPARENT ADDRES : IPV4 **.**.***.***:4673 CALLING NUMBER : 5111 CALLING NAME: Ade Vickers (home) Note in particular: Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 04050 DCall: 16174 [**.**.***.***:4673] Whenever this happens, the phone loses connection until a REGACK is received. This started happening when I upgraded Asterisk to v 1.4.22 (from an earlier v1.4.x), on a new machine. Any ideas what I need to do to fix the issue? Phone is a Quartel 710E, in case that's of any use, and it worked fine with my previous Asterisk setup. Cheers, Ade. ___ -- 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] OpenVox A400P01 vs Digium TDM401B
Gordon Henderson wrote: Other than the price (nearly £150 difference), is there any particular reason not to pick an OpenVox A400-based solution for my UK Asterisk needs? None whatsoever. I think the new digium cards are better at interrupt sharing, but if that's not an issue for you, then go for it. I've installed many OpenVox cards. Use Oslec too - works a treat. Excellent, ta for that. I don't think interrupts will be a problem, this will be the only PCI card in the system... Caller ID is the only thing the AX-100P gave me hassle with; does the A400 handle it any better? Remembering that UK CLID is presented between 1st 2nd rings, using V22.bis tones IIRC. I currently use a Pace modem (which has UK CLID capability built in) to capture CLID info... Any thoughts much appreciated, I think you're wrong about UK caller ID.. There is a line polarity reversal, then caller ID is transmitted, then the line rings. You're right, I'd forgotten the polarity reversal; that's what stumpst he AX-100P card, which simply doesn't register the initial reversal. I thought CLID came after the 1st ring, though? I'll check next time I'm on-site with the Asterisk box... You'll need a patch for Zaptel to make it work reliably - same problem with both Digium TDM400 and OpenVox A400 cards too. (ie. it's a driver issue) Look for zaptel-ring.diff if stuck, email me and I'll email my copy. I think I already have it patched with every UK CLID patch I could find... From the time before I gave up getting the AX100P to work. But thanks for the offer anyway! Cheers, Ade. ___ -- 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] Asterisk and rawplayer
BJ Weschke wrote: Ade Vickers wrote: -Original Message- Hi Folks, I'm using the rawplayer program to provide my music-on-hold, and it works very well, with one small drawback: every time I reset Asterisk, for any reason, the MoH resets to the beginning of the list. Since MoH isn't used that often, it basically means the same track is played over over again... What I'd like to do is have rawplayer continuously playing away in the background, even if it's playing to itself only, so there's an excellent chance that any caller who will be given the pleasure of my MoH choices, will get a different tune to the one s/he heard last time... This would probably involve some kind of IPC named pipe or other inter process method of getting the data from pt A to pt B to work. While technically possible, it's not a trivial amount of work to get it going in the codebase. You might be better off with something like streaming MP3 over http or something else like that if you're looking for something with no code modifications. Hm, I was ideally looking for something with no code modifications; e.g. a phantom channel which simply played music to itself, setup when Asterisk starts, or even with manual intervention (e.g. I dial a number, and rawplayer starts up). Are you really resetting Asterisk that much that this becomes a problem? If so, why? My Asterisk install is mainly used for inter-office communications, allowing the Spanish branch to use the UK landline, and testing/experimentation. As such, I frequently do things which require a full restart, or I get it tangled up to the point where it needs a restart. The hold music rarely plays, but because rawplayer always picks the files in the same order, it's almost always track 1 that's playing when I *do* put someone on hold, or whatever; I'd prefer it to be a random start point. Cheers! Ade. ___ -- 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] Asterisk and rawplayer
-Original Message- Hi Folks, I'm using the rawplayer program to provide my music-on-hold, and it works very well, with one small drawback: every time I reset Asterisk, for any reason, the MoH resets to the beginning of the list. Since MoH isn't used that often, it basically means the same track is played over over again... What I'd like to do is have rawplayer continuously playing away in the background, even if it's playing to itself only, so there's an excellent chance that any caller who will be given the pleasure of my MoH choices, will get a different tune to the one s/he heard last time... Any ideas? Asterisk is v1.4.18.1, running on Ubuntu 2.6.20-15.27-server. I'm still stuck with this, and would appreciate any thoughts... Thanks in advance! Ade. ___ -- 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] Asterisk and rawplayer
