[asterisk-users] bandwidth required for Asterisk running on T1
Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? Thanks in advance for your input. Regards, Mark ___ -- 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] bandwidth required for Asterisk running on T1
mark morreny wrote: Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? What kind of T1? And what codec? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - How to check HPET is on and working before installing Asterisk ?
Hi, I would like to improve our installation process. One of my requirement is to enable High Precision Event Timer support. I'm working with Debian Lenny which is now 2.6.24-based. Before installating Asterisk, zaptel and so on (and independently of those), I would like to check HPET is on and working. I ggogled and couldn't find anything useful on this. Do you have any clue ? 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] OT - How to check HPET is on and working before installing Asterisk ?
On Fri, 11 Apr 2008 08:40:20 +0200, Olivier [EMAIL PROTECTED] wrote: Before installating Asterisk, zaptel and so on (and independently of those), I would like to check HPET is on and working. $ zgrep HPET /proc/config.gz CONFIG_HPET_TIMER=y CONFIG_HPET=y CONFIG_HPET_RTC_IRQ=y CONFIG_HPET_MMAP=y Or, if your config is not exposed under /proc, then this: $ grep HPET /usr/src/linux/.config CONFIG_HPET_TIMER=y CONFIG_HPET=y CONFIG_HPET_RTC_IRQ=y CONFIG_HPET_MMAP=y As a last resort, if the kernel's config is available under /proc and you don't have the kernel source installed: $ grep hpet /proc/timer_list Clock Event Device: hpet set_next_event: hpet_legacy_next_event set_mode: hpet_legacy_set_mode HPET showing up as not working means a kernel rebuild. -- Godwin Stewart - Horwich IT services ___ -- 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] bandwidth required for Asterisk running on T1
Hi, The T1 is 32 x 64Kbps channels ; Codec is GSM. Thank you for your suggestions. Regards, Mark On Thu, Apr 10, 2008 at 11:25 PM, Alex Balashov [EMAIL PROTECTED] wrote: mark morreny wrote: Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? What kind of T1? And what codec? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Digium T1 Card Crashing Server (Dell 2950)
I contacted the T1 Card manufacturer (Digium), This problem seems similar to a known issue whose resolution is currently in progress. One of their driver engineers has some new code in Zaptel that may help in this case. I did implement it and hopeful that should resolve it. Digium has excellent customer service and quick response time. I usually don't get that from a company. On Thu, Apr 10, 2008 at 6:45 PM, Matt Florell [EMAIL PROTECTED] wrote: Hello, It might not be Digium's fault, I ran into similar problems with Dell 2950 servers and other PCIexpress cards. I even went so far as to have several components replaced by Dell on one of the affected servers to no avail. After many months of banging my head against a wall I stumbled across the following posts on the Trixbox forums: http://www.trixbox.org/forums/trixbox-forums/open-discussion/acpi-default-install-2-4-0 http://www.trixbox.org/forums/trixbox-forums/open-discussion/tb-2-4-crashing-asus-amd-and-new-dell-server-spec After talking to some computer engineers at a few companies I learned that It seems Dell does not have very good quality control on the power control chipsets that they use and so on some machines you have to disable acpi(or enable it) at the kernel level. If you do not set it correctly, when the power saving functions trigger there is a higher likelyhood that an error will occur leading to a kernel panic. This is most likely the same problem so take a look at the forum postings and try disabling/enabling acpi in your grub startup. Of course it could be something else entirely, but this problem does seem to be common with Dell 2950, and this did fix the problem for me on more than one Dell 2950. MATT--- On 4/10/08, broadband Voice [EMAIL PROTECTED] wrote: We're using PAE Kernel. On Thu, Apr 10, 2008 at 4:30 PM, Michael L. Young [EMAIL PROTECTED] wrote: BUG: soft lockup detected on CPU#1! [c044b2a4] softlockup_tick+0x96/0xa4 [c042e214] update_process_times+0x39/0x5c [c04196ff] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 . You don't happen to be running a XEN Kernel are you? I saw this problem while running CentOS 5.1 XEN kernel and if you search their bug tracking system you will see some reports about this bug. A search on google revealed some possible solutions. This was the first thought that came to my mind when I saw this. Regards, Michael L. Young (elguero) ___ -- 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 ___ -- 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 Dialtone problem
Did you provide power to the card? FXS extensions need power. On Fri, 2008-04-11 at 08:11 +0300, Murithi Martin wrote: Hi Guys, I have a TDM400P (2FXS and 2FXO) installed on my asterisk (1.4.18.1) Zaptel (1.4.9.2) running on Fedora Core 7, below are my configurations and some diagnosis I did. The problem is when I connect an analogue phone on either of the FXS channels I don't get a dialtone, I can't call any of the sip clients or even call my echo test, number, which is an context included in that of the FXS port. How can I solve this problem? Please assist. After doing a /sbin/ztcfg -vv, after listing the zaptel channels configuration it says 4 channels to configure instead of 4 channels configured. is there any additional configuration or dependency I need to load (I've done modprobe wctdm and modprobe zaptel before doing /sbin/ztcfg -vv). zaptel.conf # Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 (MASTER) fxoks=1 fxoks=2 fxsks=3 fxsks=4 # Global data loadzone = us defaultzone = us zapata.conf [trunkgroups] ; define any trunk groups [channels] ; hardware channels ;language=en ;context=from-zaptel ;signalling=fxs_ks ;rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; usedistinctiveringdetection=yes ; default usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;echotraining=800 echotraining=yes rxgain=0.0 txgain=0.0 ;group=0 callgroup=1 pickupgroup=1 immediate=no context=phone signalling=fxo_ks channel = 1 ;callerid= ;mailbox= ;group= context=phone ;;; line=2 WCTDM/0/1 FXOKS (In use) signalling=fxo_ks ;context=phone channel = 2 ;callerid= ;mailbox= ;group= context=incoming ;;; line=3 WCTDM/0/2 FXSKS (In use) signalling=fxs_ks ;callerid=asreceived ;group=0 channel = 3 context=incoming ;;; line=4 WCTDM/0/3 FXSKS (In use) signalling=fxs_ks ;callerid=asreceived ;group=0 ;context=incoming channel = 4 ;context=default [EMAIL PROTECTED] /]# /sbin/ztcfg -vv Zaptel Version: 1.4.9.2 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels to configure. Asterisk CLI core score show channeltypes Type Description Devicestate Indications Transfer -- --- --- --- Feature Feature Proxy Channel Driver no yes no IAX2 Inter Asterisk eXchange Driver (Ver 2) yes yes yes Local Local Proxy Channel Driver yes yes no SIP Session Initiation Protocol (SIP) yes yes yes Phone Standard Linux Telephony API Driver no yes no MGCP Media Gateway Control Protocol (MGCP) yes yes no Agent Call Agent Proxy Channel yes yes no -- 7 channel drivers registered. *CLI module reload chan_zap.so No such module 'chan_zap.so' *CLI -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz ___ -- 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] Strange CLI behaviour
Hi, I'm using Asterisk 1.4.17 and 1.4.19 versions, some time ago I've noticed that cli command 'core show channels' does not show all data. It returns only header or one line of data. After that, auto completition of commands (hitting TAB) freezes cli... Does anybody has the same problem? regards, Lokotes. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to check HPET is on and working before installing Asterisk ?
