Re: [asterisk-users] Queue and Interface time out
Okay, that makes sense. I wasn't thinking about the SIP driver needing to be told to track the peer's status. I assumed it just did that. So now there's a new problem. The Queue application doesn't always clear the member interface's status after completing a call. The SIP peer no longer has an active channel but the queue will still show the member 'In use'. The occurrence of this is erratic and I have been unable to determine any commonalities among the callers or members other than that it happens to all members. Connecting to the peer outside of the queue will clear the status. Any ideas? Thanks, James Watkins, Bradley wrote: What it actually does is tell the SIP channel driver to track whether or not any given peer has a call to it. It can then subsequently inform the Queue application so that another call will not be given to that user. If you did not have the ringinuse=no in your queue definition, you would then be able to receive up to 5 simultaneous calls (after five, then the SIP channel driver would return busy and Queue wouldn't be able to dial that peer). Regards, - Brad From: [EMAIL PROTECTED] on behalf of James Fromm Sent: Fri 1/19/2007 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out That worked. I don't understand what call-limit has to do with this. I set it to 5. Why does that keep the member interface from getting a second call from the Queue application? I would think it would allow the member interface to get up to 5 calls. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnect Supervision UK / BT solution?
thanks for your helpful investigation! I await news :-) -- Chris - Original Message - From: Matt Brown [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, January 20, 2007 7:55 AM Subject: Re: [asterisk-users] Disconnect Supervision UK / BT solution? Well, I have just phoned BT today who said they can increase the CPC value on the line - however it needs to be done at the exchange - and has been booked for Tues. I suppose I will know wether this worked on Tues :-) - I shall post my findings. Regards -- Matt Brown On 19 Jan 2007, at 16:46, Chris Earle ((CBL)) wrote: Hi all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : TDM400P amp; Not Detecting Hangups: Got a TDM400P installed and having problems with Asterisk not detecting hangups? Using BT? If so, contact BT and ask what the Disconnect Clear Time setting is for your phone line. Odds are it's probably 100. Increasing it to 800 fixed the issue for me. Disconnect Clear Time is BT's name for CPC. Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Comments appreciated before I get on the phone with BT -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content and is believed to be clean. -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with rxfax
Dear list, The company I'm working for is trying to use app_rxfax to receive faxes on an Asterisk machine. Our initial tests looked very promising, but unfortunately we've encountered some problems. We've been trying to solve these problems for quite some time now, but we're running out of options. So I really hope that somebody can give some help here. Basically our set-up is this: We have an Asterisk server (version 1.2.7) with an ISDN trunk (Sangoma A104D), we've configured asterisk to run rxfax on a specified extension. Originally we started out with spandsp 0.0.2pre26 and the original app_rxfax for spandsp 0.0.2. Some of the faxes were coming in perfectly, but soon we noticed that quite often there were substantial pieces of the fax missing in the resulting tif file. We've tried the following to solve these problems: - We've checked the timing settings for the ISDN trunk, these seem to be ok. - We've tried several versions of libtiff (currently we are using 3.7.2). - We've tried using 0.0.3 versions of spandsp (since we're using asterisk 1.2.7 we had to modify app_rxfax.c to work). - We've created a custom dialplan application to disable the echo cancellation on the isdn channel on which the fax is received. - We've tried various settings for t30_set_supported_compressions, t30_set_supported_image_sizes, t30_set_supported_modems and t30_set_supported_resolutions (I must confes that I didn't really know which settings to use here, but we have tried a lot of them). - We've tried several fax machines to send the faxes, ranging from simple fax-modems to large dedicated fax machines. But still a lot of faxes give problems, either the tif is missing large portions, or the fax isn't received at all. At the bottom of this mail are 2 examples of the logging when it goes wrong. I really hope that somebody can give a few pointers. Thanks in advance, Kind regards, Ardjan Zwartjes, Telecats === Example 1 = Jan 19 14:49:14 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:14 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:17 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:17 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:19 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:19 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:20 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-22.18dBm0) Jan 19 14:49:21 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:21 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-17.84dBm0) Jan 19 14:49:38 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:38 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:39 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:39 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:41 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:41 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:42 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-19.13dBm0) Jan 19 14:49:48 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:48 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: == Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Pages transferred: 1 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image size: 1728 x 1192 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image resolution8037 x 7700 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Transfer Rate: 9600 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Bad rows0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Longest bad row run 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Compression type3 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image size (bytes) 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: == Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 1 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: == Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: Fax successfully received. Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: Remote station id: Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: Local station id: Test Fax Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: Pages transferred: 1 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: Image resolution: 8037 x 7700 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: Transfer Rate:
[asterisk-users] QueueMemberStatus/Status Field
Hi all, Where can I find the status definitions for the status field below? Googled /web/voip-info.org and can't seem to find anything. Event: QueueMemberStatus Privilege: agent,all Queue: support Location: SIP/111 Membership: dynamic Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 Thank you! -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN30 and TDM400P + FAXing ...
On Sun, 21 Jan 2007, Robbie Hughes wrote: So it sounds like it should just work. I'll let you know in a few weeks time :) TDM400P to E1/T1 card faxing fails by design. The lack of synchronisation between cards means it can *never* work with any reliability. The hardware will not permit it. (And I know I've asked this before, but any recommendations for a basic quality PRI card in the UK? I only need a single port and it'll be servicing 15 incoming lines on the ISDN30 it'll be connected to) Works perfectly* for me in the UK as described with 2 DDI's going to 2 ports on a TDM400P from the digium TE110P card which also works fine. No echo, no problem with faxes. We use SNOM 190/300/320 phones for voice. *worked immediately on setup at client site and haven't had a single complaint about faxing with 3 different kinds of fax machine in 9 months of continuous operation. That's exactly what I need and great to know, thanks! Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: One way choppy sound
On 1/19/07, Martin Joseph [EMAIL PROTECTED] wrote: On 2007-01-17 10:29:43 -0800, Yelson Vivas [EMAIL PROTECTED] said: Hi Guys I'm conecting 2 astersk servers using this arquitecture (Ext softphone)==sip==(asterisk 1)iax2 trunk (asterisk 2) ===alaw==(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call from Ext to the pstn, i can hear perfect but they tell me that sound really choppy, i tried using several codecs (same problem) but i don't understand why the sound is bad in only one way. Any sugestions to solve it more than welcome Usually sounds can be choppy one way due to constrained upstream bandwidth. There might be plenty of room for the audio to get to you, but that doesn't mean the reverse is at all true. Jitter buffering can help this, or using a more compact format (like GSM or g729) is also a potential helper. Good luck, hope this helps, Marty On 21 Jan 2007, at 03:46, Andrew Joakimsen wrote: I've actually found in many cases a lower bandwidth codec doesn't improve at all and however it oftentimes makes the issue worse. What other traffic do you have on the IAX trunk link ? Even if it isn't 'full' you may be hearing your IAX packets being delayed behind 'bigger' packets, or sitting in a low priority queue on a router. You might want to look into applying a QoS to the link. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT:spa942 provisioning
Benko wrote: Hello! Sorry for the OT-thread, but i don't know where else too ask... Has anyone done http-provisioning of a Linksys SPA942 with client side ssl-authentication? Where do i get the CA from? I'm aware of the Sipura mass deployment howto on voip-info.org, but it doesn't cover the authentification part. Thanks Christian You get the server certificate from Linksys. You'll need to be a reseller or service provider though, or the reseller/service provider you buy from may be able to request one on your behalf. You need to use something like OpenSSL to generate a CSR to send to Linksys. cheers, Paul. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT solutions
On 21 Jan 2007, at 07:55, Brad Templeton wrote: Some NAT problems you can solve, some you never will. Many modern phones have NAT support in them, via STUN, or a static external IP address. Most NATs also offer port forwarding, so you can open a hole for the SIP port in the NAT so all outside can reach it. (With port forwarding, you need a constant address for each SIP phone, so that means either static IP for the phone, or a DHCP server with the ability to always bind a device to the same address - the latter is preferable because you can move your phone to other networks more easily.) Many devices also feature NAT keep alive on the SIP port. That is a must if you can't open ports, but it sure generates a lot of annoying debug output when you turn on sip debug. Nothing beats a permanent NAT entry point though. Some devices, notably Ciscos, just don't support NAT as well. They don't have STUN, and while they may have a static external IP mapping, that's no good if your NAT itself has a dynamic address, as most home broadband NATs do. Asterisk, if you set nat=yes (or often even without that) will take incoming packets from a natted phone, and look at the incoming address, and send back to it regardless of what the phone says in its SIP headers. That's handy, but unfortunately it does not do the same thing for the SDP, so if the phone hands out an SDP with an unreachable address, Asterisk handles it badly. Some SIP gateways are smarter, and if they see an unreachable address in the SDP, ignore it and send to whatever address they get incoming RTP from. You'll have better luck connecting to such endpoints. Many termination providers do this, so you may find your phones can talk to the term provider, but not to other phones on the same * box. Many consumer nats will not hairpin audio. That means if you do all this work to rewrite the addresses in your SIP headers/SDP via STUN so you look like an externally routable device, and Asterisk hooks you up with another device behind your same NAT, you will get one way audio. I get this problem -- I have a * box at one location, with most of the phones (no problem for those) and some other phones at another location behind NAT. These phones can talk to the main location, but not to one another, due to the hairpin. What fun. A new method, called ICE, was drafted a while ago but is getting slow adoption. In ICE, devices are given a list of possible ways they could reach one another (directly, through nats, via RTP forwarders etc.) They try them all and pick the best. In the end it will always work through the RTP forwarders, but that costs bandwidth and latency. So far, however, support is limited. In the meanwhile, use IAX, which understands about NAT pretty well. If you have multiple SIP phones on a LAN behind a NATing router, just put a small asterisk box on the LAN. It can manage your hairpin calls internally, save you bandwidth by trunking the IAX traffic to the central asterisk and avoid all the NAT hassle by using a single port (outgoing) and refreshing it often enough for the router to hold it open. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco CallManager and Asterisk VoiceMail
Dear All : Anyone can help me to get Cisco Call Manager connected to Asterisk Voicemail Feature ? Simply I have a SIP link between Cisco CM and Asterisk Box - as I am using Asterisk as a Feature BOX to supply the CM with Meetme rooms and with call redirection ... We are now looking to add VoiceMail Feature - knowing that we are using Cisco Phones ... Mohamed Farid ,, Telecommunication Security Section Head ,, Mediterranean Smart Cards Company ( MSCC ) Website: www.mscc.com.eg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnect Supervision UK / BT solution?