Hi Folks, I'm using the rawplayer program to provide my music-on-hold, and it works very well, with one small drawback: every time I reset Asterisk, for any reason, the MoH resets to the beginning of the list. Since MoH isn't used that often, it basically means the same track is played over over again... What I'd like to do is have rawplayer continuously playing away in the background, even if it's playing to itself only, so there's an excellent chance that any caller who will be given the pleasure of my MoH choices, will get a different tune to the one s/he heard last time... Any ideas? Asterisk is v1.4.18.1, running on Ubuntu 2.6.20-15.27-server. Cheers, Ade. ___ -- 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] Zaptel 1.4.10.1 and OSLEC on Ubuntu 8.04
Guillermo Salas M. wrote: [ 830.118287] zaptel: Unknown symbol oslec_echo_can_identify Make sure you get the latest version of OSLEC from SVN - the downloadable tarball has a bug in it which prevents it from compiling properly (although it acts like it worked just fine); which then prevents zaptel from loading. If it all still fails, try going back to a slightly earlier version of Zaptel (1.4.9.2). Basically, follow the instructions here: http://www.rowetel.com/ucasterisk/oslec.html (the HowTo - Run OSLEC with Asterisk/Zaptel section) HTH! Cheers, Ade. No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.3.0/1505 - Release Date: 16/06/2008 07:20 ___ -- 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] IAX2 phones, BRI and Analogue cards
Hi Hans, Can't you leave the picking up of the cli to the isdn line? Even if it is an ISDN1 (just a B-channel and a D-channel), the chances of tranferring channel info, like CLI, is better. If a call comes in over the POTS line, then I still need to get CLI over it. I'm not sure if the ISDN can be specified to replace the POTS analogue line, whilst retaining the analogue line + ADSL. Cheers, Ade. Internal Virus Database is out-of-date. Checked by AVG. Version: 7.5.524 / Virus Database: 269.24.6 - Release Date: 03/06/2008 00:00 ___ -- 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] IAX2 phones, BRI and Analogue cards
bilal ghayyad wrote: I would like just to know one thing: Where did u find a good IAX IP Phone? I am looking in the market since long time to buy such device and did not find a reliable one till now. Any advise? I haven't tried any yet; but http://x100p.eu have a few for sale; plus there are some on eBay, one of which I intend to try out, as it looks very similar (identical) to the 6050 for some £30 less... Cheers, Ade. No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.3.0/1498 - Release Date: 11/06/2008 19:13 ___ -- 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] IAX2 phones, BRI and Analogue cards
Hi, I've been asked to spec up a small Asterisk system, which needs to: - Connect to ISDN2e (I'm thinking of using a B100P card here) - Connect to the POTS (A400P with 1 FXO) - Allow remote phones (thinking of an ETC 6050 utilising IAX2) It is a requirement that the POTS analogue card picks up CLI information - and I'm in the UK which, historically, has lousy CLI support certainly, my AX100P doesn't do it... does anyone have any good news about the A400P, or do I need to be hunting down a genuine Digium card? I'm further assuming that an IAX2 phone will work far more reliably through firewalls non-static IP addresses (Asterisk box will be on a static IP, remote/roaming office may not be) than a SIP phone, based on my experiences of getting IAX2 between Asterisks to work. So -- am I on the right lines with the hardware I've specced above, or should I be looking at alternatives? Cheers, Ade. Internal Virus Database is out-of-date. Checked by AVG. Version: 7.5.524 / Virus Database: 269.24.6 - Release Date: 03/06/2008 00:00 ___ -- 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] Call signalling on BT FeatureLine Compact(Sangoma A200)
Paul Goodyear wrote: I have had a BT phone plugged into these lines for about 3 week prior to testing on asterisk, and all the lines are fine. Even the first line, it rings and answers ok. Apologies if this seems dumb, but have you done the swap the cables around test? i.e. swap the cables plugged into BT1 BT2 to make sure the fault stays on BT1? If it does - then it's probably something on BT's end; if it moves, you've eliminated BT from the equation... From what's been posted so far, I'd anticipate a cable fault (either between Asterisk the BT socket, or on the other side of the BT socket...) Cheers, Ade. No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.21.7/1331 - Release Date: 16/03/2008 10:34 ___ -- 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] Telemarketer Torture....
