[asterisk-users] Which GUI for ACD edition ?
Hello, I want to safely delegate ACD edition to a system administrator who has no knowledge of Linux nor Asterisk. More precisely, I want him to be able to edit and change menus such as : Type 1 for management; 2 for support; 3 for sales department. I could teach this administrator what Asterisk config files are but I'm wondering if any GUI exists for such task (editing a vocal menu tree). Maybe something not related to telephony could be used for that. Any idea ? Best regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
On Tue, 21 Aug 2007, Vidura Senadeera wrote: Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? With my other hat on I build and maintain many servers with disk capacities ranging from 80GB to over 6TB... All using Linux software RAID. I've been using Linux s/w RAID for over 8 years now. So with RAID-1 done in hardware, the impact to the system, CPU, etc. should be no more (or less) than running a single SCSI or SATA drive. You write the data over the (PCI) bus once and the hardware takes care of writing it to both drives behind your back. Similarly for reading (where it might only read from one drive or from alternative drives) you only see one transaction over the PCI bus. You do (sometimes) need the hardware RAID controller to be supported by Linux and this is a weak area. Some controllers just look like a standard drive, so they are transparent to the system, but then you need to use either the BIOS utilities to set it up in the first place, or (typically) a Windows utility, although some controllers are now being supported by Linux with user-land tools to manage and check the arrays. Doing it in software requires double the PCI bandwidth for writes, but the same as a single drive or hardware controller for reads. AIUI, the current software RAID-1 reads alternatively from the disks. So on writes. The overhead in terms of CPU power is minimal - write the same block twice, and if the hardware is good, then both writes can be transfered over the PCI bus rapidly, into the cache on the drives and the writes then take place in parallel, so performance wise, it's really no worse than single drive (and it's important to note than it's no better than a single drive on reads too, despite many threads on the linux-raid list suggesting otherwise!) RAID-1 doesn't require parity calculations, so the software overhead really is quite small (especially when you compare it to the relatively huge times it takes to actually get the data to/from the disks) So things that are important: Make sure the hardware to each drive is as independent as possible. Hard to do these days as there is probably only one SATA controller chip on the motherboard. You also need to see what happens when a drive dies - is it going to crowbar the entire SATA chip and block the other drive? Is the driver going to recognise it quickly enough and so on. (Some early SATA drives weren't good at this) And the usual - make sure all the hardware has it's own interrupts. For the absolute maximun performance, (and minimum overheard) then you need a motherboard with multiple PCI buses - put the disks on one bus, the PRI card on another. If terms of disk b/w needed - if we're using g711, then it's 64KB/sec, and 20 calls streaming to voicemail is 1.3MB/sec. A single modern drive ought to be able to sustain 60MB/sec read or writes, so there is plenty of overhead, as long as asterisk is relatively sensible about buffering disk write/reads (which I think it is) So I'd say go for it, but do take the time, if possible to build a custom kernel for your hardware, and at the BIOS level, turn off all drivers that you won't be using - eg. on-board sound, then 2nd network port, USB (if you're not using it, don't enable it!) and so on, and make sure you have a custom compiled kernel for your exact hardware requirements with no modules loaded other than the Zap/TDM, etc., ones. And I'd also say go for it because I have similarly specd. servers doing similar tasks also running asterisk. I won't put a server in a remote data centre these days without it either booting off flash, or using at least RAID-1. Remember to put your swap on RAID-1 too. Here is one of my servers in a similar setup to yours: $ cat /proc/mdstat Personalities : [raid0] [raid1] md1 : active raid1 hdc1[1] hda1[0] 248896 blocks [2/2] [UU] md2 : active raid1 hdc2[1] hda2[0] 995904 blocks [2/2] [UU] md3 : active raid1 hdc3[1] hda3[0] 200 blocks [2/2] [UU] md5 : active raid1 hdc5[1] hda5[0] 38081984 blocks [2/2] [UU] md6 : active raid1 hdc6[1] hda6[0] 38708480 blocks [2/2] [UU] unused devices: none $ df -h
Re: [asterisk-users] Realtime Queue Members
I have it working fine in 1.4.x, but I also have the queues defined in the Realtime database and not in the queues.conf -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Anthony Francis Enviado el: martes, 21 de agosto de 2007 1:46 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Realtime Queue Members Peder @ NetworkOblivion wrote: Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general] realtime_family=queue_members persistentmembers = yes autofill = yes monitor-type = MixMonitor [queue2280] music = default strategy = roundrobin timeout = 15 wrapuptime=10 announce-frequency = 30 announce-holdtime = no joinempty = yes extconfig.conf: [settings] queue_members=mysql,ASTERISK,queue_member_table MYSQL: [EMAIL PROTECTED]:/etc/asterisk# mysql -u myuser -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 7 to server version: 5.0.24a-Debian_9ubuntu2-log Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql use ASTERISK; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Database changed mysql select * from queue_member_table; ++---+-+ | queue_name | interface | penalty | ++---+-+ | queue2280 | SIP/2224 | 1 | | queue2280 | SIP/2223 | 1 | | queue2280 | SIP/ | 2 | ++---+-+ 3 rows in set (0.00 sec) I don't see any log info for mysql, except when I manually enter the info above. I've stopped an restarted * many times. I've even tried this on two separate boxes and I get the same thing. sipeers and voicemail work, but queue members does not. Any idea? I am running 1.4.10.1. Thanks. Peder ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There is no queue_members file, asterisk doesnt know hat you are talking about, you would have to #include queue_members from inside that queue definition. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE405/TE410P help updating from 1.0 to 1.4
On Mon, Aug 20, 2007 at 09:23:20PM -0400, Jerry Geis wrote: I have a TE405/TE410P card that was working on 1.0.X I upgraded the OS to Centos 4.5, Updated asterisk to 1.4 and zaptel to 1.4.5 and libpri. I copied all the zaptel and zapata and extensions.conf files from 1.0 I did update extensions.conf from 1.0 to 1.4 commands. I cannot get the card to work in 1.4.10. AHHH! I see with zttool that the T1 is in Green, I see calls coming in as the bits go high on channel 8, zaptel doesnt respond so it tries channel 7 then gives up. Any ideas what this might be??? zaptel modules load, asterisk loads. ztcfg gives correct reply everything looks good just not working. What is the output of a call trace from the CLI: core set verbose 3 And report what you see when you try to call out. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE405/TE410P help updating from 1.0 to 1.4
Tzafrir Cohen wrote: On Mon, Aug 20, 2007 at 09:23:20PM -0400, Jerry Geis wrote: I have a TE405/TE410P card that was working on 1.0.X I upgraded the OS to Centos 4.5, Updated asterisk to 1.4 and zaptel to 1.4.5 and libpri. I copied all the zaptel and zapata and extensions.conf files from 1.0 I did update extensions.conf from 1.0 to 1.4 commands. I cannot get the card to work in 1.4.10. AHHH! I see with zttool that the T1 is in Green, I see calls coming in as the bits go high on channel 8, zaptel doesnt respond so it tries channel 7 then gives up. Any ideas what this might be??? zaptel modules load, asterisk loads. ztcfg gives correct reply everything looks good just not working. What is the output of a call trace from the CLI: core set verbose 3 And report what you see when you try to call out. Good advice. Also try pri intense debug span X. Before any of that, replace your T1 cable, especially if it is home made job. It is always surprising how much time I have wasted on something, when just swapping out a new cable fixed it instantly. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing Variables to Voicemail's Email Notification
You could do more of a hack and find where in the code that the callerID
name and number are found in the voicemail code and use the seldom used
RDNIS variable.
C F's solution is clean and will work across upgrades but I would
probably do the above.
Thanks,
Steve
C F wrote:
While I don't have an answer on how to access channel variables from
voicemail.conf, for the problem you mention this should help.
Change CALLERID(name) to your ticket number and then use VM_CIDNAME in
the subject line.
If you don't want to lose the original CIDNAME then just add your
ticket number like this:
Set(CALLERID(name)=${CALLERID(name)} TICKETNUMBER:12345)
On 8/20/07, 0xception [EMAIL PROTECTED] wrote:
Is there a way, other then recoding the entire voicemail application, to
pass dialplan variables to the voicemail application and to the email
notifications of new voicemail.
For example in our small tech support queue i would like to pass the ticket
number with the email notification that a new support voicemail was left.
I've tried simply replacing the ${VM_WHATEVER} w/ the actual variable name
inside the voicemail.conf file, I've also tried setting the VM variables
directly before the voicemail application call in the dial plan... both of
these fail.
Anyone else know of another way?
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Re: [asterisk-users] compatibility of PRI Two B channel transfers TBTC/2BTC
Matt Florell wrote: Hello, A client has asked for Two B channel Transfer capability (known as TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG Path Replacement) in a new Asterisk system and so I researched the capability and came up with quite a few gaps in documentation. From what I've gathered, the official Digium statement is that is works with DMS100 only, and only in Asterisk 1.4.X : http://kb.digium.com/entry/26/140/ Although in a bugtracker posting with a patch from over two years ago, Matt Fredrickson from Digium says that it works with 5ESS under Asterisk 1.2.X: http://bugs.digium.com/view.php?id=3554 There are also bounties and claims of this feature working on NI2 protocol(although no patches posted) on the voip-info.org Wiki: http://www.voip-info.org/wiki/view/Asterisk+bounty+PRI+2B+channel+transfer+for+NI2+PRI+line http://www.voip-info.org/wiki/index.php?page=Asterisk%20bounty%20PRI%202B%20channel%20transfer As for actually using this feature, you apparently need to add the following lines to the zapata.conf section that you want to be able to use 2BCT: facilityenable = yes transfer=yes To execute the transfer, you need to use the Transfer cmd within Asterisk: http://voipinfo.org/wiki/view/Asterisk+cmd+Transfer And according to this post, you can only do 2BCT transfers if the first call is inbound: http://www.mail-archive.com/[EMAIL PROTECTED]/msg25131.html Does 2BCT work with DMS100 and 5ESS right now? Are there people using this in production right now that can shed some more light on exactly how they are using it, and executing the transfers? Any input would be greatly appreciated. Thanks, MATT--- Sounds like the early days of Asterisk. Do us all a favor, after you have tried everything, pulled out your hair, banged your head, had a breakthrough and solved the issue (jumping up and down excitedly saying YES, YES YES, post your experience to the wiki, and or reply to this thread. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Heavy duty environment - Is TDM2400P suits?