Sorry -- you're right, I didn't express the scenario properly .. The disconnect supervision problem is when I 'forward'/divert an incoming POTS call out another FXO channel to a mobile phone or POTS line. (POTS - Sangoma|Asterisk - POTS/mobile) When the incoming POTS hangs up and/or the mobile the person was connected to .. Asterisk/Sangoma doesn't hang the Zap channels up. I have tried busydetect and busycounts and a number of settings are enabled for UK CallerID support (polarity switch stuff) ... but I had some sketchy side effects with busydetect etc and am wary of premature hangups Thanks for your query -- Chris - Original Message - From: Ed W [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 19, 2007 1:26 PM Subject: Re: [asterisk-users] Disconnect Supervision UK / BT solution? Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Just to be clear, what is the exact disconnect problem that you see? I have three TDM cards in two different systems, one using PBX lines and one on a private BT line. Both of them have trouble detecting a caller who is ringing, but then hangs up before being answered by the asterisk system However, *all* of them are absolutely fine at spotting a normal hangup once the call is connected and I see no random disconnects during calls either. Can you confirm that this is what you mean, or whether it's something else? Ed W -- This message has been scanned for viruses and dangerous content and is believed to be clean. -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No Audio for Extension to Extension
I am at a loss, I can terminate and receive calls via any of my providers with both IAX and SIP. I use GSM, G729a, and ulaw for those carriers. If I make an extension to extension call - there is no audio at all in either direction. All my extensions are set to use G729a (I have tried changing that though to see if it would fix it). I am fairly sure it is not a transcoding issue - as the server transcodes for the inbound/outbound calls. Has anybody come across this before? Regards. Troy Disclaimer - This email and any files transmitted with it are confidential and contain privileged or copyright information. You must not present this message to another party without gaining permission from the sender. If you are not the intended recipient you must not copy, distribute or use this email or the information contained in it for any purpose other than to notify us. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the views of Purple Oranges Pty Ltd. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] STUN and SNMP
Hi all, I read somewhere that asterisk v 1.4 can make Stun and SNMP. I tried to find more information on these features but I didn't find any clues. Someone find a way to use it? Thanks, Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Requirements for faxes to work properly
Hello, everyone. I'm reading about the asterisk new features. One is T.38 protocol support. I used faxes before with asterisk 1.2 and everything was working quite well. Could anyone explain what have changed in the way faxes are handled. Another thing is, in order for asterisk to work over T.38 with my fax machine do I also need a T.38 support from my ATA and my SIP provider? Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP registration problem w/ SBC
Thanks Andrew, I see the resolved bug report. I'll get the patch fix. Sorry for the unnecessary mail. -Tom On 1/20/07, Andrew Joakimsen [EMAIL PROTECTED] wrote: http://www.google.com/search?q=423+%22Interval+Too+Brief%22start=0ie=utf-8oe=utf-8client=firefox-arls=org.mozilla:en-US:official Hint: Who develops Asterisk? On 1/20/07, Thomas Madler [EMAIL PROTECTED] wrote: Hi, I'm trying to get my * server connected to a softswitch through an SBC. I get the following error when * trys to register. Got SIP response 423 Interval Too Brief back from xxx.xxx.xxx.xxx Jan 20 12:43:54 NOTICE[2138]: chan_sip.c:5473 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED] ' timed out, trying again (Attempt #9) Is there something I can tweak on my end to fix this? TIA, -Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detecting Disconnected Numbers - PRI
I am trying to automatically detect disconnected numbers when using the outbound dialer I have written. * Some numbers hang up immediately with a Cause Code 0 and no voice treatment * Some numbers get voice treatment with a PROGRESS indication and an associated Cause Code 0 * Some numbers get voice treatment with a PROGRESS indication and no associated cause code (CC=0) My application can pick up the PROGRESS indication (if I get one) and handle the hangup, but not if I don't get a cause code! Is there anything I can do to ensure that I always get a PROGRESS indication with cause code or a hangup with cause code? Behaviour of the PRI seems to differ across telcos and also across numbers. I don't want to just assume hangup on PROGRESS indication as this may not be a disconnected number - it might be a forwarded or redirected number. I need to achieve consistency and this is proving very difficult. Has anyone else had this issue and if so, which tree should I be barking up? cheers, Mark. -- regards, Mark P. Edwards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] On what distribution is www.asterisknow.com based on ?
What is the package manager used? And what is the added value compared to the well maintained debian based asterisk ? Hi Maxim, AsteriskNOW is built on top of the R-Path linux distribution which uses conary as the package manager. There is no difference between the version of Asterisk included with AsteriskNOW and the source code obtainable from asterisk.org. It is meant for those who do not wish to or know how to administer their own linux server. William ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] On what distribution is www.asterisknow.com based on ?
On Sat, 2007-01-20 at 17:33 +0200, Maxim Veksler wrote: Hello Asteriskies, Has someone tried www.asterisknow.com ? What is the package manager used? I haven't tested it fully, but from first look it seems like a Fedora/RHEL/CentOS derivative - it uses the anaconda installer, and I think it uses yum over RPM as the package manager. And what is the added value compared to the well maintained debian based asterisk ? As well maintained the debian based asterisk is, AsteriskNOW is maintained by Digium itself. I would think that that counts for something. In addition, AsteriskNOW attempts to delivery an out-of-the-box usable Asterisk PBX, which isn't what the debian packages Asterisk is doing. If you want a simple straight forward install that gives you an easy to use Asterisk box with little other software involved, then you get AsteriskNOW. If you want a flexible server that can do a lot of other things, then you probably want to use something else - like Debian. -- Oded Arbel Atelis [EMAIL PROTECTED] Tel: +972-54-7340014 ::.. The universe is a big place, perhaps the biggest. -- Kilgore Trout ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IP of SIP server changing
I've got my 1.2.x Asterisk server registering with a SIP provider using their servers DNS entry, not their IP address. My server is behind a NAT. The setup works like a champ for weeks then I get a call reporting inbound calls are failing. When my server isn't registered, inbound callers get a disconnected message; very bad. When it happens, I've found that the IP address in the sip show peers output isn't correct; it doesn't match what host sipserver reports. Can someone explain to me what Asterisk does in the way of hostname-to-address lookups when it's registering with an external SIP server? What happens when the IP for the name changes? The provider claims they've changed nothing and a little cron job I setup to look up the DNS entry every 5 minutes didn't catch anything when the fault happened again today. I figure I should see what the expected behavior of Asterisk is in this situation before going back to the carrier. Thanks in advance for any info. pd Paul Dugas Computer Engineer Dugas Enterprises, LLC 522 Black Canyon Park Canton, GA 30114 phone: 404.932.1355 fax: 866.751.6494 [EMAIL PROTECTED] http://DugasEnterprises.com This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco AS5300
Andrew Pogrebennyk wrote: Hello Yusuf yusuf wrote: Hi all, I realize this is OT. I just got a Cisco AS5300, and I need to configure it like such: Asterisk -(H323/SIP)-- Cisco - (E1/PRI)---Telco So calls originate from the Asterisk side (registered users on SIP or just ZAP phones), and they go out H323 or SIP to Cisco, where they go out PRI. I have the Asterisk side sorted :) (either H323 or SIP), I need help in the Cisco side. Can anyone give me a brief HOW-TO or tutorial on getting this (either SIP or H323) done on the Cisco side. The link with sample Cisco config Hoah has sent is fine. It's well commented etc, but... I do not recommend you to copy it entirely :) [...skipped...] How do I specify that H323 or SIP must be for incoming calls, and outgoing must go out on the E1. Cisco is running IOS 12.1.5-12.2.13a I realize this is alot of questions, so please bear with me :) You seem to need a clear-cut explanation of dial-peer matching process like http://www.cisco.com/warp/public/788/voip/in_dial_peer_match.html or more complete guides: http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/int_c/dpeer_c/dp_confg.htm and http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122cgcr/fvvfax_c/vvfpeers.htm I think I can help you deal with Cisco side once you have drafted a clear setup. Hi, thanks for all the replies. We have got it mainly working, where we have Asterisk dial SIP to the Cisco and Cisco goes E1 to the Telco. However, we can only make one call at a time, following calls just hang. We have to reboot the Cisco to make another call :( . On the cisco, sc says this: ID: starths.index +connect pid:peer_id dir addr state dur hh:mm:ss tx:packets/bytes rx:packets/bytes IP ip:udp rtt:timems pl:play/gapms lost:lost/early/late delay:last/min/maxms codec MODEMPASS method buf:fills/drains loss overall% multipkt/corrected last buf event times dur:Min/Maxs FR protocol [int dlci cid] vad:y/n dtmf:y/n seq:y/n sig:on/off codec (payload size) ATM protocol [int vpi/vci cid] vad:y/n dtmf:y/n seq:y/n sig:on/off codec (payload size) Tele int: tx:tot/v/faxms codec noise:l acom:l i/o:l/l dBm Proxy ip:audio udp,video udp,tcp0,tcp1,tcp2,tcp3 endpt: type/manf bw: req/act codec: audio/video tx: audio pkts/audio bytes,video pkts/video bytes,t120 pkts/t120 bytes rx: audio pkts/audio bytes,video pkts/video bytes,t120 pkts/t120 bytes Total call-legs: 2 11DB : 30199hs.1 +-1 pid:0 Answer dj1 connecting dur 00:00:00 tx:335/53441 rx:337/53920 IP 192.168.0.149:10612 rtt:0ms pl:3580/0ms lost:0/2/0 delay:64/64/65ms g711ulaw 11DB : 30200hs.1 +-1 pid:1 Originate 0847889425 connecting dur 00:00:00 tx:337/53920 rx:335/53441 Tele 0:0 (6): tx:6730/669/0ms g711ulaw noise:-60 acom:1 i/0:-58/-36 dBm Is there something obvious we are missing? -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and VoIP @ Southern California Linux Expo (SCALE 5x)