On Fri, Apr 11, 2008 at 08:47:16AM +0100, Horwich IT Services wrote: On Fri, 11 Apr 2008 08:40:20 +0200, Olivier [EMAIL PROTECTED] wrote: Before installating Asterisk, zaptel and so on (and independently of those), I would like to check HPET is on and working. $ zgrep HPET /proc/config.gz CONFIG_HPET_TIMER=y CONFIG_HPET=y CONFIG_HPET_RTC_IRQ=y CONFIG_HPET_MMAP=y Or, if your config is not exposed under /proc, then this: $ grep HPET /usr/src/linux/.config CONFIG_HPET_TIMER=y CONFIG_HPET=y CONFIG_HPET_RTC_IRQ=y CONFIG_HPET_MMAP=y Most people build Zaptel as a module. Thus the above two will not show. -- Tzafrir Cohen icq#16849755 jabber:[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] tdm410p w/ echo - no full duplex
Michael Check your /etc/asterisk/zapata.conf and if you have echocancelwhenbridge=yes, remove Ruben Michael J. Liberatore escribió: hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] OT - How to check HPET is on and working before installing Asterisk ?
2008/4/11, Godwin Stewart Horwich IT Services [EMAIL PROTECTED]: On Fri, 11 Apr 2008 08:40:20 +0200, Olivier [EMAIL PROTECTED] wrote: Before installating Asterisk, zaptel and so on (and independently of those), I would like to check HPET is on and working. $ zgrep HPET /proc/config.gz CONFIG_HPET_TIMER=y CONFIG_HPET=y CONFIG_HPET_RTC_IRQ=y CONFIG_HPET_MMAP=y I don't have any config.gz file in proc/ (nor any other directory), at the moment Or, if your config is not exposed under /proc, then this: $ grep HPET /usr/src/linux/.config CONFIG_HPET_TIMER=y CONFIG_HPET=y CONFIG_HPET_RTC_IRQ=y CONFIG_HPET_MMAP=y I installed a plain Debian Lenny linux so I don't have headers nor sources installed. I choose Lenny because I hoped it included and configured HPET, by default. As a last resort, if the kernel's config is available under /proc and you don't have the kernel source installed: $ grep hpet /proc/timer_list Clock Event Device: hpet set_next_event: hpet_legacy_next_event set_mode: hpet_legacy_set_mode I can't see any hpet string within /proc/timer_list file. HPET showing up as not working means a kernel rebuild. -- Godwin Stewart - Horwich IT services So my question remains : how can I be certain HPET is included and enabled without messing with zaptel and subsequent operations ? Sure, next step will be to install Asterisk and Zaptel, but at this point of my installation process, I would like to check HPET without going any further. ___ -- 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] queue logging
Flash Operator Panel http://www.asternic.org/ regards, Drew Arjan Kroon | Mobillion wrote: Hi, I’m not looking for a programma that show the queue logging. But is there a way to check during a call, which member is connected to the caller. Kind Regard, Arjan Kroon * From: * [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] ] *On Behalf Of *Scott Wolfe *Sent:* woensdag 9 april 2008 17:19 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] queue logging You could ASTassistant to see this. Its Freeware. www.astassistant.com http://www.astassistant.com - Original Message - * From: * Arjan Kroon | Mobillion mailto:[EMAIL PROTECTED] * To: * Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com * Sent: * Wednesday, April 09, 2008 1:01 AM * Subject: * [asterisk-users] queue logging Hi, I’ using with asterisk a queue with tree members and round robin. When a caller enters this queue and it is connecting to one of the members, is there a possibility to see which member the caller is connected to? And is there a way to see in de application to see if the connection from the caller to one of the members was successful of not successful? I know you can see it in de queue. log. But I want to know if I can see it also in the hangup (h) in de application? Kind 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 ___ -- 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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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
Re: [asterisk-users] question about queue
BJ Weschke wrote: Rilawich Ango wrote: Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. Is this option active in 1.2.24? I thought it was only in 1.4 It's not mentioned in the queues.conf.sample. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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
Re: [asterisk-users] bandwidth required for Asterisk running on T1
Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? Thanks in advance for your input. Regards, Mark Check out http://www.asteriskguru.com/tools/bandwidth_calculator.php it should help you figure out how much bandwidth you will need. Ryan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to check HPET is on and working before installing Asterisk ?
On Fri, 11 Apr 2008 14:32:36 +0200, Olivier [EMAIL PROTECTED] wrote: So my question remains : how can I be certain HPET is included and enabled without messing with zaptel and subsequent operations ? HPET is part of the Linux kernel. Messing with zaptel and subsequent operations is not going to get it working. If none of the tests I described reveal it then it is not included in your kernel and you need to build a new one which includes it. -- Godwin Stewart - Horwich IT services ___ -- 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] Phantom Rings
I'm fairly certain the problem is with the phone line. I have all callerID settings disabled as the Telco is unable to provide it along with our rollover line setup due to limitations in their antiquated switch. The CLI and Logs all plainly show the calls as if they were normal calls with the exception of a message about Failed to write frame and no DTMF attempts, then the call is routed into the operator queue. The calls always came in on Zap1-1 so I tried swapping the 2 lines to see if it stayed on port 1 or if the phantom followed the line. As expected, the phantom rings followed the line and began showing up on Zap2-1. So it pretty has to be something in the telco, but I'm not sure what. Putting WaitForRing(3) before the Answer command in my IVR menu eliminates most of them, but sometimes more of them slip through. I get a similar problem with a domestic analogue line in the UK. I *speculate* that there is a short half ring being sent for some reason (line test or similar), but my card (Digium) seems to need about 5 seconds to detect hangup on the remote end, so I get a phantom 2 rings at my end and then it stops... No solution, but thought it might give you something to consider... Ed W ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to check HPET is on and working before installing Asterisk ?