I am using TDM2400 with FXO modules to handle 16 concurrent calls to PSTN for more than 12 hours a day with no problem at all. On 8/16/07, Chan Jason [EMAIL PROTECTED] wrote: Hi all, I am planning to have a new TDM2400P to replace all Planet 450 SIP gateways. Can TDM2400P survive in heavy duty environment where there will be 4 concurrent calls in within the same second? Thanks! Yours sincerely, Jason Chan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk (Vidura Senadeera)
Dear all, Thanks for the greate explanation regaing Software/H/W Raid. This details better but on voip-info.org/wiki pages. Thanks lot agian. Regs, Vidura Senadeera. == Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? With my other hat on I build and maintain many servers with disk capacities ranging from 80GB to over 6TB... All using Linux software RAID. I've been using Linux s/w RAID for over 8 years now. So with RAID-1 done in hardware, the impact to the system, CPU, etc. should be no more (or less) than running a single SCSI or SATA drive. You write the data over the (PCI) bus once and the hardware takes care of writing it to both drives behind your back. Similarly for reading (where it might only read from one drive or from alternative drives) you only see one transaction over the PCI bus. You do (sometimes) need the hardware RAID controller to be supported by Linux and this is a weak area. Some controllers just look like a standard drive, so they are transparent to the system, but then you need to use either the BIOS utilities to set it up in the first place, or (typically) a Windows utility, although some controllers are now being supported by Linux with user-land tools to manage and check the arrays. Doing it in software requires double the PCI bandwidth for writes, but the same as a single drive or hardware controller for reads. AIUI, the current software RAID-1 reads alternatively from the disks. So on writes. The overhead in terms of CPU power is minimal - write the same block twice, and if the hardware is good, then both writes can be transfered over the PCI bus rapidly, into the cache on the drives and the writes then take place in parallel, so performance wise, it's really no worse than single drive (and it's important to note than it's no better than a single drive on reads too, despite many threads on the linux-raid list suggesting otherwise!) RAID-1 doesn't require parity calculations, so the software overhead really is quite small (especially when you compare it to the relatively huge times it takes to actually get the data to/from the disks) So things that are important: Make sure the hardware to each drive is as independent as possible. Hard to do these days as there is probably only one SATA controller chip on the motherboard. You also need to see what happens when a drive dies - is it going to crowbar the entire SATA chip and block the other drive? Is the driver going to recognise it quickly enough and so on. (Some early SATA drives weren't good at this) And the usual - make sure all the hardware has it's own interrupts. For the absolute maximun performance, (and minimum overheard) then you need a motherboard with multiple PCI buses - put the disks on one bus, the PRI card on another. If terms of disk b/w needed - if we're using g711, then it's 64KB/sec, and 20 calls streaming to voicemail is 1.3MB/sec. A single modern drive ought to be able to sustain 60MB/sec read or writes, so there is plenty of overhead, as long as asterisk is relatively sensible about buffering disk write/reads (which I think it is) So I'd say go for it, but do take the time, if possible to build a custom kernel for your hardware, and at the BIOS level, turn off all drivers that you won't be using - eg. on-board sound, then 2nd network port, USB (if you're not using it, don't enable it!) and so on, and make sure you have a custom compiled kernel for your exact hardware requirements with no modules loaded other than the Zap/TDM, etc., ones. And I'd also say go for it because I have similarly specd. servers doing similar tasks also running asterisk. I won't put a server in a remote data centre these days without it either booting off flash, or using at least RAID-1. Remember to put your swap on RAID-1 too. Here is one of my servers in a similar setup to yours: $ cat /proc/mdstat Personalities : [raid0] [raid1] md1 : active raid1 hdc1[1] hda1[0] 248896 blocks [2/2] [UU] md2 : active raid1 hdc2[1] hda2[0] 995904 blocks [2/2] [UU] md3 : active raid1
Re: [asterisk-users] Heavy duty environment - Is TDM2400P suits?
You should have no problems. Make sure you put surge protection and ground your POTS lines. It is a small investment. I have had SEVERAL FXO modules die or behave strangely after thunderstorms. I cannot prove it was a surge, but logic would indicate that was the issue. Thanks, Steve Totaro O.Kamal wrote: I am using TDM2400 with FXO modules to handle 16 concurrent calls to PSTN for more than 12 hours a day with no problem at all. On 8/16/07, *Chan Jason* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I am planning to have a new TDM2400P to replace all Planet 450 SIP gateways. Can TDM2400P survive in heavy duty environment where there will be 4 concurrent calls in within the same second? Thanks! Yours sincerely, Jason Chan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 99 bottles of beer
Russell Bryant wrote: Steve Murphy wrote: How about this one: from an extensions.conf that someone posted on the internet, I think, and I converted to AEL; I'm sorry, but I can't find the original author. (If anybody can find his post, I'd love to give him credit.) I did test this out, and it works; just put a call to the macro ( guessgame(); ) in an extension in your dialplan Nice! While we're on the subject of silly but fun dialplan bits, check out my TV remote extension. When I moved a few months ago, there was a while when I couldn't find the wireless keyboard that I usually use as my TV remote to control MythTV. So, I built dialplan so I could use a wireless phone as my remote, instead. The dialplan reads digits from the phone and sends the correct commands to a MythTV network control interface for the frontend application. I posted my tested .conf version and the untested AEL version to the MythTV wiki. The AEL version would probably be prettier with macros, now that I think of it ... http://www.mythtv.org/wiki/index.php/Controlling_MythTV_from_any_phone_using_Asterisk Wow... that's just wow. Words fail me. I'm not saying it isn't cool... just... wow. ;) N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE405/TE410P help updating from 1.0 to 1.4
Tzafrir Cohen wrote: / On Mon, Aug 20, 2007 at 09:23:20PM -0400, Jerry Geis wrote: // // I have a TE405/TE410P card that was working on 1.0.X // // I upgraded the OS to Centos 4.5, Updated asterisk to 1.4 and zaptel to // 1.4.5 and libpri. // // I copied all the zaptel and zapata and extensions.conf files from 1.0 // // I did update extensions.conf from 1.0 to 1.4 commands. // // I cannot get the card to work in 1.4.10. AHHH! // // I see with zttool that the T1 is in Green, I see calls coming in as the // bits go high on channel 8, // zaptel doesnt respond so it tries channel 7 then gives up. // // Any ideas what this might be??? zaptel modules load, asterisk loads. // ztcfg gives correct reply // everything looks good just not working. // // // What is the output of a call trace from the CLI: // // core set verbose 3 // // // And report what you see when you try to call out. // // /Good advice. Also try pri intense debug span X. Before any of that, replace your T1 cable, especially if it is home made job. It is always surprising how much time I have wasted on something, when just swapping out a new cable fixed it instantly. Thanks, Steve Totaro Guys, THanks for the reply. Last night tried backing down to 1.2, didnt work, I then backed down to 1.0 and that is working. If I can get back in and get time I will try 1.4 again the above commands. Was really disappointed 1.4 didnt just work. The customer also has a cisco call manager so I may just try and migrate them to SIP. Thanks again, hope I have a chance to at least try the commands above and see what it says. Jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Heavy duty environment - Is TDM2400P suits?
Steve Totaro wrote: You should have no problems. Make sure you put surge protection and ground your POTS lines. It is a small investment. I have had SEVERAL FXO modules die or behave strangely after thunderstorms. I cannot prove it was a surge, but logic would indicate that was the issue. Steve, How are you providing surge protection? I have lost a couple of cards to storms also. -- Chris Mason -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Redundancy / Failover
Dears Any one succeeded to make Redundancy / Failover with asterisk 1.4.9 on centos with kernel 2.6.9-55.EL. Can you please send me the documentation link on how to or write down how to . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
C F wrote: ~# cat /proc/mdstat Personalities : [raid1] md0 : active raid1 sdb2[2](F) sda2[1] 76139968 blocks [2/1] [_U] unused devices: none The above is from an active system that one hdd failed. It would take way longer to find such a thing on a hardware raid. Unless it came with a program that emails me notification on such a failure. Also, speaking of email notifications, the mdadm program that controls Linux software RAID has a monitor mode with that capability. One of my own systems is a file server that had four 120GB SATA drives in software RAID5 configuration, using a pair of PCI controller cards. A few months back, one of the drives failed (shortly after the three-year warranty expired, hm), and I got a note in my inbox about it. The array continued running in degraded mode, so I made one last backup, replaced the old drives with three 500 GB units (again in RAID5), and restored the contents onto the new array. Of course, once this array gets full enough, I'm going to have to get one of those 1 TB drives in an eSATA enclosure to back things up. USB2 is too slow now. :/ At least the new drives have five-year warranties - hopefully they should hold up at least that long. Russ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TC400B and show transcoder
Hi All, I have recently installed a TC400B card into a system and am trying to get it to work. As far as I ca tell from the docco on Digiums website, there is no config as such unless you want to enable / disable only 1 codec, otherwise by default it runs as 92 channels of either. I have tried asterisk 1.4.9, 1.4.10 and 1.4.10.1 along with zaptel 1.4.4 and addons 1.4.2. The zaptel modules all apear to be loaded correctly (loading wctc4xxp loads up zttranscode and zaptel). Dmesg shows that the card has been found: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) Zaptel DTE (g.729a / g.723.1 5.3kbps) Transcoder support LOADED (firm ver = 56) Found and successfully installed a Wildcard TC: Wildcard TC400P+TC400M and the card has its own interrupt - 193:18715321896779 IO-APIC-level tc400b But when ever we need to do a transcode, ie playing back a wav file on a g729 channel, the audio is complete rubbish, with a lot of stutters in it (sounds like a recording does when you upload a file in the wrong sample rate etc) - the file that we are playing back is a wav file that has existed on the system and has been successfully played back with the soft g729 transcoding and also plays back fine when the channel is alaw, just not when the channel is g729. The same issue occurs when a transcode has to happen from a handset to a IP trunk, eg alaw on the handset and g729 on the trunk channel, the audio stream is non comprehensible. The other issue is that whilst all the modules apear to be loaded ocrrectly, and a show translation shows that the codes are supported without the presence of a g729 key: pbxla*CLI core show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723- 3113 32 32 - 12 3- gsm3 -222 21 23 - 11 2- ulaw1 2-12 21 21 - 11 2- alaw1 21-2 21 21 - 11 2- g726aal23 222- 21 23 - 11 1- adpcm3 2222 -1 23 - 11 2- slin2 1111 1- 12 - 10 1- lpc103 2222 21 -3 - 11 2- g7292 3113 32 3- - 12 3- speex- ---- -- -- -- -- ilbc4 3333 32 34 -- 3- g7263 2221 21 23 - 11 -- g722- ---- -- -- -- -- The show transcoder command listed in the documentation does not exist. There is no show transcoder or core show transcoder command available on the system. I have checked the menu options for the build and cannot see any specific item that needs to be enabled for this command to be available but have a feeling that the lack of this command and the horrible transcoded audio quality are related. Or is it just that the show transcoder command is only available in 1.2 and not in 1.4? Another quick (hopefully) question - does the TC400 card provide a zaptel timing source, or do you still need to load ztdummy in the case of not having another card in the system? Any info or experiences would be great. Thanks in advance. Ben ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Heavy duty environment - Is TDM2400P suits?
Here is a good read as far as what your risk is and how to mitigate it. They clamp on top of your terminated 66 blocks and you also want to properly ground. Here is a vendor with a good selection and pricing. I have no idea if they are any good, I have never used them. http://www.twacomm.com/catalog/dept_id_602.htm I usually use Graybar. Thanks, Steve Totaro Chris Mason (Lists) wrote: Steve Totaro wrote: You should have no problems. Make sure you put surge protection and ground your POTS lines. It is a small investment. I have had SEVERAL FXO modules die or behave strangely after thunderstorms. I cannot prove it was a surge, but logic would indicate that was the issue. Steve, How are you providing surge protection? I have lost a couple of cards to storms also. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redundancy / Failover
Please stop posting this repeatedly. There are pointers on www.voip-info.org If you post to the biz list, and pay for someone's time and effort, you may have better luck. If you keep posting the same annoying message complete with HTML to the user's list, I seriously doubt anyone will ever help you. Thanks, Steve Totaro Khaled Chehab wrote: Dears Any one succeeded to make _Redundancy* / Failover with asterisk 1.4.9 on centos with kernel 2.6.9-55.EL. ***_ Can you please send me the documentation link on how to or write down how to . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Heavy duty environment - Is TDM2400P suits?