Hello, Asterisk and VoIP will again have a presence at SCALE 5x, the 2007 Southern California Linux Expo this February. On the exhibit hall floor Trixbox will have a booth demonstrating their asterisk related products. Additionally, a number of other open-source projects will be using Asterisk as part of their demos. The event will be held on Feb 10th and 11th at the Los Angeles Airport Westin. In addition to Asterisk related booths, we will have 2 presentations on VOIP and open-source VoIP in our seminar tracks. * Brian Deenhardt (SwitchVox Four Loop Technologies) - Every thing you wanted to know about voip. * Dave Neary (OpenWengo) - Unifying VoIP, video conferencing and instant messaging. * BoF's on Trixbox, OpenPBX, and Asterisk. Other speakers include: Jono Bacon, Chris Dibona , Don Marti, Jay Pipes, and more.. For further details see the conference website at: http://www.socallinuxexpo.org Those interested in attending the show can use the promo code AST07 to get 40% off full access passes. (http://www.socallinuxexpo.org/order/) Regards, Ilan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnect Supervision UK / BT solution?
Chris Earle (CBL) wrote: Sorry -- you're right, I didn't express the scenario properly .. The disconnect supervision problem is when I 'forward'/divert an incoming POTS call out another FXO channel to a mobile phone or POTS line. (POTS - Sangoma|Asterisk - POTS/mobile) When the incoming POTS hangs up and/or the mobile the person was connected to .. Asterisk/Sangoma doesn't hang the Zap channels up. Just to clarifydoes it all work ok if you are using SIP or IAX for the forwarded channels? Eg local SIP phones? I only have incoming zap lines in my config and with the exception of hangup on ringing I have found hangup detection to work fine. I have a fax machine forwarding in my config as well and again no problems yet with hangup on that. Does it fail to work *every* time, or just intermittently? Does CallerId work ok in your setup? (can be a clue to help diagnose your setup) Ed W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] tdm400p not working with brazilian lines
Hi, I'm installing an Asterisk box with a TDM2400P in Brazil. I can make analog phones work while lines are not working. Since I do not know anything about brazilian lines, is there anybody who can tell me what is wrong/missing in my conf files (below)? TIA Giorgio _zaptel.conf:_ fxoks=9-16 fxsks=17-24 defaultzone=br loadzone=br* * _zapata.conf:_ context = inbound_zap echocancel = 128 echocancelwhenbridged = yes echotraining = 200 language = br signalling = fxo_ks callerid = Christina 102 channel = 9-16 context = outbound_zap canpark = yes echocancel = 128 echocancelwhenbridged = yes echotraining = 200 faxdetect = both language = br musiconhold = native signalling = fxs_ks callerid = asreceived channel = 17-24 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Videopodcast about Asterisk
Just yesterday i ran by the webpage Revision3.com, which houses a number of video podcasts. One of them i called Systm, and is hosted by Digg's Kevin Rose, who recently did an episode about Asterisk http://revision3.com/systm/asterisk Sorry if you already knew, i haven't been on this list for very long. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ebay Unwired Buyer, Using Asterisk?
Jens Vagelpohl wrote: There was a link posted to an interview with Allison a few weeks back. She mentioned eBay as a customer, and how she used eBay unwired before and and listened to herself speak. It doesn't mean they use Asterisk, though. They do. The ExternalIVR application was developed in cooperation with them. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] agi script as member in queue
Hi i want to put an AGI script in a queue, to serve once at time the callers. Example: Queue (8 callers waiting) Agi script / IVR (serving the caller) can i do that? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Load Balancing
Hello Users, How can I perform the load Balancing in My SIP server of Both OpenSER and Asterisk , Currently I have One OpenSER server and Asterisk Server, For OpenSER is to need use these modules, and is any 1) LCR and Dispatcher modules, 2) OSP Modules ( also need ) Please can anyone help me .. -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX call limit
Gordon Henderson wrote: On Sun, 21 Jan 2007, Cristian Draghici wrote: IDEfisk is based on iaxclient (iaxclient.sourceforge.net) which will reject a call with the BUSY signal if there is no available line in the softphone to take the call. This means you need to configure IDEfisk to use only one line (call context). I don't know if this is possible. Somewhere in IDEfisk is this call that initialised iax client: iaxclient.h:EXPORT int iaxc_initialize(int audType, int nCalls); what you want is nCalls to be 1. There is a configurable parameter in sip.conf: call-limit, but this seems to be missing from the iax channel setup. Maybe this is deliberate for other reasons though. But switching to a SIP client and using this might work, if SIP is an option for you. Gordon Hope this helps, Cristi -- Cristian Draghici http://www.loudhush.ro On 1/21/07, Nir Simionovich [EMAIL PROTECTED] wrote: Hi Philipp, Thanks for the tip, but that is not what I initially meant. I'm using IDEfisk, and I would like it when a call comes Into IDEfisk to generate a BUSY signal, if there is already a call in the client. Any ideas ? Nir S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Thursday, January 18, 2007 12:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX call limit Nir Simionovich wrote: Stupid and silly question - is there a way to limit the number of concurrent calls an IAX client can make? something in the similar sense of incominglimit and outgoing limit on SIP? It can be done in the dial plan: http://www.voip-info.org/wiki/view/Asterisk+Queue+with+limited+calls+per+IAX+agent Best regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You may consider using the macro superdial. There you can specify the maximum number of concurrent calls per group, so that the next one recieves the busy tone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with rxfax
Ardjan Zwartjes wrote: Dear list, The company I'm working for is trying to use app_rxfax to receive faxes on an Asterisk machine. Our initial tests looked very promising, but unfortunately we've encountered some problems. We've been trying to solve these problems for quite some time now, but we're running out of options. So I really hope that somebody can give some help here. Save yourself some time and frustration and type iaxmodem and HylaFAX+ on the Asterisk machine, you won't regret it. http://iaxmodem.sourceforge.net http://hylafax.sourceforge.net Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with rxfax
Just out of curiosity. Would you mind sharing that app_rxfax.c file that you modified to work with SpanDSP 0.0.3? TIA, Darren From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ardjan Zwartjes Sent: Monday, January 22, 2007 2:12 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problems with rxfax Dear list, The company I'm working for is trying to use app_rxfax to receive faxes on an Asterisk machine. Our initial tests looked very promising, but unfortunately we've encountered some problems. We've been trying to solve these problems for quite some time now, but we're running out of options. So I really hope that somebody can give some help here. Basically our set-up is this: We have an Asterisk server (version 1.2.7) with an ISDN trunk (Sangoma A104D), we've configured asterisk to run rxfax on a specified extension. Originally we started out with spandsp 0.0.2pre26 and the original app_rxfax for spandsp 0.0.2. Some of the faxes were coming in perfectly, but soon we noticed that quite often there were substantial pieces of the fax missing in the resulting tif file. We've tried the following to solve these problems: - We've checked the timing settings for the ISDN trunk, these seem to be ok. - We've tried several versions of libtiff (currently we are using 3.7.2). - We've tried using 0.0.3 versions of spandsp (since we're using asterisk 1.2.7 we had to modify app_rxfax.c to work). - We've created a custom dialplan application to disable the echo cancellation on the isdn channel on which the fax is received. - We've tried various settings for t30_set_supported_compressions, t30_set_supported_image_sizes, t30_set_supported_modems and t30_set_supported_resolutions (I must confes that I didn't really know which settings to use here, but we have tried a lot of them). - We've tried several fax machines to send the faxes, ranging from simple fax-modems to large dedicated fax machines. But still a lot of faxes give problems, either the tif is missing large portions, or the fax isn't received at all. At the bottom of this mail are 2 examples of the logging when it goes wrong. I really hope that somebody can give a few pointers. Thanks in advance, Kind regards, Ardjan Zwartjes, Telecats === Example 1 = Jan 19 14:49:14 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:14 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:17 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:17 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:19 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:19 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:20 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-22.18dBm0) Jan 19 14:49:21 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:21 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-17.84dBm0) Jan 19 14:49:38 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:38 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:39 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:39 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:41 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:41 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:42 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-19.13dBm0) Jan 19 14:49:48 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:48 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: == Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Pages transferred: 1 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image size: 1728 x 1192 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image resolution8037 x 7700 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Transfer Rate: 9600 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Bad rows0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Longest bad row run 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Compression type3 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image size (bytes) 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: == Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 1 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: ==
[asterisk-users] 7 points of comparison Polycom 430/501 and Aastra 480i. Which one to choose ?