James Finstrom wrote: Anyone have the telemarketer torture prompts? I would seriously like to revive this. I created a simple hold forever loop, thus: [tele_loop] exten = s,1,Answer() exten = s,2,Set(DEVSTATE(Custom:telepark)=INUSE) exten = s,3,WaitMusicOnHold(15) exten = s,4,Wait(1) exten = s,5,Playback(pls-hold-while-try) exten = s,6,Wait(0.25) exten = s,7,Goto,3 exten = h,1,Set(DEVSTATE(Custom:telepark)=NOT_INUSE) exten = h,n,Hangup() ; If anything goes wrong, quit the loop exten = i,1,Set(DEVSTATE(Custom:telepark)=NOT_INUSE) exten = i,n,Hangup() When I get a call from a telemarketer, I either manually dump them into the loop (via an extension defined elsewhere) , or I can drop them into it using CLID. This plays music for 15 seconds, then asks them to wait while we try to put you through, repeat... The 1 second gap between the music ending the announcement beginning is designed to make them think someone's answered. I may need to tune it up to 2 seconds, I'm not sure yet. The devstate Custom:telepark is used so I can monitor if someone's in the loop using a lamp on my phone :) It's not perfect: the loop can hold an indefinite number of calls, but the lamp goes out when the first one exits. Mind you, having the lamp causes its own problems: I tend to find that instead of getting on with my work (which was the plan), I end up timing how long they wait until they give up (record: approx. 18 minutes) instead. So maybe I need to make Asterisk put some tracking information into a database, so I can just run a report at the end of each day :-) Cheers, Ade. (PS: The pls-hold-while-try is either a standard Asterisk sound, or comes with the Asterisk-Addons package, I forget which.) (PPS: I use Asterisk 1.4 with the DEVSTATE patch applied) No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.21.7/1328 - Release Date: 13/03/2008 11:31 ___ -- 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] Asterisk in the call center - how do you do it?
Hi folks, If you are running a call centre (large or small) using Asterisk, I'd be interested to know how you log your agents in out: E.g. - Do you use AgentLogin (to force calls onto the agents, perhaps)? - Do you still use AgentCallbackLogin? - If you use AddQueueMember, are you - running it through the agent's phones (i.e. in the dialplan)? - through a manager interface some software (homebrew or otherwise)? - Do you allow agent hot-desking? - if so, how do you determine which agent is logged in at which desk at what time? - how do you deal with authentication, or don't you bother? It'd also be useful if you could tell me what version of Asterisk you're using. And, finally, a pure fishing expedition: - What kind of reporting (if any) do you currently get out of the Asterisk, and are you happy with it? The reason I'm asking this stuff is because since 2003 I've been working on an ACD reporting product for Nortel Meridians (and, more recently, Avaya and Cisco systems, although that's all early days); and I'm thinking that as Asterisk gains a toe-hold in the call centre market, there maybe a market for this reporting tool for Asterisk users too. The only downside is I just know that my client (who owns the IPR) will never allow the s/w to be opensourced, or even available for free :( But I guess I shouldn't be too unhappy, as it puts the bread butter on my table too... All the above said - I should add that I'm a complete convert to Asterisk, use it daily (albeit at a fairly low simplistic level), e.g. I've only just got around to using a queue on my main POTS line, so I can login at any of the 4 Asterisk boxes I use around Europe, without having horridly complicated dialplans... Many thanks in advance for any responses, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.21.4/1312 - Release Date: 04/03/2008 21:46 ___ -- 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] How do I do this?
I have 2 asterisk servers - serverA and serverC - connected via IAX2. On serverA, I have a telemarketer hold extension which, if I transfer a caller into it, loops around playing music please wait messages, until they give up hang up the phone. Also on serverA, I have a custom devstate, which lights a lamp on a phone connected to serverA, which tells me if someone is currently held in that loop. When they hang up, the devstate is re-set the lamp goes out. On serverC, I have a similar devstate, and a couple of extensions - one to turn the lamp on one to turn it off. What happens is this: 1) A call arrives @ Asterisk, and calls a phone on serverA, and a phone on serverC. 2) I answer on serverC, determine it's a telemarketer, and transfer to the telemarketer hold extension on serverA 3) The call enters the loop, and the devstate is set on serverA. As it enters the loop, it calls the turn on extension on serverC, which sets the serverC devstate, and hangs up with an all extensions are busy response. 4) The call, then, stays parked on serverA until the caller hangs up. 5) The h extension on serverA detects the hangup, and re-sets the serverA devstate. 6) Simultaneously, it calls the turn off extension on serverC, which re-sets the devstate returns a all extensions are busy response. 7) serverA then hangs up the call 'officially' by calling Hangup() Unfortunately: Step 6 doesn't do anything on serverC... you can see it being executed on serverA, but the call never arrives at serverC. I'm guessing this is because the caller has already hung up; so, in effect, there's no call to transfer... My question, then, is how to get Asterisk to generate a new call, to tell serverC to switch off it's lamp? Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.1/1182 - Release Date: 12/12/2007 11:29 ___ --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] How do I do this?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Totaro
Sent: 13 December 2007 14:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How do I do this?
- Original Message -
From: Ade Vickers [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, December 13, 2007 7:49 AM
Subject: [asterisk-users] How do I do this?
