RE: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P
Thanks. One of the replies was that the TE110P does have some issues with certain motherboards. If I remember well the problem you are having can only be solved by power cycling the complete system. But I can very well imagine that inserting 2 Digium cards into the 2850 will lead to problems. The 2850 has three PCI slots, the first is sharing it's interrupt with the Perc RAID controller, the other 2 slots each share their interrupt with an onboard NIC. In your case you will never be able to give each card their own interrupt, you would need another PCI slot. Right now zttest is showing me pretty consistent results. The result is always exactly 99.987793% nothing more and in about 1 of 35 probes I get a 100% score. Because 99.987793% is still regarded as the minimum acceptable I might leave things as they are. Or maybe (because the cost is limited) I may start looking for a 2 port, gbit ethernet card and disable the onboard nic's in the 2850 to make sure the digium card has it's own irq. When I have time I will try disabling the onboard nic's first to see if that actually improves the scores, the last time I was messing with the 2850 I think I didn't see any improvement but then again I can't remember exactly anymore what I have / haven't tried. Good luck, hope they will find the problem soon! Cheers! Remco On Fri, 26 May 2006, Mario Montiel wrote: I have many problems with this server and two cards TE4110P, because after several minutes one of two cards stays out without sending anyone alarm and then offer a NMI alarm i suppose that it is to cause the sharing IRQ, it´s a ticket for a DIGIUM -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de stoffell Enviado el: Sábado, 20 de Mayo de 2006 07:29 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P On 5/18/06, Remco Barende [EMAIL PROTECTED] wrote: Also the 2850 is *always* sharing IRQ's on every PCI slot, you need to buy a dual port ethernet adapter which will use only one irq to free up an IRQ on another slot. This just totally sucks and irq sharing in a box with only 3 pci slots is totally unnecessary Because we only needed 1 NIC, we disabled the 2nd onboard NIC. That made 1 pci slot free of IRQ sharing, making the system stable and performing very well. cheers ___ --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] Snom firmwares suck --additional datapoint to consider
Changing firmware revs did not help, so that left the LAN. I looked long and hard at the LAN and it was basically narrowed down to the switches. In this smaller install, several cheapo Dlink ($30) switches de-aggregate a Cisco Catalyst switch. What I noticed was that any phone plugged direcly into the Catalyst did *not* lock up or reboot. Any phone plugged into the crap switches experienced the lockup. So now we are down to the cheap switches themselves. We are nuking the Dlink switches and replacing them with 3com workgroup switches, same as what we use in the large install to good effect, and I fully expect the problem to dissapear. It's unfortunate that Snoms have a propensity to freak out in certain environments but I don't think it would preclude me from using Snom in the future. As long as one is aware of this issue, it should be easy enough to work around. Thanks for your input! Previously I was using Nortel 10/100 switches, I replaced them some weeks ago with 3C16479 gbit switches. The phones are connected directly to the gbit switches. By coincidence I dit notice on one phone that in a split second a message appeared 'Ethernet cable disconnected'. Because I have cable unplug set to ignore the conversation was not interrupted and the conversation could continue. But that still doesn't solve the occasional lockup. One phone was giving me *lots* more reboots than others but that was due to it running firmware 6.0.4 without having the ramdisk converted to jffs. Apparently the firmware didn't like that at all or just runs out of memory and decides to reboot. ___ --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] Snom firmwares suck --additional datapoint to consider
On Fri, 26 May 2006, Dave Cotton wrote: On Fri, 2006-05-26 at 10:11 +0200, Remco Barende wrote: Thanks for your input! Previously I was using Nortel 10/100 switches, I replaced them some weeks ago with 3C16479 gbit switches. The phones are connected directly to the gbit switches. By coincidence I dit notice on one phone that in a split second a message appeared 'Ethernet cable disconnected'. Because I have cable unplug set to ignore the conversation was not interrupted and the conversation could continue. But that still doesn't solve the occasional lockup. Looks like you're getting somewhere now. That was my real complaint xyz sucks helps no one. As I said in my reply I've never had such problems with SNOM, perhaps it's because I've always used decent switches. You mean that 3Com switches are not to be regarded as decent switches? At least Snom could have put some remark then that you need a certain brand of switches. If 3Com is not good enough for the phones I would have bought different phones. ___ --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] Snom firmwares suck --additional datapoint to consider
On Fri, 26 May 2006, Rich Adamson wrote: You mean that 3Com switches are not to be regarded as decent switches? At least Snom could have put some remark then that you need a certain brand of switches. If 3Com is not good enough for the phones I would have bought different phones. Blaming the 3com switch is very likely to be the wrong root cause. High probability the 3com was not configured properly for the phone. The 3C16479 is a non-configurable, non-managed 3Com workgroup gbit switch. It is directly connected to the asterisk server with one cable, the phones are connected to the other ports. There is nothing to configure on the switch. Maybe I need to change my opinion, it's not only the firmware that sucks, if the ethernet chip on the phone is this oversensitive I guess the same would apply for the hardware. There is just no valid reason why the phone would need to lockup or reboot even if the network connection would be problematic, no matter what. That is just poor design, not a feature. ___ --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] A call from a call file always does a redial?
I have an issue with the Snom 360's (any firmware) and asterisk call files. When you setup a call using a call file from Asterisk and the call is connected, Asterisk will start to redial the call after about 5 minutes when the conversation is already ongoing. (Annoying and it can only be avoided by disabling call waiting) I tried to reproduce the problem with a GrandStream phone and a Sipura ATA, it doesn't occur. I guess these are 2 problems : 1) The callfile specifies that a call should not be retried, still * does a redial 2) I *guess* the Snom is returning a different signal than other phones when the call is answered up making Asterisk believe that the call never succeeded. I registered this as a bug in mantis previously but nobody was able to reproduce, I know found out that it is only happening when using a Snom 360 as client. ___ --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] Snom firmwares suck
On Fri, 19 May 2006, Steve Davies wrote: On 5/19/06, Remco Barende [EMAIL PROTECTED] wrote: Most people seem quite positive about Snom phones, I cannot share this opinion. The displays are dying quite often, and firmware is buggy. I have tried every firmware from 4.5 up to 5.x and 6.04 but keep having problems with phones locking up or rebooting during an ongoing conversation. REALLY annoying for a phone that is advertised / targeted as a business class phone Hmmm... A random statement out of the blue... I assume that you meant to add Does any kind soul have a suggestion to help out? :-) I find that the snom phones can be over-sensetive to network glitches, which with the default configuration can cause a reboot (usually caused by cheap switches). Try changing the reboot on ethernet unplug setting to ignore. Tried setting the phone to ignore, it didn't help. Still the phone reboots occasionally during a conversation. But thanks for the suggestion! ___ --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] SIP useragent?
Hi list ! Is it possible to show the used Useragent of a peer that registered with Asterisk? It's being saved obviously because the console says so when a phone is registering but sip show peers doesn't show it? Is there any other way to view it? 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] Snom firmwares suck
Most people seem quite positive about Snom phones, I cannot share this opinion. The displays are dying quite often, and firmware is buggy. I have tried every firmware from 4.5 up to 5.x and 6.04 but keep having problems with phones locking up or rebooting during an ongoing conversation. REALLY annoying for a phone that is advertised / targeted as a business class phone ___ --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] Snom firmwares suck
The displays are dying quite often, and firmware is buggy. I have tried every firmware from 4.5 up to 5.x and 6.04 but keep having problems with phones locking up or rebooting during an ongoing conversation. Yes, as someone asked earlier on the list the displays do die at some random moment without any apparent reason REALLY annoying for a phone that is advertised / targeted as a business class phone Hmmm... A random statement out of the blue... I assume that you meant to add Does any kind soul have a suggestion to help out? :-) Heh :) It is pure frustration I find that the snom phones can be over-sensetive to network glitches, which with the default configuration can cause a reboot (usually caused by cheap switches). Try changing the reboot on ethernet unplug setting to ignore. Good idea, I will change the settings to ignore. The switches are 3Com gbit switches. Not sure if that would qualify as cheap :) I have about 40 Snom 360's and I experienced this problem on my phone at home and some at the office using different firmware versions. Thanks! Remco ___ --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] DELL PowerEdge 2850 and TE4110P and TE110P
On Wed, 17 May 2006, Rodney G. McDuff wrote: Hi All Before I go out and buy a DELL PowerEdge 2850 has anyone had problems (or any other useful experience) getting a TE411P to work with it. I also have a legacy TE110P. Has anyone had problems with this combo. I tried using a TE110P and a TE210P. The irq hit percentage on the Dell with zttest is quite disappointing, even after tweaking. It's just above the minimum requirement Also the 2850 is *always* sharing IRQ's on every PCI slot, you need to buy a dual port ethernet adapter which will use only one irq to free up an IRQ on another slot. This just totally sucks and irq sharing in a box with only 3 pci slots is totally unnecessary I'm now using the 2850 with one TE210P, system is in production use since the 1st of May and no lockups or apparent problems so far. I would have expected better from the 2850 however and would not buy one again Sangoma cards are said to be better but I preferred to support Digium and I don't want to mess around with additional drivers Just my $0.02 ___ --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] Extension '' in context 'whatever' from '123456789' does not exist.
Hi list! I have a Legacy PBX hooked up to one port of a TE210P. One is connected to I want to use the PBX for incoming and outgoing fax. I managed to get incoming working but I don't understand how to feed calls from the PBX into the FreePBX setup. It seems that when you press zero on the Legacy PBX the PBX just switches over to the PRI to pass each digit nicely one by one as it is received to the PRI. When I press zero for an outside line on a legacy phone I immediately get this error message : !! Unexpected Channel selection 3 -- Extension '' in context 'legacypbx' from '(the number of the phone on the pbx)' does not exist. Rejecting call on channel 0/31, span 2 legacypbx is the context I put on the port connected to the Legacy pbx: zapata.conf : group=1 context=legacypbx switchtype=euroisdn overlapdial=yes pridialplan=unknown prilocaldialplan=unknown signalling=pri_net echocancel=yes echocancelwhenbridged=no channel = 32-46 channel = 48-62 Does anyone have an idea how to solve this? Basically I want Asterisk to accept the call from the PBX. Can I set the extension that * is expecting hard in a config file and also have asterisk wait till all the digits have arrived and then dial the number? Basically after pressing 0 I want to feed the call into the standard dial plan. Many thanks for any hints / pointers! Remco ___ --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: Extreme delay before * processes call files
Found it! It seems that Asterisk is looking at the date / time stamp of the call file to process the call?? I was simply moving the call files hoping it would just work (tm) I guess that the call files created on the samba share I created carried the time/date stamp of the local machine (workstation) and not the asterisk server causing a time difference. Now I run a touch * on the asterisk server before moving the call files, all the calls are now processed immediately. Is this intended behaviour for the call files?? Or just a bug? Thanks On Fri, 28 Apr 2006, Remco Barende wrote: I guess that I'm the only one experiencing this problem is there any way to debug this problem? Does anyone know how to debug this particular item in *? (Or should I open a bug in Mantis?) Thanks!! On Thu, 27 Apr 2006, Remco Barende wrote: Hi list! I'm using Asterisk 1.2.7.1. with FreePBX 2.0.1 on a CentOS 3.7 box. On the * box I also have a samba share where our CRM app can dump call files and a cron script is moving the call files every second to the asterisk directory. Everything goes really quickly, the call file is placed on the samba share and very quickly moved to the asterisk dir, so far so good. But then the call file just keeps sitting in the /var/spool/asterisk/outgoing directory and it seems that * is doing nothing with it?? Only after 10-30 seconds sometimes even much longer the call file is picked up. There is no message on the * console about a call file being present. Does anyone have a clue why asterisk fails to pick up call files within a reasonable amount of time? The load on the box is 0.05 at most. Thanks!! Remco ___ --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] FreePBX in production?
FreePBX is catching up reasonably quickly. There are still some basic things missing (for example if you don't use voicemail it is not possible to set a destination for the call if not answered, you have to create a ring group for each extension to work around it, this is a major issue) and some smaller minor issues. Other than that I think the basic framework is already pretty impressive but in my opinion good enough for production use. On Mon, 1 May 2006, Craig Guy wrote: Wouldn't use it in production for a customer personally. Too many limitations in terms of having a flexible diaplan. What would be nice though is if they were to produce a 'lite' version that gave a gui interface to add/change/move things - sip.conf, voicemail.conf, meetme.conf but staying well away from extensions.conf Craig - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users-List asterisk-users@lists.digium.com Sent: Monday, May 01, 2006 5:19 AM Subject: [Asterisk-Users] FreePBX in production? Has anyone attempted to use FreePBX for a business in production mode? Initial take is there are lots of things scripted but a lot of limitations in terms of supporting basic business functions. Inability (or lack of flexibility) is handling multiple incoming pstn lines, dialplan limitations, poor/no documentation, etc, to mention a few. Maybe its just me, but it appears its no where near usable even with the latest beta1 code. Is it just me or what? Rich ___ --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] FreePBX in production?