I need to provide a 80 people office with VOIP. I want to commit to one vendor Polycom or Aastra. Price of the phones is not a factor in the decision. The quality of the phones is the factor. Some of the features that I am evaluating on are: (arranged in order of priority) 1. Sound quality 2. complete product line with conference phone and receptionist phone (not on Aastra) 3. cordless (not on 501/430) 4. backlit LCD (not on 501/430) 5. Inbuilt POE (not on 501) 6. speaker phone 7. 2 network ports. Which one will you choose ? Vikas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Load Balancing
use LCR is really good. On 1/22/07, raviprakash sunkara [EMAIL PROTECTED] wrote: Hello Users, How can I perform the load Balancing in My SIP server of Both OpenSER and Asterisk , Currently I have One OpenSER server and Asterisk Server, For OpenSER is to need use these modules, and is any 1) LCR and Dispatcher modules, 2) OSP Modules ( also need ) Please can anyone help me .. -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
Thanks Moises, I was trying to find some consistence, but the only similarity I could find is that much of the calls that fail are long distance ones or international. It fails in both, Telmex and Meridian link. I 'll try looping. I'll be posting results soon. I hope I can manage to get it work. Thanks for your help. Regards. On 1/19/07, Moises Silva [EMAIL PROTECTED] wrote: Similar probles I had were fixed incrementing one of the timers, but if you have already done that, I have no idea of your problem, you require to debug the problem and try to find some consistence in the failures, find if the failure is on the Asterisk - telco Link, or in the Asterisk - meridian link? find if putting in loop your own asterisk still fails, etc etc. Kind Regards On 1/18/07, Facundo Ameal [EMAIL PROTECTED] wrote: Thanks for your help, but I've already adjusted timers on the source code. I found your document a week ago and read it. Do you really think that is a matter of timers only? Greets! On 1/18/07, Moises Silva [EMAIL PROTECTED] wrote: Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users
[asterisk-users] X100P how do i recieve incomming calls?
I have just purchased a 2nd hand X100P, if I do a ztcfg -vv I get: Zaptel Version: 1.4.0 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. My understanding of the above is that the zaptel driver has detected the card. What do I now need to do, in order to get an incoming call to work with asterisk? I assume I need to make some sort of change to /etc/asterisk/zapata.conf in order to tell asterisk about the card? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with rxfax
I would install Hylafax(opensource too) with Asterisk via IAXModem, it worth! You can keep all the features, hylafax running in same server or a separated one, and IAXmodem will be a Modem on Hylafax, and a simple extension on you *. It works great for me, and there are more users using this architecture, you can setup as much IAXModem as your servers can handle, so it's very scalable. Best regards, Marco Mouta On 1/22/07, Ardjan Zwartjes [EMAIL PROTECTED] wrote: Dear list, The company I'm working for is trying to use app_rxfax to receive faxes on an Asterisk machine. Our initial tests looked very promising, but unfortunately we've encountered some problems. We've been trying to solve these problems for quite some time now, but we're running out of options. So I really hope that somebody can give some help here. Basically our set-up is this: We have an Asterisk server (version 1.2.7) with an ISDN trunk (Sangoma A104D), we've configured asterisk to run rxfax on a specified extension. Originally we started out with spandsp 0.0.2pre26 and the original app_rxfax for spandsp 0.0.2. Some of the faxes were coming in perfectly, but soon we noticed that quite often there were substantial pieces of the fax missing in the resulting tif file. We've tried the following to solve these problems: - We've checked the timing settings for the ISDN trunk, these seem to be ok. - We've tried several versions of libtiff (currently we are using 3.7.2). - We've tried using 0.0.3 versions of spandsp (since we're using asterisk 1.2.7 we had to modify app_rxfax.c to work). - We've created a custom dialplan application to disable the echo cancellation on the isdn channel on which the fax is received. - We've tried various settings for t30_set_supported_compressions, t30_set_supported_image_sizes, t30_set_supported_modems and t30_set_supported_resolutions (I must confes that I didn't really know which settings to use here, but we have tried a lot of them). - We've tried several fax machines to send the faxes, ranging from simple fax-modems to large dedicated fax machines. But still a lot of faxes give problems, either the tif is missing large portions, or the fax isn't received at all. At the bottom of this mail are 2 examples of the logging when it goes wrong. I really hope that somebody can give a few pointers. Thanks in advance, Kind regards, Ardjan Zwartjes, Telecats === Example 1 = Jan 19 14:49:14 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:14 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:17 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:17 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:19 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:19 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:20 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-22.18dBm0) Jan 19 14:49:21 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:21 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-17.84dBm0) Jan 19 14:49:38 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:38 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:39 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:39 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:41 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:41 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:42 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-19.13dBm0) Jan 19 14:49:48 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:48 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: == Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Pages transferred: 1 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image size: 1728 x 1192 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image resolution8037 x 7700 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Transfer Rate: 9600 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Bad rows0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Longest bad row run 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Compression type3 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image size (bytes) 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: == Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 1 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c:
[asterisk-users] 2 ring delay before asterisk answer
I am a little green when it comes to all this but I am trying to connect
our PBX to an asterisk server using a TDM400 with 4 FXO modules. I am able
to dial an extension on my PBX handset and I get a dialtone from the PBX.
After 2 rings I then hear the asterisk server connect and I get a dialtone
from asterisk. I am then able to dial an extension on another asterisk
server.
My question is: How do I get asterisk to connect immediately without the
annoying 2 ring wait before I can start dialing a number.
snippets of extensions.conf
[net_incoming]
exten = s,1,DISA(no-password,net_outgoing)
[net_outgoing]
exten = _2XXX,1,Dial(${PYRMONT}/${EXTEN:1})
exten = _2XXX,n,Hangup()
logging:
Jan 23 07:39:47 NOTICE[16526]: chan_zap.c:6184 ss_thread: Got event 18 (Ring
Begin)...
Jan 23 07:39:49 NOTICE[16526]: chan_zap.c:6184 ss_thread: Got event 2
(Ring/Answered)...
Jan 23 07:39:50 NOTICE[16526]: chan_zap.c:6184 ss_thread: Got event 18 (Ring
Begin)...
-- Executing DISA(Zap/1-1, no-password|net_outgoing) in new stack
--
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Suite 54, Jones Bay Wharf Fax: +612 95525549
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Re: [asterisk-users] tdm400p not working with brazilian lines
On Mon, Jan 22, 2007 at 03:16:54PM +0100, Giorgio Incantalupo wrote: Hi, I'm installing an Asterisk box with a TDM2400P in Brazil. I can make analog phones work while lines are not working. What does happend when you try to ring or when a call comes in? Since I do not know anything about brazilian lines, is there anybody who can tell me what is wrong/missing in my conf files (below)? TIA Giorgio _zaptel.conf:_ fxoks=9-16 fxsks=17-24 defaultzone=br loadzone=br* * _zapata.conf:_ context = inbound_zap echocancel = 128 echocancelwhenbridged = yes echotraining = 200 language = br signalling = fxo_ks callerid = Christina 102 channel = 9-16 context = outbound_zap canpark = yes echocancel = 128 echocancelwhenbridged = yes echotraining = 200 faxdetect = both language = br musiconhold = native signalling = fxs_ks callerid = asreceived channel = 17-24 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Ebay Unwired Buyer, Using Asterisk?