Oooops forgot first link. http://www.sandman.com/surge.html Steve Totaro wrote: Here is a good read as far as what your risk is and how to mitigate it. They clamp on top of your terminated 66 blocks and you also want to properly ground. Here is a vendor with a good selection and pricing. I have no idea if they are any good, I have never used them. http://www.twacomm.com/catalog/dept_id_602.htm I usually use Graybar. Thanks, Steve Totaro Chris Mason (Lists) wrote: Steve Totaro wrote: You should have no problems. Make sure you put surge protection and ground your POTS lines. It is a small investment. I have had SEVERAL FXO modules die or behave strangely after thunderstorms. I cannot prove it was a surge, but logic would indicate that was the issue. Steve, How are you providing surge protection? I have lost a couple of cards to storms also. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
On Tue, 21 Aug 2007 07:33:23 +0530 Vidura Senadeera [EMAIL PROTECTED] wrote: Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? I will be highly appreciate your feedback on this regard. 1: Software RAID on Linux is way less than impressive. Plus last a I checked Linux can't handle mirroring a entire disk. Last I looked at it around a year ago you were limited to only mirroring partitions, which is a joke from a administrative standpoint. 2: No real impact other than a bad disk won't mean a reinstall. 3: On Linux, go hardware. On FreeBSD it is personal choice. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
Remco Barendse a écrit : Has anyone ever tried using a Nokia phone with SIP client as channel for Asterisk? I mean i would like to receive calls to the mobile on asterisk and use the Nokia phone to place calls to cell destinations. E70 and E65 are working perfectly as SIP client through WIFI. I have enough Nokia E60's to do that and it would circumvent the need for chan_bluetooth or something similar!! :) Should work too -- Daniel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
Thank you all for your post, i've found them quite interesting and will give work for some time :) Thanks again. Cheers, Jonathan GF On 8/20/07, Eric Chamberlain [EMAIL PROTECTED] wrote: Using the phone itself as a GSM-SIP gateway is not possible with the native VoIP application, but it looks like it should be possible with a custom application for the phone. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Remco Barendse Sent: Monday, August 20, 2007 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Nokia cell connected to Asterisk Has anyone ever tried using a Nokia phone with SIP client as channel for Asterisk? I mean i would like to receive calls to the mobile on asterisk and use the Nokia phone to place calls to cell destinations. I have enough Nokia E60's to do that and it would circumvent the need for chan_bluetooth or something similar!! :) On Mon, 20 Aug 2007, Steve Totaro wrote: Well chan_bluetooth is really amazing (especially if your phone does not support SIP). You connect your phone via bluetooth to your asterisk box and it becomes a channel type. You can use it as an extension(FXS) or a phone line (FXO). I believe you can send and receive SMS through the phone/Asterisk as well. Chan_bluetooth README is in the asterisk-addons trunk and gives you basic instruction on setting it up. You get several added pieces of functionality with this setup. SMS send and receive through your phone using Asterisk?, FXO failover or LCR, FXS where your cell phone becomes an extension. Thanks, Steve Jonathan GF wrote: Thanks Steve and Mitcheloc, in fact i was think in something more obsolet like connect via serial/usb cable the cell to the asterisk box. Never thought in the SIP stack of new Nokia's but i will start looking for info about this. If you [Steve] know of a good written material of interest please let me know. Probably Mitcheloc is right too, there are a lot of manners to achieve this and the problem is mine that i don't know how to search what i want. Anyway, thank you for your inputs. Any others will be welcomed, for sure. Regards, Jonathan GF On 8/20/07, *mitcheloc* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Jonathon, Are you talking about using the built in SIP client on some Nokia phones? I'm using an E90 with Asterisk and it works very well. I used Google for help and it returned plenty of results. Cheers, Mitchel On 8/19/07, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If it is bluetooth and you don't mind running Asterisk 1.4 trunk, you should look at chan_mobile. Thanks, Steve Totaro From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] on behalf of Jonathan GF Sent: Sun 8/19/2007 6:26 PM To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Nokia cell connected to Asterisk Hi folks, i've been looking for in many sources but i cannot see clear if the options i'm chasing is feasible with Asterisk. I understand that should be. I would like to connect a nokia cell to Asterisk but i don't know how exactly. Any ideas, inputs, docs or refs will be welcomed. Thanks in advance. Jonathan GF ___ --Bandwidth and Colocation Provided by http://www.api-digital.com- - asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com http://www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
Zane C.B. wrote: On Tue, 21 Aug 2007 07:33:23 +0530 Vidura Senadeera [EMAIL PROTECTED] wrote: Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? I will be highly appreciate your feedback on this regard. 1: Software RAID on Linux is way less than impressive. Plus last a I checked Linux can't handle mirroring a entire disk. Last I looked at it around a year ago you were limited to only mirroring partitions, which is a joke from a administrative standpoint. 2: No real impact other than a bad disk won't mean a reinstall. 3: On Linux, go hardware. On FreeBSD it is personal choice. You can (sort of) run raid on an entire disk, but you have to use LVM. You basically create a single partition on the disk, run raid on that partition and then use LVM with the /dev/md? device as a physical volume that you can then partition with LVM. -Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redundancy / Failover
Any one succeeded to make _Redundancy* / Failover with asterisk 1.4.9 on centos with kernel 2.6.9-55.EL. ***_ Can you please send me the documentation link on how to or write down how to . hint yum -y install heartbeat (on node1 and node2) configure heartbeat if you have configuration in mysql then set up master-to-master replication (- www.mysql.com) or generate ssh keys priodically copy /etc/asterisk and /var/lib/asterisk/astdb from master node to slave node (astdb is needed because of sip registrations) question1: do you someone know how to _easily_ find out which node is master? (heartbeat) - now i have custom script for this question2: it's possible read registration data from astdb from python/php (or it is possible write sip registrations to mysql/sqlite? i do not want realtime because of NAT issues) --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk in Soekris 5501: Is Astlinux the only able solution?
Hello, i would like the forum to help or advice me if my feeling is correct or not. Is Astlinux the only distribution able to run on Soekris 5501 hardware or other can run also (trixbox, freepbx, o maybe a manual installation of asterisk). My question is easy: i'd need to install it on that hardware for a very small office and the further administrator do not understand so much about Asterisk, although they can handle unix/linux boxes. Any help would be really appreciated. Thank in advance for you help. Regards, Jonathan GF ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
On Mon, 20 Aug 2007, Steve Totaro wrote: Well chan_bluetooth is really amazing (especially if your phone does not support SIP). You connect your phone via bluetooth to your asterisk box and it becomes a channel type. You can use it as an extension(FXS) or a phone line (FXO). I believe you can send and receive SMS through the phone/Asterisk as well. Chan_bluetooth README is in the asterisk-addons trunk and gives you basic instruction on setting it up. You get several added pieces of functionality with this setup. SMS send and receive through your phone using Asterisk?, FXO failover or LCR, FXS where your cell phone becomes an extension. Does FSX really work? Can I really use my mobile as an extension? How do I make my mobile phone dial out over bluetooth rather than it's GSM connection? If this really is the case, does it then create the holy grail of one phone for everything? Does it support one to many? I'm imagining an office where I connect a bluetooth dongle on the end of a long USB cable up to the middle of the room, into the PBX which many mobile phones can then access and let the punters use their mobiles to make/take calls via the PBX when in the office and use them as normal mobile when out of the office... So in the office, mobile rings via bt, when no bt connection, then it rings out via the PSTN to the mobile. (or via another GSM gateway) But I'm really clueless on bluetooth use - other than sending cheeky messages to other peoples mobiles and connecting my borg implant to my mob when driving! I also know I can probably do this with mob's that have WiFi and SIP clients too, however ... Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Queue Members
Peder @ NetworkOblivion wrote: Anthony Francis wrote: There is no queue_members file, asterisk doesnt know hat you are talking about, you would have to #include queue_members from inside that queue definition. Huh? How is including a file going to make realtime access the queue_members database via mysql? Because the extconfig.conf is a mapping of files to database tables. -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
Gordon Henderson a écrit : On Mon, 20 Aug 2007, Steve Totaro wrote: Well chan_bluetooth is really amazing (especially if your phone does not support SIP). You connect your phone via bluetooth to your asterisk box and it becomes a channel type. You can use it as an extension(FXS) or a phone line (FXO). I believe you can send and receive SMS through the phone/Asterisk as well. Chan_bluetooth README is in the asterisk-addons trunk and gives you basic instruction on setting it up. You get several added pieces of functionality with this setup. SMS send and receive through your phone using Asterisk?, FXO failover or LCR, FXS where your cell phone becomes an extension. Does FSX really work? Can I really use my mobile as an extension? How do I make my mobile phone dial out over bluetooth rather than it's GSM connection? If this really is the case, does it then create the holy grail of one phone for everything? Does it support one to many? I'm imagining an office where I connect a bluetooth dongle on the end of a long USB cable up to the middle of the room, into the PBX which many mobile phones can then access and let the punters use their mobiles to make/take calls via the PBX when in the office and use them as normal mobile when out of the office... So in the office, mobile rings via bt, when no bt connection, then it rings out via the PSTN to the mobile. (or via another GSM gateway) But I'm really clueless on bluetooth use - other than sending cheeky messages to other peoples mobiles and connecting my borg implant to my mob when driving! I also know I can probably do this with mob's that have WiFi and SIP clients too, however ... Yes, and it's working great, particulary with Nokia's: you tell them to try to call at first through Internet, if it fails, fallback to GSM. Once you're in the office with WIFI, device get automatically connected to the net and you can pass/receive calls, if outside office -better say, not near a WIFI or HotSpot-, you pass calls through GSM and receive the calls from your asterisk through a GSM gateway (other BT or WIFI gsm phone, GSM gw device, ...) in your office -- Daniel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call queue problem
Hi all, We have an 8 agent support desk setup with 2 call queues running Asterisk 1.4.5. Every so often agents will receive a call from the queue that only rings once not allowing them time to answer. The call doesn't seem to be dropped, just seems to go to voicemail. The agents are also mentioning they do not receive the 30 second wrapuptime I have specified in queues.conf. We're using polycom 501 phones and I'm adding agents to the queues using Addqueuemember(). I believe I have the call limits and limitonpeer settings right in sip.conf. The only difference between the two queues is one has a higher weight. Any suggestions would be greatly appreciated. [our-support-queue] musicclass = default strategy = leastrecent timeout = 12 retry = 15 wrapuptime=30 weight=0 autopause=yes maxlen=0 joinempty=strict leavewhenempty=strict ringinuse=no context=queue-out periodic-announce-frequency=60 announce-holdtime=no periodic-announce=my-prompt-29 Thanks, Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing Variables to Voicemail's Email Notification
On 8/20/07, 0xception [EMAIL PROTECTED] wrote:
Okay for a quick report back, that all seems to work...
I am assuming that means that when doing ${VM_CIDNAME:15} you got just
the ticket number.
Thanks for reporting back.
Thanks a lot.
Not much to report back other then that :)...
On 8/20/07, C F [EMAIL PROTECTED] wrote:
After rethinking.
I'm not sure if this works, but please report back after testing.
The idea would be that the CIDNAME should not be in the subject just
the ticket number, and the ticket number should not be in the email
body just the CIDNAME.
Please try the following and report back.
exten = _X.,1,Set(BLANKS= );actual 15 spaces, since
CIDName on PSTN should never be longer, and should realy be padded
with blank spaces.
exten =
_X.,n,Set(CALLERID(name)=${CALLERID(name)}${BLANKS:${LEN(${CALLERID(name)})}})
;the above just pads the CIDNAME with blanks so you know for sure it's
at least 15 char long, yes I know if the len of cidname is longer than
blanks then blah.
exten =
_X.,n,Set(CALLERID(name)=${CALLERID(name):0:15}=TicketNum:1234)
;this makes sure that it is not longer than 15 plus the ticketnumber.
exten = _X.,n,Voicemail(blah)
In voicemail.conf
emailsubject=${VM_CIDNAME:15}
If this should work then the subject should be: TicketNum:1234
emailbody=New voicemail from ${VM_CIDNAME:0:15} balh.
Again, I'm not sure this will work, please test and report back.
Thank you
On 8/20/07, C F [EMAIL PROTECTED] wrote:
While I don't have an answer on how to access channel variables from
voicemail.conf, for the problem you mention this should help.
Change CALLERID(name) to your ticket number and then use VM_CIDNAME in
the subject line.