On Mon, 1 May 2006, Kerry Garrison wrote: There are still some basic things missing (for example if you don't use voicemail it is not possible to set a destination for the call if not answered, you have to create a ring group for each extension to work around it, this is a major issue) Remco - take a look at the Follow Me module I added. It is basically a presonal ring group for each extension. If you want to do the above, just define the Follow-Me settings to ring your own extension (or more if you want) and then choose any destination you want. It effectively does 'creat a ring group for each extensions' that wants one, but it does it in such a way as to be separate and work side by side with normal ringgroups, and there is a direct link between it and the extension (or user) so that navigation is very easy as you can bounce back and forth with a single mouse click. Sounds very interesting, and looks like a good solution to the problem. I believe that many of the shortcomings of FreePBX are caused by lack of documentation, i.e. you do not immediately see how to solve a certain problem. Thanks to all the develeopers for all the work they are putting in FreePBX! I'll have a look at the new functions, 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] Legacy PBX integration
Hi list! I managed to come reasonably far (farther than I thought I would) but have two problems. I still need to pass calls to the Legacy PBX for Fax (I need it as a channel bank). I have calls coming in into asterisk, that works fine. Based on the DID I can route calls to the Legacy PBX but I'm puzzled how. I guess I need a new dial command for that? All fax calls are now coming in a new context which I called topbx. If I issue a dial command there the legacy PBX treats it as a local extension call and not a call from the outside. Which dial command do I need to use to make the old PBX believe the call came from outside? (All the pages I found on this subject mention something about retaining caller ID which is nice but now I need to retain DID info on the call I guess?) Thanks for any help! ___ --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] Digium cards, so disappointing !
On Sun, 30 Apr 2006, Boris Bakchiev wrote: I must say, spending just a little extra to get good hardware pays off in the long run. If you have any questions, email. Wow, impressive results must say. Thanks for the specs and test results. I had hoped that with the Dell 2850 I would have bought a decent piece of hardware, it isn't. I e-mailed Dell support and asked them if it is possibel to assign a unique IRQ to one of the three PCI slots. Their reply was, not possible, you are ALWAYS sharing IRQ's, I guess this is the reason for the poor results I'm seeing. I will try to find a solution. Thanks again! Remco ___ --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] Legacy PBX integration
Duh.. sorry for this dumb mistake, basicaly the connection is: PRI -TE210 Port 1- * -TE210 Port 2- Legeacy PBX Basically I need * to send whatever the telco used to send to the pri Thanks! On Sun, 30 Apr 2006, Jerry Jones wrote: You do not say how you have the two connected/ Are you connecting the * to stations via fxo or to lines via fxs on the legacy? On Apr 30, 2006, at 11:22 AM, Remco Barende wrote: Hi list! I managed to come reasonably far (farther than I thought I would) but have two problems. I still need to pass calls to the Legacy PBX for Fax (I need it as a channel bank). I have calls coming in into asterisk, that works fine. Based on the DID I can route calls to the Legacy PBX but I'm puzzled how. I guess I need a new dial command for that? All fax calls are now coming in a new context which I called topbx. If I issue a dial command there the legacy PBX treats it as a local extension call and not a call from the outside. Which dial command do I need to use to make the old PBX believe the call came from outside? (All the pages I found on this subject mention something about retaining caller ID which is nice but now I need to retain DID info on the call I guess?) Thanks for any help! ___ --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] Legacy PBX integration
Thanks! The PBX is a Alcatel Novo Supreme. All calls go straight into the auto attendant no matter whoch extension I dial on the Zap group the PBX is connected to. I tried dialling in by hand using several combinations but I always get the auto attendant. How do you transfer the call straight to the extension the fax is on? I guess using a Dial command fro *? I suspect that the PBX is missing some signalling. Thanks! On Sun, 30 Apr 2006, Lacy Moore - Aspendora wrote: Also, what is the legacy PBX? On the Merlin Legend, for instance, there are special Class of Services that can be setup to go straight to the auto attendant. I'm not sure if that's what you need or not. The other question is, why can't you transfer the call straight to the extension the fax is on? Again, on the Legend, it defaults (I guess) to receving the extension number. For example, if my fax machine is located on ext. 170, then I just dial 170 from Asterisk on the PRI that is connected to the Legend. I didn't have to do half of what the manual says you have to do, because Asterisk takes care of all the translations. On 4/30/06, Jerry Jones [EMAIL PROTECTED] wrote: You do not say how you have the two connected/ Are you connecting the * to stations via fxo or to lines via fxs on the legacy? On Apr 30, 2006, at 11:22 AM, Remco Barende wrote: Hi list! I managed to come reasonably far (farther than I thought I would) but have two problems. I still need to pass calls to the Legacy PBX for Fax (I need it as a channel bank). I have calls coming in into asterisk, that works fine. Based on the DID I can route calls to the Legacy PBX but I'm puzzled how. I guess I need a new dial command for that? All fax calls are now coming in a new context which I called topbx. If I issue a dial command there the legacy PBX treats it as a local extension call and not a call from the outside. Which dial command do I need to use to make the old PBX believe the call came from outside? (All the pages I found on this subject mention something about retaining caller ID which is nice but now I need to retain DID info on the call I guess?) Thanks for any help! ___ --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 -- Lacy Moore Aspendora, Inc. ___ --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] Legacy PBX integration
Yes indeed I suspect that * is not passing any DID information on the call. This could be because my Dial command is wrong or I may need to use different signalling settings. (Is there any other setting with pri_net?) When doing pri debug I noticed a line that * was thinking that the other side was not ISDN equipment (but there is a lot of output and I never read debug output before) DID was working, the PRI was passing it on to the Alcatel. I'll have a look on the forum you mentioned. Thanks! On Sun, 30 Apr 2006, Lacy Moore - Aspendora wrote: Hmm... In my case, it could be just dumb luck. I found some instructions on setting up DID on my pbx, and started that. Part way through, I wasn't sure what the rest of the instructions were talking about and felt I was getting in too deep. So, I decided to see what would happen if I just tried it. It worked. Since this is only a temporary solution until we move completely off the pbx to Asterisk, I felt like I didn't need to find out why it worked, just be thankful that it worked. It sounds like your system is just answering the line and not paying attention to any DID information. I was able to find a lot of information on the tek-tips.com forums for the Merlin Legend. You may try a search on there and see. Was DID working in the past, or have you just added it with the addition of the Asterisk system? On 4/30/06, Remco Barende [EMAIL PROTECTED] wrote: Thanks! The PBX is a Alcatel Novo Supreme. All calls go straight into the auto attendant no matter whoch extension I dial on the Zap group the PBX is connected to. I tried dialling in by hand using several combinations but I always get the auto attendant. How do you transfer the call straight to the extension the fax is on? I guess using a Dial command fro *? I suspect that the PBX is missing some signalling. Thanks! On Sun, 30 Apr 2006, Lacy Moore - Aspendora wrote: Also, what is the legacy PBX? On the Merlin Legend, for instance, there are special Class of Services that can be setup to go straight to the auto attendant. I'm not sure if that's what you need or not. The other question is, why can't you transfer the call straight to the extension the fax is on? Again, on the Legend, it defaults (I guess) to receving the extension number. For example, if my fax machine is located on ext. 170, then I just dial 170 from Asterisk on the PRI that is connected to the Legend. I didn't have to do half of what the manual says you have to do, because Asterisk takes care of all the translations. On 4/30/06, Jerry Jones [EMAIL PROTECTED] wrote: You do not say how you have the two connected/ Are you connecting the * to stations via fxo or to lines via fxs on the legacy? On Apr 30, 2006, at 11:22 AM, Remco Barende wrote: Hi list! I managed to come reasonably far (farther than I thought I would) but have two problems. I still need to pass calls to the Legacy PBX for Fax (I need it as a channel bank). I have calls coming in into asterisk, that works fine. Based on the DID I can route calls to the Legacy PBX but I'm puzzled how. I guess I need a new dial command for that? All fax calls are now coming in a new context which I called topbx. If I issue a dial command there the legacy PBX treats it as a local extension call and not a call from the outside. Which dial command do I need to use to make the old PBX believe the call came from outside? (All the pages I found on this subject mention something about retaining caller ID which is nice but now I need to retain DID info on the call I guess?) Thanks for any help! ___ --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 -- Lacy Moore Aspendora, Inc. ___ --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 -- Lacy Moore Aspendora, Inc. ___ --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: Extreme delay before * processes call files
I guess that I'm the only one experiencing this problem is there any way to debug this problem? Does anyone know how to debug this particular item in *? (Or should I open a bug in Mantis?) Thanks!! On Thu, 27 Apr 2006, Remco Barende wrote: Hi list! I'm using Asterisk 1.2.7.1. with FreePBX 2.0.1 on a CentOS 3.7 box. On the * box I also have a samba share where our CRM app can dump call files and a cron script is moving the call files every second to the asterisk directory. Everything goes really quickly, the call file is placed on the samba share and very quickly moved to the asterisk dir, so far so good. But then the call file just keeps sitting in the /var/spool/asterisk/outgoing directory and it seems that * is doing nothing with it?? Only after 10-30 seconds sometimes even much longer the call file is picked up. There is no message on the * console about a call file being present. Does anyone have a clue why asterisk fails to pick up call files within a reasonable amount of time? The load on the box is 0.05 at most. Thanks!! Remco ___ --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] Extreme delay before * processes call files
Hi list! I'm using Asterisk 1.2.7.1. with FreePBX 2.0.1 on a CentOS 3.7 box. On the * box I also have a samba share where our CRM app can dump call files and a cron script is moving the call files every second to the asterisk directory. Everything goes really quickly, the call file is placed on the samba share and very quickly moved to the asterisk dir, so far so good. But then the call file just keeps sitting in the /var/spool/asterisk/outgoing directory and it seems that * is doing nothing with it?? Only after 10-30 seconds sometimes even much longer the call file is picked up. There is no message on the * console about a call file being present. Does anyone have a clue why asterisk fails to pick up call files within a reasonable amount of time? The load on the box is 0.05 at most. Thanks!! Remco ___ --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] Dreadful results from zttest with TE210P and Dell 2850?
Hi list! I'm trying to setup a new * server with a TE210P in a Dell 2850 box. I'm using zaptel version 1.2.5, the linux flavour I'm using is CentOS 4. In a previous thread I read about the results I should expect from zttest. On my home box (using the crappy Asus A7V600) I got really bad results from zttest (just over 97.5) but I know that this motherboard just sucks. To my (huge) disappointment however the results from zttest are equally as bad as from my home box?? (just over 97.5) lspci -vb reveals that the card is sharing IRQ 3 with the second Gbit LAN controller. The box is only idling I'm the only user shh'ing into it. Does anyone have a clue why the results from zttest are this horrible? Looking at the wiki I don't even need to try and put the box into production with such results. 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] Dreadful results from zttest with TE210P and Dell2850?
Thanks for the hints and tips. While you are familiar with the 2850, I am using the PERC raid controller but guess this shouldn't make any real difference. I used the middle PCI slot for the TE210P, do you use any particular slot. I will disable HyperThreading and the box was already running an SMP kernel (there were no irq conflicts shown by lspci -v) in runlevel 3. Are you using the onboard e1000 ethernet controllers? The wiki is advising not to. Thanks for your input! Remco On Mon, 24 Apr 2006, Craig Guy wrote: Using an SMP kernel will fix the interrupt sharing, you could also disable hyperthreading and set runlevel 3. FWIW I almost exclusively use Poweredge 850 for my * servers with a third party sata raid controller if raid is required. Never had any problems. Craig - Original Message - From: Remco Barende [EMAIL PROTECTED] To: Asterisk Users List asterisk-users@lists.digium.com Sent: Monday, April 24, 2006 6:38 PM Subject: [Asterisk-Users] Dreadful results from zttest with TE210P and Dell2850? Hi list! I'm trying to setup a new * server with a TE210P in a Dell 2850 box. I'm using zaptel version 1.2.5, the linux flavour I'm using is CentOS 4. In a previous thread I read about the results I should expect from zttest. On my home box (using the crappy Asus A7V600) I got really bad results from zttest (just over 97.5) but I know that this motherboard just sucks. To my (huge) disappointment however the results from zttest are equally as bad as from my home box?? (just over 97.5) lspci -vb reveals that the card is sharing IRQ 3 with the second Gbit LAN controller. The box is only idling I'm the only user shh'ing into it. Does anyone have a clue why the results from zttest are this horrible? Looking at the wiki I don't even need to try and put the box into production with such results. 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 ___ --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] problem with TE205
Hello, I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have my jumper set (i.e closed to use the E1 facility.) Does the TE205P use jumpers for T1 / E1 setting? I thought jumpers were completely obsolete now? ___ --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] problem with TE205
Weird, I just received a new TE210P card (should be identical only 3.3v) but I cannot find any info on jumper settings om the Digium site? But then again the installation info on the Digium site really sucks. On Fri, 21 Apr 2006, Rob Lith wrote: Jumpers must still be on for E1 mode. Rob On 21/04/06, Remco Barende [EMAIL PROTECTED] wrote: Hello, I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have my jumper set (i.e closed to use the E1 facility.) Does the TE205P use jumpers for T1 / E1 setting? I thought jumpers were completely obsolete now? ___ --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: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)
On Fri, 14 Apr 2006, Kevin P. Fleming wrote: What about a new line of Digium cards that have bridge cables that run between the various cards and bypass the PCI bus? Since one of the best aspects of using Asterisk is standards. This bridge cable should be standardized and published so that other companies can adopt the standard. For example an ISDN card could bridge to a Digium T1 card. Or a card that supported legacy digital phones could bridge to other cards. That is called H.100, and it has existed for many years. It's also ludicrously expensive to implement, so you won't see it on Digium cards any time soon :-) I heard that Junghanns is working on such an interconnection. It is already possible to connect their PRI cards, and they are working on BRI-PRI. I ise their bristuff for an HFC-S BRI card and am not happy at all with the way they implemented timing, without applying the florz patch I have lots of problems (lockups, lost line etc.) My hesitation is with the driver, I think florz only fixes HFC-S if I would run into similar trouble with PRI I would be in deep trouble. But it would certainly fix faxing. I have now ordered a TE210P and will try native bridging using a legacy PBX for my faxing. I hope this will solve my faxing nightmares. (I don't care if a solution costs money, it just needs to work). I think it would be a good idea if Digium would put some articles about fax and possible solutions or work-arounds to fax problems on their website. ___ --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: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)
So, to document this, the likelihood of a fax working goes in this order best to worse: 1. POTS - fax 2. POTS - FXO-TDM400P-FXS - fax 3. T1 - TE410P - channel bank - fax 4. T1 - TE110P - PCI - TE110P - channel bank - fax 5. T1 - TE110P - PCI - TDM400P-FXS - fax 6. T1 - TE110P - PCI - Ethernet/IP - IAXy - fax 7. FXO-TDM400P - PCI - Ethernet/IP - IAXy - fax Is this a correct? If it's not a PCI problem then there shouldn't be much of a difference between options 3 and 4. If it's a card issue then it would be nice to know which T1 cards handle fax better than others. Yes, BUT!!! be aware that if you have an E1 pri from your telco a T1 channel bank will not help anything. In this case (your option 3) native bridging will be possible and asterisk will have to transcode giving you the some problems again. ___ --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] Digium cards, so disappointing !