They do. The ExternalIVR application was developed in cooperation with them. lol so the $2.6b they spent on Skype was well worth it, then. It'd be nice, though to have an OSS chan_skype. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ebay Unwired Buyer, Using Asterisk?
I can honestly say Unwired Buyer's competitive advantage is definitely not the fact that they use Asterisk or Allison. Those two things were/are definitely development advantages though. On 1/22/07, Kevin P. Fleming [EMAIL PROTECTED] wrote: Jens Vagelpohl wrote: There was a link posted to an interview with Allison a few weeks back. She mentioned eBay as a customer, and how she used eBay unwired before and and listened to herself speak. It doesn't mean they use Asterisk, though. They do. The ExternalIVR application was developed in cooperation with them. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Detecting Disconnected Numbers - PRI
original message I am trying to automatically detect disconnected numbers when using the outbound dialer I have written. * Some numbers hang up immediately with a Cause Code 0 and no voice treatment * Some numbers get voice treatment with a PROGRESS indication and an associated Cause Code 0 * Some numbers get voice treatment with a PROGRESS indication and no associated cause code (CC=0) My application can pick up the PROGRESS indication (if I get one) and handle the hangup, but not if I don't get a cause code! Is there anything I can do to ensure that I always get a PROGRESS indication with cause code or a hangup with cause code? Behaviour of the PRI seems to differ across telcos and also across numbers. I don't want to just assume hangup on PROGRESS indication as this may not be a disconnected number - it might be a forwarded or redirected number. I need to achieve consistency and this is proving very difficult. Has anyone else had this issue and if so, which tree should I be barking up? /original message Yep, I experienced this frequently. I have several PRI vendors and they all give me the same line of crap: Well, PRI is good, but it's not perfect... Sad but true. I feel comfortable in saying that there is no 100% guaranteed way of detecting disconnected numbers on a PRI. I've done lots of testing and come to the conclusion that you have to do your best to work around it. For example, I know that phone number xxx-yyy- is disconnected. I dial it 25 times with Asterisk. 18 times I get one cause code (like 'invalid' or fast busy - I don't recall the exact cause code number), 6 times I get PROGRESS indicating ring-no answer and 1 time I get traditional busy. All calls to same phone number, same provider, made one right after the other. Oddly enough, if I call the number on a POTS line I *ALWAYS* get the disconnect message. It's one case where advanced technology yields poorer results than the old stuff. I know that doesn't help but I wanted you to know that you're not alone. -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CentOS and 1.4
Has anyone had success in installing Asterisk 1.4 on a CentOS 3.8 system? John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ebay Unwired Buyer, Using Asterisk?
Steve Totaro wrote: Just got a call from Ebay's unwired buyer and The Voice is Allison Smith. Adoption is wide but who is willing to give away their competitive edge (although ebay doesn't really have any real competition). Thanks, Steve Steve, Thanks for trying UnWired Buyer! :) Yes, we're very happy with Allison's work. I just checked and while I think we used some of the existing sounds at the start, I believe (and ls -lurt concurs) that we've had Allison record everything we use now. I recorded most of our original sounds, but, as many people refer to my voice as the voice of a songbird.strangling in hot tar!, it was decided to contact Allison. As UWB is a real time, auction is almost over tool, we had her rerecord existing sounds that we needed just a little faster and instead of using sixty and five, we got her to record sixty-five, etc. As for asterisk, yes we use it. We designed and funded the ExternalIVR app (so don't blame Kevin for my ugly protocol design) and worked with Kevin on uncovering some deeper bugs that our high volume use of asterisk uncovered (long live valgrind). I believe that Digium talked about us using Asterisk at Astricon 2 years ago, but that could just be a trick of my fading memory. Digium and the community: Thank you for Asterisk! Also, just to clarify, although we work tightly with eBay, we are a separate entity. -- Wayne Walker Operations Manager UnWired Buyer, Inc. http://www.unwiredbuyer.com Note: My opinions are not necessarily the opinions of my employer. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7 points of comparison Polycom 430/501 and Aastra 480i. Which one to choose ?
IMO, the 480i, by a LONG shot. The 480i is easier to use, looks nicer, has better audio quality, easier to read, and has a great speakerphone. The web-interface is also leagues better than the tripe the Polycom phones have. The only issue I have with the 480i, is that it's a little unintuitive in how to disable the X missed calls option. There's no option in the web-interface (I'm told one is coming, however), so you have to manually edit a .cfg file and send the info back to the phone. Other than that, I have had zero problems with my 480i's, and nothing but frustration with any of the Polycoms I have on hand. HTH, Jay Vikas wrote: I need to provide a 80 people office with VOIP. I want to commit to one vendor Polycom or Aastra. Price of the phones is not a factor in the decision. The quality of the phones is the factor. Some of the features that I am evaluating on are: (arranged in order of priority) 1. Sound quality 2. complete product line with conference phone and receptionist phone (not on Aastra) 3. cordless (not on 501/430) 4. backlit LCD (not on 501/430) 5. Inbuilt POE (not on 501) 6. speaker phone 7. 2 network ports. Which one will you choose ? Vikas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] X100P how do i recieve incomming calls?
Dmesg reports: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.0 Echo Canceller: MG2 ACPI: PCI interrupt :00:0c.0[A] - GSI 16 (level, low) - IRQ 185 wcfxo: DAA mode is 'FCC' Found a Wildcard FXO: Wildcard X100P Capability LSM initialized When I run zttool it says that alarms: red. The card is connected to the phone line. Anybody any idea what the problem is? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charlie Grosvenor Sent: 22 January 2007 20:08 To: asterisk-users@lists.digium.com Subject: [asterisk-users] X100P how do i recieve incomming calls? I have just purchased a 2nd hand X100P, if I do a ztcfg -vv I get: Zaptel Version: 1.4.0 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. My understanding of the above is that the zaptel driver has detected the card. What do I now need to do, in order to get an incoming call to work with asterisk? I assume I need to make some sort of change to /etc/asterisk/zapata.conf in order to tell asterisk about the card? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with rxfax
On Monday 22 January 2007 20:19, Marco Mouta wrote: I would install Hylafax(opensource too) with Asterisk via IAXModem, it worth! You can keep all the features, hylafax running in same server or a separated one, and IAXmodem will be a Modem on Hylafax, and a simple extension on you *. It works great for me, and there are more users using this architecture, you can setup as much IAXModem as your servers can handle, so it's very scalable. Just a 'me too' - we send about 3500 faxes per day via Hylafax (I appreciate this thread is about fax reception), until recently using the onboard DSPs of an Eicon Diva Server quad-BRI card, and the success rate we've achieved using 8 instances of IAXModem instead of the Diva Server card is just as high :) The thing that really impressed me was that we're not using any kind of dedicated (v)LAN for the link between the Hylafax and Asterisk servers. The IAX data is just shuffling across the same noisy LAN with chatter from a hundred Windows boxes.. HTTP traffic.. Samba file sharing.. all kinds of nonsense, and any problems are negligible! Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ebay Unwired Buyer, Using Asterisk?
Unwired Buyer paid for ExternalIVR. At this point, they're not owned by eBay. On 1/22/07, Colin Anderson [EMAIL PROTECTED] wrote: They do. The ExternalIVR application was developed in cooperation with them. lol so the $2.6b they spent on Skype was well worth it, then. It'd be nice, though to have an OSS chan_skype. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] X100P how do i recieve incomming calls?
From:"Charlie Grosvenor" [EMAIL PROTECTED]I have just purchased a 2nd hand X100P, if I do a ztcfg -vv I get:... Channel 01: FXS Kewlstart (Default) (Slaves: 01)1 channels configured."My understanding of the above is that the zaptel driver has detected thecard. What do I now need to do, in order to get an incoming call to workwith asterisk?I assume I need to make some sort of change to /etc/asterisk/zapata.confin order to tell asterisk about the card? You want todefine a contextto use withthis channel, match the signaling with your channel driver (see examples in zapata.conf); if you create a new context, you'll need to also create a dial plan for it. Yuan Liu Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P how do i recieve incomming calls?
On Mon, Jan 22, 2007 at 08:08:16PM -, Charlie Grosvenor wrote: I have just purchased a 2nd hand X100P, Is there another X100P card in the same system? if I do a ztcfg -vv I get: Zaptel Version: 1.4.0 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. My understanding of the above is that the zaptel driver has detected the card. What do I now need to do, in order to get an incoming call to work with asterisk? I assume I need to make some sort of change to /etc/asterisk/zapata.conf in order to tell asterisk about the card? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Streaming audio file while working in background ?