I have 2 asterisk servers - serverA and serverC - connected via IAX2.
On serverA, I have a telemarketer hold extension which,
if I transfer a
caller into it, loops around playing music please wait
messages, until
they give up hang up the phone.
Also on serverA, I have a custom devstate, which lights a
lamp on a phone
connected to serverA, which tells me if someone is
currently held in that
loop. When they hang up, the devstate is re-set the lamp goes out.
On serverC, I have a similar devstate, and a couple of
extensions - one to
turn the lamp on one to turn it off.
What happens is this:
1) A call arrives @ Asterisk, and calls a phone on serverA,
and a phone on
serverC.
2) I answer on serverC, determine it's a telemarketer, and
transfer to the
telemarketer hold extension on serverA
3) The call enters the loop, and the devstate is set on
serverA. As it
enters the loop, it calls the turn on extension on
serverC, which sets
the
serverC devstate, and hangs up with an all extensions are
busy response.
4) The call, then, stays parked on serverA until the caller
hangs up.
5) The h extension on serverA detects the hangup, and re-sets the
serverA
devstate.
6) Simultaneously, it calls the turn off extension on
serverC, which
re-sets the devstate returns a all extensions are busy response.
7) serverA then hangs up the call 'officially' by calling Hangup()
Unfortunately: Step 6 doesn't do anything on serverC... you
can see it
being
executed on serverA, but the call never arrives at serverC.
I'm guessing this is because the caller has already hung up; so, in
effect,
there's no call to transfer...
My question, then, is how to get Asterisk to generate a
new call, to
tell
serverC to switch off it's lamp?
Use the h exten? Would you mind sharing more details about
your setup such
as the dialplan or/or apps you are using? I guess you really hate
telemarketers ;-)
Hi Steve,
It's not just telemarketers; I find it's a useful dumping ground for any
caller I don't particularly want to speak to ;)
OK: There are 2 servers involved:
serverA
- Located in the UK, has a connection to a POTS line via an AX100P card.
- Handles any 5xxx extension locally, plus a couple of others
- Talks to serverC via IAX2 channel
- Running Asterisk v1.4.5 + custom devstate patch
serverC
- Located in Spain, has only an internet connection
- Handles any 62xx extension locally, plus the special teledeath_on and
teledeath_off extensions
- Talks to serverA via IAX2 channel
- Running Asterisk v1.4.11 + custom devstate patch
So; in serverA, the following bits of the dialplan are relevant:
[default]
exten = ,hint,custom:telepark
;
-
; When an internal phone dials, this section defines what happens to the
calls
;
-
[internal]
;other destinations cut from here
;Death to telemarketers
exten = ,1,Goto,teledeath|s|1
[teledeath]
exten = s,1,Answer()
exten = s,2,Set(DEVSTATE(Custom:telepark)=INUSE)
exten = s,3,Dial(IAX2/serverC/teledeath_on)
exten = s,4,WaitMusicOnHold(15)
exten = s,5,Wait(1)
exten = s,6,Playback(pls-hold-while-try)
exten = s,7,Wait(0.25)
exten = s,8,Goto,4
exten = h,1,Set(DEVSTATE(Custom:telepark)=NOT_INUSE)
exten = h,n,Dial(IAX2/serverC/teledeath_off)
exten = h,n,Hangup()
; If anything goes wrong, quit the loop
exten = i,1,Set(DEVSTATE(Custom:telepark)=NOT_INUSE)
exten = i,n,Dial(IAX2/serverC/teledeath_off)
exten = i,n,Hangup()
Thus; when I transfer the call to ; it jumps into teledeath|s, which
sets the devstate locally; dials the special extension
IAX2/serverC/teledeath_on to set the serverC busy lamp; then loops around
music/announcements.
When the caller hangs up, teledeath|h is executed; turning off the local
lamp calling IAX2/serverC/teledeath_off - which SHOULD turn off the
serverC lamp, but doesn't - because the call never arrives on serverC...
Here is serverC's extensions.conf file (again, non-pertinent bits removed):
[default]
exten = ,hint,custom:telepark
[internal]
include = outbound
include = default
;Internal phones (local (62xx) remote (everything else)
exten = _[57]XXX,1,Goto,external_extensions|6000${EXTEN}|1
;Death to telemarketers status marker
exten = teledeath_on,1,Set(DEVSTATE(Custom:telepark
Re: [asterisk-users] How do I do this?