On Sat, 15 Apr 2006, Kevin P. Fleming wrote: Actually, I did. During a FAX transmission, there are many shifts to different carriers and signaling rates as pages are transmitted and acknowledged. It is _not_ as simple as a single carrier, like a normal data modem connection. In addition to those shifts occurring, they are very strictly timed and must occur within fairly short windows. Hi Kevin, I think the biggest problem is that almost any more modern fax machine persistently tries to connect at the highest possible speed. To solve the problem I suggested a workaround to this earlier on the list, no idea if it is technically possible or dfficult to implement, this is what I wrote : record the sound fax machines make when negotiating (specifically the part where they try to negotiate anything above 9600 baud) and make a provision in asterisk (an extra letter added to the Dial command?) that will make Asterisk monitor the channel and listen for the fax nego sounds and have Asterisk distort or mute the audio. This way all fax machines would be forced to lower their speeds. I suspect that such a solution would greatly improve reliability for faxing without the need for drastic changes in the way asterisk works. If you could lower the speed further down to 4800 or even 2400 baud that might even be an interesting option. Instead of faxing at 9600 or 14k4 through a normal (expensive) landline it could be cheaper to fax even at 2400 baud via a voip line depending on where you need to fax to. None of my fax machines are able to reduce their TX/RX speeds, if any devices capable of capping the speed it would be a nice addition to the wiki, I would instantly buy some all-in-one machines that could do that The lower connection speeds wouldn't bother me, reliable faxing would make up for the lost connection speed! ___ --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] Digium cards, so disappointing !
record the sound fax machines make when negotiating (specifically the part where they try to negotiate anything above 9600 baud) and make a provision in asterisk (an extra letter added to the Dial command?) that will make Asterisk monitor the channel and listen for the fax nego sounds and have Asterisk distort or mute the audio. This way all fax machines would be forced to lower their speeds. Complex, clunky, and solves nothing. Hmm not so sure of that. I have an HP all-in-one thingy. It is not possible to set the TX/RX speed hard in the config at a certain speed. Through the developers menu in the beast it is possible to do this temporary. Faxing at max 9600 bps works, anything higher fails miserably after the second or third page. None of my fax machines are able to reduce their TX/RX speeds, if any devices capable of capping the speed it would be a nice addition to the wiki, I would instantly buy some all-in-one machines that could do that Actually, they are all able to reduce their speed when they need to. They figure out for themselves what the path is capable of. Often in a slightly quirky way, but bugs are the mainstay of the FAX industry. Indeed but asking HP to implement a cap on the TX/RX speeds is like talking to a brick wall. I guess we will have to look for alternative solutions! ___ --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] Digium cards, so disappointing !
Hmm not so sure of that. I have an HP all-in-one thingy. It is not possible to set the TX/RX speed hard in the config at a certain speed. Through the developers menu in the beast it is possible to do this temporary. Faxing at max 9600 bps works, anything higher fails miserably after the second or third page. This doesn't make sense. The known problems are all timing related, and 9600 (I presume you mean V.29 at 9600) is no more or less sensitive to timing slips than V.17. Actually, on a poor line V.17 at 9600bps should perform considerably better than V.29 at 9600bps. Can you tell me your exact setup? There must be something else wrong. I tried lots of different settings but none really seemed to help. The line is ISDN BRI with an HFC-S card. Software is bristuff with florz patch. Echo can, silence suppr. etc all disabled. The HP is connected to a Sipura SPA 2000 with the correct settings for fax and the region i'm in. Still consistently faxes fail after the first or second page. The HP is a LaserJet 3330 mfp. Setting it back 9600 did help a bit. I solved the problem now by connecting an old Digital - Analog converter to the BRI line, bypassing Asterisk. 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] Config with TE210P, Asterisk and Legacy PBX and FreePBX?
Hi list! Has anyone ever tried the following installation : I want to replace our legacy PBX with Asterisk but... I still need the legacy PBX as a 'channel bank' for fax (I need E1 not T1) I will put a dual port PRI card in the Asterisk box, and for incoming and outgoing faxes I want to use native bridging on the TE210P and route fax calls (based on DID and prefix when dialling) to / from the legacy PBX. I guess I do not need to modify anything in the PBX (Alcatel Novo Supreme) because I can simply use dialling prefixes to catch outbound calls. Does anyone have example config files how to implement this config? This would be the setup : PRI - Asterisk -- Legaxy PBX on TE210P |- SIP phones Would it be possible to use FreePBX to setup such routing (inbound and outbound), if anyone could guide me in the basic direction for this I would be most grateful. 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] Fax over 2 bridged TE110P channels
I suspect that in your case the fax channels are not natively bridged. I'm
not sure whether native bridging will work if you are using 2 cards. It
should work if the ports are on one card i.e. TE205P or TE210P in which
case there will be no intervention of Asterisk at all.
You might want to check with Digium support to verify
Let me know the result.
Cheers!
Remco
On Tue, 4 Apr 2006, Alessio Focardi wrote:
Hi,
I have an asterisk installation with 2 E1 cards
Software version is
Asterisk 1.2.6
Libpri 1.2.2
Zaptel 1.2.5
I'm having problem with fax transmission, let me explain better my
setup:
My fist TE110P E1 card is connected to the telco line
the second TE110P E1 one to an Nexspan PBX
so the server is basically sitting between the line, and the pbx.
every call coming from the line is simply redialed in the pbx
every call from pbx is simply redialed to the line
no answer is done
All is working great with voice, but faxing often results in error, both
receiving and sending.
I have disabled echo cancel, and also checked for interrupts problems
and other common misconfiguration problems.
Would someone please help me sort this out ?
I'm suspecting sync problems ...
Tnx for any help!
Following are some debug and config files
zaptel.conf
loadzone = it
defaultzone = it
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,0,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
zapata.conf
[channels]
switchtype = euroisdn
;line
signalling=pri_cpe
pridialplan=unknown
switchtype=euroisdn
priindication = outofband
echocancel=no
overlapdial=yes
immediate=no
nationalprefix=
internationalprefix=
resetinterval=300
context=pri1
group=1
channel = 1-15
channel = 17-31
;pbx
signalling=pri_net
pridialplan=international
switchtype=euroisdn
priindication=outofband
echocancel=no
overlapdial=yes
immediate=no
nationalprefix=
internationalprefix=
resetinterval=300
context=pri2
group=2
channel = 32-46
channel = 48-62
pri1 context
exten=_X.,1,Dial(Zap/g2/${EXTEN}||j)
exten=_X.,2,Congestion()
exten=_X.,102,Busy()
pri2 context
exten=_X.,1,Dial(Zap/g1/${EXTEN}||j)
exten=_X.,2,Congestion()
exten=_X.,102,Busy()
cat /proc/interrupts
CPU0
0: 1114420235 XT-PIC timer
1: 8 XT-PIC i8042
2: 0 XT-PIC cascade
5: 1114083499 XT-PIC t1xxp
8: 1 XT-PIC rtc
9: 0 XT-PIC acpi
10:2531734 XT-PIC eth0
12: 1114121836 XT-PIC t1xxp
14: 306435 XT-PIC ide0
NMI: 0
lspci -v
00:00.0 Host bridge: Silicon Integrated Systems [SiS] SiS645 Host Memory
AGP Controller (rev 01)
Flags: bus master, medium devsel, latency 32
Memory at e000 (32-bit, non-prefetchable) [size=64M]
Capabilities: [c0] AGP version 2.0
00:01.0 PCI bridge: Silicon Integrated Systems [SiS] Virtual PCI-to-PCI
bridge (AGP) (prog-if 00 [Normal decode])
Flags: bus master, fast devsel, latency 64
Bus: primary=00, secondary=01, subordinate=01, sec-latency=0
Memory behind bridge: dde0-dfef
Prefetchable memory behind bridge: d9c0-ddcf
00:02.0 ISA bridge: Silicon Integrated Systems [SiS] SiS961 [MuTIOL Media
IO]
Flags: bus master, medium devsel, latency 0
00:02.1 SMBus: Silicon Integrated Systems [SiS] SiS961/2 SMBus Controller
Flags: medium devsel
I/O ports at 0c00 [size=32]
00:02.5 IDE interface: Silicon Integrated Systems [SiS] 5513 [IDE] (rev d0)
(prog-if 80 [Master])
Subsystem: Silicon Integrated Systems [SiS] SiS5513 EIDE Controller
(A,B step)
Flags: bus master, fast devsel, latency 128
I/O ports at ff00 [size=16]
00:03.0 Ethernet controller: Silicon Integrated Systems [SiS] SiS900 PCI
Fast Ethernet (rev 90)
Subsystem: Silicon Integrated Systems [SiS] SiS900 10/100 Ethernet
Adapter
Flags: bus master, medium devsel, latency 64, IRQ 10
I/O ports at dc00 [size=256]
Memory at dfffc000 (32-bit, non-prefetchable) [size=4K]
Expansion ROM at dffa [disabled] [size=128K]
Capabilities: [40] Power Management version 2
00:08.0 ISDN controller: Cologne Chip Designs GmbH: Unknown device 16b8 (rev
01)
Subsystem: Cologne Chip Designs GmbH: Unknown device b562
Flags: medium devsel, IRQ 11
I/O ports at d800 [size=8]
Memory at d000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
00:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
Subsystem: Unknown device 6159:0001
Flags: bus master, medium devsel, latency 64, IRQ 5
I/O ports at d400 [size=256]
Memory at dfffe000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
00:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
Subsystem: Unknown device 6159:0001
Flags: bus master, medium devsel, latency
Re: [Asterisk-Users] Asterisk 1.2.3 CentOS 4.x RPMS
Hi! I'm trying to install the RPMS, in the installation document the following module is not mentioned: perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm But the RPM is in the CentOS 4 directory. On CentOS 4 the rpm is even already present albeit an older version: [EMAIL PROTECTED] rpms]# rpm -qa | grep -i perl-dbd perl-DBD-MySQL-2.9004-3.1 Update of the stock rpm doesn't work: [EMAIL PROTECTED] rpms]# rpm -Uvh perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm warning: perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm: V3 DSA signature: NOKEY, key ID 0f0d98a3 Preparing...### [100%] file /usr/share/man/man3/Bundle::DBD::mysql.3pm.gz from install of perl-DBD-mysql-3.0002-1.RHEL4.LSE conflicts with file from package perl-DBD-MySQL-2.9004-3.1 etc. What should I do with the new module, is it safe to ignore (not install) the LSE rpm? Thanks!! On Thu, 26 Jan 2006, Andrew McRory wrote: Available in the usual place. ftp://ftp.linuxsys.com/pub/releases/CentOS-4.0 This release includes minor spec changes, spandsp 0.0.2pre23, a new Sangoma wanpipe RPM for use with the LSE kernel rpm and an AMP installation document. Best Regards, ___ --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] Kirk IP600
Hi! Yes, it works (sort of) but I still have some issues. When using more than 2 handsets some of them do not always ring on an incoming call. This might be because I use only 2 Kirk handsets and the rest are Siemens, maybe it's the driver I created a howto for it, you can find it here: http://www.ecem-it.nl/hardware/Asterisk-Kirk-IP600.txt Let me know if you find any errors / omissions, or the solution to the ringing problem :) On Mon, 30 Jan 2006, Giordano Grandis wrote: Hi all, has anyone tryied to configure asterisk with Kirk IP600 Dect-IP gateway? Could it works using the skinny channel ? Thanks Giordano ___ --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] E1 - T1 native bridging for fax, will it work?