2008/4/11, Godwin Stewart Horwich IT Services [EMAIL PROTECTED]: On Fri, 11 Apr 2008 14:32:36 +0200, Olivier [EMAIL PROTECTED] wrote: So my question remains : how can I be certain HPET is included and enabled without messing with zaptel and subsequent operations ? HPET is part of the Linux kernel. Messing with zaptel and subsequent operations is not going to get it working. If none of the tests I described reveal it then it is not included in your kernel and you need to build a new one which includes it. You're certainly right. I thought Lenny defaulted with HPET support. Either, this is not true or my hardware doesn't support it or my configuration doesn't enable it. I fished this on Lesswatts.org *Which chipsets support HPET timers?* If you have an ICH6 or higher chipset, you should be fine. Some support exists on ICH5 chipsets. *How do I know if HPET is really active on my system?* First, HPET must be compiled in the kernel. However, having HPET compiled in the kernel and a hardware chipset supporting HPET doesn't guarantee that HPET is active. You can verify this with the command: grep hpet /proc/timer_list If this doesn't show the word hpet then it's not active. The BIOS may hide this functionality as well. You should try the force-enable HPET patch from http://linuxpowertop.org/known.php.; I tried it but it doesn't show anything such as HPET is disabled or Your hardware can't support HPET. Though this raises several questions (mainly, Why is HPET no enabled ?), I've got the answer to my initial question (grep hpet /proc/timer_list). Thank you very much. -- Godwin Stewart - Horwich IT services ___ -- 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
[asterisk-users] Friday April 11th @ 12 Noon EDT VoIP Users Conference
Hi, There are a lot of people who can and will answer questions for newbies, live on the conference. Just make sure you already Googled and read The Book ;) Details on how to hook up with us are here: http://voipusersconference.org Conference mailing list is here: http://groups.google.com/group/VOIP-Users-Conference Forums and blogs: http://food4wine.ning.com Ok, I'm stalling on today's subject because I am in the middle of a difficult move and I haven't received concrete confirms from various people. * I think Terry from Pika will be available to talk about their fax solution * I think Dean from Cognation will be around to talk about licensing * I know I'll be there to whine about how hard moving is... but that's OT Join us! /r ___ -- 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] Friday April 11th @ 12 Noon EDT VoIP Users Conference
Just this minute got the confirm from Terry for today. We'll be talking about this: http://www.pikatechnologies.com/english/View.asp?x=539 Note the free port. Try it and let us know how it goes. r ___ -- 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] odd error compiling zaptel-1.4.10 - XPP
Jerry Geis wrote: CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/vpmadt032.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/GpakApi.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/../voicebus.o LD [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/wcte12xp.o /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/zconfig.h:91:41: error: missing binary operator before token ( CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/xpp/card_fxo.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/xpp/card_fxs.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/xpp/card_pri.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/xpp/xbus-core.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/xpp/xbus-sysfs.o Havent seen that before... Any ideas. I am running centos 5.1 amd x86_64. Jerry is there a ./configure option to NOT compile in xpp code? This is the problem. When I do a menu select and disable the XPP stuff it compiles fine. If not a ./configure option how can I automatically remove the XPP code at compile time. I dont want to have to remember to go in and disable it. THanks, 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] Phantom Rings
Quoting Ed W [EMAIL PROTECTED]: one thing I thought about, but never actually did was to install a damping circuit across the line - a phone plugged in the line never actually rang or if it did it was so short it was imperceptible. I figured just a load across the might damp down the test pulse enough to not be tricked into ringing the channel bank. social engineering rather than technical engineering eventually solved the problem though. I'm fairly certain the problem is with the phone line. I have all callerID settings disabled as the Telco is unable to provide it along with our rollover line setup due to limitations in their antiquated switch. The CLI and Logs all plainly show the calls as if they were normal calls with the exception of a message about Failed to write frame and no DTMF attempts, then the call is routed into the operator queue. The calls always came in on Zap1-1 so I tried swapping the 2 lines to see if it stayed on port 1 or if the phantom followed the line. As expected, the phantom rings followed the line and began showing up on Zap2-1. So it pretty has to be something in the telco, but I'm not sure what. Putting WaitForRing(3) before the Answer command in my IVR menu eliminates most of them, but sometimes more of them slip through. I get a similar problem with a domestic analogue line in the UK. I *speculate* that there is a short half ring being sent for some reason (line test or similar), but my card (Digium) seems to need about 5 seconds to detect hangup on the remote end, so I get a phantom 2 rings at my end and then it stops... No solution, but thought it might give you something to consider... Ed W ___ -- 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 ___ -- 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] manger hangup call
Is there a way to tell the difference in an agi between the person actually hanging up the phone and the manager interface doing a hangup command? Thanks, 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
[asterisk-users] testing the list
I'm having problems sending e-mails to the list. Please ignore this message, I'm just testing. sorry for the inconvenience. Thiago Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.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
[asterisk-users] problems in REFER request to a different machine
Hi everyone, Sorry if I'm repeating the e-mail, but I'm having problems with the list. I'm currently trying to enable call transfer to different domains in asterisk box (Asterisk 1.2.13 running on Debian etch). I have a configuration that requires me to transfer call to separate domains like [EMAIL PROTECTED]:5050. My calls come from a R2 channels in a board installed in the machine. When the call comes in I dial a sip address in another machine and I need to receive REFER from this other machine to transfer the call to a third sip URI, that may be or not in any of the two machines . My machines change all the time, so registering them in my asterisk box is not an option. The big picture here is this: I have a asterisk box to receive calls from PSTN and I send this calls to sip application that I made that will transfer the call to a different sip application depending on user input. And this other application also needs the ability to transfer calls to different sip URI. The applications are conferences, voice mail and others, each running on a different sip uri ([EMAIL PROTECTED]:port) and the user needs to jump between them. So I need my asterisk box to accept arbitrary sip URI in a REFER (xfer) command. Right now it always tries to send the call to a local extension, for example, if I have a call from my asterisk to [EMAIL PROTECTED]:5060 and this application asks