Steve Totaro wrote: snippage I suppose you could create a new context on server C, include it in your internal context, and create an h exten on that box to handle it locally. I am unsure why what you have does not work but I assume the unable to transfer is a hint. Except that, once I've transferred the call hung up the serverC end; the call should be entirely handled by serverA; the only further contact the two severs should have in relation to that call is serverA telling serverC to reset the devstate. As you say, the unable to transfer sounds like a clue I wonder if it's due to codecs? Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.1/1182 - Release Date: 12/12/2007 11:29 ___ --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] Can I emulate SIP presence for an extension?
Philipp Kempgen wrote: http://www.asterisk.org/node/48325 http://www.asterisk.org/node/48360 Brilliant, that works a treat, thanks! :) Now, for my next question I have 2 remote sites; 1 @ home, and 1 which I will shortly be transporting to Spain. I've already set up my dialplan so that when a call comes in to the POTS line, it automatically rings on 2 or 3 sites; to do this, it uses an IAX2 channel between the local remote servers. So, for example, a call comes in on the ZAP channel, and rings xtns 5100 and 6100 (5100 is local, 6100 is remote via IAX2). I answer on 6100 (remote) can talk to the caller no problem. Having determined it's a telemarketer, I then transfer them to extension (which is back on the local Asterisk). The appropriate lamp lights on extension 5100's phone, as per the DEVSTATE above. However, because I answered on 6100, I now don't know (from that phone) whether xtn 5500 is live or not... So, is there any way of monitoring the status of a device on a remote server, perhaps utilising an IAX2 channel? Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.15.3 - Release Date: 19/10/2007 00:00 ___ --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] Can I emulate SIP presence for an extension?
I wrote: So, is there any way of monitoring the status of a device on a remote server, perhaps utilising an IAX2 channel? To answer my own question, I did it :) In case anyone's remotely interested in a similar setup/idea, here's the relevant bits of my dialplan. Assume that a call, having been answered, now needs dumping in the telemarketer tarpit of death Up in default, I have the following defined: exten = ,hint,custom:telepark So, that sets up the custom:telepark device, which will be either busy or idle. On serverA (local), I have this extension defined: ;Death to telemarketers exten = ,1,Goto,teledeath|s|1 That jumps into the teledeath context, defined as: [teledeath] exten = s,1,Answer() exten = s,2,Set(DEVSTATE(Custom:telepark)=INUSE) exten = s,3,Dial(IAX2/serverC/teledeath_on) exten = s,4,WaitMusicOnHold(15) exten = s,5,Wait(1.5) exten = s,6,Playback(pls-hold-while-try) exten = s,7,Wait(0.25) exten = s,8,Goto,4 exten = h,1,Set(DEVSTATE(Custom:telepark)=NOT_INUSE) exten = h,n,Dial(IAX2/serverC/teledeath_off) exten = h,n,Hangup() Step 2 sets the custom:telepark to be in use (lighting the lamp on anything connected to serverA(local) which is watching extension ). Step 3 calls the extension teledeath_on on the remote server (serverC) Steps 4-8 keep the telemarketer busy... When they finally give up, exten h is called: Step 1 simply re-sets the device state Step 2 calls extension teledeath_off on the remote server Meanwhile, on my remote server, I simply define a SIP hint: exten = ,hint,Custom:teledeath and two extensions: ;Death to telemarketers status marker exten = teledeath_on,1,Set(DEVSTATE(Custom:teledeath)=INUSE) exten = teledeath_on,n,Set(HANGUPCAUSE=17) exten = teledeath_on,n,Hangup() exten = teledeath_off,1,Set(DEVSTATE(Custom:teledeath)=NOT_INUSE) exten = teledeath_off,n,Hangup() Note that the middle step of teledeath_on is required to prevent the local server from seeing a caller hung up event; which would otherwise cause the music to stop the telemarketer to be disconnected; which is not what we want to happen... The only problem with this scheme occurs if I do something stupid: 1) Dial from any of the connected phones (goes to teledeath loop) 2) Hit transfer, dial send 3) Now I've got a call locked in the system, constantly trying to play music to itself Any ideas on how I could prevent it doing that? Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.15.3 - Release Date: 19/10/2007 00:00 ___ --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] Can I emulate SIP presence for an extension?