Hi list! The stability of * with fax (or the lack of it) is causing me headaches. To solve it, I was thinking to put a TE205P card in the * box, connect the E1 pri on one port and a channelbank (I was thinking of the Rhino) on the other port in T1 mode. (Has anyone tried this??) The TE205P supports channel bridging which sounded like the ideal (but not cheap) solution when combined with a channel bank. When checking with Digium support if E1-T1 bridging will work they replied: It is unlikely to work consistently with that configuration, because E1s typically use alaw encoding and T1s typically use ulaw encoding, and the ulaw-alaw conversion would be done in software, thereby making a native bridge not possible. This makes sense but raises several questions: - even if * will do the software conversion will the quality of the channel remain sufficient to provide rock solid faxing. Maybe the channel bank is considerably better than the sipura's I have been using so far and there would be no ethernet involved. - are there any T1 channel banks that support alaw or would I be forced to either buy an E1 channel bank or use a legacy PBX as channel bank - who decides which encoding is used, zaptel or the channelbank (in other words, if a T1 channel bank would support alaw, can I force alaw only to guarantee native bridging? Any help / input / thoughts / experiences greatly appreciated!! Thanks, Remco ___ --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] Eid Mubarak
Is there no list moderation, could somebody please kick all these spammers off the list? This just sucks, the lists volume is very high already even without the spam ___ --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] Fax baud rate
yes i know, it's not i have been battling with this problem myself I thought of a theoretical solution (but I have zero programming skills which means no possibility to code / try it) : record the sound fax machines make when negotiating (specifically the part where they try to negotiate anything above 9600 baud) and make a provision in asterisk (an extra letter added to the Dial command?) that will make Asterisk monitor the channel and listen for the fax nego sounds and have Asterisk distort or mute the audio. This way all fax machines would be forced to lower their speeds. I suspect that such a solution would greatly improve reliability for faxing without the need for drastic changes in the way asterisk works. If you could lower the speed further down to 4800 or even 2400 baud that might even be an interesting option. Instead of faxing at 9600 or 14k4 through a normal (expensive) landline it could be cheaper to fax at 2400 baud via a voip line depending on where you need to fax to. None of my fax machines are able to reduce their TX/RX speeds, if any devices capable of capping the speed it would be a nice addition to the wiki, I would instantly buy some all-in-one machines that could do that On Sun, 8 Jan 2006, Mark Ackroyd wrote: Does anyone know if it's possible to set the incoming and or outgoing fax baud rate in asterisk ? Mark ___ --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] GSM Gateway / Terminal for sale
Not really, their suggested retail price is USD 300 for the analog unit, probably because of the intelligent stuff in the box (which we do not need when using *). At USD 300 you can find SIP capable devices, for an analog unit the SIPCE is 3x more expensive than the unit we were discussing. But thanks for the tip! On Thu, 5 Jan 2006, Cory Andrews wrote: SICPE has a new product called the GSM Call Director that may be of interest to GSM enthusiasts. http://www.sipcpe.com/fx300GSM.html Cory Andrews Purchasing Manager ++ VOIPSupply.com A Division of b2 Technologies 454 Sonwil Drive Buffalo, NY 14225 direct - 716.250.3402 mobile - 716.907.4054 email - [EMAIL PROTECTED] AIM - b2Cory - Original Message - From: Sam Tam [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, January 05, 2006 3:30 PM Subject: RE: [Asterisk-Users] GSM Gateway / Terminal for sale We have ran out of stock in our office in UK. All GSM Gateway are now being send from HK therefore the shipping will be more expensive than usual. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bails Sent: Friday, January 06, 2006 12:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GSM Gateway / Terminal for sale Chris Bagnall wrote: Single port GSM Gateway support 900 / 1800 GSM mode with external antenna. Brand new unit and all of them will be tested before dispatch. Extremely easy to setup and can be used out of the box without any configuration. So should be good alternatively of phonecell or nokia pbx etc.. Units are located in UK and £60 GBP per unit excluding shipping. Has anyone bought one of these and able to offer some feedback? I'm seriously considering a GSM gateway to take advantage of the spare SIM cards lying around still inside their 12-month contracts. Looking at the website in question, delivery is £17.37 for a 6-day delivery, or £10 for a 30+ day delivery, both of which seem a bit high for an item apparently located in the UK. Regards, Chris We were working in the area (Reading) and offered to pay cash and collect from their site, but the response was; that they could only be sent direct from the far east We weren't prepared to take the risk, I mean they turned down cash! Bails ___ --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
RE: [Asterisk-Users] GSM Gateway / Terminal for sale
VOIP - GSM calls may be cheap if you call to China. When you call a cell in The Netherlands it will cost you USD 0.25 per minute. I am located in NL therefore a lot of calls go to NL mobiles. You can buy sim cards that offer minutes for USD 0.02 per minute, if you can recommend a carrier that offers VOIP - NL GSM calls for that amount I will be very happy :) On Fri, 6 Jan 2006, JCC wrote: I don't get it. What is the advantage of using a GSM gateway? VOIP calls are pretty inexpensive as they are now. Is the use of a gateway intended as a backup incase a wired network connection goes down? I have being looking around the net for information on this. Anyone out there using it and if so you can please share with me how you use this technology? Any information will be appreciated. Thanks, Jay -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Friday, January 06, 2006 5:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GSM Gateway / Terminal for sale Remco Barende wrote: Not really, their suggested retail price is USD 300 for the analog unit, probably because of the intelligent stuff in the box (which we do not need when using *). At USD 300 you can find SIP capable devices, for an analog unit the SIPCE is 3x more expensive than the unit we were discussing. Where can I find the $300 SIP capable units? ___ --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] Fax with Asterisk and Sipura 2100
I tried to get it working for a very long time (over a year) with every possible set of config parameters I could find both for * as well as for the Sipura's. Echo cancelling etc. etc. all changed but still problems. I tried to get it working on an * box with a BRI line. Finally I have given up and attached a traditional ISDN - Analog (A/B) converter to the ISDN line for the faxing bit next to Asterisk. I have yet to find a similar solution for faxing with a PRI, I'm afraid it will be impossible because as far as I know it's not possible to hook up some sort of A/B adapter next to the * box on one pri line. I think it can work if your fax machines are capable of capping fax tx/rx speeds to 9600 baud maximum without error correction. However it seems that not a single producer of FAX equipment (be it modems, all-in-one devices or even dedicated fax machines) offer such an option. HP doesn't seem very interested in capping the fax speeds for their all-in-one thingies. All fax products keep trying to transmit/receive at higher speeds after which the fax will fail completely or after the second page. Maybe there is a solution coming for PRI faxing. Junghanns informed me some time ago that they were working on a PRI card with a possibility to sync the clock to other cards. If this works in theory you could use a Junghanns PRI card and a Junghanns BRI card, sync the clocks and keep the path fully digital without lost frames. On their website however they only mention the possibility to interconnect the PRI cards, not (yet?) PRI - BRI. On Thu, 5 Jan 2006, Darrell Long wrote: I know the subject of faxing has been covered in some detail, but I was wondering if anyone has a hardware configuration similar to ours that has faxes working successfully and would be willing to share any settings/insight. We are unable to fax reliably with a Sipura 2100 connected to Asterisk. We do not route calls over the Internet and our network has very low latency. The Asterisk servers connect to Cisco Routers that have PRIs from various carriers. We have all the recommended settings in the Sipura ATA, with Echo Cancellation and Silence Suppression off, uLaw only for the codec, etc. While I realize that no faxes going through passthrough like this will work 100% of the time, we currently have a less than 40% success rate with inbound faxes being the worst. Any insight anyone has would be greatly appreciated! Best Regards, ___ --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] snom Firmware 5.0.
Thanks for the new firmware, finally some of the features are becoming available that make the phone more usable with Asterisk. One question though, ringer tone #2 on the Snom 360 firmware has been replaced? How can I get the old ringtone back? I was using the ringtone on phones in locations like meeting rooms. The ringtone wasn't intrusive at all, yet well audible. Now when a phone rings everybody is disturbed with a loud noise. Thanks! Remco On Thu, 22 Dec 2005, Usman Tahir wrote: Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, - Usman Tahir snom technology AG www.snom.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] snom Firmware 5.0.
Hi Usman, Thanks for the explanation. Could you make the old Ringer 2 available in some form, preferable already in the format the phone understands? That would solve the problem too :) Thanks!! Remco On Mon, 2 Jan 2006, Usman Tahir wrote: Hi Remco, Old Ringer 2 is not there on the phone anymore, perhaps you can use another ring melody or a suitable custom melody: The wav file itself should be a PCM encoded 8 KHz file at 16bit mono. The time for loading the file should not be longer then 3 seconds ! And the size should be below 250KB. To create this format from mp3: /usr/bin/mpg123 -m -r 8000 -w - -n 190 -q test.mp3 test.wav To convert an existing WAV file: sox GENERIC.wav -c 1 -r 8000 -w SNOM.wav * The -c 1 flag makes the output mono. * The -r 8000 flag makes the output a 8kHz sample. * The -w flag uses 16 bits (word) per sample. Regards, Usman. - Usman Tahir snom technology AG Gradestraße 46 D-12347 Berlin. Tel: +49 30 398330 Fax: +49 30 39833111 [EMAIL PROTECTED] www.snom.com This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail könnte vertrauliche und/oder rechtlich geschützte Informationen enthalten. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail sind nicht gestattet. - -Original Message- From: Remco Barende [mailto:[EMAIL PROTECTED] Sent: Monday, January 02, 2006 2:29 PM To: Usman Tahir Cc: Asterisk Users List Subject: Re: [Asterisk-Users] snom Firmware 5.0. Thanks for the new firmware, finally some of the features are becoming available that make the phone more usable with Asterisk. One question though, ringer tone #2 on the Snom 360 firmware has been replaced? How can I get the old ringtone back? I was using the ringtone on phones in locations like meeting rooms. The ringtone wasn't intrusive at all, yet well audible. Now when a phone rings everybody is disturbed with a loud noise. Thanks! Remco On Thu, 22 Dec 2005, Usman Tahir wrote: Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, - Usman Tahir snom technology AG www.snom.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] Problem with octobri and x100p clone
On Thu, 22 Dec 2005, Tzafrir Cohen wrote: On Thu, Dec 22, 2005 at 11:06:50AM +0100, Agustin Gudiño wrote: the thing is: I first had problems loading the drivers and was the zaptel init file that I will post below What's wrong with a simple 'modprobe qozap; modprobe wcfxo; ztcfg' ? (With no sleeps). I need even longer sleeps (TDM11B only box), if I don't use the sleeps my RedHat Enterprise Linux 4.x rebuild will fail to load the second module after zaptel. I don't know why, it seems that the second module is very impatient and also that RHEL is slow setting up the proper udev stuff. ___ --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] Change time when * is running
On Fri, 9 Dec 2005, Tzafrir Cohen wrote: On Fri, Dec 09, 2005 at 02:02:53PM +, Julian Lyndon-Smith wrote: We've just seen that one of our servers is an hour out (it reckons that it's 15:02 instead of 14:02). Can I change the time when * is running ? I don't want to try just in case it causes * some grief. keep clocks in sync with ntp . Or set the system clock with time. I do run ntp on my servers, when ntp corrects the time I start getting error messages on the * console about music on hold events occurring in the past. It doesn't seem to cause any problems though so I just ignore 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
[Asterisk-Users] Asterisk 1.2.1 released
It seems that Asterisk 1.2.1 is on the Digium FTP, but no posts to the users lists, nothing in the wiki? Everybody still asleep? Looking forward to the changelogs :) ___ --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] ZapHFC cards not maintaining sync?!