asterisk to transfer this call to [EMAIL PROTECTED]:5070 asterisk will try to send the to the local extension 666. Bellow I have a sip debug from the messages. My asterisk box is running in the IP 201.73.67.5, and my first application (the one that asterisk dials directly) is at the address 201.73.67.7:5080 and it transfers the calls to 201.73.67.7:5070, but it fails. All help is very much welcome. Thanks in advance, Thiago Sip debug: -- SIP read from 201.73.67.7:5080: REFER sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 201.73.67.7:5080;rport;branch=z9hG4bKPj3r0RqvljQLyTKpBVXgbhce5dADV20tVx Max-Forwards: 70 From: sip:[EMAIL PROTECTED];tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA To: 3130296800 sip:[EMAIL PROTECTED];tag=as26b5df58 Contact: sip:201.73.67.7:5080 Call-ID: [EMAIL PROTECTED] CSeq: 15651 REFER Event: refer Expires: 300 Accept: message/sipfrag;version=2.0 Allow-Events: presence, refer Refer-To: sip:[EMAIL PROTECTED]:5070 Referred-By: sip:[EMAIL PROTECTED] Content-Length: 0 --- (15 headers 0 lines) --- Transfer to 5070 in from-sip-external Transfer from 0778 in from-sip-external Transmitting (no NAT) to 201.73.67.7:5080: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 201.73.67.7:5080;branch=z9hG4bKPj3r0RqvljQLyTKpBVXgbhce5dADV20tVx;received=201.73.67.7;rport=5080 From: sip:[EMAIL PROTECTED];tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA To: 3130296800 sip:[EMAIL PROTECTED];tag=as26b5df58 Call-ID: [EMAIL PROTECTED] CSeq: 15651 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing sip:201.73.67.7:5080 for address/port to send to set_destination: set destination to 201.73.67.7, port 5080 Reliably Transmitting (no NAT) to 201.73.67.7:5080: NOTIFY sip:201.73.67.7:5080 SIP/2.0 Via: SIP/2.0/UDP 201.73.67.5:5060;branch=z9hG4bK26db8c59;rport From: 3130296800 sip:[EMAIL PROTECTED];tag=as26b5df58 To: sip:[EMAIL PROTECTED]:5080;tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=15651 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- set_destination: Parsing sip:201.73.67.7:5080 for address/port to send to set_destination: set destination to 201.73.67.7, port 5080 Reliably Transmitting (no NAT) to 201.73.67.7:5080: BYE sip:201.73.67.7:5080 SIP/2.0 Via: SIP/2.0/UDP 201.73.67.5:5060;branch=z9hG4bK1e66e326;rport From: 3130296800 sip:[EMAIL PROTECTED];tag=as26b5df58 To: sip:[EMAIL PROTECTED]:5080;tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA Call-ID: [EMAIL PROTECTED] CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing Content-Length: 0 --- -- SIP read from 201.73.67.7:5080: SIP/2.0 200 OK Via: SIP/2.0/UDP 201.73.67.5:5060;rport=5060;received=201.73.67.5;branch=z9hG4bK26db8c59 Call-ID: [EMAIL PROTECTED] From: 3130296800 sip:[EMAIL PROTECTED];tag=as26b5df58 To: sip:[EMAIL PROTECTED];tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA CSeq: 103 NOTIFY Contact: sip:201.73.67.7:5080 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub Content-Length: 0 --- (10 headers 0 lines) --- -- SIP read from 201.73.67.7:5080: SIP/2.0 200 OK Via: SIP/2.0/UDP 201.73.67.5:5060;rport=5060;received=201.73.67.5;branch=z9hG4bK1e66e326 Call-ID: [EMAIL PROTECTED] From: 3130296800 sip:[EMAIL PROTECTED];tag=as26b5df58 To:
Re: [asterisk-users] bandwidth required for Asterisk running on T1
That sounds like an E1 to me. Is that 32 DS0 channels or 24? On Fri, Apr 11, 2008 at 4:18 AM, mark morreny [EMAIL PROTECTED] wrote: Hi, The T1 is 32 x 64Kbps channels ; Codec is GSM. Thank you for your suggestions. Regards, Mark On Thu, Apr 10, 2008 at 11:25 PM, Alex Balashov [EMAIL PROTECTED] wrote: mark morreny wrote: Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? What kind of T1? And what codec? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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 -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] TuxTone Inc. http://www.TuxTone.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
[asterisk-users] Cisco 7905 / 7911G Reviews
I hope these phone / asterisk compatibility questions are not considered OT for this list. I am currently in Grandstream hell and need a cost-effective way out :) Just wanted to know if anybody has experience with the Cisco 7905 / 7911 Running SIP with Asterisk. These seem like a good replacement for the GXP2000. My basic requirement at this point is a good IP phone that talks gsm/g729/ulaw and does not crash :) I have used the Cisco 7940G in the past without problems on Sip 8.1.X from cisco. Also it appears that Cisco does not support gsm as a codec. I'm assuming there are no grandstream-like problems with these cisco's. Cisco is much more co-operative than Linksys/Polycom and is working a feasible deal for me to replace GXPs with Cisco 7905/06G -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz ___ -- 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] tdm410p w/ echo - no full duplex
Michael J. Liberatore wrote: hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. It sounds like you need the new revision of the firmware. Please contact technical support and they should be able to get it to you. Matthew Fredrickson thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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 -- 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] bandwidth required for Asterisk running on T1
Hi Andrew, Yes, it is actually a E1. Your suggestion will be greatly appreciated. Thanks, Mark On Fri, Apr 11, 2008 at 7:50 AM, Andrew Latham [EMAIL PROTECTED] wrote: That sounds like an E1 to me. Is that 32 DS0 channels or 24? On Fri, Apr 11, 2008 at 4:18 AM, mark morreny [EMAIL PROTECTED] wrote: Hi, The T1 is 32 x 64Kbps channels ; Codec is GSM. Thank you for your suggestions. Regards, Mark On Thu, Apr 10, 2008 at 11:25 PM, Alex Balashov [EMAIL PROTECTED] wrote: mark morreny wrote: Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? What kind of T1? And what codec? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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 -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] TuxTone Inc. http://www.TuxTone.com http://www.tuxtone.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] bandwidth required for Asterisk running on T1
Using the online calculator mentioned in this thread will help. There is a lot to bandwidth and even more to VoIP network traffic than can be answered with your question. On an E1 that is dedicated to IAX terminating to a provider that does trunking I would say that you could get a large number of concurrent calls through On the other hand if the calls where SIP u.law and going to different network destinations you may only get a few concurrent calls to work. Its like a good bottle of wine, the bottle is just the container On Fri, Apr 11, 2008 at 11:15 AM, Pete Kay [EMAIL PROTECTED] wrote: Hi Andrew, Yes, it is actually a E1. Your suggestion will be greatly appreciated. Thanks, Mark On Fri, Apr 11, 2008 at 7:50 AM, Andrew Latham [EMAIL PROTECTED] wrote: That sounds like an E1 to me. Is that 32 DS0 channels or 24? On Fri, Apr 11, 2008 at 4:18 AM, mark morreny [EMAIL PROTECTED] wrote: Hi, The T1 is 32 x 64Kbps channels ; Codec is GSM. Thank you for your suggestions. Regards, Mark On Thu, Apr 10, 2008 at 11:25 PM, Alex Balashov [EMAIL PROTECTED] wrote: mark morreny wrote: Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? What kind of T1? And what codec? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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 -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] TuxTone Inc. http://www.TuxTone.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 -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] TuxTone Inc. http://www.TuxTone.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
Re: [asterisk-users] Loosing SIP registration.