I recently implemented a simple spam trap extension for telemarketers - once identified as a telemarketer (usually they ask to speak to the person in charge of recruiting/website/purchasing/etc.), I simply offer to put them through to the person in question, dump them on a special extension which plays music for 15 seconds, then 1.5s silence, then a please wait, we're trying to put you through message; repeat until they give up waiting. I'm using a Grandstream GXP2000 phone, so I've got 7 presence lights; of which I'm only using a couple at the moment. Is it possible in Asterisk 1.4.x to issue a dialplan command which will set a phantom SIP extension to busy for as long as a caller is in the spam trap, back to idle when they finally give up hang up? The basic reason is twofold: 1) I want to see just how long they're willing to wait, and 2) For a sense of personal amusement (yes, I am a bad man) :) Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.15.1/1078 - Release Date: 18/10/2007 17:47 ___ --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] Dell, HP, Digium, homebrew - what do you use
Compaq P3 1GHz server (about 6 or 7 yrs old) running 2gb RAM, 40(?)G HDD, single AX100P. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.14.3/1054 - Release Date: 06/10/2007 19:12 ___ --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] Changing contexts on the fly
Hi,
Many thanks all for the useful tips - I've gone with a (simple!) mySQL table
with a flag in it, indicating the day/night mode, adding the following into
the dialplan:
[external]
; other stuff in here, excluded for clarity
; Include the SJS phone line controls
include = sjs_ctrl
[sjs_ctrl]
; Determine if we're in or out of the office, and divert accordingly
; Note - callerID is set because it doesn't get it from the line :(
exten = s,1,NoOp(-- ${CALLERID(number)} calling on ZAP channel)
exten = s,2,Set(CALLERID(number)=unknown)
exten = s,3,Set(CALLERID(name)=SJS Line 1)
exten = s,4,MYSQL(Connect connid db_server login_id super_secret_password
db_name)
exten = s,5,MYSQL(Query resultid ${connid} SELECT\ currentStatus\ FROM\
myStatus)
exten = s,6,MYSQL(Fetch fetchid ${resultid} MyStatus)
exten = s,7,MYSQL(Clear ${resultid})
exten = s,8,MYSQL(Disconnect ${connid})
exten = s,9,GotoIf($[${MyStatus} = y]?10:12)
exten = s,10,GoTo(sjs,s,1)
exten = s,12,Goto(sjs-ooh,s,1)
[sjs]
exten = s,1,NoOp(-- ${CALLERID(number)} calling on ZAP channel)
exten = s,n,Set(CALLERID(number)=unknown)
exten = s,n,Set(CALLERID(name)=SJS Line 1)
exten = s,n,Dial(SIP/5100,30)
exten = s,n,Answer()
exten = s,n,Wait(0.75)
exten = s,n,Voicemail(5100,u)
exten = s,n,Hangup()
[sjs-ooh]
exten = s,1,Answer()
exten = s,n,Wait(0.75)
exten = s,n,Playback(thank-you-for-calling [etc - lots more soundfiles
here])
exten = s,n,Voicemail(5100,s)
exten = s,n,Hangup()
Then, in the internal extensions config, I've added the following:
; Switch SJS day/night modes
;Daytime (star star D)
exten = **3,1,NoCdr()
exten = **3,n,Answer()
exten = **3,n,MYSQL(Connect connid db_server login_id super_secret_password
db_name)
exten = **3,n,MYSQL(Query resultid ${connid} UPDATE\ myStatus\ SET\
currentStatus\ = \ \'n\')
exten = **3,n,MYSQL(Clear ${resultid})
exten = **3,n,MYSQL(Disconnect ${connid})
exten = **3,n,Playback(daytime)
exten = **3,n,Hangup()
;Nighttime (star star N)
exten = **6,1,NoCDR()
exten = **6,n,Answer()
exten = **6,n,MYSQL(Connect connid db_server login_id super_secret_password
db_name)
exten = **6,n,MYSQL(Query resultid ${connid} UPDATE\ myStatus\ SET\
currentStatus\ = \ \'n\')
exten = **6,n,MYSQL(Clear ${resultid})
exten = **6,n,MYSQL(Disconnect ${connid})
exten = **6,n,Playback(nighttime)
exten = **6,n,Hangup()
So I can switch between day night modes with **D or **N (3 or 6
respectively) :) Dead simple stuff so far, I may get more whizzy with it
later on... At some point, I'll probably switch over to a fully realtime
config, so I can DIY my own user interface.
Cheers!
Ade.