Hey All, Is there an app available, or another method, to stream an audio file to a caller while performing additional actions in the background? Regardless of whether DTMF is received or not from the caller. I had originally thought that I could use the Background app for this but after further investigation found that Background is primarily for playing audio and waiting for DTMF, and it seems it won't do what I need in this situation. Ideally I would like to be able to play an audio file to the caller while making outbound calls in the background (via the Dial app) and then discontinue the audio file stream and bridge the calls once an outbound call is connected. Can anyone point me in the right direction on how to do this with * ? Thanks in advance, Darren Nay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with rxfax
We've helped a lot of customers with fax for VoIP...often turns out to be audio quality issues (eg: test fax from within office is ok, but fax from customer fails). One solution is to optimize route (minimize latency) if you have that control. Moving fax line retry/resend control up a level (to Hylafax) helps too. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, January 22, 2007 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems with rxfax Ardjan Zwartjes wrote: Dear list, The company I'm working for is trying to use app_rxfax to receive faxes on an Asterisk machine. Our initial tests looked very promising, but unfortunately we've encountered some problems. We've been trying to solve these problems for quite some time now, but we're running out of options. So I really hope that somebody can give some help here. Save yourself some time and frustration and type iaxmodem and HylaFAX+ on the Asterisk machine, you won't regret it. http://iaxmodem.sourceforge.net http://hylafax.sourceforge.net Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting Disconnected Numbers - PRI
Michael Collins wrote:
original message
I am trying to automatically detect disconnected numbers when using the
outbound dialer I have written.
* Some numbers hang up immediately with a Cause Code 0 and no voice treatment
* Some numbers get voice treatment with a PROGRESS indication and an associated
Cause Code 0
* Some numbers get voice treatment with a PROGRESS indication and no associated
cause code (CC=0)
My application can pick up the PROGRESS indication (if I get one) and handle the hangup, but not if I don't get a cause code!
Is there anything I can do to ensure that I always get a PROGRESS indication with cause code or a hangup with cause code?
Behaviour of the PRI seems to differ across telcos and also across numbers.
I don't want to just assume hangup on PROGRESS indication as this may not be a disconnected number - it might be a forwarded or redirected number.
I need to achieve consistency and this is proving very difficult.
Has anyone else had this issue and if so, which tree should I be barking up?
/original message
Yep, I experienced this frequently. I have several PRI vendors and they all give me the
same line of crap: Well, PRI is good, but it's not perfect... Sad but true.
I feel comfortable in saying that there is no 100% guaranteed way of detecting
disconnected numbers on a PRI. I've done lots of testing and come to the conclusion that
you have to do your best to work around it. For example, I know that phone number
xxx-yyy- is disconnected. I dial it 25 times with Asterisk. 18 times I get one
cause code (like 'invalid' or fast busy - I don't recall the exact cause code number), 6
times I get PROGRESS indicating ring-no answer and 1 time I get traditional busy. All
calls to same phone number, same provider, made one right after the other. Oddly enough,
if I call the number on a POTS line I *ALWAYS* get the disconnect message. It's one case
where advanced technology yields poorer results than the old stuff.
I know that doesn't help but I wanted you to know that you're not alone.
The correct way to determine the ending cause of a call is the
${HANGUPCAUSE} variable that Dial creats. Just to be sure, set
priindication=outofband in /etc/asterisk/zapata.conf. HANGUPCAUSE
should always be set.
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Re: [asterisk-users] 2 ring delay before asterisk answer
From memory, the 2 ring is pretty standard - that's how long the card
takes to answer the card and take over the call.
I'm not sure it can be shortened to zero rings.
later,
PaulH
On Tue, 2007-01-23 at 07:54 +1100, Daryl Sayers wrote:
I am a little green when it comes to all this but I am trying to connect
our PBX to an asterisk server using a TDM400 with 4 FXO modules. I am able
to dial an extension on my PBX handset and I get a dialtone from the PBX.
After 2 rings I then hear the asterisk server connect and I get a dialtone
from asterisk. I am then able to dial an extension on another asterisk
server.
My question is: How do I get asterisk to connect immediately without the
annoying 2 ring wait before I can start dialing a number.
snippets of extensions.conf
[net_incoming]
exten = s,1,DISA(no-password,net_outgoing)
[net_outgoing]
exten = _2XXX,1,Dial(${PYRMONT}/${EXTEN:1})
exten = _2XXX,n,Hangup()
logging:
Jan 23 07:39:47 NOTICE[16526]: chan_zap.c:6184 ss_thread: Got event 18 (Ring
Begin)...
Jan 23 07:39:49 NOTICE[16526]: chan_zap.c:6184 ss_thread: Got event 2
(Ring/Answered)...
Jan 23 07:39:50 NOTICE[16526]: chan_zap.c:6184 ss_thread: Got event 18 (Ring
Begin)...
-- Executing DISA(Zap/1-1, no-password|net_outgoing) in new stack
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[asterisk-users] how to make a video phone call
hi, i am trying to make video phone call between Grandstream endpoints which support video feature, but failed to get video on the call. The calling party has attached m=video SDP field in INVITE message, but did not get m=video SDP from the peer 200 OK response message. Could you tell me how to configure the Asterisk to get video call? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] LDAP get and Asterisk 1.4
app_ldap-1.0rc6 is not compiling on my CentOS 4.4 with Asterisk 1.4. I have it working successfully using Asterisk 1.2. Can anyone give me any hints? make: *** [app_ldap.o] Error 1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 7 points of comparison Polycom 430/501 and A astra 480i. Which one to choose ?
IMO, the 480i, by a LONG shot. Yeah I have a rollout of 36 480i's right now and they are cat's ass, but I have found that the cordless will cause interference with Bluetooth headsets - static. Otherwise, yeah, my favorite phone. Good implementation all around. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7 points of comparison Polycom 430/501 and Aastra 480i. Which one to choose ?
If I commit to Aastra what do I for the: A. Receptionist B. Conference room Aastra does not seem to have a phone for these two functions. Any suggestions ? Vikas On 1/22/07, Jay Moore [EMAIL PROTECTED] wrote: IMO, the 480i, by a LONG shot. The 480i is easier to use, looks nicer, has better audio quality, easier to read, and has a great speakerphone. The web-interface is also leagues better than the tripe the Polycom phones have. The only issue I have with the 480i, is that it's a little unintuitive in how to disable the X missed calls option. There's no option in the web-interface (I'm told one is coming, however), so you have to manually edit a .cfg file and send the info back to the phone. Other than that, I have had zero problems with my 480i's, and nothing but frustration with any of the Polycoms I have on hand. HTH, Jay Vikas wrote: I need to provide a 80 people office with VOIP. I want to commit to one vendor Polycom or Aastra. Price of the phones is not a factor in the decision. The quality of the phones is the factor. Some of the features that I am evaluating on are: (arranged in order of priority) 1. Sound quality 2. complete product line with conference phone and receptionist phone (not on Aastra) 3. cordless (not on 501/430) 4. backlit LCD (not on 501/430) 5. Inbuilt POE (not on 501) 6. speaker phone 7. 2 network ports. Which one will you choose ? Vikas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LDAP get and Asterisk 1.4
On Tue, Jan 23, 2007 at 11:37:36AM +1100, Klaverstyn, David C wrote: app_ldap-1.0rc6 is not compiling on my CentOS 4.4 with Asterisk 1.4. I have it working successfully using Asterisk 1.2. Can anyone give me any hints? make: *** [app_ldap.o] Error 1 The real error message is a bit above that line. Could you give a more complete error log? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7 points of comparison Polycom 430/501 and Aastra 480i. Which one to choose ?
On Monday 22 January 2007 5:33 pm, Jay Moore wrote: The 480i is easier to use, looks nicer, has better audio quality, easier to read, and has a great speakerphone. The web-interface is also leagues better than the tripe the Polycom phones have. I have a very hard time believing that the 480i has better sound or a better speakerphone than Polycom. They've been in the business of high quality audio for many, many years. Aastra has, OTOH, been in the business of POTS telephones for about the same amount of time. Web interface: yes, absolutely correct. However nobody, and I mean nobody who has any sanity at all will be using a web interface for anything more than a onesie-twosie deployment. Polycom's win hands down (IMO) in this department. It's just amazing how you can deploy these things without ever touching the actual phone. I'm waiting for a polycom 802.11 phone, and praying the damn thing has bluetooth. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] LDAP get and Asterisk 1.4
Thanks for the reply. Here are the lines. cc -g -Wall -D_REENTRANT -D_GNU_SOURCE -fPIC -DCHANNEL_HAS_CID -DNEW_CONFIG -c -o app_ldap.o app_ldap.c app_ldap.c:53: warning: type defaults to `int' in declaration of `STANDARD_LOCAL_USER' app_ldap.c:53: warning: data definition has no type or storage class app_ldap.c:54: warning: type defaults to `int' in declaration of `LOCAL_USER_DECL' app_ldap.c:54: warning: data definition has no type or storage class app_ldap.c: In function `ldap_exec': app_ldap.c:69: warning: implicit declaration of function `LOCAL_USER_ADD' app_ldap.c:109: warning: assignment discards qualifiers from pointer target type app_ldap.c:112: warning: assignment discards qualifiers from pointer target type app_ldap.c:115: warning: assignment discards qualifiers from pointer target type app_ldap.c:118: warning: assignment discards qualifiers from pointer target type app_ldap.c:121: warning: assignment discards qualifiers from pointer target type app_ldap.c:122: warning: assignment discards qualifiers from pointer target type app_ldap.c:123: warning: assignment discards qualifiers from pointer target type app_ldap.c:126: warning: assignment discards qualifiers from pointer target type app_ldap.c:129: warning: assignment discards qualifiers from pointer target type app_ldap.c:132: warning: assignment discards qualifiers from pointer target type app_ldap.c:135: warning: assignment discards qualifiers from pointer target type app_ldap.c:213: warning: implicit declaration of function `LOCAL_USER_REMOVE' app_ldap.c: In function `unload_module': app_ldap.c:218: error: `STANDARD_HANGUP_LOCALUSERS' undeclared (first use in this function) app_ldap.c:218: error: (Each undeclared identifier is reported only once app_ldap.c:218: error: for each function it appears in.) app_ldap.c: In function `usecount': app_ldap.c:232: warning: implicit declaration of function `STANDARD_USECOUNT' app_ldap.c: In function `replace_ast_vars': app_ldap.c:331: warning: assignment discards qualifiers from pointer target type make: *** [app_ldap.o] Error 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, 23 January 2007 11:09 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] LDAP get and Asterisk 1.4 On Tue, Jan 23, 2007 at 11:37:36AM +1100, Klaverstyn, David C wrote: app_ldap-1.0rc6 is not compiling on my CentOS 4.4 with Asterisk 1.4. I have it working successfully using Asterisk 1.2. Can anyone give me any hints? make: *** [app_ldap.o] Error 1 The real error message is a bit above that line. Could you give a more complete error log? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7 points of comparison Polycom 430/501 and Aastra 480i. Which one to choose ?