Already HAVE Florz patch installed! :-( What version of * and BRIstuff are you using? Strange, sounds like the florz patch has not been effectively applied or it's broken. I'm using an old version of bristuff : Asterisk 1.0.9-BRIstuffed-0.2.0-RC8n built by [EMAIL PROTECTED] on a x86_64 running Linux I had an issue compiling it for x86_64 though, but that's a different question. I assumed as much when I saw your last name... :-) Whereabouts in NL? I'm in Zoetermeer (ZH)... Amsterdam, but the ISDN setup I installed near Leiden (ZH) :) 1) Every 10 seconds () the D channel gets torn down, which That's too slow, it should happen about every 1-2 seconds or so. The d channel going down and up again is normal behaviour. I know it is. Used to work for a Networking Competence Centre, and we had the same kind of issues with 3Com Netbuilders. The first call attempt after the D Channel was torn down always failed... The only solution was to get KPN to turn on the D Channel permanently... Strange, I never had that problem before. When the * box gets up I can immediately make calls. Also the standard KPN A/B equipment doesn't have this problem, sounds like it's more 3Com related. One problem I have found with bristuff (and no solution yet), if you disconnect the ISDN line from the * box (or the ISDN line is out of order for a short while), bristuff will not re-establish the connection. It is then required to unload all the modules and re-load them or even worse, reboot the box. I guess that is a specific bristuff problem. All calls to the ISDN line fail and it's not possible to make any calls. Even after several hours bristuff doesn't setup the line connection. 2) Results in the CRC error, which means that 3) Every 3 minutes, the D channel goes down for EXACTLY 1 minute. I could try to get the KPN to give me a permanent D channel, but are there any tricks to try that would/could make asterisk somehow keep up the D channel?... I noticed that the 'deactivated' issue doesn't happen for a while after a call has been placed. I am now testing placing a call every minute, with a 100 ms timeout using the manager api. This means it never actually gets a chance to get through, or be picked up, but it does cause activity on the D channel. This has been running for half an hour now, and I haven't seen the channel go down for extended periods since. I'm not sure whether the KPN will like it, but it's an interesting test to run! G Good luck with our Royal Dutch KPN, but I would try florz first :) Tell me about it! Like I said above, we had *extensive* experience with them over the D Channel issue! Weird, I checked with KPJ before and he mentioned it is normal behaviour for ISDN. My console is filled with messages like this : == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down and it doesn't cause me any issues. It would be nice to 'hide' these messages when not in very verbose mode to avoid cluttering up the console. The messages indeed do appear about every 10 seconds or so. ___ --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] Can't compile chan_zap.c on fresh cvs 1.20 checkout
On Sat, 3 Dec 2005, Andrew Kohlsmith wrote: On Saturday 03 December 2005 04:09, Remco Barendse wrote: chan_zap.c:9318: warning: implicit declaration of function `pri_get_debug' chan_zap.c:10602: error: `PRI_SWITCH_QSIG' undeclared (first use in this chan_zap.c:10870: warning: passing arg 1 of `pri_set_error' from chan_zap.c:10871: warning: passing arg 1 of `pri_set_message' from PRI is involved in every one of these messages... I'd start looking to see if you've got libpri installed, including the libpri headers. :-) Hmmm, guess you are right. But this is a home PBX, I'm never going to need a PRI here and in the past libpri was never a requirement or dependency for any asterisk installation. Has this now changed or do I (somewhere, someplace) have some stuff in a config file which make(s) :) asterisk believe it should do something with PRI stuff? 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] Can't compile chan_zap.c on fresh cvs 1.20 checkout
On Sat, 3 Dec 2005, Rich Adamson wrote: On Saturday 03 December 2005 04:09, Remco Barendse wrote: chan_zap.c:9318: warning: implicit declaration of function `pri_get_debug' chan_zap.c:10602: error: `PRI_SWITCH_QSIG' undeclared (first use in this chan_zap.c:10870: warning: passing arg 1 of `pri_set_error' from chan_zap.c:10871: warning: passing arg 1 of `pri_set_message' from PRI is involved in every one of these messages... I'd start looking to see if you've got libpri installed, including the libpri headers. :-) Hmmm, guess you are right. But this is a home PBX, I'm never going to need a PRI here and in the past libpri was never a requirement or dependency for any asterisk installation. Has this now changed or do I (somewhere, someplace) have some stuff in a config file which make(s) :) asterisk believe it should do something with PRI stuff? You might review each statement in your zapata.conf file to ensure those that are used pertain to whatever card/interface you are using. In past sample configs that folks have posted, some have included things like 'switchtype=national' for their x100p/tdm card, and that parameter (as one example only) is associated with PRI's. For the past two years, I've always compiled and installed zaptel, libpri, and asterisk at the same time. The libpri modules only get loaded _if_ a card is detected in the system that requires them. Certainly doesn't hurt anything. It worked. Strange, but true. I checked through all my configs and there is nothing there that even remotely hints to a PRI. Thanks for the solution :) ___ --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 1.2 and weird ZAP interface behaviour
I just upgraded my config from * 1.0.10 to 1.2 I removed caller ID from my configs because when I try to use CallerID (new style) on my IAX provider (magrathea) but whenever I try to make a call I get a message from the provider that You are not registered to use this service. Removing the callerid stuff seems to solve this. I guess they are not ready for the new updated IAX protocol? Anyways, now to my real problem. I have a TDM11B card. Obviously one connection to the phone line, one connection to an analog phone. I just used the exact same config files as with * 1.0.10 I have this in my /etc/asterisk/zapata.conf: callerid=202 signalling=fxo_ks group=1 context=intern-all channel = 1 Whenever I pick up that phone I get on the console: Dec 3 16:37:36 WARNING[19551]: pbx.c:2347 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' Okay, but I want to make an OUTGOING call so I don't need this phone to be in default context, do I?? When I add the default context (with s extension) to intern-all whenever I pick up the analog phone it starts ringing my default context like a bat phone. Nice but not what I wanted.. I just want it to give me a dial tone and wait for the number I want to dial. What am I overlooking here?? ___ --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 1.2 and weird ZAP interface behaviour
On Sat, 3 Dec 2005, Begumisa Gerald M wrote: On Sat, 3 Dec 2005, Remco Barende wrote: Whenever I pick up that phone I get on the console: Dec 3 16:37:36 WARNING[19551]: pbx.c:2347 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' Have you by chance set immediate to yes? IIRC, there's a feature that will send you to the configured context as soon as you pick up your phone (this is in zapata.conf). Might be worth checking that out. I have but only for the phone line, it is immediately after: signalling=fxs_ks immediate=yes I did some further testing, this happens only after I have done a RELOAD on the console. When I exit asterisk and start asterisk again all seems to be working as normal. Maybe it's an * bug? ___ --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] ZapHFC cards not maintaining sync?!
On Sat, 3 Dec 2005, Francesco Peeters (Asterisk) wrote: On Sat, December 3, 2005 16:40, Francesco Peeters (Asterisk) said: On Fri, December 2, 2005 22:54, Francesco Peeters said: Watching the console for a while I see regular messages, which I could also find in /var/log/messages: Dec 3 16:37:15 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: empty HDLC frame received. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. This is not normal. Run the florz patch over your bristuff install (I'm assuming you are using bristuff). These problems will cause your box to hang after anything beteen 5 and 48 hours. Can anyone with one or more HFC-PCI card(s) (esp. in The Netherlands) check if they see these on a regular basis as well? (And I am talking many times an hour here!) I am in NL :) 1) Every 10 seconds () the D channel gets torn down, which That's too slow, it should happen about every 1-2 seconds or so. The d channel going down and up again is normal behaviour. 2) Results in the CRC error, which means that 3) Every 3 minutes, the D channel goes down for EXACTLY 1 minute. I could try to get the KPN to give me a permanent D channel, but are there any tricks to try that would/could make asterisk somehow keep up the D channel?... Good luck with our Royal Dutch KPN, but I would try florz first :) ___ --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] Dutch callerid and x100p
I've been trying for a while now to get callerid working with an x100p on the Dutch KPN phone network, but been completely unsuccessful. The bounty on the wiki appears to be closed, and all the information I've found _implies_ that it works, but I just can't get it working, with either 1.0.9 or a recent CVS of * and zaptel. The various patches on the bug tracker seem to be completely inconsistent. Is there any definitive info on whether this does work, and how to get it going? It doesn't. The x100p is only capable of getting callerid according to the us system. Get a TDM11B and it works flawlessly. As a bonus you'll have less problems with echo and sound quality :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Softphone with Bluetooth support for *
Can anyone recommend a proper Softphone? I tried Firefly (the Virbiage stuff) but it's a pile of crap. It reports to have registered by displaying the number in the top left corner, when I dial something I hear a ringing sound but the thing hasn't even registered. And no place where you can see if and why registration failed. For the Snom thing I guess you need a license for it to work, how nice Finally DIAX wil not install, complains about MSCOMCTL.OCX missing Thanks for any hints/tips! On Sat, 19 Nov 2005, Dan wrote: Hi Actually, I was hoping not use any phone at all. My idea was to install a softphone on the laptop and use a standard BT headset without using any cell phone at all. This is the way I use it too..:-) My laptop doesn't have a decent quality speaker or mic, that's where the headset comes in. I can then use the BT headset to walk around the room and still make/receive VOIP calls from my regular number :) Yes, but for the moment you will not be able to answer or hangup a call from the BT headset. It will be possible in a future version. Bet regards, Dan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Softphone with Bluetooth support for *
I have seen some options for road warriors to connect a DECT phone to a SIP device or use a WIFI VOIP phone when travelling but I was wondering if it would be possible to use a soft phone and a standard bluetooth headset to connect to * Most newer laptops have bluetooth support built in and a bluetooth headset is lots cheaper than a WIFI VOIP phone, not to mention easier to carry! Has anyone ever tried such a setup? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Softphone with Bluetooth support for *
In order to clarify some afirmations : - You can control DIAX from a BT enabled phone (dial, display callerID, on/off hook), but you cannot use the phone for the audio part. For this a BT headset can be used in the same time. - DIAX cannot control (yet) the bluetooth headset for on/off hook signals. You must connect the BT headset manually before using DIAX. Hope that some of those limitations will be eliminated in a future version of DIAX. Thanks!! Actually, I was hoping not use any phone at all. My idea was to install a softphone on the laptop and use a standard BT headset without using any cell phone at all. My laptop doesn't have a decent quality speaker or mic, that's where the headset comes in. I can then use the BT headset to walk around the room and still make/receive VOIP calls from my regular number :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Opinions
I'm not too pleased with the phones, I have about 40 of them, some of the displays tend to die and the dial pad feels to 'mushy' IMHO, just like the keys on a good old ZX80 computer Also I'm having some issues with sound quality on some phones, but I still need to switch some phones to see if that is really an issue of the phone. Also if you want to use * call files, with the 360 you will run into a big where the call is being redialled as if it failed while in fact the call is ongoing. Annoying and haven't found out if that is an * bug or Snom bug. The Snom 190's do not have this problem. Just my $0.02 (which is really not a lot these days!) :) On Sat, 12 Nov 2005, Curren C. Calhoun wrote: I¹m looking to add in some Snom 360 phones, could anyone give thoughts or opinions about the speakerphone, general quality... Also the phone would need to be powered over Ethernet... I like some of the listed features and the expandability of the phone but am open to any other suggestions as well... Thanks Curren ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff-0.2.0-RC8n problems and kernel panic
It's a problem with bristuff that has been there for quite some time. If you load the modules in the wrong order it will kernel panic the box. I have been bitten by it many times, very frustrating if you are working on a remote box I manually load all modules now too On Tue, 8 Nov 2005, gincantalupo wrote: Hi, this is my /etc/modprobe.d/zaptel: options torisa base=0xd alias char-major-196 torisa install tor2 /sbin/modprobe --ignore-install tor2 /sbin/ztcfg install torisa /sbin/modprobe --ignore-install torisa /sbin/ztcfg install wcusb /sbin/modprobe --ignore-install wcusb /sbin/ztcfg install wcfxo /sbin/modprobe --ignore-install wcfxo /sbin/ztcfg install wcfxs /sbin/modprobe --ignore-install wcfxs /sbin/ztcfg install ztdynamic /sbin/modprobe --ignore-install ztdynamic /sbin/ztcfg install ztd-eth /sbin/modprobe --ignore-install ztd-eth /sbin/ztcfg install wct1xxp /sbin/modprobe --ignore-install wct1xxp /sbin/ztcfg install wct4xxp /sbin/modprobe --ignore-install wct4xxp /sbin/ztcfg install wcte11xp /sbin/modprobe --ignore-install wcte11xp /sbin/ztcfg alias wctdm wcfxs and this is my /etc/init.d/asterisk made by me: #!/bin/sh ztcfg -s # unload wcfxs module because I must load # qozap module first! /sbin/rmmod wcfxs /sbin/rmmod zaptel # Now I load all the modules in the right order /sbin/insmod /lib/modules/2.6.8-2-386/misc/zaptel.ko /sbin/insmod /lib/modules/2.6.8-2-386/misc/qozap.ko /sbin/insmod /lib/modules/2.6.8-2-386/misc/wcfxs.ko ztcfg -vv # this is to exec asterisk as asterisk user chown --recursive asterisk:asterisk /dev/zap chmod --recursive u=rwx,g=rx /dev/zap chown asterisk /dev/tty9 sudo -u asterisk /usr/sbin/safe_asterisk and it magically works (!!!) even if modifying debian zaptel and wcfxs modules loading sequence should be a better way to solve the problem but I don't know where to find that damned sequence. Giorgio Incantalupo This Tzafrir Cohen wrote: On Mon, Nov 07, 2005 at 03:43:03PM +0100, gincantalupo wrote: Hi, I had some problems to with a quadBRI with a 2.6 kernel debian distro. Have you tried to insmod the zaptel.ko module instead of modprobing? It worked for me, hope it will work for you too. Giorgio Incantalupo Could you please give more details? One thing you should try to do is remove the automatic run of ztcfg at module load time. Practically: rem-out all the lines in /etc/modprobe.d/zaptel . There is some black-magic claim that if you un ztcfg more than once it may cause a problem to a configured zaphfc module. Don't forget to run ztcfg manually (or in an init.d script) later. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk dropping call file without *any* notice
I'm trying to debug the old call file redial bug I prepared a call file and trying to setup a call from my remote asterisk server to my home number. However whenever I dump a call file to /var/spool/asterisk/outgoing it is just deleted without *any* action Nothing in the logs, nothing on the console... GRRR I already increased the verbosity and enabled sip debug... nothing It would be nice if * would at least report that it found a call file and ditched it for whatever reason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.8 and TDM stop acking inbound calls?