How about a change in IP from the IP provider? es (just a calculated guess, but it was a 286 calculator. :) ) Klaverstyn, David C wrote: Hi All, I am having problems with some SIP peers. I seem to loose registration. If I reload SIP the registration comes back. They usually stay registered for about 2 days before they drop. The problem is not all of them drop usually just the list 2 in the list. The other strange thing is that the 2 the do drop their registration do not occur at the exact same time. It could be many hours between them. I am using Asterisk 1.4.18.1 Any help would be greatly appreciated. My parents server is having the problems. My server does not exhibit this problem. I just took my router/firewall down to them as I have just purchased a new one and they are still experiencing the problem. sip show registry Host Username Refresh State Reg.Time 202.168.56.133:5060 61990xx 105 Registered Fri, 11 Apr 2008 15:15:58 sip.pennytel.com:5060 61289xx 105 Request Sent Thu, 10 Apr 2008 21:38:54 sip2.bbpglobal.com:5060 617000xxx 105 Request Sent Thu, 10 Apr 2008 20:43:20 sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/sip-register.conf': Found == Parsing '/etc/asterisk/sip-klavo.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found sip show registry Host Username Refresh State Reg.Time 202.168.56.133:5060 61990xx 105 Registered Fri, 11 Apr 2008 15:16:15 sip.pennytel.com:5060 61289xx 105 Registered Fri, 11 Apr 2008 15:16:16 sip2.bbpglobal.com:5060 617000xxx 105 Registered Fri, 11 Apr 2008 15:16:16 ___ -- 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] manger hangup call
On Friday 11 April 2008 09:23:09 Jerry Geis wrote: Is there a way to tell the difference in an agi between the person actually hanging up the phone and the manager interface doing a hangup command? Not in AGI, no. In the core, there's a bitfield that contains a bit for every reason, but it's not exposed to the AGI interface. -- Tilghman ___ -- 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] bandwidth required for Asterisk running on T1
On Fri, 2008-04-11 at 01:18 -0700, mark morreny wrote: The T1 is 32 x 64Kbps channels ; Codec is GSM. That's incorrect... a T1 is 24 channels, and each channel is 64kbps. There are also a few extra bits for framing, which adds up to 1.544 megabits per second in each direction. The audio comes across a T1 as G.711 (not GSM as stated above), and on a T1 it's usually using ulaw companding. An E1 is 32 channels, and each channel is the same 64kbps. This adds up to 2.048 megabits per second. Again, the audio is in G.711 format, but alaw companding is typically used on an E1. -- Jared Smith Community Relations Manager 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] Asterisk trunk/1.6 and nvfaxdetect
I'll begin working on full cross-version support (Asterisk 1.2, 1.4, and 1.6) in early May for nvfaxdetect and a handful of other modules. Justin Newman Hi, we are using the app_nvfaxdetect from Newman Telecom with Asterisk 1.4 and tried to build the trunk/next release 1.6 with this application, but it failed (We are using fax stuff with iaxmodem/Hylafax). I remember that we had the same issue switching from 1.2 to 1.4 and someone made the port (We don't have the necessary knowledge to do it). Has anyone port this application to last trunk and would share the port? Or is the native Asterisk fax feature (spandsp) stable enough to replace faxdetect? Regards -- Daniel TOOTAi Networks __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: [asterisk-users] SIP KEEPALIVES, with QUALIFY and without making UNREACHABLE
Need SIP KEEPALIVES in Asterisk, but QUALIFY won't presently work for you (due to it's channel disabling behavior)? Someone posted on the list that they would like to split keepalives and qualify into different features. Sounds like a good plan, but until that is done you can turn qualify= into a keepalive mechanism, without disabling your channels. Here's a quick fix: 1) Open chan_sip.c. 2) Replace lastms = -1 with lastms = 0. 3) Save. 4) #make 5) #make install I've used it in the past without problems. Not perfect (or even close), but it works. Justin __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing
Did this just start happening with the 1.4 tree? Have you made any progress on getting it resolved? Justin Newman Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen wrote: Let's be more specific here, folks: What version numbers? Asterisk, spandsp, agx-addons / rx-tx-fax? Asterisk: yesterday's 1.4 SVN SpanDSP: tried with pre 15, 16 and 18 AGX-Addons: tried with 1.4.5 and svn trunk rx/txfax: supplied by AGX Addons - although they seem to build the files and stick them into the modules directory, rather than adding to the apps directory and modifying the Makefile. i have Asterisk 1.4.18, SpanDSP 0.0.4pre16, AGX addons 1.4.5 linux kernel 2.6.18 AMD64. it (Asterisk) segfault on rxfax when i enable faxdetect in zapata.conf. since then it disabled faxdetect and use nvfaxdetect function in dialplan, it works fine afterward. also it seems to works fine using regular 32bit kernel. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: [asterisk-users] Need good voicemail documentation
Dave, Docos for Comedian Mail? Justin From: dave cantera [EMAIL PROTECTED] Subject: Re: [asterisk-users] Need good voicemail documentation An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080208/501668f8/attachment.htm __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: [asterisk-users] nvfaxdetect, nvvoicemail, and others
I'll open the source repository soon for envy and nv suite of tools, including nvfaxdetect. I have a few handfuls of useful Asterisk add-ons. Starting on module updates to fully support Asterisk 1.2, 1.4, and 1.6 in May. Maybe we can get some of these in agx-ast-addons. Also, I am interested to see how the 3rd party tools community develops. Justin Newman Hi Justin, On Thu, 2007-12-27 at 15:38 -0800, Justin Newman wrote: Yes, I wrote nvfaxdetect and a number of other modules. I don't have any nvfaxdetect updates planned for public release unless someone would like to integrate some of my changes in the GPL version...we could do this though. Perhaps you could send the diff to Antonio Gallo who started the agx-ast-addons project which includes faxdetect and backgrounddetect ported to 1.4. He seems open to enhancements/additions. His email is agx at users.sourceforge.net The project can be found at: http://sourceforge.net/projects/agx-ast-addons http://agx-ast-addons.svn.sourceforge.net/viewvc/agx-ast-addons/trunk/ Regards, Patrick - Original Message From: Matt Riddell [EMAIL PROTECTED] Justin Newman wrote: We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. Also, are you the guy who wrote nvfaxdetect et al? Any chance of an update for 1.4 etc? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[asterisk-users] Asterisk temporary hangs when no internet connection