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.488 / Virus Database: 269.13.35/1040 - Release Date: 30/09/2007
21:01
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[asterisk-users] Changing contexts on the fly
Hi folks, I've been playing around with an Asterisk server in my office for a few weeks now, and I've got it pretty much nailed down the way I want it, which is nice. One of the features I'm using is the ability to switch different contexts in out of the dialplan on a schedule. So, for example, I've got the official tel number ringing my desk phone between 9.00-17.30 mon-fri; and out of those hours any caller gets a recorded message + sent to voicemail. However, I'm quite often working later than 17.30, and would quite like to be able to easily flick a switch which tells Asterisk that, actually, I'm here in the office, and I'd quite like to receive calls. Currently, I have to alter dialplans.conf, comment out a couple of lines uncomment another; save then re-load the dialplan. I'm guessing I've got 3 options open to me: 1) Convert from using the various .conf files, to using a realtime config, then write a small front-end to the DB so I can access the settings from a simple switch on my Windows desktop 2) Write some kind of script which I can execute on the Asterisk box which makes the same changes I'm currently making manually 3) Some other option I've not thought of... What's the panel's opinion on the best way to do this? For info: Asterisk 1.4.5 running on Ubuntu 7.04 Digium-compatible AX100P card providing connection to POTS line (this is the one that needs controlling) 2 SIP extensions (Grandstream GXP2000) Numerous SIPGATE lines (these are configured as I like them already) Much appreciated in advance. Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.13.33/1034 - Release Date: 27/09/2007 17:00 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] [OT] IAX2 WiFi phone?
Does such a beastie exist? I've tried a couple of UT Starcom WiFi SIP phones (the F1000g and F3000 respectively), and found them both to be seriously lacking - regular crashes (especially the F3000), poor battery life, and poor reception in particular. However, whilst SIP phones are great, I'd really like an IAX2 phone if there is one, as I can make that work natively though the firewall, connected directly to a remote Asterisk server (remote = the other end of a broadband link). Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.1/965 - Release Date: 21/08/2007 16:02 ___ --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] [OT] IAX2 WiFi phone?
Brandon Kruse wrote: /me goes to work. There are none that I know of. There are only a couple of IAX(2) hard phones, and none of them, that I know of, are manufactured in the US anyways, and have problems. (Of course, what is manufactured in the US these days) That would be a great device, would love to see it come about. That was pretty much what I'd concluded from my googling :( Of course, for me, I'd like to see a device available in the UK/Europe the US is fine, but your power supply is a bit incompatible with ours, and the shipping cost is a nightmare ;) If I could convert my Grandstream GXP-2000's to IAX2, i'd be as happy as a pig in, er, you get the picture; I could then drop the one Asterisk per site setup I'm currently stuck with - although I suppose the advantage of that particular setup is all internal calls are truly internal... Anyway - if anyone hears of an IAX2 WiFi phone in the works, please do drop a line in here Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.1/965 - Release Date: 21/08/2007 16:02 ___ --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] 2 asterisk servers, how to connect them together?
Hi... I have what is, I am sure, a relatively common straightforward problem (no, NOT that kind of problem!)... I'm trying to hook two asterisk servers together so I can make a distributed PBX. Here's the scenario: [MASTER] is in the office. It has unrestricted access to the internet, and a fixed IP address. It has 3 SIP hardphones configured working, plus a couple of softphones which log in/out as necessary. The phones are on extensions 5100-5104, with a special extension 5999 which just plays music. [HOME] is at home. It has internet access only through a Microsoft ISA 2003 firewall, and has a dynamic IP address. It has 1 SIP hardphone configured, and working, on extension 5110. I can add a second hardphone to verify that this (new build) server is working OK, but all of the messages indicate it's fine. What I want to do, obviously, is have ALL of the extensions (5XXX) pretending to be on the same PBX. i.e. if I dial 5100 (on [MASTER]) from 5110 (on [HOME]), the call goes through everyone's happy; and vice versa, calling 5110 from 5100. I know I need to use IAX to achieve this (as IAX can negotiate its way past the firewall), but I can't find the magic incantations for IAX.CONF (on either server) to make them talk nicely to each other. They did, very briefly, as the [MASTER] server spotted the IP address of [HOME], added it to the peer list, my heart rose; but, now it's dead again. Rather than post my broken conf files here, can anyone suggest a nice'n'easy way to get this to work? Many thanks in advance. Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.0/959 - Release Date: 17/08/2007 17:43 ___ --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] 2 asterisk servers, how to connect them together?