If you don't want to lose the original CIDNAME then just add your
ticket number like this:
Set(CALLERID(name)=${CALLERID(name)}
TICKETNUMBER:12345)
On 8/20/07, 0xception [EMAIL PROTECTED] wrote:
Is there a way, other then recoding the entire voicemail application,
to
pass dialplan variables to the voicemail application and to the email
notifications of new voicemail.
For example in our small tech support queue i would like to pass the
ticket
number with the email notification that a new support voicemail was
left.
I've tried simply replacing the ${VM_WHATEVER} w/ the actual variable
name
inside the voicemail.conf file, I've also tried setting the VM
variables
directly before the voicemail application call in the dial plan...
both of
these fail.
Anyone else know of another way?
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Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
I thought that was what the flashing LEDs on the front of the server's HDs were for (besides showing activity). Some I have seen also have an LED near the power button to indicate HD problems. I guess if you are building your own boxen and not using enterprise grade servers, this is not the case. Thanks, Steve Totaro C F wrote: While hardware RAID tend to be more reliable, it is not always possible to properly monitor hardware raid in a linux system, unless you write your own code. Consider this: ~# cat /proc/mdstat Personalities : [raid1] md0 : active raid1 sdb2[2](F) sda2[1] 76139968 blocks [2/1] [_U] unused devices: none The above is from an active system that one hdd failed. It would take way longer to find such a thing on a hardware raid. Unless it came with a program that emails me notification on such a failure. On 8/20/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? I will be highly appreciate your feedback on this regard. -- Thanks Regards, Vidura Senadeera, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CLI Question
When I use the CLI (asterisk -r) I get all sorts of info scrolling past about current activity such as... -- Executing Macro(SIP/7110-b1d316e0, callrecord|7134) in new stack -- Executing NoOp(SIP/7110-b1d316e0, Call Record Macro REC7134 ) in new stack -- Executing GotoIf(SIP/7110-b1d316e0, 0?4:3) in new stack -- Goto (macro-callrecord,s,3) -- Executing GotoIf(SIP/7110-b1d316e0, 0?4:8) in new stack -- Goto (macro-callrecord,s,8) -- Executing NoOp(SIP/7110-b1d316e0, ) in new stack -- Executing GotoIf(SIP/7110-b1d316e0, 1?8:150) in new stack -- Goto (macro-simpleexten,s,8) and soforth... I'm trying to learn the CLI and so I type something like: sip showtabtab and I get a list of other options. BUT, before I get through reading what is on the screen, a call comes and and scrolls up the screen with the info above. Is there a flag to pass to rasterisk to tell it only show info related to my queries and don't keep showing me all the current call status? (less verbose?) Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which GUI for ACD edition ?
Olivier wrote: Hello, I want to safely delegate ACD edition to a system administrator who has no knowledge of Linux nor Asterisk. More precisely, I want him to be able to edit and change menus such as : Type 1 for management; 2 for support; 3 for sales department. I could teach this administrator what Asterisk config files are but I'm wondering if any GUI exists for such task (editing a vocal menu tree). Maybe something not related to telephony could be used for that. Any idea ? Best regards Oliver, You can check out DialplanPro if you like. Its very easy and graphical to create dialplans for our users. http://www.datatrakpos.com/pos/datatalk/Default.aspx Unfortunately, roles/permission groups are not yet implemented (another reason why its still in beta) so the user would have the full gambit of functionality. However, if you feel comfortable with the admin, then you can simply create the initial menus and train her to modify only those menus. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with Dynanic Users (BUG?)
Tim Groeneveld wrote: On Tuesday 21 August 2007 12:32:12 am Mark Michelson wrote: When users call 510 then, it actually does ring everyone who has called 511. The problem is when the operator comes to pick up the call. The operator hears nothing, and the user still hears the Music on Hold. Not only that, but after about 5 seconds, the operators call gets dropped. Is there anything that I am doing wrong? Remove the Answer() before the call to Queue(). See if that corrects the problem. No, that did not help at all. Maybe I should use AgentLoginCallback? Thanks a million, Tim Groeneveld AgentCallbackLogin is NOT recommended as it has always been very buggy. In the interest of sanity, I just tried the same setup you have using 1.4 and then using trunk. It worked in 1.4, but I had the same problem as you when trying it with trunk. I'd recommend opening a bug report so that this can be further analyzed. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
On Tue, 21 Aug 2007, Steve Totaro wrote: I thought that was what the flashing LEDs on the front of the server's HDs were for (besides showing activity). Some I have seen also have an LED near the power button to indicate HD problems. I guess if you are building your own boxen and not using enterprise grade servers, this is not the case. Some of us take enterprise grade servers, but decide to not use some of the facilities they offer - for a whole host of reasons. Eg. right now Dell[1] don't offer RAID-6 in hardware, so I do it in software, bypassing their on-board RAID-5 controllers. And when a server's 300 miles away (as some of mine are) a blinky light isn't much use )-: Gordon [1] Not that I particulatly regard Dell as enterprise grade but that's for another list, and was just using it as an example here. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
Administrator TOOTAI wrote: Gordon Henderson a écrit : On Mon, 20 Aug 2007, Steve Totaro wrote: Well chan_bluetooth is really amazing (especially if your phone does not support SIP). You connect your phone via bluetooth to your asterisk box and it becomes a channel type. You can use it as an extension(FXS) or a phone line (FXO). I believe you can send and receive SMS through the phone/Asterisk as well. Chan_bluetooth README is in the asterisk-addons trunk and gives you basic instruction on setting it up. You get several added pieces of functionality with this setup. SMS send and receive through your phone using Asterisk?, FXO failover or LCR, FXS where your cell phone becomes an extension. Does FSX really work? Can I really use my mobile as an extension? How do I make my mobile phone dial out over bluetooth rather than it's GSM connection? If this really is the case, does it then create the holy grail of one phone for everything? Does it support one to many? I'm imagining an office where I connect a bluetooth dongle on the end of a long USB cable up to the middle of the room, into the PBX which many mobile phones can then access and let the punters use their mobiles to make/take calls via the PBX when in the office and use them as normal mobile when out of the office... So in the office, mobile rings via bt, when no bt connection, then it rings out via the PSTN to the mobile. (or via another GSM gateway) But I'm really clueless on bluetooth use - other than sending cheeky messages to other peoples mobiles and connecting my borg implant to my mob when driving! I also know I can probably do this with mob's that have WiFi and SIP clients too, however ... Yes, and it's working great, particulary with Nokia's: you tell them to try to call at first through Internet, if it fails, fallback to GSM. Once you're in the office with WIFI, device get automatically connected to the net and you can pass/receive calls, if outside office -better say, not near a WIFI or HotSpot-, you pass calls through GSM and receive the calls from your asterisk through a GSM gateway (other BT or WIFI gsm phone, GSM gw device, ...) in your office I should correct myself, it was called chan_bluetooth but there was an abandoned project with the same name. Just for clarity, the app you should be researching is chan_mobile. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI Question
For 1.4: core set verbose 2 For 1.2: set verbose 2 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Andersen Sent: Tuesday, August 21, 2007 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CLI Question When I use the CLI (asterisk -r) I get all sorts of info scrolling past about current activity such as... -- Executing Macro(SIP/7110-b1d316e0, callrecord|7134) in new stack -- Executing NoOp(SIP/7110-b1d316e0, Call Record Macro REC7134 ) in new stack -- Executing GotoIf(SIP/7110-b1d316e0, 0?4:3) in new stack -- Goto (macro-callrecord,s,3) -- Executing GotoIf(SIP/7110-b1d316e0, 0?4:8) in new stack -- Goto (macro-callrecord,s,8) -- Executing NoOp(SIP/7110-b1d316e0, ) in new stack -- Executing GotoIf(SIP/7110-b1d316e0, 1?8:150) in new stack -- Goto (macro-simpleexten,s,8) and soforth... I'm trying to learn the CLI and so I type something like: sip showtabtab and I get a list of other options. BUT, before I get through reading what is on the screen, a call comes and and scrolls up the screen with the info above. Is there a flag to pass to rasterisk to tell it only show info related to my queries and don't keep showing me all the current call status? (less verbose?) Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI Question
On Tue, 21 Aug 2007, Bill Andersen wrote: When I use the CLI (asterisk -r) I get all sorts of info scrolling past about current activity such as... -- Executing Macro(SIP/7110-b1d316e0, callrecord|7134) in new stack -- Executing NoOp(SIP/7110-b1d316e0, Call Record Macro REC7134 ) in new stack -- Executing GotoIf(SIP/7110-b1d316e0, 0?4:3) in new stack -- Goto (macro-callrecord,s,3) -- Executing GotoIf(SIP/7110-b1d316e0, 0?4:8) in new stack -- Goto (macro-callrecord,s,8) -- Executing NoOp(SIP/7110-b1d316e0, ) in new stack -- Executing GotoIf(SIP/7110-b1d316e0, 1?8:150) in new stack -- Goto (macro-simpleexten,s,8) and soforth... I'm trying to learn the CLI and so I type something like: sip showtabtab and I get a list of other options. BUT, before I get through reading what is on the screen, a call comes and and scrolls up the screen with the info above. Is there a flag to pass to rasterisk to tell it only show info related to my queries and don't keep showing me all the current call status? (less verbose?) Either start asterisk with no -v's or type: set verbose 0 at the prompt. Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [hylafax-users] asterisk, iaxmodem, hylafax quality problem
Lee Howard wrote: Artifex Maximus wrote: zttest is often on 99.975586% with final result: --- Results after 67 passes --- Best: 99.987793 -- Worst: 99.951172 -- Average: 99.973764 This is unacceptable for faxing, and it is evidence of the underlying problem also causing your faxes to come through with poor quality. Sadly both my production machine and a test machine I have here (both with TDM-400P's in them) have results that match this. (Shame really, I'd like to replace the real modem on a line on the production server with an IAXmodem process). 0: 2087872259IO-APIC-edge timer 7: 0IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 1 IO-APIC-level acpi 14: 18440124IO-APIC-edge ide0 15:4456445IO-APIC-edge libata 169:4878102 IO-APIC-level eth0 177: 2086847525 IO-APIC-level wctdm24xxp 185: 2086810653 IO-APIC-level wct4xxp Notice the priorities here... and that your Zaptel cards come *last*, after eth0, after IDE. Each of those Zap cards are going to generate an interrupt once every millisecond when in use. You can hopefully imagine how IDE or eth0 activity would interfere, since they have a higher priority than the Zap cards. The weird thing is, looking at the motherboard manual for my test machine, The lower the Interrupt does not neccesarily mean the higher the priority. Eg. 8 to 15 have a higher priority than 3 to 7. On the bright side on that machine there is an IRQ - slot allocation system in the BIOS. On the down side, it appears to do bugger all. (as below) 0: 63 IO-APIC-edge timer 1: 2 IO-APIC-edge i8042 6: 5 IO-APIC-edge floppy 7: 0 IO-APIC-edge parport0 8: 1 IO-APIC-edge rtc 9: 0 IO-APIC-fasteoi acpi 12: 3 IO-APIC-edge i8042 14: 4744 IO-APIC-edge ide0 15: 11412 IO-APIC-edge ide1 17:428 IO-APIC-fasteoi eth0 19: 40266 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb2, uhci_hcd:usb3, uhci_hcd:usb4, uhci_hcd:usb5 20: 0 IO-APIC-fasteoi VIA8237 21:1284151 IO-APIC-fasteoi wctdm I've also noticed that on the production server, the card not only has the lowest priority but is now sharing an IRQ (probably happened last time I saw fit to shut the machine down). 0: 2931864860 XT-PIC timer 1: 1659 XT-PIC i8042 2: 0 XT-PIC cascade 3:270 XT-PIC uhci_hcd:usb3 4:3231957 XT-PIC serial 5: 0 XT-PIC uhci_hcd:usb4 6: 13146224 XT-PIC dpti0 7: 475736174 XT-PIC eth0, eth1 8: 4 XT-PIC rtc 10: 0 XT-PIC uhci_hcd:usb2, uhci_hcd:usb5 11: 2931432215 XT-PIC ehci_hcd:usb1, wctdm 14:759 XT-PIC ide0 Guess I'll try disabling the USB controllers and moving cards round again. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which GUI for ACD edition ?