Hello all, I have the following problem: if there is a temporary loss of Internet connectivity, the asterisk server 'hangs' if it has external SIP trunks configured. By hanging I mean that any calls between the local extensions and any calls to the voicemail extension stop working. Everything works fine again when the internet connectivity returns (I tested this by removing and reinserting the network cable from the cable modem). My guess is that the asterisk server tries resolving the names of the SIP providers when it tries to re-register to them and because there is no internet connectivity it hangs there for a while. However in that time all the local calls to the asterisk server stop working. Has anybody else encountered this problem? Thanks! ___ -- 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 temporary hangs when no internet connection
Marius Muja wrote: My guess is that the asterisk server tries resolving the names of the SIP providers when it tries to re-register to them and because there is no internet connectivity it hangs there for a while. However in that time all the local calls to the asterisk server stop working. Try using a local DNS server. Sounds like its waiting on DNS lookups... ___ -- 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] Correlating queue_logs and cdr for abandoned calls
Hi, I am using asterisk 1.4.19, my requirement is to find out which agents were ringed by the queue when a call is abandoned (or connected) in a call center. While this information is available in parts in queue_logs and cdr, there is no way to correlate this information. For example this is the queue_log entries for a call that was abandoned 1207935049|1207935049.6|queue|NONE|ENTERQUEUE||_0_ 1207935060|1207935049.6|queue|Agent/3501|RINGNOANSWER|1 1207935078|1207935049.6|queue|Agent/3501|RINGNOANSWER|1 1207935090|1207935049.6|queue|Agent/3501|RINGNOANSWER|1 1207935093|1207935049.6|queue|NONE|ABANDON|1|1|44 and cdr during this time: ,3501,5501,sip,3501 3501,Local/[EMAIL PROTECTED],2,SIP/5501-082495a8,Dial,SIP/5501,2008-04-11 17:30:49,,2008-04-11 17:31:00,11,0,NO ANSWER,DOCUMENTATION,1207935049.8, ,3501,5501,sip,3501 3501,Local/[EMAIL PROTECTED],2,SIP/5501-0824b5f8,Dial,SIP/5501,2008-04-11 17:31:08,,2008-04-11 17:31:18,10,0,NO ANSWER,DOCUMENTATION,1207935068.11, ,3501,5501,sip,3501 3501,Local/[EMAIL PROTECTED],2,SIP/5501-0824b5f8,Dial,SIP/5501,2008-04-11 17:31:19,,2008-04-11 17:31:30,11,0,NO ANSWER,DOCUMENTATION,1207935079.14, ,3501,5501,sip,3501 3501,Local/[EMAIL PROTECTED],2,SIP/5501-0824b5f8,Dial,SIP/5501,2008-04-11 17:31:31,,2008-04-11 17:31:33,2,0,NO ANSWER,DOCUMENTATION,1207935091.17, there are no common fields in both logs to correlate them. Also missing in the cdr is the entry for the call coming in. ie record if the call from the customer to the callcenter. This entry is present when the call is completed. Just wondering how others are dealing with this requirement and if missing cdr entry is a bug? regards, raj ___ -- 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] Correlating queue_logs and cdr for abandoned calls
On Friday 11 April 2008 12:57:17 Rajkumar S wrote: I am using asterisk 1.4.19, my requirement is to find out which agents were ringed by the queue when a call is abandoned (or connected) in a call center. While this information is available in parts in queue_logs and cdr, there is no way to correlate this information. For example this is the queue_log entries for a call that was abandoned 1207935049|1207935049.6|queue|NONE|ENTERQUEUE||_0_ 1207935060|1207935049.6|queue|Agent/3501|RINGNOANSWER|1 1207935078|1207935049.6|queue|Agent/3501|RINGNOANSWER|1 1207935090|1207935049.6|queue|Agent/3501|RINGNOANSWER|1 1207935093|1207935049.6|queue|NONE|ABANDON|1|1|44 and cdr during this time: ,3501,5501,sip,3501 3501,Local/[EMAIL PROTECTED],2,SIP/5501-082495a8,Dial,SIP/5501,2008 -04-11 17:30:49,,2008-04-11 17:31:00,11,0,NO ANSWER,DOCUMENTATION,1207935049.8, ,3501,5501,sip,3501 3501,Local/[EMAIL PROTECTED],2,SIP/5501-0824b5f8,Dial,SIP/5501,2008 -04-11 17:31:08,,2008-04-11 17:31:18,10,0,NO ANSWER,DOCUMENTATION,1207935068.11, ,3501,5501,sip,3501 3501,Local/[EMAIL PROTECTED],2,SIP/5501-0824b5f8,Dial,SIP/5501,2008 -04-11 17:31:19,,2008-04-11 17:31:30,11,0,NO ANSWER,DOCUMENTATION,1207935079.14, ,3501,5501,sip,3501 3501,Local/[EMAIL PROTECTED],2,SIP/5501-0824b5f8,Dial,SIP/5501,2008 -04-11 17:31:31,,2008-04-11 17:31:33,2,0,NO ANSWER,DOCUMENTATION,1207935091.17, there are no common fields in both logs to correlate them. Actually, there are. Check the second field in the queue log and the second-to-last field in the CDR. This is the uniqueid field, and it's present in both. Also missing in the cdr is the entry for the call coming in. ie record if the call from the customer to the callcenter. This entry is present when the call is completed. Just wondering how others are dealing with this requirement and if missing cdr entry is a bug? Check your settings in cdr.conf. Logging unanswered calls is an option available to the PBX administrator. -- Tilghman ___ -- 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 temporary hangs when no internet connection
It is using a local DNS server. On Fri, Apr 11, 2008 at 10:43 AM, Steven Kurylo [EMAIL PROTECTED] wrote: Marius Muja wrote: My guess is that the asterisk server tries resolving the names of the SIP providers when it tries to re-register to them and because there is no internet connectivity it hangs there for a while. However in that time all the local calls to the asterisk server stop working. Try using a local DNS server. Sounds like its waiting on DNS lookups... ___ -- 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] Asterisk temporary hangs when no internet connection
This is a very common issue with Asterisk. There is no good fix, but if you make sure ALL IP addresses of the server are listed in /etc/hosts on the server it may help. Marius Muja wrote: It is using a local DNS server. On Fri, Apr 11, 2008 at 10:43 AM, Steven Kurylo [EMAIL PROTECTED] wrote: Marius Muja wrote: My guess is that the asterisk server tries resolving the names of the SIP providers when it tries to re-register to them and because there is no internet connectivity it hangs there for a while. However in that time all the local calls to the asterisk server stop working. Try using a local DNS server. Sounds like its waiting on DNS lookups... ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 temporary hangs when no internet connection
Marius Muja wrote: Hello all, I have the following problem: if there is a temporary loss of Internet connectivity, the asterisk server 'hangs' if it has external SIP trunks configured. By hanging I mean that any calls between the local extensions and any calls to the voicemail extension stop working. Everything works fine again when the internet connectivity returns (I tested this by removing and reinserting the network cable from the cable modem). My guess is that the asterisk server tries resolving the names of the SIP providers when it tries to re-register to them and because there is no internet connectivity it hangs there for a while. However in that time all the local calls to the asterisk server stop working. Has anybody else encountered this problem? Yes, this is a common issue. A workaround is to use IP Addresses or enter hostnames manually in /etc/hosts Andres http://www.neuroredes.com Thanks! ___ -- 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
[asterisk-users] ZD Net article