On Tue, 4 Oct 2005, Patrick Friedel wrote: Cirelle Enterprises wrote: we also experienced this with asterisk 1.0.9 and rev H of the tdm with 4 fxo modules we were restarting asterisk every night via cron and this still happened in our case, 3 out of 4 fxo modules (2,3,4) crapped out and stopped ack'ing incoming calls (outgoing calls were fine) it took a reboot of the server to get the card operational again and answering calls As a certain Zippy would say: Yow! I assume you reached that point because unloading and reloading the wctdm modules didn't do anything? Do the digital interfaces have these sorts of problems? Is there an alternate FXO solution? I've heard nothing but trouble with the TDM, and I know that's probably because the 99% of satisfied users are generally quiet but still... I have the same problem, after about a month the card doesn't report any incoming calls anymore to the console. Don't know the rev of my card yet, unloading asterisk and unloading the modules and then restarting everything does seem to help though, no need to reboot. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] kill a .call file
On Mon, 19 Sep 2005, jltaylor wrote: From my CLI: Attempting call on SIP/gw/19857749166 for [EMAIL PROTECTED]:1 (Retry 114) Attempting call on SIP/gw/19037747603 for [EMAIL PROTECTED]:1 (Retry 83) Attempting call on SIP/gw/19857747603 for [EMAIL PROTECTED]:1 (Retry 80) I want to stop it from any future attempts. Any idea about a command to kill or where the data is stored? This is an asterisk bug. I already filed it but they need a full trace and I haven't had the time yet to do it. It seems that * keeps retrying the call, even when it was succesfully completed. Annoying. James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of trixter http://www.0xdecafbad.com Sent: Monday, September 19, 2005 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] kill a .call file On Mon, 2005-09-19 at 13:43 -0500, jltaylor wrote: Any means of killing a .call file that is in progress? You mean once the call has begun? You prolly want to hangup the call ... asterisk -rx soft hangup callid Or is there something else that you wanted? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] kill a .call file
A workaround is to disable call waiting on your phones (nasty workaround tho) On Mon, 19 Sep 2005, Rene Kluwen wrote: I have the same problem. Asterisk always makes two calls, even when the first one went through successfully. Rene Kluwen Chimit -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Remco Barende Sent: maandag 19 september 2005 22:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] kill a .call file On Mon, 19 Sep 2005, jltaylor wrote: From my CLI: Attempting call on SIP/gw/19857749166 for [EMAIL PROTECTED]:1 (Retry 114) Attempting call on SIP/gw/19037747603 for [EMAIL PROTECTED]:1 (Retry 83) Attempting call on SIP/gw/19857747603 for [EMAIL PROTECTED]:1 (Retry 80) I want to stop it from any future attempts. Any idea about a command to kill or where the data is stored? This is an asterisk bug. I already filed it but they need a full trace and I haven't had the time yet to do it. It seems that * keeps retrying the call, even when it was succesfully completed. Annoying. James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of trixter http://www.0xdecafbad.com Sent: Monday, September 19, 2005 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] kill a .call file On Mon, 2005-09-19 at 13:43 -0500, jltaylor wrote: Any means of killing a .call file that is in progress? You mean once the call has begun? You prolly want to hangup the call ... asterisk -rx soft hangup callid Or is there something else that you wanted? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limiting call minutes on a GSM SIM
On Tue, 13 Sep 2005, trixter http://www.0xdecafbad.com wrote: On Wed, 2005-09-14 at 07:01 +0200, Remco Barende wrote: Hi! I'm considering to buy a GSM bridge to save on GSM calls. Right now they are offering subscriptions with 200 minutes each month for almost nothing, however the 400 minutes subscriptions are considerably more expensive. Most GSM bridges can cater for 2 SIM cards, is there a way for Asterisk to run the first SIM card to it's max and then switch to the second? (If one call would overlap I wouldn't mind). Asterisk would have to keep track of the minutes called each month for a SIM (channel?). On most bridges you can select the SIM you want by a dial prefix. I do not know about the specifics, but it seems to me that you would need an AGI that would track the usage and compare that before placing a call. To switch I do not know how you tell the sim adapter which one to use, but surely there must be a command somewhere, the mere fact that agi allows you to script something like this fairly easily means that it shouldnt be a big problem, assuming you code :) And you can even pick your favourite language given how the AGI talks to asterisk even 'unsupported' languages can be used. Thanks for the tip. I was actually thinking in the direction of putting the asterisk calling card application to use. I've never used it and wonder if it is at all possible to use it from within the dial plan instead of normally from an extension. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Limiting call minutes on a GSM SIM
Hi! I'm considering to buy a GSM bridge to save on GSM calls. Right now they are offering subscriptions with 200 minutes each month for almost nothing, however the 400 minutes subscriptions are considerably more expensive. Most GSM bridges can cater for 2 SIM cards, is there a way for Asterisk to run the first SIM card to it's max and then switch to the second? (If one call would overlap I wouldn't mind). Asterisk would have to keep track of the minutes called each month for a SIM (channel?). On most bridges you can select the SIM you want by a dial prefix. Thanx! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Motherboard and processor recommendations
On Fri, 9 Sep 2005, Domjan Attila wrote: Hi, I sucked the TE410 in a Siemens dual Xeon machine... lot of irq problems, digium support said: try the card in another machine. A cheap amd64 + via K8T800 and TE405 works perfectly... I'm looking for a good, reliable and upgradeable solution too. I don't care to spend a lot of money if the hardware is reusable. A Dell 2850 is useless after 3 years, no way to upgrade it. A quality Intel SC5300 for example is not cheap at all but will last you a lifetime. But now the difficult task, to find the right mobo. I prefer to go with an AMD CPU because it is not as power hungry as Intel which only improves runtime on ups power. In the category professional mobo's I like the HDAMA from Rioworks which features an AMD 8111 [HyperTransport I/O Hub] and AMD 8131. [PCI-X Tunnel]. Has anyone ever tried Digium hardware on this chipset or even mobo? Alternatively I'm considering a regular mobo with an nForce3 or nForce4 chipset. Will these work with Digium hardware? (Looking at the repsonses most people seem to use Intel stuff?) Recommendations for quality barebones also welcome :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk overheating on VIA Epia MSeriesmoth erboard
Just out of interest, how do you run the VIA boards? They only have one network connection and if you add a PRI card you cannot have both a LAN and NET connection? (Highly offtopic, sorry!) On Tue, 6 Sep 2005, Nathan C. Smith wrote: Is that a bios setting (I don't recall seeing it) or an OS setting? I run a lot of Via C3 machines (they are so nifty) but don't remember seeing this. -Original Message- From: Technical Support [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 06, 2005 10:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk overheating on VIA Epia MSeriesmotherboard You can dramatically reduce the heat from your EPIA board by turning on CPU scaling! Once we turned it on, the heatsink was cool to the touch. (Even with asterisk running). MD ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
My experience with auto-rebooting schemes is that reliability doesn't improve. I also reported the non-registration of the Snom phones as a bug. On a few occasions I found that the phone lost registration, and rebooting or power cycling the phone didn't help (athorization failed at the * console) but re-entering the password in the phone did. Would this match your problem too? I guess it would be nice if we could make * log each password with which a SIP client tries to register (yes I am aware of the security implications of it). My guess is that the password disappears from the phone or it is corrupted. On Fri, 2 Sep 2005, altus wrote: SO if I do a reboot of the system each night at 12,it should be up and working again at 8 in the morning? On Thu, 2005-09-01 at 08:49 -0500, Jeff Brownlee wrote: IS there a way to make the phone reboot each day at a time? You could do it via a cron job by wget'ting the reboot uri (on the advanced page again), but there really shouldn't be any need to do so. The only time subscriptions should disappear is when you do a reload or restart on asterisk. Even after a reload or restart the subscriptions will come back, but it usually takes ~30 minutes or so depending on when the last subscriptions were sent. -Jeff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax trouble with HP 3330mfp (again)
I'm using Asterisk 1.09 with bristuff 0.2.0-RC8n, one BRI line and a
Sipura SPA-2000. I have a HP LaserJet 3330mfp all-in-one. I can receive
faxes but not send them. The faxes start whistling to each other but the
transmission is stopped with a communication error
To receive a fax I have this in my dialplan:
exten = 00,1,Ringing
exten = 00,2,zapEC(off) ; disable EC on the incoming channel
exten = 00,3,SetCIDNum(${PRI_NETWORK_CID})
exten = 00,4,LookupCIDName
exten = 00,5,Dial(${FAX})
To send a fax I use these options:
exten = _9.,1,Dial(ZAP/g1md/${EXTEN:1},70,rdT)
exten = _9.,2,Macro(fastbusy)
I'm puzzled why I can receive faxes but not send them. The HP is capable
of 14k4 transmission speeds (and I think even higher) but why wouldn't
that be a show stopper to receive faxes?
In the firmware of the HP I cannot discover an option to limit the tx/rx
speed.
Any hints on what I am doing wrong greatly appreciated. Is there another
way to force the faxes to try 9k6 as the max speed or does my dialplan
have mistakes?
Thanks!!
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[Asterisk-Users] Call file always redials (grrrrr)
Hi list! Our CRM app is creating call files for outgoing calls which is working great I just have one problem. I am using this as my call file: Channel: SIP/228(my phone) MaxRetries: 0 Context: from-internal (the context to dial from) Extension: 003120531234 (the phone number) Priority: 1 Callerid: Myfinecustomer 003120531234 so the external number is connected to my sip phone. However after speaking for approx 5 minuted, Asterisk always does a retry and I see the external number in my display on the second line. It does this on every call. When I'm finished I also see 2 records in the log files. Any idea why Asterisk is trying to place the call again even though the first attempt was succesful and the call is still in progress? I didn't specify a redial anywhere. I'm running the latest cvs stable (of this morning), Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI voice one way only
The easiest thing is probably to get a card that is more widely supported. Any cheap pci HFC-S card will do, they are sold for anthything between 9 and 15 eur. With an hfc-s card you can then use bristuff or chan_capi On Sun, 21 Aug 2005, Klemens Kasemaa wrote: hi PSTN -- [Teles ISDN / Asterisk] -- SIP client When call is made through ISDN, no matter if taken from PSTN or Asterisk side, person in PSTN side can hear perfectly but in Asterisk side I only hear a very scrambled or very low quality voice, words repeated several times. Same is with echo test (call taken from PSTN) Get a CAPI module for your Teles and try chan_capi-cm from http://sourceforge.net/projects/chan-capi/ Accordingly capi.org this card does have capi support but not under linux. Because of it's a ISA card, I can't use zaphfc also. So any help is appreciated. with rgrds, klem ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PrivacyManager not working Asterisk 1.0.9-BRIstuffed-0.2.0-RC8n
Hi list!
I'm trying to get PrivacyManager working but for some reason it always
thinks that CallerID is present (when it isn't). I get this on the
console:
== Primary D-Channel on span 1 up
-- Accepting voice call from '' to '0711234567' on channel 0/2, span 1
-- Executing Ringing(Zap/2-1, ) in new stack
-- Executing Zapateller(Zap/2-1, answer|nocallerid) in new stack
-- Executing PrivacyManager(Zap/2-1, ) in new stack
-- CallerID Present: Skipping
-- Executing SetCIDNum(Zap/2-1, ) in new stack
-- Executing LookupCIDName(Zap/2-1, ) in new stack
I have this in my extensions.conf:
exten = 0711234567,1,Ringing
exten = 0711234567,2,Zapateller(answer|nocallerid)
exten = 0711234567,3,PrivacyManager
exten = 0711234567,4,SetCIDNum(${PRI_NETWORK_CID})
exten = 0711234567,5,LookupCIDName
exten = 0711234567,6,Dial(${EVERYONE},45,t)
exten = 0711234567,7,Answer
I need SetCIDNum(${PRI_NETWORK_CID}) to get CallerID working but I guess
it needn't be before Zapateller or PrivacyManager to trigger the
PrivacyManager.
Other question, would it be possible to use my own (custom recorded)
message instead of a standard prompt?
Thanks!!
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Re: [Asterisk-Users] DECT gateways
Hi! I have the Kirk IP600, technically it's great and *the* best solution for wireless phones and VOIP. WiFi phones plainly suck for a dozen of reasons and IMHO are totally useless. However, here comes the bad part. Currently you need to use the chan_sccp driver to talk to it and that is not the most stable driver. I'm still battling to get it to work properly using 4 phones. Kirk (sold as Tiptel in NL) is rumoured to support standard SIP, the firmware was due to be released right about now. Haven't heard anything from Kirk support on this however and the last thing I heard was that they still had to start working on it in July. But that's just rumours. Cheers! Remco On Wed, 17 Aug 2005, Michiel van Baak wrote: Heya list, I need some advice/experience. Some of our customers are asking us about DECT solutions for their asterisk install. Some others will not go to asterisk if there won't be a DECT solution. They now have a Siemens or a Samsung PBX. Those PBX-es come with a DECT basestation and optionally repeaters etc. All those basestations speak some own protocol to the PBX, so we cannot use them with asterisk. I been looking around on the internet and found the Kirk gear. Anyone has any experience with them ? The website states they are recognized as Cisco 7970 in CCM. Does chan-sccp handel those Kirk emulated devices ? Is there any other solution like this out there that works with asterisk ? Thanks for your input, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DECT gateways
Hi Michiel, I have a Kirk set which should be able to do H323, but I haven't had time yet to try it. They have SCCP and H323 types, and ofcourse there are sets which can be connected via an E1 link. I tried H323, it has some severe problems (at least with asterisk). First of all it is complicated, you need a gatekeeper too and even worse, it is impossible to transfer a call. Maybe acceptable in a home environment but certainly not anywhere were you need to hold/transfer a call I should still have my config files somewhere if interested mail me offlist BTW every Kirk set can do H323, for Skinny (Cisco) you need an extra license (just a pincode). The Cisco version is just as expensive and can do both. If you buy a new one, get the Cisco version :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DECT gateways
On Thu, 18 Aug 2005, Andreas Sikkema wrote: [EMAIL PROTECTED] wrote: Nice looking device. Does it support DECT repeaters? I cannot rely on 1 basestation for my handset when I walk around in the location. The Siemens stuff has 10 antennas/repeaters/extenders in the building. It's a bit overkill, but the Siemens guys tend to love doing it BIG. I think 3 to 6 repeaters will be way enough for the cases we have open now. Buildings do strange things with radio. What might seem like overkill could be mandated by liftcages, firestairs etc. Also if you want to add dect coverage in a busy area where lots of people with dect handsets gather (meeting rooms, canteens) you need lots of basestations. I've worked in a building where there seemed to be an overkill of basestations every hallway had 3 or 4, (every 20 meters or so) and still there were areas with insufficient coverage... In such areas where you would also need voip the Kirk IP1500 DECT could be a solution. Unfortunately the same driver trouble applies for the IP1500 as for the IP600. The firmwares of the IP600 and the IP1500 do look very similar however maybe Kirk will support SIP in the 1500 too. Cheers! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] florz patch for bristuff breaks compile on x86_64?