Replying to myself, as I've just discovered AsteriskNow screenshots (in German !), AsteriskNow seems to offer interesting features for that. I thought I should let this list readers know that. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call queue problem
Nick Whitaker wrote: Hi all, We have an 8 agent support desk setup with 2 call queues running Asterisk 1.4.5. Every so often agents will receive a call from the queue that only rings once not allowing them time to answer. The call doesn't seem to be dropped, just seems to go to voicemail. The agents are also mentioning they do not receive the 30 second wrapuptime I have specified in queues.conf. We're using polycom 501 phones and I'm adding agents to the queues using Addqueuemember(). I believe I have the call limits and limitonpeer settings right in sip.conf. The only difference between the two queues is one has a higher weight. Any suggestions would be greatly appreciated. [our-support-queue] musicclass = default strategy = leastrecent timeout = 12 retry = 15 wrapuptime=30 weight=0 autopause=yes maxlen=0 joinempty=strict leavewhenempty=strict ringinuse=no context=queue-out periodic-announce-frequency=60 announce-holdtime=no periodic-announce=my-prompt-29 Thanks, Nick I hate to say it but for any call center (or even PBX) that is not dev or does not absolutely need the functionality in 1.4.x, I would use the latest release of 1.2.x. One very nice function in 1.4 is whisper coaching but I can live without that in place of stability. I am rolling one server back as we speak. After running for a few hours, the stop now command does nothing. Ctrl-C stops it but I cannot be sure what other bugs are there so it is 1.2.X for me. I will use one box running 1.4 trunk for the purpose of chan_mobile unless that can be back ported to 1.2.x but I am not finding any docs on that. This box will ONLY handle chan_mobile functions with a separate box for SMS (Kannel). If it proves solid enough, maybe I will eliminate the Kannel box. Much less headaches with 1.2.x. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI Question
On Tuesday 21 August 2007 10:58:16 Bill Andersen wrote: When I use the CLI (asterisk -r) I get all sorts of info scrolling past about current activity such as... -- Executing Macro(SIP/7110-b1d316e0, callrecord|7134) in new stack -- Executing NoOp(SIP/7110-b1d316e0, Call Record Macro REC7134 ) in new stack -- Executing GotoIf(SIP/7110-b1d316e0, 0?4:3) in new stack -- Goto (macro-callrecord,s,3) -- Executing GotoIf(SIP/7110-b1d316e0, 0?4:8) in new stack -- Goto (macro-callrecord,s,8) -- Executing NoOp(SIP/7110-b1d316e0, ) in new stack -- Executing GotoIf(SIP/7110-b1d316e0, 1?8:150) in new stack -- Goto (macro-simpleexten,s,8) and soforth... I'm trying to learn the CLI and so I type something like: sip showtabtab and I get a list of other options. BUT, before I get through reading what is on the screen, a call comes and and scrolls up the screen with the info above. Is there a flag to pass to rasterisk to tell it only show info related to my queries and don't keep showing me all the current call status? (less verbose?) Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users core set verbose 0 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [hylafax-users] asterisk, iaxmodem, hylafax quality problem
Thomas Kenyon wrote: The weird thing is, looking at the motherboard manual for my test machine, The lower the Interrupt does not neccesarily mean the higher the priority. Eg. 8 to 15 have a higher priority than 3 to 7. Correct. IRQ 2 bridges to IRQ 8. Thus the priority order is: 0, 1, 2, 8, 9, 10, 11, 12, 13, 14, 15, 3, 4, 5, 6, 7 This is one reason why on modern Linux kernels where the ATA (IDE hard drive) driver is permitted to be very resource-greedy the serial ports on IRQs 3 and 4 can lose requisite attention for high-throughput serial devices (like Class 2.1 fax modems). And just think of those poor, poor printers on the LPT port, IRQ 7... The end-result is that the already slim pickings on IRQs gets reduced even further to a very narrow band for add-on PCI devices, usually just 9, 10, and 11 on many systems. This is one reason for APIC, although it's quite buggy in many kernels and motherboard BIOSes. Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
Steve Totaro wrote: I should correct myself, it was called chan_bluetooth but there was an abandoned project with the same name. Just for clarity, the app you should be researching is chan_mobile. Thanks, Steve Totaro It was actually never called chan_bluetooth. That was one of the suggestions we got, but we couldn't use it because it was already in use. The original name (which was never committed) was chan_cellphone. -- Jason Parker Digium ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with overlap dial and Xorcom Astribank BRI
I have a strange problem with overlap dialing. I installed an asterisk server between a Siemens HiCom PBX and our telephony provider. Everything is working fine except some strange problems with the dialing of the fax (connected to the HiCom PBX). It seems to me that if dialing takes too long Asterisk just hangs up the channel without recognizing that the fax machine is still dialing: (Fax gets HiCom dialton) (Fax dials 0 to get Asterisk dialtone) -- Accepting overlap voice call from 'xxx' to 'unspecified' on channel 0/2, span 8 -- Starting simple switch on 'Zap/23-1' (Fax starts dialing...) -- Hungup 'Zap/23-1' (Fax still dialing) The time between 'Starting...' and 'Hungup' is about 5 secs. If the number is dialed in this interval everything is fine. The phones (also connected to the HiCom PBX) also use overlapdialing, but there I can dial as slow as I want as long as I press a digit every 5 secs. (But the phones are not connected to an analog line like the fax is.) I thought that I could fix this with a DigitTimeout of 10 secs. But this does not work and interestingly it does not affect the time when Asterisk hangs up on the fax. This time is still 5 secs. But I can see in the logs that the digit timeout was modified and it affects the timeout for dialing from the phones. How can I fix this behaviour? I'm using the Asterisk package 1:1.2.19~dfsg-0.4227 from the Xorcom debian repository. Thanks, Lars -- Life would be so much easier if we could just look at the source code. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] compatibility of PRI Two B channel transfers TBTC/2BTC
Matt Florell wrote: Hello, A client has asked for Two B channel Transfer capability (known as TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG Path Replacement) in a new Asterisk system and so I researched the capability and came up with quite a few gaps in documentation. From what I've gathered, the official Digium statement is that is works with DMS100 only, and only in Asterisk 1.4.X : http://kb.digium.com/entry/26/140/ This definitely works. I wrote it and tested it myself. Although in a bugtracker posting with a patch from over two years ago, Matt Fredrickson from Digium says that it works with 5ESS under Asterisk 1.2.X: http://bugs.digium.com/view.php?id=3554 There's an implementation I scrubbed out a couple of years ago, but I think there was a bug in it that I was not able to fix. When push came to shove, and I needed a switch to debug it on (and when I had more time to work on it), nobody offered switch access so that I could debug it. So I don't think it is working right now. There are also bounties and claims of this feature working on NI2 protocol(although no patches posted) on the voip-info.org Wiki: http://www.voip-info.org/wiki/view/Asterisk+bounty+PRI+2B+channel+transfer+for+NI2+PRI+line http://www.voip-info.org/wiki/index.php?page=Asterisk%20bounty%20PRI%202B%20channel%20transfer Yeah, well, they're really old :-) Try getting a hold of the authors. As for actually using this feature, you apparently need to add the following lines to the zapata.conf section that you want to be able to use 2BCT: facilityenable = yes transfer=yes Yes, that is correct. To execute the transfer, you need to use the Transfer cmd within Asterisk: http://voipinfo.org/wiki/view/Asterisk+cmd+Transfer This is incorrect. If you have transfer=yes and facility=yes in zapata.conf for both channels, and both channels meet all the other criteria for TBCT (on the same PRI, and a few other switch dependent rules), when a native bridge is attempted, it automatically attempts to pass the calls up to the upstream switch. If it is successful, your calls will remain up, but you will get a hangup in asterisk on both calls. And according to this post, you can only do 2BCT transfers if the first call is inbound: http://www.mail-archive.com/[EMAIL PROTECTED]/msg25131.html That's a rule only for DMS100. Does 2BCT work with DMS100 and 5ESS right now? Last I heard (a couple of years ago) it doesn't. Are there people using this in production right now that can shed some more light on exactly how they are using it, and executing the transfers? I hope I answered your questions :-) -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
Hi, Are all models with bluetooth capabiilty able to dial using bluetooth ??? In Brazil some telephony companies offer a little box to conect your fixed land line. Probably a bluetooth to Analog line gateway. However, only cellphones with especial firmware can be used. So, what cellphones can I use to do it ?? Searching I found this one that seems a very good option... HP iPAQ 514 Luis A P Barbosa 2007/8/21, Administrator TOOTAI [EMAIL PROTECTED]: Gordon Henderson a écrit : On Mon, 20 Aug 2007, Steve Totaro wrote: Well chan_bluetooth is really amazing (especially if your phone does not support SIP). You connect your phone via bluetooth to your asterisk box and it becomes a channel type. You can use it as an extension(FXS) or a phone line (FXO). I believe you can send and receive SMS through the phone/Asterisk as well. Chan_bluetooth README is in the asterisk-addons trunk and gives you basic instruction on setting it up. You get several added pieces of functionality with this setup. SMS send and receive through your phone using Asterisk?, FXO failover or LCR, FXS where your cell phone becomes an extension. Does FSX really work? Can I really use my mobile as an extension? How do I make my mobile phone dial out over bluetooth rather than it's GSM connection? If this really is the case, does it then create the holy grail of one phone for everything? Does it support one to many? I'm imagining an office where I connect a bluetooth dongle on the end of a long USB cable up to the middle of the room, into the PBX which many mobile phones can then access and let the punters use their mobiles to make/take calls via the PBX when in the office and use them as normal mobile when out of the office... So in the office, mobile rings via bt, when no bt connection, then it rings out via the PSTN to the mobile. (or via another GSM gateway) But I'm really clueless on bluetooth use - other than sending cheeky messages to other peoples mobiles and connecting my borg implant to my mob when driving! I also know I can probably do this with mob's that have WiFi and SIP clients too, however ... Yes, and it's working great, particulary with Nokia's: you tell them to try to call at first through Internet, if it fails, fallback to GSM. Once you're in the office with WIFI, device get automatically connected to the net and you can pass/receive calls, if outside office -better say, not near a WIFI or HotSpot-, you pass calls through GSM and receive the calls from your asterisk through a GSM gateway (other BT or WIFI gsm phone, GSM gw device, ...) in your office -- Daniel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [hylafax-users] asterisk, iaxmodem, hylafax quality problem
Lee Howard wrote: Correct. IRQ 2 bridges to IRQ 8. Thus the priority order is: 0, 1, 2, 8, 9, 10, 11, 12, 13, 14, 15, 3, 4, 5, 6, 7 My zttest results weren't quite as bad as the previous poster. Home Machine. --- Results after 113 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.994452 Work Machine. --- Results after 115 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.993920 Presumably the work machine will marginally improve once the card has it's IRQ to itself again. Are these results good enough to be able to use TxFax/RxFax/iaxmodem? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisks addon make problem
Hi on debian iam try to make i get this problem any suggestions. make res_config_mysql.so cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o res_config_mysql.o res_config_mysql.c res_config_mysql.c:75: warning: data definition has no type or storage class res_config_mysql.c:77: warning: data definition has no type or storage class res_config_mysql.c: In function âconfig_mysqlâ: res_config_mysql.c:430: error: too few arguments to function âast_config_internal_loadâ res_config_mysql.c: At top level: res_config_mysql.c:463: warning: initialization from incompatible pointer type res_config_mysql.c: In function âunload_moduleâ: res_config_mysql.c:503: error: âSTANDARD_HANGUP_LOCALUSERSâ undeclared (first use in this function) res_config_mysql.c:503: error: (Each undeclared identifier is reported only once res_config_mysql.c:503: error: for each function it appears in.) res_config_mysql.c: In function âparse_configâ: res_config_mysql.c:541: warning: assignment discards qualifiers from pointer target type res_config_mysql.c:548: warning: assignment discards qualifiers from pointer target type res_config_mysql.c:555: warning: assignment discards qualifiers from pointer target type res_config_mysql.c:562: warning: assignment discards qualifiers from pointer target type res_config_mysql.c:569: warning: assignment discards qualifiers from pointer target type res_config_mysql.c:576: warning: assignment discards qualifiers from pointer target type make: *** [res_config_mysql.o] Error 1 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with overlap dial and Xorcom Astribank BRI