Just came across a ZDnet article on the cogoblue appliance that was launched last week. http://blogs.zdnet.com/Greenfield/?p=215 Not commenting on the article or the appliance. But just wanted to highlight that it's good to see asterisk vendors reaching out beyond the usual geek marketing areas. Yes ZDnet has a tech focus but it's pretty mainstream, so should be reaching an audience not currently being reached. Lol - though nothing is going to top the Forbes article about Mark Spencer this week http://www.forbes.com/forbes/2006/0410/063.html Half of their readers wouldn't even know what a PABX actually is. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ -- 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] tdm410p w/ echo - no full duplex
Matthew, I have just emailed support. Do you know what the latest revision is? Also, is it ok for mg2 to be in zconfig.h and echocancel=yes ? It will know automatically to use the hw ec rather than the software one? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Friday, April 11, 2008 11:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm410p w/ echo - no full duplex Michael J. Liberatore wrote: hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. It sounds like you need the new revision of the firmware. Please contact technical support and they should be able to get it to you. Matthew Fredrickson thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. -- -- ___ -- 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 -- 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] tdm410p w/ echo - no full duplex
Ok I will remove it, may I ask what that will do or how that will help? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ruben Zamora Sent: Friday, April 11, 2008 7:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm410p w/ echo - no full duplex Michael Check your /etc/asterisk/zapata.conf and if you have echocancelwhenbridge=yes, remove Ruben Michael J. Liberatore escribió: hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. -- -- ___ -- 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] tdm410p w/ echo - no full duplex
Michael J. Liberatore wrote: Matthew, I have just emailed support. Do you know what the latest revision is? Also, is it ok for mg2 to be in zconfig.h and echocancel=yes ? It will Yes. Chan_zap and zaptel know how to automatically use the hardware echo canceller. The configuration options like echocancel and echocancelwhenbridged apply the same to hardware and software echo cancellers. Matthew Fredrickson Digium, Inc. know automatically to use the hw ec rather than the software one? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Friday, April 11, 2008 11:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm410p w/ echo - no full duplex Michael J. Liberatore wrote: hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. It sounds like you need the new revision of the firmware. Please contact technical support and they should be able to get it to you. Matthew Fredrickson thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. -- -- ___ -- 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 -- 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 -- 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] ZD Net article
On 4/11/08, Dean Collins [EMAIL PROTECTED] wrote: Lol – though nothing is going to top the Forbes article about Mark Spencer this week http://www.forbes.com/forbes/2006/0410/063.html That article is over two years old... -- Kristian Kielhofner ___ -- 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] tdm410p w/ echo - no full duplex
You can read these information in the zapata.conf. Most of the time when you use hardware cancelation echo these paramater make worse echo. Its better when you use HPEC that is a software no hardware for that parameter. Michael J. Liberatore escribió: Ok I will remove it, may I ask what that will do or how that will help? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ruben Zamora Sent: Friday, April 11, 2008 7:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm410p w/ echo - no full duplex Michael Check your /etc/asterisk/zapata.conf and if you have echocancelwhenbridge=yes, remove Ruben Michael J. Liberatore escribió: hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. -- -- ___ -- 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 ___ -- 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] best way for call detail logging
On Thursday 10 April 2008 12:30:24 Michiel van Baak wrote: On 10:00, Thu 10 Apr 08, Pete Kay wrote: I would like to be able to log call details in Asterisk. The kind of logs that I like to generate is like this: From To Forward Time Incoming Call604-343-3334 503-233-4454 13:33:32 Extension Routing 503-233-4454 Extension 403 13:33:32 Forwarding 503-233-4454 454-444-2334 13:33:32 where 503-233-4454 is my DID number. Basically, I would like to log how calls are being handled in Asterisk. I understand I can use AGI to log the information in database, but I am wondering if this is scalable enough for large number of users. I am using realtime CDR but it does not record the kind of detail that I am looking for. If I don't use AGI, what would be the best way to do it? Can someone please give me some advice or inputs? Thank you very much in advance for your suggestion. Maybe write something that connects to the AMI and listens to what happens there. Or he could read up on cdr_adaptive_odbc and use the backport. -- Tilghman ___ -- 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] ZD Net article
Hmm weird for some reason it showed up in my google alerts box on asterisk this week -I saved the url and didn't even notice the date so thought it was this week. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Friday, 11 April 2008 4:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ZD Net article On 4/11/08, Dean Collins [EMAIL PROTECTED] wrote: Lol - though nothing is going to top the Forbes article about Mark Spencer this week http://www.forbes.com/forbes/2006/0410/063.html That article is over two years old... -- Kristian Kielhofner ___ -- 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] odd error compiling zaptel-1.4.10 - XPP
On Fri, Apr 11, 2008 at 09:58:08AM -0400, Jerry Geis wrote: Jerry Geis wrote: CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/vpmadt032.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/GpakApi.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/../voicebus.o LD [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/wcte12xp.o /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/zconfig.h:91:41: error: missing binary operator before token ( http://bugs.digium.com/12426 There's also a fix there that I don't fully understand (and I'm not sure that that fix does not cause damage, so don't just apply it). -- Tzafrir Cohen icq#16849755 jabber:[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] Connecting Asterisk to Nortel Succession 4.0 sip...
What type of Nortel? How are you connected to the Nortel? CP Eugen Soare wrote: Well I am entering into a realm that I don't know. 3 sites with Asterisk 1 site with Nortel Asterisk/Sip calls working fine between the 3 sites. Asterisk to Nortel set calls working fine. (call comes from asterisk to nortel and rings telephone, people answer and talk happens, hangup call clears) Nortel to Asterisk. Set on Nortel gets a busy signal. Any suggestions on what to look for? Much appreciated! Eugen ___ -- 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] Connecting Asterisk to Nortel Succession 4.0 sip...