I found it!! (Heh, probably to easy to mention for most of you but being unable to read any code at all I'm quite happy) :) After applying florz patch it seems that the Makefile is ignoring the linux26 option. The Makefile should change the $KSRC to the proper directory i.e. /usr/src/linux-2.6 but it doesn't and keeps looking for /usr/src/linux. The error message is correct however. I'm not sure who should fix this, florz or kapejod. Anyways, the 'fix' is to change the first line of the makefile to read like this: KSRC=/usr/src/linux-2.6/ Cheers! Remco On Wed, 17 Aug 2005, Remco Barende wrote: On Wed, 17 Aug 2005, Tzafrir Cohen wrote: On Wed, Aug 17, 2005 at 06:57:19AM +0200, Remco Barende wrote: After upgrading a CentOS 3.x box to CentOS 4.1 (both x86_64 with an Athlon64) I also wanted to get the latest bristuff. Unfortunately bristuff without florz causes the box to kernel panic within hours (console will complain about bad frame received something). Then merge the fix into the bristuff patch if it has not been merged yet! That's what I did when I patched bristuff :) It seems however that the florz patch will not work for x86_64 arch. Bristuff -0.2.0-RC8j compiles fine without the florz patch, but after applying the patch zaphfc will not compile anymore (the patch applies cleanly). Latest bristuff is RC8n, BTW. What exactly is the florz patch? It seems to have been onchanged since January or so. I have never ever been able to keep a bristuffed box up for more than a few hours or 2 days at best without the florz patch. It seems that KPJ is trying various approaches to solve timing problems but I guess it's not stable yet. Florz fixes a lot of timing issues, reduces interrupt load and makes bristuff stable. You can find more info here: http://zaphfc.florz.dyndns.org/ Anyone managed to get bristuff with florz working on x86_64 arch? It is part of the debian packages and they are built on amd64 as well. http://packages.debian.org/zaptel http://packages.debian.org/libpri http://packages.debian.org/asterisk I would guess thet are without bristuff and/or florz? Bristuff compiles without florz, but zaphfc doesn't after applying florz. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Limit fax tx speed of 'dumb' faxes??
Hi! Still trying to get fax to work in a reliable manner. I have some dumb fax machines that do not have a setting to limit tx/rx speeds. For some reason when training they often think the line is good enough for 14k4 or higher after which the fax will fail. Is there any other way of forcing the faxes to lower their speeds? Would it be possible to monitor certain extensions for the sound pattern a fax machine generates during training to try 14k4 or higher and maybe jam the line or mute the audio for a moment? That way the fax will think 14k4 failed and would switch to 9k6. I'm looking to connect some fax servers (for various reasons spandsp is not an option) for incoming / outgoing faxes but even on recent modems like the MultiTech MT5634ZPX (PCI) it's not possible to cap the speed AFAIK. Any hints greatly appreciated! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DECT gateways
On Thu, 18 Aug 2005, Michiel van Baak wrote: On 08:48, Thu 18 Aug 05, Remco Barende wrote: Hi! I have the Kirk IP600, technically it's great and *the* best solution for wireless phones and VOIP. WiFi phones plainly suck for a dozen of reasons and IMHO are totally useless. However, here comes the bad part. Currently you need to use the chan_sccp driver to talk to it and that is not the most stable driver. I'm still battling to get it to work properly using 4 phones. Kirk (sold as Tiptel in NL) is rumoured to support standard SIP, the firmware was due to be released right about now. Haven't heard anything from Kirk support on this however and the last thing I heard was that they still had to start working on it in July. But that's just rumours. Cheers! Remco Hi, Do you have trouble to get the Kirk phones working or just any SCCP device ? I use it with 7905's and it is rockstable here. WAY better then the SIP firmware/chan_sip combi (loosing registration, bad audio, sucky monitor). I don't know about other Cisco stuff, I don't like Cisco 'support' and their licensing crap, I'm not buying it :) The problems I had (have): - if one phone is talking and a new call is coming in audio is muted on the first call (I'm now trying Sergio's version to see if it's better) - * segfaults when you pressed the on hook key on a handset but changed your mind and hung up again without pressing any digit (but again Sergio's driver may change that - i have 4 phones registered to it, some phones do not ring, sometimes they do, sometimes they don't (even with Sergios sccp) - caller id is only displayed on the phones for a split second (which may be a Kirk fw bug) The best version (for me) of chan_sccp so far is Sergio's version chan_sccp-20050715 Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaphfc.ko module error
Hi! You didn't state what distro you are running but my guess is that you have autoupdate / up2date running. Before the powerfailure the kernel was updated and after the powerfailure the box booted the new kernel for which you need to recompile the module. Cheers! Remco On Thu, 18 Aug 2005, Terry Wade wrote: Hi Guys I have been running a test server for a few days now with * 1.0.9 bristuff RC8n. I had a power failure and the test machine was not on the ups. When power was restored I found the following error: FATAL: Error inserting zaphfc (/lib/modules/2.6.11.4-20a-default/misc/zaphfc.ko): Unknown symbol in module, or unknown parameter (see dmesg) My dmesg output: zaphfc: unsupported module, tainting kernel. ^^ that makes me believe you are now running a newer kernel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] florz patch for bristuff breaks compile on x86_64?
On Wed, 17 Aug 2005, Tzafrir Cohen wrote: On Wed, Aug 17, 2005 at 06:57:19AM +0200, Remco Barende wrote: After upgrading a CentOS 3.x box to CentOS 4.1 (both x86_64 with an Athlon64) I also wanted to get the latest bristuff. Unfortunately bristuff without florz causes the box to kernel panic within hours (console will complain about bad frame received something). Then merge the fix into the bristuff patch if it has not been merged yet! That's what I did when I patched bristuff :) It seems however that the florz patch will not work for x86_64 arch. Bristuff -0.2.0-RC8j compiles fine without the florz patch, but after applying the patch zaphfc will not compile anymore (the patch applies cleanly). Latest bristuff is RC8n, BTW. What exactly is the florz patch? It seems to have been onchanged since January or so. I have never ever been able to keep a bristuffed box up for more than a few hours or 2 days at best without the florz patch. It seems that KPJ is trying various approaches to solve timing problems but I guess it's not stable yet. Florz fixes a lot of timing issues, reduces interrupt load and makes bristuff stable. You can find more info here: http://zaphfc.florz.dyndns.org/ Anyone managed to get bristuff with florz working on x86_64 arch? It is part of the debian packages and they are built on amd64 as well. http://packages.debian.org/zaptel http://packages.debian.org/libpri http://packages.debian.org/asterisk I would guess thet are without bristuff and/or florz? Bristuff compiles without florz, but zaphfc doesn't after applying florz. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] florz patch for bristuff breaks compile on x86_64?
After upgrading a CentOS 3.x box to CentOS 4.1 (both x86_64 with an Athlon64) I also wanted to get the latest bristuff. Unfortunately bristuff without florz causes the box to kernel panic within hours (console will complain about bad frame received something). It seems however that the florz patch will not work for x86_64 arch. Bristuff -0.2.0-RC8j compiles fine without the florz patch, but after applying the patch zaphfc will not compile anymore (the patch applies cleanly). Anyone managed to get bristuff with florz working on x86_64 arch? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: florz patch for bristuff breaks compile on x86_64?
On Wed, 17 Aug 2005, Remco Barende wrote: After upgrading a CentOS 3.x box to CentOS 4.1 (both x86_64 with an Athlon64) I also wanted to get the latest bristuff. Unfortunately bristuff without florz causes the box to kernel panic within hours (console will complain about bad frame received something). It seems however that the florz patch will not work for x86_64 arch. Bristuff -0.2.0-RC8j compiles fine without the florz patch, but after applying the patch zaphfc will not compile anymore (the patch applies cleanly). Anyone managed to get bristuff with florz working on x86_64 arch? Thanks! Sorry for replying to my own message, I forgot to include the error: rm -f zaphfc.o *.ko *.mod.c *.mod.o .*o.cmd *~ rm -rf .tmp_versions Link /usr/src/linux-2.6 to your kernel sources first! make: *** [linux26] Error 1 install -D -m 644 zaphfc.ko /lib/modules/`uname -r`/misc/zaphfc.ko install: cannot stat `zaphfc.ko': No such file or directory make: *** [installlinux26] Error 1 hfc-pci driver installed. Press Enter to continue, or CTRL + C to abort. All other packages from bristuff compile fine after florz, just not zaphfc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to remove standard ISDN drivers from RedHat
I have newly installed a RedHat 4.0 EL rebuild. The install was done without the ISDN card present. After disabling kudzu and haldaemon I inserted the card. Stil that *($^%$($^!! kudzu shit modified my config and is loading hisax, crc_ccit and isn modules. Even worse, they do not appear in /etc/modprobe.conf which means that that f*cking kudzu added the modules to initrd. I have googled for hours and browsed through all the redhat docs but I cannot find how to remove these modules. All the docs mention is to 'simply comment them out from modprobe.conf' well, they aren't there. Does anybody know how I can remove these modules? They are really a pain in the ass because now I cannot start asterisk from the init scripts only from rc.local Thanx a 1,000,000 :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bristuff-0.2.0-RC8n problems and kernel panic
Hi list! On a newly installed RHEL 4 box I'm trying to install bristuff-0.2.0-RC8n. Everything did compile but I am running into some problems with the zaphfc driver. First of all when I load zaphfc *before* zaptel (yes I know I shouldn't do that) I get a kernel panic and the box hangs. Not so nice, especially when you are trying to fix stuff from remote locations. But ok. Now for the real trouble, when I do make load in zaphfc I get this: make -C /usr/src/linux-2.6 SUBDIRS=/tmp/bristuff-0.2.0-RC8n/zaphfc ZAP=-I/tmp/bristuff-0.2.0-RC8n/zaptel-1.0.9 modules make[1]: Entering directory `/usr/src/kernels/2.6.9-11.EL-x86_64' Building modules, stage 2. MODPOST *** Warning: zt_register [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! *** Warning: zt_receive [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! *** Warning: zt_transmit [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! *** Warning: zt_ec_chunk [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! *** Warning: zt_unregister [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! make[1]: Leaving directory `/usr/src/kernels/2.6.9-11.EL-x86_64' modprobe zaptel insmod ./zaphfc.ko ztcfg -v Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) 3 channels configured. Notice: Configuration file is /etc/zaptel.conf line 8: Unable to open master device '/dev/zap/ctl' I guess (hope) the warnings are nothing serious but the message about /dev/zap/ctl is. (I did read README.udev and added the lines.) Rebooting the box didn't help. And when I try to start asterisk: Aug 15 23:25:51 WARNING[6454]: chan_zap.c:933 zt_open: Unable to specify channel 1: No such device or address Aug 15 23:25:51 ERROR[6454]: chan_zap.c:6484 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Aug 15 23:25:51 ERROR[6454]: chan_zap.c:10329 setup_zap: Unable to register channel '1-2' Aug 15 23:25:51 WARNING[6454]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Aug 15 23:25:51 WARNING[6454]: loader.c:440 load_modules: Loading module chan_zap.so failed! Ideas anyone? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.voip-info.org/ front page taken out by spammer
Today the front page of http://www.voip-info.org/ was taken out by a spammer. It also seem the history page for http://www.voip-info.org/ was also nuked. I've restored the best I could using google cache, but still missing some information. Who is an admin on http://www.voip-info.org/ and can fix it? Google cache is a hard way to fix wiki-busting -- the easiest way is to click on history at the top of the page, go right to the version before the spam, copy it, then paste it into an edit of the page.. Of course, now, it's harder, because since the page was restored, people have since modified it.. (also, for some reason, when I click on history, nothing seems to happen).. Highly offtopic but weird that the wiki software doesn't have an option to undo all the changes that one user or one ip address made. Wouldn't be too hard to implement IMHO, just keep a copy of all the stuff that was changed / deleted / added for an x number of days and build an option to automagically undo the changes. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and firmware 4.0 problem
Does this explain why they would not register, or do I have to worry that there is some new setting which caused the problem? I do not want to go through the pain of upgrading the customer's phones (probably with a site visit) only to find that I have to downgrade them and go though it again with 4.1. I guess you haven't read my earlier post to your question, it would have answered your question Also, is there any way to tell the phone, *before a reboot* that I want it to update the firmware? I do most of my maintenance remotely, and I can tell the phones where to find new firmware and clicking Load will start the reboot process. However, I need a person there to press the Check button so that it will really update the firmware. Is there any way around this so that I can update the phone after-hours and remotely? Thank you... Not just for this answer, but for all the answers I get from If you would have read the manual of the phone or took 5 seconds to browse through the html interface of the phone itself you would have found the parameter to get the behaviour you want of the phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and firmware 4.0 problem
Did you install the license file as per the instructions? Starting version 4.0 Snom thought it would be nice to be a real pain in the ass to people with a lot of Snom phones. You need to go to a webpage and enter the mac address of the phone and they will send you a license file. Only after you install the license file the phone will make a SIP connection. How nice if you have approx 40 phones that need upgrading :( Snom never replied to my complaint about the license thing, i have 40 users that are unable to do that themselves unless I take 3 hours per user to explain it and I can do it quicker myself.. I have the same problems too btw, low memory errors and broken displays on some phones (one or two lines are completely dead). I still have to RMA the phones. On Fri, 5 Aug 2005, Michael George wrote: I have a pair of snom 360s at a customer and they were giving me Low Memory errors. The distributor suggested updating the firmware. I did that, to the one just below 4.0 (which wasn't released yet). One of the phones is still giving the Low Memory error every 3-4 days. The other one had a broken display that was just RMA'd, so it' hasn't been up long enough to know if the error occurs on that one, too. The distributor's latest suggestion was to go to the newest firmware, 4.0. I did that on the new 360 (from the RMA) and with the same account settings as the one it was replacing, it could not register with *. Since I was in a pinch, I updated the firmware down to the latest below 4.0 and the phone works just fine. Does anyone with more knowledge than I know what might be going on? Maybe a new default setting in 4.0 that's breaking things? Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wengo config and G729(a)