On Tue, Aug 21, 2007 at 06:08:03PM +0200, Lars Bensmann wrote: I have a strange problem with overlap dialing. I installed an asterisk server between a Siemens HiCom PBX and our telephony provider. Everything is working fine except some strange problems with the dialing of the fax (connected to the HiCom PBX). It seems to me that if dialing takes too long Asterisk just hangs up the channel without recognizing that the fax machine is still dialing: (Fax gets HiCom dialton) (Fax dials 0 to get Asterisk dialtone) -- Accepting overlap voice call from 'xxx' to 'unspecified' on channel 0/2, span 8 Not sure what the problem is, but a way around it: Any chance you could disable the overlap dialing and get the PBX to send the whole number in one go? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 99 bottles of beer
SIP wrote: Russell Bryant wrote: Steve Murphy wrote: How about this one: from an extensions.conf that someone posted on the internet, I think, and I converted to AEL; I'm sorry, but I can't find the original author. (If anybody can find his post, I'd love to give him credit.) I did test this out, and it works; just put a call to the macro ( guessgame(); ) in an extension in your dialplan Nice! While we're on the subject of silly but fun dialplan bits, check out my TV remote extension. When I moved a few months ago, there was a while when I couldn't find the wireless keyboard that I usually use as my TV remote to control MythTV. So, I built dialplan so I could use a wireless phone as my remote, instead. The dialplan reads digits from the phone and sends the correct commands to a MythTV network control interface for the frontend application. I posted my tested .conf version and the untested AEL version to the MythTV wiki. The AEL version would probably be prettier with macros, now that I think of it ... http://www.mythtv.org/wiki/index.php/Controlling_MythTV_from_any_phone_using_Asterisk Wow... that's just wow. Words fail me. I'm not saying it isn't cool... just... wow. ;) It's a nerd explosion in your mouth! -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisks addon make problem
On 23:08, Tue 21 Aug 07, ram wrote: Hi on debian iam try to make i get this problem What version of Debian? What version of asterisk-addons? Is this an upgrade? We need more info -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 99 bottles of beer
On Tue, 21 Aug 2007, Russell Bryant wrote: Nice! While we're on the subject of silly but fun dialplan bits, check out my TV remote extension. When I moved a few months ago, there was a while when I couldn't find the wireless keyboard that I usually use as my TV remote to control MythTV. So, I built dialplan so I could use a wireless phone as my remote, instead. The dialplan reads digits from the phone and sends the correct commands to a MythTV network control interface for the frontend application. I posted my tested .conf version and the untested AEL version to the MythTV wiki. The AEL version would probably be prettier with macros, now that I think of it ... http://www.mythtv.org/wiki/index.php/Controlling_MythTV_from_any_phone_using_Asterisk And practical :) Almost every room in my house has a phone -- if I could teach my kids to put them back where they belong. This could easily be extended to recognize which phone was used so it could control the Myth FE in that room. Also, it could/should be extended to control x10 devices as well... To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [hylafax-users] asterisk, iaxmodem, hylafax quality problem
Thomas Kenyon wrote: My zttest results weren't quite as bad as the previous poster. Home Machine. --- Results after 113 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.994452 This should be perfectly fine. Work Machine. --- Results after 115 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.993920 This should work fine as well. Are these results good enough to be able to use TxFax/RxFax/iaxmodem? I can't really speak for TxFax/RxFax, but they should be fine for iaxmodem. Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SET EXTENSION
Hello All,
How can I SET EXTENSION from context?
This is my context: -
[docall-usa]
exten = _NXXNXX,1,Answer
exten = _NXXNXX,n,Set() ; What do I need to set here
exten = _NXXNXX,n,DeadAGI(dousacall.php|1)
exten = _NXXNXX,n,Hangup
I need to add 1 in front of ${EXTEN} and then send the call to dousa.php.
Set(CALLERID(number)=1${EXTEN}) will set the callerID to that
extension... But I want to add '1' to my extension.
Can anyone please put some light... what I am missing here...
Cheers,
Nitesh
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Re: [asterisk-users] TC400B and show transcoder
and the card has its own interrupt - 193:18715321896779 IO-APIC-level tc400b But when ever we need to do a transcode, ie playing back a wav file on a g729 channel, the audio is complete rubbish, with a lot of stutters in it (sounds like a recording does when you upload a file in the wrong sample rate etc) - Try to compare the frame size you are receiving from asterisk and set your phone to transmit the same frame size. I would guess the card appears to have problems when the frame size is different. Please try and report back. I am curious about this. the file that we are playing back is a wav file that has existed on the system and has been successfully played back with the soft g729 transcoding and also plays back fine when the channel is alaw, just not when the channel is g729. The same issue occurs when a transcode has to happen from a handset to a IP trunk, eg alaw on the handset and g729 on the trunk channel, the audio stream is non comprehensible. Andres http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
Jason Parker wrote: Steve Totaro wrote: I should correct myself, it was called chan_bluetooth but there was an abandoned project with the same name. Just for clarity, the app you should be researching is chan_mobile. Thanks, Steve Totaro It was actually never called chan_bluetooth. That was one of the suggestions we got, but we couldn't use it because it was already in use. The original name (which was never committed) was chan_cellphone. Thank you for correcting my correction ;-P Bottom line, the correct app name is chan_mobile. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
Hi, Gustavo: [EMAIL PROTECTED] wrote: Hi all and thanks for every suggest about my problem, I found that my TDM400P was sharing IRQ with onboard sound device using cat /proc/interrupts, lspci -v and lspci -vb. When I disable all unnecessary hardware on my machine and test it, clicking sounds continue on the line with the same intensity; again using lspci -vb i found that: 01:00.0 VGA compatible controller: VIA Technologies, Inc. Unknown device 3230 (rev 11) (prog-if 00 [VGA]) Subsystem: Micro-Star International Co., Ltd. Unknown device 7253 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 11 Memory at c000 (32-bit, prefetchable) Memory at dd00 (32-bit, non-prefetchable) Capabilities: [60] Power Management version 2 Capabilities: [70] AGP version 3.0 04:04.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0003 Flags: bus master, medium devsel, latency 64, IRQ 11 I/O ports at be00 Memory at dfaff000 (32-bit, non-prefetchable) Capabilities: [40] Power Management version 2 Now TDM card share IRQ 11 with onboard vga controller. I have a sata raid 1 level running on the box too and cat /proc/interrupts show me: 0: 23572057 0IO-APIC-edge timer 1:196 0IO-APIC-edge i8042 6: 3 0IO-APIC-edge floppy 7: 0 0IO-APIC-edge parport0 8: 0 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 14: 66 0IO-APIC-edge ide0 209:3663990 0 IO-APIC-level eth0 217: 403070 0 IO-APIC-level libata 225: 95602389 0 IO-APIC-level wctdm NMI: 3824180 LOC: 23572106 23572083 ERR: 0 You must ignore the IRQ flag in the lspci output when your system uses IO-APIC. Your /proc/interrupts doesn't seem to show a shared IRQ... are we sure this is the real cause of the problem? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIf not working with ${EXTEN} for me in 1.4.8
On Sat, 2007-08-18 at 21:11 -0700, Ira wrote:
At 08:29 PM 8/18/2007, you wrote:
exten = _1NXXNXX,1,GotoIf($[${EXTEN} = 15554441212]?100)
Where?
I the only variable I am using is ${EXTEN} and as far as I can see I
have a dollar sign on each ${EXTEN}.
I think it's this one.
GotoIf($[${EXTEN} = 15554441212]?100)
Ira
Both issues have to be fixed for this to work right, the $ issue and the
quote issue:
exten = _1NXXNXX,1,GotoIf($[${EXTEN} = 15554441212]?100)
or you can, in this case, drop the quotes on both sides:
exten = _1NXXNXX,1,GotoIf($[${EXTEN} = 15554441212]?100)
Best of luck,
murf
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Steve Murphy
Software Developer
Digium
smime.p7s
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Re: [asterisk-users] CDR Disposition Value with ODBC
On Sat, 2007-08-18 at 22:11 -0700, Douglas Warren Garstang wrote: It looks like when you use odbc for CDR storage, rather than getting a Dispositon string like ANSWERED, CONGESTION etc, you'll get an integer (1,2,4,8). Does anyone know where I can find what strings (ANSWERED etc) these integers map to? Doug. Sure, see include/asterisk/cdr.h: #define AST_CDR_NULL0 #define AST_CDR_FAILED (1 0) #define AST_CDR_BUSY(1 1) #define AST_CDR_NOANSWER(1 2) #define AST_CDR_ANSWERED(1 3) So, FAILED = 1 BUSY = 2 NOANSWER = 4 ANSWERED = 8 0 means that no disposition was set. murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisks addon make problem
On 8/21/07, Michiel van Baak [EMAIL PROTECTED] wrote: On 23:08, Tue 21 Aug 07, ram wrote: Hi on debian iam try to make i get this problem What version of Debian? What version of asterisk-addons? Is this an upgrade? We need more info Hi no its fresh installation. asterisk-addons-1.2.7 asterisk-addons-1.2-current.tar.gz Debian 4.0 uname -a Linux 2.6.18-5-686 #1 SMP Sun Aug 12 21:57:02 UTC 2007 i686 GNU/Linux ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phone as control interface (was 99 bottles of beer)
Steve Edwards wrote: Almost every room in my house has a phone -- if I could teach my kids to put them back where they belong. This could easily be extended to recognize which phone was used so it could control the Myth FE in that room. Also, it could/should be extended to control x10 devices as well... To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... A while back I was thinking along the lines of using a phone as a home automation interface, though I was thinking of it in combination with a voice recognitition system such as Lumenvox. It occured to me that when you want to turn the lights on, you don't really want to pick up a phone, dial a special extension, and then start using menus. What I was thinking about was what if instead of a dialtone you are brought directly to a home automation voice menu which works in parallel with your normal dial plan. If you wanted to make a call, just ignore the voice menu and dial normally. If you wanted to turn on the lights, just say lights on. or somesuch. Having a traditional dialtone seems unnecessary when you can get more function instead. The trick is doing this without giving up on the use of nice existing GUIs to manage the dialplan that we have now. I'd like some way of merging in the voice dialtone function with the existing dialplan such that initially both are active, but as soon as either a phrase is recognized or a button is pressed the system branches to one or the other, but that button or phrase is passed through to the rest of the processing and not just an extra prompt getting in the way. Does this spark anyone's imagination or ideas to implement? Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialogic support
Can someone share pointers to Asterisk's Dialogic support? Which boards are supported, driver status, and etc. Thnx ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phone as control interface (was 99 bottles of beer)
Quoting Steve Prior [EMAIL PROTECTED]: shutting off the dialtone should be pretty simple, then what is really needed is an audio Bidirectional Tee almost like a 3 way call, well I guess exactly like a 3 way call but not dialed. you have the dsp that is going to process audio on the channel, yourself, and a listener/talker interface that listens for voice, recognizes it and then converts to touchtones and dials them into the dsp (possibly muting audio to you while its doing that.) this would allow the conventional dialplan logic to support menus etc for the control. maybe something like answer immediate, bridge 3 way call to an extension context that expects dialing along with an extension that does voice recognition in a 3 way call. Either one acts on what it gets and both hang up when you do. just don't call a real person and start talking about turning lights on and off :) Steve Edwards wrote: Almost every room in my house has a phone -- if I could teach my kids to put them back where they belong. This could easily be extended to recognize which phone was used so it could control the Myth FE in that room. Also, it could/should be extended to control x10 devices as well... To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... A while back I was thinking along the lines of using a phone as a home automation interface, though I was thinking of it in combination with a voice recognitition system such as Lumenvox. It occured to me that when you want to turn the lights on, you don't really want to pick up a phone, dial a special extension, and then start using menus. What I was thinking about was what if instead of a dialtone you are brought directly to a home automation voice menu which works in parallel with your normal dial plan. If you wanted to make a call, just ignore the voice menu and dial normally. If you wanted to turn on the lights, just say lights on. or somesuch. Having a traditional dialtone seems unnecessary when you can get more function instead. The trick is doing this without giving up on the use of nice existing GUIs to manage the dialplan that we have now. I'd like some way of merging in the voice dialtone function with the existing dialplan such that initially both are active, but as soon as either a phrase is recognized or a button is pressed the system branches to one or the other, but that button or phrase is passed through to the rest of the processing and not just an extra prompt getting in the way. Does this spark anyone's imagination or ideas to implement? Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SET EXTENSION
Nitesh Divecha wrote:
How can I SET EXTENSION from context?