Your list seems to lean heavily to the Aastra, while I choose the Polycom 501/601 over the Aastra, I did like the unit I tested and the cordless. In the end the fact that most of the people using the phones would use the speaker phone, Polycom and their history of conference phones made the choice. We rolled 75 phones at one site and another 30 now at remote locations. As far a a receptionist phone, we choose to use a software operator panel instead of a phone that took up most of the desk, there were initial concerns but the results have been excellent. If you have not already done so grab a few people from different parts of the office and have them give their 2 cents, it will help to have their perspectives on the quality and feel of the phones. On 1/22/07, Vikas [EMAIL PROTECTED] wrote: I need to provide a 80 people office with VOIP. I want to commit to one vendor Polycom or Aastra. Price of the phones is not a factor in the decision. The quality of the phones is the factor. Some of the features that I am evaluating on are: (arranged in order of priority) 1. Sound quality 2. complete product line with conference phone and receptionist phone (not on Aastra) 3. cordless (not on 501/430) 4. backlit LCD (not on 501/430) 5. Inbuilt POE (not on 501) 6. speaker phone 7. 2 network ports. Which one will you choose ? Vikas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Hater
-- Forwarded message -- From: Jeremy McNamara [EMAIL PROTECTED] Date: Jan 22, 2007 5:22 PM Subject: Hater To: [EMAIL PROTECTED] Continue to Hate on NuFone - Every time you post something people already know that you don't have a clue. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music on Hold on IP Phones with FreePBX 2.2.0
Hello everyone, We just installed a new Trixbox 2.0 server and updated FreePBX to 2.2.0 from 2.2.0rc3. We are having some problems with regards to Music on Hold on IP phones. When we press the Hold button, the caller doesn't hear the MOH sound. This functionality used to work with the older [EMAIL PROTECTED] installation on the same hardware and configuration. However, we don't have any problems with softphones only on IP phones. Is there anyone also having the same problem? Best regards, Matt -- Stand before it and there is no beginning. Follow it and there is no end. Stay with the ancient Tao, Move with the present. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Hater
On Mon, Jan 22, 2007 at 08:35:23PM -0500, Andrew Joakimsen wrote: -- Forwarded message -- From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Thanks for once again posting off-topic private mail to a crowded mailing list. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Streaming audio file while working in background ?
Dial(ZAP/g1/18005551212,90,m) m will play music on hold while the call rings. On 1/22/07, Darren Nay [EMAIL PROTECTED] wrote: Hey All, Is there an app available, or another method, to stream an audio file to a caller while performing additional actions in the background? Regardless of whether DTMF is received or not from the caller. I had originally thought that I could use the Background app for this but after further investigation found that Background is primarily for playing audio and waiting for DTMF, and it seems it won't do what I need in this situation. Ideally I would like to be able to play an audio file to the caller while making outbound calls in the background (via the Dial app) and then discontinue the audio file stream and bridge the calls once an outbound call is connected. Can anyone point me in the right direction on how to do this with * ? Thanks in advance, Darren Nay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Thomson ST2030S and BLF
Perhaps it was a hardware defect. Besides that alot of buttons/features that don't work and cant be turned off TrMail and Pickup soft keys. If I am not mistaken they are using Broadcom CallCtrl which is the same thing Aastra uses and they support asterisk very well. Also get a Wrong number! after each time the call is dialed, yet it still connects! Where is Thomson, they seemed to try to keep up for a while... On 12/21/06, Alberto Pastore [EMAIL PROTECTED] wrote: Andrew Joakimsen ha scritto: I too am wondering if someone has a contact at Thomson, some of the softkeys need to either be fixed or have the option to remove (like FwdVM and Pickup keys). In addition, has anyone notice a humming noise when using the handset? I can hear it and so can the person that I am calling. Honestly, I'm experiencing a good audio quality, no humming noise or hiss. Well, I'm using g711a... Alberto. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weird names vs. correct agent's ext.
Hello all! I've Queue Log Analyzer installed [http://www.micpc.com/qloganalyzer] to check Asterisk main stats. There's an option of CALLS COMPLETED [ALL] where I can see the completed calls that entered any of the queues. There, there's a column that displays the extention number of the Agent who received each call, for example, it displays: Agent/202 So I know who that Joe which has that ext. got it. However, sometimes [like on 2% of the records] it displays weid names such as: Local/[EMAIL PROTECTED],1 SIP/202-007bc0f0 My questions are: a. Why?? b. Can I safely assume, eventhough the display is like that for whatever reason, that by reading the extention number between the line that such agent received such call? For example, in the two samples above it says Local/[EMAIL PROTECTED],1 and SIP/202-007bc0f0 so I assume agent on ext 202 got the call, is this correct? Thanks for your help. Best! Danny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why app_rx and app_tx when we have IAXModem and Hylafax and hy-email2fax? Should we reinvent the wheel?
I'm just wondering here, in our community... Why not HylaFax for Fax solutions? I'm not saying bad about the big efforts on app_tx and app_rx with spandsp. But as me and many other asterisk users, we find out that Hylafax+IAXModem+Asterisk is very reliable. And from what i've been testing and reading, Hylafax is an extremely powerful: HylaFAX is an enterprise-class system for sending and receiving facsimiles as well as for sending alpha-numeric pages. The software is designed around a client-server architecture. Fax modems may reside on a single machine on a network and clients can submit an outbound job from any other machine on the network. Client software is designed to be lightweight and easy to port. HylaFAX is designed to be very robust and reliable. The fax server is designed to guard against unexpected failures in the software, in the configuration, in the hardware and in general use. HylaFAX can support multiple modems and a heavy traffic load. http://www.hylafax.org/content/Main_Page Using this architecture and bash script hy-email2fax, i've been able to setup Email2Fax and Fax2Email. Of course, we the community, could improve and simplify the integration and documentation, as well as the process handled currently by hy-email2fax: http://wpkg.org/email2fax/index.php/Download Is a bad idea to bet on this solution? My apologies, if someone thinks that i'm complaining about app_tx and app_rx, that's not the case! What I mean is that the Real Enterprise Fax Server is not Asterisk, but some one had done it already and it's widely used Hylafax... Please let me know if i'm missing something on this email. Best regards to this great Community, Marco Mouta dCAP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Thomson ST2030S and BLF
Actually I noticed just three days ago there is a new release, and the releae notes seem to address Disable TrMail and Pickup keys Disable call progress indication ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7 points of comparison Polycom 430/501 and Aastra 480i. Which one to choose ?