Hi list! Again Wengo has made changes to their servers that require modifications to * configs. Is there anyone that has the 'new' wengo working with asterisk that could post their configs? Also they switched codecs, now G720a is required to connect. I can only find an (open) G729 codec, is this the same as G729a? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wengo config and G729(a)
One way would do for me, I only use wengo for my outbound calls since they are a lot cheaper than our Royal Dutch KPN :) Which codec did you use and could you post your config lines? Thanks!! Remco On Mon, 25 Jul 2005, Wilson Pickett wrote: Also they switched codecs, now G720a is required to connect. I can only find an (open) G729 codec, is this the same as G729a? I only have it working one-way, no incoming calls. Ironically, when Mark was here we caould have gone to meet them and straighten it out once and for all :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sccp new realease
Hurray! Thanks for this new release! I've been plagued by segfaults for a long time, hope this is now solved. Does this version of chan_sccp replace the version at sourceforge or is this Yet Another Fork(tm) :) Thanks!! On Wed, 6 Jul 2005, Sergio Chersovani wrote: http://chan-sccp.berlios.de/ 20050705 ftp://ftp.berlios.de/pub/chan-sccp/chan_sccp-20050705.tar.gz - Added support for distinctive rings on stable: SetVar(ALERT_INFO=inside) or outside or feature on head: SetVar(_ALERT_INFO=inside) or outside or feature - Added support for native transfer incoming call-answer- hit transfer (incoming call is now on hold and marked as a transfer) - dial a new number - hit transfer (you can wait to talk to the user (consultative transfer) or just hit transfer (blind transfer) - fixed a segmentation fault when dialing a not configured line (Thanks Mark for the report) - fixed switching lines softkey state, hold/resume issues (thanks to Stefan and Joseph) - fixed segmentation faults on hangups For testers: http://www.voip-info.org/wiki-Asterisk+debugging run asterisk -vvvcg I need the bt full log Sergio Chersovani ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT : Wengo sucks
Would just like to warn everybody for Wengo.fr Once you sign up there is no possibility to remove your credit card and even though you send them resignation letters they keep charging your credit card. Now I understand what they mean when they say `unlimited subscription'. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Proper way to start * and load modules on a RedHat box
Hi list! I have several boxes running asterisk as I want, no problems there but the one thing I haven't sorted out is how to properly start asterisk on boot time. This is probably a n00b class question but how do I properly set this up (I didn't find any docs on this). The zaptel script alone does not load the proper driver on boot time, I guess I need to do some thing with the alias stuff in modules.conf? Also how can I make the startup scripts appear in ntsysv? Even when I copy the scripts to rc.d they do not show up in ntsysv I tried loading the modules manually from rc.local but that doesn't work, even if I use delays. For some reason ztcfg doesn't work when run from rc.local and therefore asterisk fails to load. If I run ztcfg manually then ztcfg starts properly. Thanks for any hints / tips! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Proper way to start * and load modules on a RedHatbox
That would work if for some reason ztcfg didn't refuse to run. I get the proper output of ztcfg on the console but at runlevel it seems as if ztcfg has never been run On Mon, 4 Jul 2005, Jimmy Smith wrote: how about /etc/rc.local #a line that would work path/to/screen -d -m path/to/asterisk -vvgfc -d -m Start screen in detached mode. This creates a new session but doesn't attach to it. This is useful for system startup scripts. On 7/4/05, Carlos Alperin [EMAIL PROTECTED] wrote: Did you check the log files looking for load errors? Carlos Alperin Senior System Engineer Seneca Communications, LLC [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: Monday, July 04, 2005 3:06 PM To: Asterisk Users List Subject: [Asterisk-Users] Proper way to start * and load modules on a RedHatbox Hi list! I have several boxes running asterisk as I want, no problems there but the one thing I haven't sorted out is how to properly start asterisk on boot time. This is probably a n00b class question but how do I properly set this up (I didn't find any docs on this). The zaptel script alone does not load the proper driver on boot time, I guess I need to do some thing with the alias stuff in modules.conf? Also how can I make the startup scripts appear in ntsysv? Even when I copy the scripts to rc.d they do not show up in ntsysv I tried loading the modules manually from rc.local but that doesn't work, even if I use delays. For some reason ztcfg doesn't work when run from rc.local and therefore asterisk fails to load. If I run ztcfg manually then ztcfg starts properly. Thanks for any hints / tips! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using 2 x DSL
How about using SER to balance the load for outgoing calls? On Thu, 23 Jun 2005, jltaylor wrote: You can't really do true bonding unless you control both ends of the link. I had a customer who tried this. It's easy to do with ATM and IMA interfaces on T1/T3 type stuff. The $300-$1000 dual wan routers will not work off the shelf. Policy based routing helped but it's tough to make it work. Now, what you can do is put the Asterisk on ONE network and use policy based routing to share other stuff like surfing, smtp, telnet, etc. You can prioritize the traffic so that the packets to and from the Asterisk are mangled to have the higher priority. If both DSL's are for Asterisk ONLY then you might try round-robin DNS or manually setup traffic. Asteris will work on multiple LAN's - I have both a PUBLIC and PRIVATE ip in the same box on different NIC's. Just set your routing. James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of VoIP-PBX Sent: Thursday, June 23, 2005 1:46 PM To: Jorge Carrasquillo; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Using 2 x DSL Hi all, my client wants to double his bandwidth by using 2 x DSL lines into one Asterisk network How can I do this ? Thanks Henry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call file calling twice
Hi list! The call files are working really great I just have one problem. I am using this as my call file: Channel: SIP/228 Context: from-internal Extension: 0090 Priority: 1 Callerid: 0090 so the external number is connected to my sip phone. However after speaking for approx 30 seconds, Asterisk does a retry and I see the external number in my display on the second line. It does this on every call. When I'm finished I also see 2 records in the log files. This is from the event log: Jun 21 14:08:15 asterisk[1760]: Queued call to SIP/228 expired without completion after 0 attempt(s) Jun 21 14:08:16 asterisk[1760]: Queued call to SIP/228 completed Any idea why Asterisk is trying to place the call again even though the first attempt was succesfull and the call is still in progress? I didn't specify a redial anywhere. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call file ignored?
Hi list! I'm trying to use call files to place outgoing calls. I want to schedule an outbound call and want it to ring on my sip phone. My sip phone is SIP/228 and the call should go out according to the LCR rules as defined in the dialplan. I don't mind waiting for the call to be answered on my phone so I don't need the functionality that my phone will ring only when the call is answered. This is what I have in the call file: Channel: SIP/228(my sip phone) MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: from-internal Extension: 003120123456 Priority: 1 When I set the permissions and move the file to /var/spool/asterisk/outgoing nothing happens. I guess * does find the file because it is gone immediately but I don't even get an error on the console. What am I doing wrong? Can I snatch a working call file from the outgoing directory? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call file ignored?
Yes, the channel should be correct. I'm using AMP and from-internal is the context the sip phones are normally in. Do you see anything on the console even if you dial a number that isn't answered? Thanks! On Mon, 20 Jun 2005, jurczak wrote: Hello, I just tried it, and it worked fine for me. Of course the context and the Extension where different. Is the Channel correct? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: Monday, June 20, 2005 2:12 PM To: Asterisk Users List Subject: [Asterisk-Users] call file ignored? Hi list! I'm trying to use call files to place outgoing calls. I want to schedule an outbound call and want it to ring on my sip phone. My sip phone is SIP/228 and the call should go out according to the LCR rules as defined in the dialplan. I don't mind waiting for the call to be answered on my phone so I don't need the functionality that my phone will ring only when the call is answered. This is what I have in the call file: Channel: SIP/228(my sip phone) MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: from-internal Extension: 003120123456 Priority: 1 When I set the permissions and move the file to /var/spool/asterisk/outgoing nothing happens. I guess * does find the file because it is gone immediately but I don't even get an error on the console. What am I doing wrong? Can I snatch a working call file from the outgoing directory? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call file ignored?
I really don't know why but after restarting asterisk it just works all of a sudden, no change to the call file I guess that doesn't offer a real solution but it works Thanks guys! On Mon, 20 Jun 2005, Marco Parmeggiani wrote: Remco Barende ha scritto: Do you see anything on the console even if you dial a number that isn't answered? i see this for a non existant number: Attempting call on Zap/g1/12345 for [EMAIL PROTECTED]:1 (Retry 1) i guess it prints out for every call originated by a call file. asterisk -cvvv ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QuadBRI: How to set the outgoing callerid (KPN - NL)
Are you sure that for the BRI outgoing callerid is allowed?
I have several BRI lines where outgoing callerid is blocked (on my
request) by KPN. No matter what you pass to the BRI line, KPN will never
pass callerid.
On Mon, 20 Jun 2005, Stijn Jonker wrote:
Hello all,
Recently I purchased an QuadBRI card from junghanns.net after some
playing around, reconfiguring dialplans etc with the exception of 1
thing everything seems to work:
I seem to be unable to set the outbound callerid. The dutch telecom
operator (KPN) provided me with 4 MSN's on 1 BRI interface. In the past
years I'm more then used to setting the MSN without the leading 0, this
always worked fine.
With the QuadBRI this didn't work, added the 0, messed around with
CallingPres, SetCallerPres and the various callerid and usecallerpres
values in zapata.conf. (Off course found via google, mailinglist,
voip-info etc) All calls resulting to be presented at the far end with
the network default caller id.
I'm lost, can somebody provide some assistance, attached is an output of
bri debug span 1 (this due to line wraps..)
But I think this is the network returning my error:
Ext: 1 Cause: Invalid information element contents (100), class =
Protocol Error (6)
My zapata is short, but as said the various caller id options have
passed through this file ;-)
zapata.conf:
--
[channels]
switchtype = euroisdn
pridialplan = dynamic
prilocaldialplan = local
nationalprefix = 0
internationalprefix = 00
musiconhold=default
echocancel = yes
echocancelwhenbridged = no
echotraining = 100
;-
signalling = bri_cpe_ptmp
context=pstn-inbound
group = 1
channel = 1-2
;-
signalling = bri_net_ptmp
context=isdn-inbound
group = 2
channel = 4-5
Part of dialplan:
--
exten = _0.,1,SetCallerID(${EDN_MAIN:1})
exten = _0.,2,Dial(${ZAP_MAIN}${EXTEN:1},30)
exten = _0.,3,Macro(dial-result)
exten = _9.,1,SetCallerID(${EDN_WORK:1})
exten = _9.,2,Dial(${ZAP_MAIN}${EXTEN:1},30)
exten = _9.,3,Macro(dial-result)
Variables:
-
ZAP_MAIN=Zap/g1/
EDN_MAIN=022929XXX0
EDN_WORK=022927XXX7
The 022929XXX0 is the default msn, even if i try to set this
explicitly it gets rejected with the network message as mentioned above.
(But because it's default all calls will originate with 22929XXX0)
If more info is needed don't hesitate to ask for it I'm kind of lost on
what to provide.
Thanks in advance.
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Re: [Asterisk-Users] QuadBRI: How to set the outgoing callerid (KPN - NL)
On Mon, 20 Jun 2005, Stijn Jonker wrote: Hello Remco Michiel, First of all thanks for the replies. On 20-Jun-2005 21:47, Michiel van Baak wrote: On 21:11, Mon 20 Jun 05, Remco Barende wrote: Are you sure that for the BRI outgoing callerid is allowed? Yep, with pain in the heart i reconnected the KPN ISDN phone directly to the NT1 again. Reprogrammed the MSN's and I can dialout with a diffrent MSN. Did you put : usecallerid=yes callerid=asreceived In zapata.conf under channels? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Intelligent maximum channels solution?
Hi list! I have an asterisk box connected to an ADSL connection that has 1 Mbit upstream. Is there any way to use max channels intelligently? For example I would like to do some checks on the outgoing calls. When it's quiet I want each and every call to go out to my IAX provider. However when more people start placing calls I would like to leave some room for the real expensive calls and switch chep (local) calls to the PSTN so expensive international calls can still be routed through the IAX provider. I.e. allow 2 local calls and 1 call to a neighbouring country and still leave 3 channels free for calls to Japan or China that would be frightfully expensive from the PSTN. This way I could squeeze the maximum benefit from the IAX / ADSL connection. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TAPI for Dos (xbase / Clipper)
Hi list! Does anyone know of a TAPI that will work wil a Clipper application (MS DOS)? Alternatively I could recompile the lot and run it on a linux box but then again I would need a TAPI I guess? I just want clickdial from our CRM app. Alternatively, I could create a samba share for the asterisk call files and have our CRM app create Asterisk call files and just dump them on that share. Anyone ever tried this approach? Thanks!! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