This is my context: -
[docall-usa]
exten = _NXXNXX,1,Answer
exten = _NXXNXX,n,Set() ; What do I need to set here
exten = _NXXNXX,n,DeadAGI(dousacall.php|1)
exten = _NXXNXX,n,Hangup
I need to add 1 in front of ${EXTEN} and then send the call to dousa.php.
Set(CALLERID(number)=1${EXTEN}) will set the callerID to that
extension... But I want to add '1' to my extension.
Can anyone please put some light... what I am missing here...
Why mess around with the dialplan when you call a PHP
script anyway? Do it in the script.
Or pass 1${EXTEN} as an argument if you really need to.
Regards,
Philipp Kempgen
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[asterisk-users] Contact: header and NAT.
Greetings, I have a problem getting Asterisk registered as a UAC against the MetaSwitch call agent, because the customer insists on running it on a NAT'd box. Thus, the Contact: field in the REGISTER request betrays the private IP address of the Asterisk box, but the source IP of the message is a public one. Most registrars don't have a problem with this, including Asterisk. However, MetaSwitch doesn't like that; it expects (whether doing IP-trust or user authentication) to contact the SIP peer at such and such IP address in the SIP binding, and expects that's what the Contact: reachability information will be too. Any way to overcome this in Asterisk? I thought about the externip= option but it did not seem to work from an internal test box that is not behind NAT. Thanks, -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phone as control interface (was 99 bottles of beer)
On 8/21/07, Steve Prior [EMAIL PROTECTED] wrote: Steve Edwards wrote: Almost every room in my house has a phone -- if I could teach my kids to put them back where they belong. This could easily be extended to recognize which phone was used so it could control the Myth FE in that room. Also, it could/should be extended to control x10 devices as well... To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... A while back I was thinking along the lines of using a phone as a home automation interface, though I was thinking of it in combination with a voice recognitition system such as Lumenvox. It occured to me that when you want to turn the lights on, you don't really want to pick up a phone, dial a special extension, and then start using menus. What I was thinking about was what if instead of a dialtone you are brought directly to a home automation voice menu which works in parallel with your normal dial plan. If you wanted to make a call, just ignore the voice menu and dial normally. If you wanted to turn on the lights, just say lights on. or somesuch. Having a traditional dialtone seems unnecessary when you can get more function instead. The trick is doing this without giving up on the use of nice existing GUIs to manage the dialplan that we have now. I'd like some way of merging in the voice dialtone function with the existing dialplan such that initially both are active, but as soon as either a phrase is recognized or a button is pressed the system branches to one or the other, but that button or phrase is passed through to the rest of the processing and not just an extra prompt getting in the way. Does this spark anyone's imagination or ideas to implement? Sparks my imagination thusly: Suppose you have a speaker phone in every room. When the phone is onhook, Asterisk automatically opens up a call to the speaker and places it in the automation context. When you pick up the phone, it grabs a different line, and drops the automation connection. Now, you can address Asterisk by saying, Computer, raise lights 20% and impress all of your trekkie friends when the lights turn up. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 99 bottles of beer
Steve Edwards wrote: Almost every room in my house has a phone -- if I could teach my kids to put them back where they belong. This could easily be extended to recognize which phone was used so it could control the Myth FE in that room. Also, it could/should be extended to control x10 devices as well... To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... Only in America ... ;) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 99 bottles of beer
On 8/21/07, Steve Edwards [EMAIL PROTECTED] wrote: To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... To control the power bar the Asterisk server is plugged into, press 5 click DAAD! The stupid phone isn't working! -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phone as control interface (was 99 bottles of beer)
On Tue, 21 Aug 2007, David Gomillion wrote: Now, you can address Asterisk by saying, Computer, raise lights 20% and impress all of your trekkie friends when the lights turn up. Sorry - it's gotta be: [1] Zen, lights up. boing Confirm. But I guess not many leftpondians might appreciate that ;-) Gordon [1] http://en.wikipedia.org/wiki/Blake%27s_7 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
My servers run in a datacenter, 50km away from my office... if a led flash, if the speaker beep... I think I'll not see/hear it ... My servers are monitored using nagios which has a plugin for software raid... so if one array goes down, I receive a mail/sms/call/... futher more, everything is on the same panel: raid, http servers, free disk space, ... I think it is better than any led flashing into the DC :-D A. On Tue, 2007-08-21 at 10:30 -0400, Steve Totaro wrote: I thought that was what the flashing LEDs on the front of the server's HDs were for (besides showing activity). Some I have seen also have an LED near the power button to indicate HD problems. I guess if you are building your own boxen and not using enterprise grade servers, this is not the case. Thanks, Steve Totaro C F wrote: While hardware RAID tend to be more reliable, it is not always possible to properly monitor hardware raid in a linux system, unless you write your own code. Consider this: ~# cat /proc/mdstat Personalities : [raid1] md0 : active raid1 sdb2[2](F) sda2[1] 76139968 blocks [2/1] [_U] unused devices: none The above is from an active system that one hdd failed. It would take way longer to find such a thing on a hardware raid. Unless it came with a program that emails me notification on such a failure. On 8/20/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? I will be highly appreciate your feedback on this regard. -- Thanks Regards, Vidura Senadeera, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialogic support
I am pretty sure you can only get Dialogic support in ABE. Thanks, Steve Totaro Wai Wu wrote: Can someone share pointers to Asterisk's Dialogic support? Which boards are supported, driver status, and etc. Thnx ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mitel 5020 IP phones
Hi: I've got a dozen Mitel 5020 IP sets and can't find out if they do SIP, or even find an administrator's manual for them. Mitel has been rather unhelpful. They only deal with partner resellers. Has anybody used these with Asterisk? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI Question
Gordon Henderson wrote: Either start asterisk with no -v's or type: set verbose 0 at the prompt. Thanks. Exactly what I needed. Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phone as control interface (was 99 bottles of beer)
Gordon Henderson wrote: On Tue, 21 Aug 2007, David Gomillion wrote: Now, you can address Asterisk by saying, Computer, raise lights 20% and impress all of your trekkie friends when the lights turn up. Sorry - it's gotta be: [1] Zen, lights up. boing Confirm. But I guess not many leftpondians might appreciate that ;-) Gordon [1] http://en.wikipedia.org/wiki/Blake%27s_7 More of a Whovian here. Have been since I was a very little guy. I guess Dr. Who would use his sonic screwdriver or bang the TARDIS with a hammer. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialogic support
On Tue, 21 Aug 2007, Wai Wu wrote: Can someone share pointers to Asterisk's Dialogic support? Which boards are supported, driver status, and etc. Which type of boards are you interested in? I don't know about other cards, but the DIVA Server ISDN cards are well supported. Armin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
I guess I am just lucky to have 24 hour manned data centers with staff that walk around looking for flashing LEDs. I am sure there is some error thrown in /var/log/messages about a failure that could be used to trigger a notification quite trivially. Thanks, Steve Arnaud Ligot wrote: My servers run in a datacenter, 50km away from my office... if a led flash, if the speaker beep... I think I'll not see/hear it ... My servers are monitored using nagios which has a plugin for software raid... so if one array goes down, I receive a mail/sms/call/... futher more, everything is on the same panel: raid, http servers, free disk space, ... I think it is better than any led flashing into the DC :-D A. On Tue, 2007-08-21 at 10:30 -0400, Steve Totaro wrote: I thought that was what the flashing LEDs on the front of the server's HDs were for (besides showing activity). Some I have seen also have an LED near the power button to indicate HD problems. I guess if you are building your own boxen and not using enterprise grade servers, this is not the case. Thanks, Steve Totaro C F wrote: While hardware RAID tend to be more reliable, it is not always possible to properly monitor hardware raid in a linux system, unless you write your own code. Consider this: ~# cat /proc/mdstat Personalities : [raid1] md0 : active raid1 sdb2[2](F) sda2[1] 76139968 blocks [2/1] [_U] unused devices: none The above is from an active system that one hdd failed. It would take way longer to find such a thing on a hardware raid. Unless it came with a program that emails me notification on such a failure. On 8/20/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? I will be highly appreciate your feedback on this regard. -- Thanks Regards, Vidura Senadeera, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisks addon make problem
On 00:18, Wed 22 Aug 07, ram wrote: On 8/21/07, Michiel van Baak [EMAIL PROTECTED] wrote: On 23:08, Tue 21 Aug 07, ram wrote: Hi on debian iam try to make i get this problem What version of Debian? What version of asterisk-addons? Is this an upgrade? We need more info Hi no its fresh installation. asterisk-addons-1.2.7 asterisk-addons-1.2-current.tar.gz Debian 4.0 uname -a Linux 2.6.18-5-686 #1 SMP Sun Aug 12 21:57:02 UTC 2007 i686 GNU/Linux Did you install libmysqlclient15-dev ? if not, please do so. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phone as control interface (was 99 bottles of beer)
Steve Prior wrote: What I was thinking about was what if instead of a dialtone you are brought directly to a home automation voice menu which works in parallel with your normal dial plan. If you wanted to make a call, just ignore the voice menu and dial normally. If you wanted to turn on the lights, just say lights on. or somesuch. Having a traditional dialtone seems unnecessary when you can get more function instead. The trick is doing this without giving up on the use of nice existing GUIs to manage the dialplan that we have now. I'd like some way of merging in the voice dialtone function with the existing dialplan such that initially both are active, but as soon as either a phrase is recognized or a button is pressed the system branches to one or the other, but that button or phrase is passed through to the rest of the processing and not just an extra prompt getting in the way. Now that the idea is coming back to me a bit, here's a possiblity. When the phone is picked up it is auto-dialed into the voice driven/home control application AGI. At this point there are three options: 1. User utters a voice command. 2. User presses a touch tone which is meant for home control. 3. User presses a touch tone meant for the dial plan. option 2 vs 3 would be determined by internal extensions starting with a given number and dial 9 to reach an outside line, so other digits could be used for home control. As soon as option 3 is detected the voice AGI stuffs the touch tone back into the processing buffer, transfers to the normal diaplan, and exits. From there the normal dialplan handles the call normally. So, does anyone know if it is possible to stuff a touch tone event back into the processing stream so it can be handled by the new dialplan context? Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contact: header and NAT.