With over 300 Polycoms, and around 80 Aastra 480i under my belt here is my $0.02. 1. Sound quality, Polycom wins but the Aastra has excellent sound quality as well. 2. Complete product line, Polycom wins. 3. Cordless Aastra, although it's not the best cordless. 4. Backlit, Aastra 5. PoE, Aastra 6. Speakerphone, they both have good speaker phones. Although in general the answer to 1 goes here as well. 7. They both have 2 network ports, but I havnt' done any tests on the speed, I did however notice that when restarting the phone, the Polycom will not shut the network ports down, while the Aastra will. On another note, in general the Polycoms give me less problems. The Aastras are not yet that stable. See my next post to the list. On 1/22/07, Bruce Reeves [EMAIL PROTECTED] wrote: Your list seems to lean heavily to the Aastra, while I choose the Polycom 501/601 over the Aastra, I did like the unit I tested and the cordless. In the end the fact that most of the people using the phones would use the speaker phone, Polycom and their history of conference phones made the choice. We rolled 75 phones at one site and another 30 now at remote locations. As far a a receptionist phone, we choose to use a software operator panel instead of a phone that took up most of the desk, there were initial concerns but the results have been excellent. If you have not already done so grab a few people from different parts of the office and have them give their 2 cents, it will help to have their perspectives on the quality and feel of the phones. On 1/22/07, Vikas [EMAIL PROTECTED] wrote: I need to provide a 80 people office with VOIP. I want to commit to one vendor Polycom or Aastra. Price of the phones is not a factor in the decision. The quality of the phones is the factor. Some of the features that I am evaluating on are: (arranged in order of priority) 1. Sound quality 2. complete product line with conference phone and receptionist phone (not on Aastra) 3. cordless (not on 501/430) 4. backlit LCD (not on 501/430) 5. Inbuilt POE (not on 501) 6. speaker phone 7. 2 network ports. Which one will you choose ? Vikas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Aastra 480i freezes
I realized today after much playing around that the Aastra 480i phone (I'm assuming the others as well), will freeze and can be reproduced if all of the following is true: 1. You enable auto answer in the configurations 2. You call from an Aastra phone to another Aastra phone using the alertinfo that auto answers the phone. 3. You hangup and then try using a softbutton programmed as a speeddial, the phone freezes right then. If I put canreinvite=no in sip.conf, then all is fine. Also even when not using auto answer, the phone will freeze after just a few phone calls if canreinvite=yes. Has anybody else seen this problem? I would realy like to disable reinvites. TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 7 points of comparison Polycom 430/501 andAastra 480i. Which one to choose ?
Here are another $0.02 We too have put in a lot of polycoms and aastras. I agree with a lot of what you noted below...but there are two big strikes against aastra: 1. Firmware bugs. Even some basic functions of the 480i are unusable/unstable due to firmware bugs. The word from support is always wait for the next firmware 2. Poor documentation. Their documentation is out of date and lacking a LOT of critical functions. (eg: Try to setup a hold button on the wireless handset using a config file) We're steering more customers towards polycom now. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, January 22, 2007 9:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 7 points of comparison Polycom 430/501 andAastra 480i. Which one to choose ? With over 300 Polycoms, and around 80 Aastra 480i under my belt here is my $0.02. 1. Sound quality, Polycom wins but the Aastra has excellent sound quality as well. 2. Complete product line, Polycom wins. 3. Cordless Aastra, although it's not the best cordless. 4. Backlit, Aastra 5. PoE, Aastra 6. Speakerphone, they both have good speaker phones. Although in general the answer to 1 goes here as well. 7. They both have 2 network ports, but I havnt' done any tests on the speed, I did however notice that when restarting the phone, the Polycom will not shut the network ports down, while the Aastra will. On another note, in general the Polycoms give me less problems. The Aastras are not yet that stable. See my next post to the list. On 1/22/07, Bruce Reeves [EMAIL PROTECTED] wrote: Your list seems to lean heavily to the Aastra, while I choose the Polycom 501/601 over the Aastra, I did like the unit I tested and the cordless. In the end the fact that most of the people using the phones would use the speaker phone, Polycom and their history of conference phones made the choice. We rolled 75 phones at one site and another 30 now at remote locations. As far a a receptionist phone, we choose to use a software operator panel instead of a phone that took up most of the desk, there were initial concerns but the results have been excellent. If you have not already done so grab a few people from different parts of the office and have them give their 2 cents, it will help to have their perspectives on the quality and feel of the phones. On 1/22/07, Vikas [EMAIL PROTECTED] wrote: I need to provide a 80 people office with VOIP. I want to commit to one vendor Polycom or Aastra. Price of the phones is not a factor in the decision. The quality of the phones is the factor. Some of the features that I am evaluating on are: (arranged in order of priority) 1. Sound quality 2. complete product line with conference phone and receptionist phone (not on Aastra) 3. cordless (not on 501/430) 4. backlit LCD (not on 501/430) 5. Inbuilt POE (not on 501) 6. speaker phone 7. 2 network ports. Which one will you choose ? Vikas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Optimum voice problems.
I'm trying to figure out if I'm the only one with these problems with them. I recently had a few customers that switched to them because of the price, of course that means that they have to use FXO ports, but it is realy cheaper, so customers don't really care. In any case, there are 2 issues that I can't get solved, and they are not interested in helping. 1. When they tell you that they are putting all your lines in a hunt, it realy is not a hunt but just CallForwarding No Answer/Busy, what that means is that if I have asterisk setup to first ring a phone for 5 times and then go to an IVR and answer the phohe, it will go to the next line and stop ringing the first line, and therefore never end up in Voicemail or my IVR. 2. No CPC, hung channles, blank voicemails, and all the other goodies that come with no hangup supervison, is a daily thing. Anybody else seen this? TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Optimum voice problems.
C F wrote: 1. When they tell you that they are putting all your lines in a hunt, it realy is not a hunt but just CallForwarding No Answer/Busy, what Some PBX implement line hunting that way. So, you need to Answer before you do anything else. Otherwise the PSTN switch will cheerfully go on its merry way while you're scrambling to route the call. that means is that if I have asterisk setup to first ring a phone for 5 times and then go to an IVR and answer the phohe, it will go to the next line and stop ringing the first line, and therefore never end up in Voicemail or my IVR. 2. No CPC, hung channles, blank voicemails, and all the other goodies that come with no hangup supervison, is a daily thing. Make sure you configure your zaptel signaling correctly. If you have loopstart (fxs_ls) then the best solution is use busydetect. On my loopstart line, it will always hangup in 4s if the other party hangs up. Of course, in my loopstart setup I'll still get the occasional blank voicemail if the other party hung up just as asterisk goes to voicemail. But, that's rather normal for most voicemail system. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Optimum voice problems.
On 1/22/07, Leo Ann Boon [EMAIL PROTECTED] wrote: C F wrote: 1. When they tell you that they are putting all your lines in a hunt, it realy is not a hunt but just CallForwarding No Answer/Busy, what Some PBX implement line hunting that way. So, you need to Answer before you do anything else. Otherwise the PSTN switch will cheerfully go on its merry way while you're scrambling to route the call. I disagree about this, this is NOT line hunting, but CallForward No Answer. It's an ignorance from Optimums side to offer it as line hunting. that means is that if I have asterisk setup to first ring a phone for 5 times and then go to an IVR and answer the phohe, it will go to the next line and stop ringing the first line, and therefore never end up in Voicemail or my IVR. 2. No CPC, hung channles, blank voicemails, and all the other goodies that come with no hangup supervison, is a daily thing. Make sure you configure your zaptel signaling correctly. If you have loopstart (fxs_ls) then the best solution is use busydetect. On my loopstart line, it will always hangup in 4s if the other party hangs up. Of course, in my loopstart setup I'll still get the occasional blank voicemail if the other party hung up just as asterisk goes to voicemail. But, that's rather normal for most voicemail system. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2-way MS-GSM support in Asterisk?
Hi all, I realise Asterisk can accept an MS-GSM (65 byte) RTP stream and convert it to GSM etc. but can Asterisk do the reverse (i.e. send *out* a 65 byte MS-GSM stream to a client?). I would like to use the MS-GSM codec for Microsoft clients - but at the moment I am getting garbled voice probably due to the 33 byte GSM RTP coming back from * to the client. Any advice on how this might be done would be welcome. Rgds, Ray ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Optimum voice problems.
C F wrote: On 1/22/07, Leo Ann Boon [EMAIL PROTECTED] wrote: C F wrote: 1. When they tell you that they are putting all your lines in a hunt, it realy is not a hunt but just CallForwarding No Answer/Busy, what Some PBX implement line hunting that way. So, you need to Answer before you do anything else. Otherwise the PSTN switch will cheerfully go on its merry way while you're scrambling to route the call. I disagree about this, this is NOT line hunting, but CallForward No Answer. It's an ignorance from Optimums side to offer it as line hunting. IMHO, Regardless of how they market or implement line hunting, you'll still need to get your Asterisk box to answer the call before ringing your user's phone. Otherwise, their switch will just assume no answer and move on to ring the next line. Flame Retardant From a user's POV, there are no perceptible differences between call forward on busy/no answer and linear line hunting. In linear hunting, the switch will try a line and move on if it's busy or there's no answer after a set time. Call forward on busy/no answer will also work pretty much in the same way, albeit more slowly. The main difference is in the provisioning: configuring a single hunt group vs individual call forwarding setting for each number. /Flame Retardant Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] weird undocumented extensions such as s-BUSY
I've seen several examples that use extensions such as; s-BUSY s-NOANSWER etc... It's more or less evident what they do, but I've searched for some FORMAL documentation everywhere and have found nothing. Do they work for anything else than s-? (I think I've seen other examples, but can't find them now) Are they standard in any way? What are the allowed values after the dash? In which version were they introduced? etc... (please no replies explaining me how s-BUSY matches when the start extension is set busy or trivial explanations like that) BarZ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