Panic over... I have a weird network problem (now solved), whereby incoming packets arrived directly to the Asterisk box (eth1); but outgoing packets attempted to leave via the LAN (eth0)... solved it by sending the IAX packets thru the firewall at both ends of the connection (i.e. binding IAX to the LAN address instead of the WAN address). Freaky? You betcha... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ade Vickers Sent: 18 August 2007 23:47 To: asterisk-users@lists.digium.com Subject: [asterisk-users] 2 asterisk servers, how to connect them together? Hi... I have what is, I am sure, a relatively common straightforward problem (no, NOT that kind of problem!)... I'm trying to hook two asterisk servers together so I can make a distributed PBX. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.0/960 - Release Date: 18/08/2007 15:48 ___ --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] Music on hold - 1.4.5
Stephen Bosch wrote: Ade Vickers wrote: Hi Richard, Thanks for those replies - I'll give them a shot shortly. That's not really what I meant by configuration -- you can choose the MOH source for Asterisk. It's only the native player that restarts the music file every time someone is put on hold. We're still using 1.2, which doesn't have this problem, so I don't know how it's done in 1.4. Ah well, whatever was meant, using rawplayer works more or less as I'd hoped (music doesn't re-cue from the start). It's not a constant stream (if no MoH is playing at all, the stream freezes so the next person to get music gets it from the point where the previous person left off... The most amusing bit was when the original raw files I'd converted from MP3 using a Windows tool played at approx 2x their normal speed! Re-converting with mpg123 fixed that... Cheers! Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.14/885 - Release Date: 03/07/2007 10:02 ___ --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] Music on hold - 1.4.5
Stephen Bosch wrote: Russell Bryant wrote: Lacy Moore - Aspendora wrote: On 6/29/07, Ade Vickers wrote: What I'd like to do is have the music streaming constantly, so the on hold caller always gets music at the current position; even if that's in the middle or near the end of a file. Many of us would like this, but the powers that be decided they didn't want that, and I don't know enough about coding to figure out how to change it. I'm not quite sure what you're referring to. I think this would be a very welcome addition. I thought this was just a matter of configuration. Hi all, thanks for the responses so far. I too understood it to be a configuration thing, with the addition of a streaming music server (which, obviously, provides the MoH stream). Asterisk should then simply pick up the stream play it whenever MoH is requested. It'd also be nice to periodically interrupt the stream with a your call is important to us (no, honestly) message, although I probably wouldn't use that on my own server. Does no-one have any suggestions for a streaming MoH setup? Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.14/884 - Release Date: 02/07/2007 15:35 ___ --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] Music on hold - 1.4.5
Hi Richard,
Thanks for those replies - I'll give them a shot shortly.
Cheers,
Ade.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Richard Lyman
Sent: 03 July 2007 16:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Music on hold - 1.4.5
Richard Lyman wrote:
Ade Vickers wrote:
*snipped
Hi all, thanks for the responses so far.
I too understood it to be a configuration thing, with the
addition of
a streaming music server (which, obviously, provides the
MoH stream).
Asterisk should then simply pick up the stream play it
whenever MoH is requested.
It'd also be nice to periodically interrupt the stream
with a your
call is important to us (no, honestly) message, although
I probably
wouldn't use that on my own server.
Does no-one have any suggestions for a streaming MoH setup?
here are my notes
http://dynx.net/ASTERISK/gnudialer/moh.txt
(replying to my own post because i am sure the next reply
will be how does this work)
when you use the above method it will load up all the files
in that dir to the rawplayer instance (like this)
... Jun18 1:09 /usr/bin/rawplayer fpm-calm-river.raw
fpm-sunshine.raw
fpm-world-mix.raw
by doing so, it will cycle those 3 files.
you can test it out using a variation of the ael snippet below.
WaitExten(1);
Playback(one-moment-please);
WaitMusicOnHold(5);
for (x=0; ${x} 3; x=${x} + 1) {
Verbose(3|x is ${x} !);
Playback(thnk-u-for-patience);
WaitMusicOnHold(5);
};
Playback(pls-hold-while-try);
i hope this helps
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Date: 02/07/2007 15:35
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Checked by AVG Free Edition.
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15:35
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[asterisk-users] Music on hold - 1.4.5
Hi, Please bear with me if I'm asking stupid questions... I'm new to Asterisk, newish to Linux, etc... I've got MoH working nicely with my new Asterisk setup using the files option; except that it always plays from the start of a (random) music file when you first put someone on hold. Take them off hold put them back, and sometimes (not always!) it will start playing a new file from the beginning If I park a call, from the point of pressing the TRNF button the caller gets music; but, when the call parks, the music starts a new file! What I'd like to do is have the music streaming constantly, so the on hold caller always gets music at the current position; even if that's in the middle or near the end of a file. The musiconhold.conf file mentions a couple of streaming options; but (rightly) doesn't go into particular detail. So, what's my best strategy? For info: - Asterisk is running on a P3 1GHz server (it's only a tiny experimental PBX setup though) - v1.4.5, compiled by myself (thanks to voip-info.org a couple of other sites) - Server is Ubuntu Fiesty Fawn, clean install (especially for Asterisk) - VoIP (SIP) only - All music files are in uLaw format, and the SIP phones are forced to use uLaw encoding. Cheers! Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.12/878 - Release Date: 28/06/2007 17:57 ___ --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