Got this figured out. externip= does work if the other NAT-related options are also enabled, plus it appears that Trixbox (which is what the end-user was using, it seems) handles this well in its config file structure regardless. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.11 released
The Asterisk development team has released version 1.4.11. This version contains numerous bug fixes. One of these is for a security issue in chan_sip. The issue is that SIP dialog history was being stored in memory regardless if the option for this was turned on or off. This could be abused to cause a system using chan_sip to run out of memory. The security issue is documented in AST-2007-020. Affected systems include any that are using chan_sip. Also, only Asterisk 1.4 is affected. Asterisk 1.2 is not vulnerable to this issue. * http://downloads.digium.com/pub/asa/AST-2007-020.pdf The name prefix for our security advisories has been changed from ASA to AST. The ASA scheme was already in use by another company before we started using it. This release is available for download from http://downloads.digium.com/pub/telephony/asterisk/. Thank you for your support! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisks addon make problem
On 8/22/07, Michiel van Baak [EMAIL PROTECTED] wrote: On 00:18, Wed 22 Aug 07, ram wrote: On 8/21/07, Michiel van Baak [EMAIL PROTECTED] wrote: On 23:08, Tue 21 Aug 07, ram wrote: Hi on debian iam try to make i get this problem What version of Debian? What version of asterisk-addons? Is this an upgrade? We need more info Hi no its fresh installation. asterisk-addons-1.2.7 asterisk-addons-1.2-current.tar.gz Debian 4.0 uname -a Linux 2.6.18-5-686 #1 SMP Sun Aug 12 21:57:02 UTC 2007 i686 GNU/Linux Did you install libmysqlclient15-dev ? if not, please do so. -- Hi i have installed that before iam making the addons ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2007-020: Resource Exhaustion vulnerability in SIP channel driver
Asterisk Project Security Advisory - AST-2007-020 ++ | Product | Asterisk | |+---| | Summary | Resource Exhaustion vulnerability in SIP channel | || driver| |+---| | Nature of Advisory | Denial of Service | |+---| | Susceptibility | Remote Unauthenticated Sessions | |+---| | Severity | Moderate | |+---| | Exploits Known | No| |+---| |Reported On | August 9, 2007| |+---| |Reported By | Jon Moldenauer (bugs.digium.com user | || jmoldenhauer) | |+---| | Posted On | August 21, 2007 | |+---| | Last Updated On | August 21, 2007 | |+---| | Advisory Contact | Russell Bryant [EMAIL PROTECTED] | |+---| | CVE Name | CVE-2007-4455 | ++ ++ | Description | The handling of SIP dialog history was broken during the | | | development of Asterisk 1.4. Regardless of whether | | | recording SIP dialog history is turned on or off, the| | | history is still recorded in memory. Furthermore, there | | | is no upper limit on how many history items will be | | | stored for a given SIP dialog. | | | | | | It is possible for an attacker to use up all of the | | | system's memory by creating a SIP dialog that records| | | many entires in the history and never ends. It is also | | | worth noting for the sake of doing the math to calculate | | | what it would take to exploit this that each SIP history | | | entry will take up a maximum of 88 bytes.| ++ ++ | Resolution | The fix that has been added to chan_sip is to restore the | || functionality where SIP dialog history is not recorded in | || memory if it is not enabled. Furthermore, a maximum of 50 | || entires in the history will be stored for each dialog | || when recording history is turned on. | || | || The only way to avoid this problem in affected versions | || of Asterisk is to disable chan_sip. If chan_sip is being | || used, the system must be upgraded to a version that has | || this issue resolved. | ++ ++ | Affected Versions| || | Product | Release | | | | Series| | |--+-+---| | Asterisk Open Source |1.0.x| Not affected | |--+-+---| | Asterisk Open Source |1.2.x| Not affected |
[asterisk-users] Call back or some voicemail notifing.
Hello PPL, someone have any idea for notifying users that they have voicemail waiting when they will register after weren't being registered on asterisk? I need this for nokia terminal e series users. I studied sms service but seems to be only for PSTN lines. I comes with idea to receive a call from asterisk and notified that you have a voicemail. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Enable Media Atribute on 180 Ringing
Hi guys, I've made some tests with a partner and when he call to me he can't hear ring back tone. My asterisk sent 180 ringing message to him. He told me that in 180 ringing there isn't a media attributes and i need to reconfigure my side to send 180 ringing with media attributes. How can i enable this on asterisk ? thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.11 released
The Asterisk Development Team wrote: The Asterisk development team has released version 1.4.11. This release is available for download from http://downloads.digium.com/pub/telephony/asterisk/. Not quite. :) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
I have this exact same problem with two different Business Edition systems. Both are using TDM400s. Do we have an answer for this yet? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Thursday, August 16, 2007 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TDM400P FXO click sounds shadowym wrote: Please explain to me how FXO tune would fix popping and clicking sounds??? As mentioned by Stephen, if the echo canceler is improperly tuning that certainly might be possible. But moreover, if there is ambient line noise that is on the line, fxotune will try to pick the best settings on the line interface to either mitigate any line noise that it receives in the audio receive path. One other possibility is you could see if it the clicking and popping correlates to hard drive activity... if that's so, you might have a hard drive or raid controller disabling interrupts for too long. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
Michael Munger wrote: I have this exact same problem with two different Business Edition systems. Both are using TDM400s. Do we have an answer for this yet? I know this sounds silly, but if there is a chance that it is an improperly tuned echo canceller, has anyone tried using oslec. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with overlap dial and Xorcom Astribank BRI
On Tue, Aug 21, 2007 at 08:42:50PM +0300, Tzafrir Cohen wrote: Not sure what the problem is, but a way around it: Any chance you could disable the overlap dialing and get the PBX to send the whole number in one go? Mmmh. The PBX is not very friendly to program. But I will have a look. Another idea that came to my mind was to omit the leading '0' and configure the PBX to directly pass the extension to Asterisk. This would save at least two seconds so the dialing process should finish in time. Not very clean, but I hope it works. Will have a look tomorrow, Lars -- If windows is the answer, it must have been a stupid question. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialogic support
My customer has tones of DM3 cards (DM/V600, DM/N1200, and D600-2E1), they want to see if they can use them in Asterisk. My advise to them is to sell those cards and buy Sangoma E1 cards, and still have money left. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Armin Schindler Sent: Tuesday, August 21, 2007 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [asterisk-users] Dialogic support On Tue, 21 Aug 2007, Wai Wu wrote: Can someone share pointers to Asterisk's Dialogic support? Which boards are supported, driver status, and etc. Which type of boards are you interested in? I don't know about other cards, but the DIVA Server ISDN cards are well supported. Armin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
Michael Munger wrote: I have this exact same problem with two different Business Edition systems. Both are using TDM400s. Do we have an answer for this yet? You need to contact Digium technical support. They provide free support for hardware issues like this. Furthermore, since you are a BE customer, that gives you even higher priority in getting attention to your problems. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TC400B and show transcoder
Andres wrote: Try to compare the frame size you are receiving from asterisk and set your phone to transmit the same frame size. I would guess the card appears to have problems when the frame size is different. Please try and report back. I am curious about this. The problem occurs when we have external (pstn) calls coming into / out of the system (via an iax trunk), in which case we have no control over frame size, as well as occurring with handsets directly connected to the system. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom behind NAT won't register to * server behind ALG
I’ve been tearing my hair out trying to get a Polycom phone (behind a NAT) to register to an * box behind a Cisco SIP ALG. With known good credentials configured on the phone and in *, I get 403 Bad Auth when trying to register. If I put the phone onto the same LAN as * it works fine without changing any authentication parameters whatsoever. If I make the secret blank (null) on the phone and *, it’s registers fine. An X-Lite softphone works fine, and I already have a Cisco 7960 working in the same scenario but no go with the Polycom (I’ve tried a few different firmware versions, currently up to 2.02). I also have configured Polycoms behind NAT to configure to * that is not behind a SIP ALG many times without trouble…it’s just with the SIP ALG that I’ve hit this wall… I found this forum thread which seems to pinpoint the issue, but I’m not really understanding the suggested fix. Does anyone out there understand the MD5 challenge/response and URI stuff that can give me some clue what to do? HYPERLINK http://forum.voxilla.com/asterisk-support-forum/sipura-asterisk-registration-failed-wrong-password-18730.htmlhttp://forum.voxilla.com/asterisk-support-forum/sipura-asterisk-registration-failed-wrong-password-18730.html Thanks! mmastera No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.1/963 - Release Date: 8/20/2007 5:44 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom and NAT
Hi All, I have a Polycom 501 that is behind a NAT. When it registers to the Asterisk server it is using the IP address on the private network and not the public IP of the NAT address. Can someone tell me what I need to do so the phone registerers using an internet address rather than the remote network NAT address. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 99 bottles of beer
On 8/21/07, Steve Edwards [EMAIL PROTECTED] wrote: To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... Before this thread I already had a Firecracker on the server, a fair assortment of lights and the sprinklers are on an X10Pro Irrigation Controller. Damn, now I'm gonna be up all night. - dbc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi, So Easy A Caveman Could Do It!
Here you go folks: ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf If someone would be so kind as to upload to the wiki, it will be much appriciated. Thank you all who replied to my poll questions. As always, I hope this help. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom behind NAT won't register to * server behind ALG
Polycom's were simply not originally built for multi location VoIP. There is no NAT support in the Polycom's. We have several networks, being an ISP, and have found that when transversing one network say 192.168.2.x with the * box on a 192.168.1.x the polycoms were able to communicate however sustained a lot of one way audio problems. Moving thim onto the same network is the only thing we have been able to reliable do. According to Polycom Support this is what they are intended for and no definitive answer as to whether they would support Stun or another method in the future. At least as of 6 months ago. Matt ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom behind NAT won't register to * server behind ALG
On Tue, 21 Aug 2007, Matthew Warren wrote: We have several networks, being an ISP, and have found that when transversing one network say 192.168.2.x with the * box on a 192.168.1.x the polycoms were able to communicate however sustained a lot of one way audio problems. Moving thim onto the same network is the only thing we have been able to reliable do. Forgive what may be a naively misplaced line of questioning, but what precisely does this have to do with NAT as such? Unless you mean to imply otherwise, it would seem to me you are referring to 192.168.1.0/24 and 192.168.2.0/24 being intermediated by way of a router -- but not necessarily NAT'd? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom and NAT
In your sip.conf, for the user: nat=yes To send keepalives for the UDP connection (depending on how flimsy the device handling NAT is): qualify=yes From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Tuesday, August 21, 2007 17:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom and NAT Hi All, I have a Polycom 501 that is behind a NAT. When it registers to the Asterisk server it is using the IP address on the private network and not the public IP of the NAT address. Can someone tell me what I need to do so the phone registerers using an internet address rather than the remote network NAT address. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom behind NAT won't register to * server behind ALG
I think what Alex was trying to say was that Polycom IP Phones are an example of immature product development. While they look very nice and have a nice display the product doesn't compete very well compared to other manufacturers. The two most obvious flaws are that they cannot be NAT'ed so they cannot be used as Off Premise eXtensions phones and the other being that they take so long to configure and re-boot. I have a golden rule with any phone that I plan on installing for a customerIf I can't get it working within 20 minutes then don't use it. I'm afraid Polycom breaks my golden rule. With such a lot of competition in this market they should have sorted this out two years ago. -- Henry L. Coleman. Alex Balashov On Tue, 21 Aug 2007, Matthew Warren wrote: We have several networks, being an ISP, and have found that when transversing one network say 192.168.2.x with the * box on a 192.168.1.x the polycoms were able to communicate however sustained a lot of one way audio problems. Moving thim onto the same network is the only thing we have been able to reliable do. Forgive what may be a naively misplaced line of questioning, but what precisely does this have to do with NAT as such? Unless you mean to imply otherwise, it would seem to me you are referring to 192.168.1.0/24 and 192.168.2.0/24 being intermediated by way of a router -- but not necessarily NAT'd? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: Direct : ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