Succession 1000SG running 4.0. Using SIP trunks. es CunningPike wrote: What type of Nortel? How are you connected to the Nortel? CP Eugen Soare wrote: Well I am entering into a realm that I don't know. 3 sites with Asterisk 1 site with Nortel Asterisk/Sip calls working fine between the 3 sites. Asterisk to Nortel set calls working fine. (call comes from asterisk to nortel and rings telephone, people answer and talk happens, hangup call clears) Nortel to Asterisk. Set on Nortel gets a busy signal. Any suggestions on what to look for? Much appreciated! Eugen ___ -- 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
[asterisk-users] Asterisk and the Mitel SX 200 integration
Hi. One of my clients has an old Mitel SX 200 with a separate computer doing the voicemail and auto attendant and integrated via a COV card which is in his case an ISA card! We would all like to migrate to asterisk, but as a first step, can asterisk integrate into the Mitel, so it can serve as auto attendant and the voicemail for the extensions? If this is successful we could gradually migrate extensions, particularly if we could get the Mitel to talk to asterisk via one of its t1 cards. Any assistance or experience along these lines would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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] Asterisk and the Mitel SX 200 integration
At 17:32 4/11/2008, John covici wrote: Hi. One of my clients has an old Mitel SX 200 with a separate computer doing the voicemail and auto attendant and integrated via a COV card which is in his case an ISA card! Is it an ActiveVoice system? We would all like to migrate to asterisk, but as a first step, can asterisk integrate into the Mitel, so it can serve as auto attendant and the voicemail for the extensions? We've got a client with the exact same setup. They have suffered long enough with this dinosaur. They are in the process of going with an all-Asterisk system. You would probably make more money trying an intermediate step using the SX-200 and Asterisk, but it would be obviously more costly for them as well as prolong their misery. It's your call, but I would recommend getting away from 30 year old technology as fast as you can run. The ActiveVoice system is a cantankerous 20 year old system in itself. You have just received 2 cents worth of advice for FREE! If this is successful we could gradually migrate extensions, particularly if we could get the Mitel to talk to asterisk via one of its t1 cards. Any assistance or experience along these lines would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and the Mitel SX 200 integration
Yep, you guessed it, an activvoice system. Anyway to make Asterisk act like that for a while? Thanks. on Friday 04/11/2008 Doug([EMAIL PROTECTED]) wrote At 17:32 4/11/2008, John covici wrote: Hi. One of my clients has an old Mitel SX 200 with a separate computer doing the voicemail and auto attendant and integrated via a COV card which is in his case an ISA card! Is it an ActiveVoice system? We would all like to migrate to asterisk, but as a first step, can asterisk integrate into the Mitel, so it can serve as auto attendant and the voicemail for the extensions? We've got a client with the exact same setup. They have suffered long enough with this dinosaur. They are in the process of going with an all-Asterisk system. You would probably make more money trying an intermediate step using the SX-200 and Asterisk, but it would be obviously more costly for them as well as prolong their misery. It's your call, but I would recommend getting away from 30 year old technology as fast as you can run. The ActiveVoice system is a cantankerous 20 year old system in itself. You have just received 2 cents worth of advice for FREE! If this is successful we could gradually migrate extensions, particularly if we could get the Mitel to talk to asterisk via one of its t1 cards. Any assistance or experience along these lines would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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
[asterisk-users] NAT issue with Fortinet Firewall
I have a customer with a Fortinet Firewall that is having stability issues with Asterisk and SIP endpoints (PAP2T) outside his network. The first issue I see is that Asterisk sees all phones as the IP address of the Fortinet. Since the parameter localnet defines the local network and that address falls in that range, how will Asterisk treat the endpoints? I have nat=yes for all phones and canreinvite=no as well. The externip parameter is set to the outside public IP address. Still we have calls with one way audio. This is the first setup with a firewall that rewrites the IP address of the endpoint so I do not know how that is affecting the packet flow. On my other servers I can always see the public IP of the endpoint. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- 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] Is Asterisk really good??
On Thu, 10 Apr 2008 11:46:48 -0700, Eugen Soare [EMAIL PROTECTED] wrote: So this is just a general question, Is Asterisk really good? Yes, but you should also look at an alternative that used Asterisk as a reference (www.freeswitch.org), and make an informed decision. ___ -- 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] NAT issue with Fortinet Firewall
Fortinets have a SIP session-helper. Sometime this causes issues, try turning it off. To do this you need to enable telnet on the forinet management interface. Telnet into the cli and type the following config system session-helper edit 12 set port 5066 end Instead of turning this off or taking it out I am changing the port so it will not affect 5060 anymore. This way you can put it back if this doesn't work for you. John Bittner Simlab.net -Original Message- I have a customer with a Fortinet Firewall that is having stability issues with Asterisk and SIP endpoints (PAP2T) outside his network. The first issue I see is that Asterisk sees all phones as the IP address of the Fortinet. Since the parameter localnet defines the local network and that address falls in that range, how will Asterisk treat the endpoints? I have nat=yes for all phones and canreinvite=no as well. The externip parameter is set to the outside public IP address. Still we have calls with one way audio. This is the first setup with a firewall that rewrites the IP address of the endpoint so I do not know how that is affecting the packet flow. On my other servers I can always see the public IP of the endpoint. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- 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] Is Asterisk really good??
I am looking at that. hmm... what to do... don't want any regrets you know! :) thanks, es Vincent wrote: On Thu, 10 Apr 2008 11:46:48 -0700, Eugen Soare [EMAIL PROTECTED] wrote: So this is just a general question, Is Asterisk really good? Yes, but you should also look at an alternative that used Asterisk as a reference (www.freeswitch.org), and make an informed decision. ___ -- 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
[asterisk-users] No compatible codecs / static noise
I'm running asterisk 1.2 with Sipura adapters. I've tried to experiment with different codes but I'm either getting No compatible codecs if I use gsm or static noise if I use g726 I was under impression that asterisk would translate between codecs according to show translation table. 2.) Does show audio codecs shows all available codesc install on the system or all the codesc that asterisk is capable to works with? -- #Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and the Mitel SX 200 integration
I second Doug advice. Migrate to Asterisk asap. We have several Asterisk auto attendant integrated with Mitel, even billing using the Mitel's smdr. But voicemail is different. The COV card emulate a SS4 phone and receive information needed for a voice mail system. With FXO/FXS ports is not possible receive such information. Jorge Mendoza John covici wrote: Yep, you guessed it, an activvoice system. Anyway to make Asterisk act like that for a while? Thanks. on Friday 04/11/2008 Doug([EMAIL PROTECTED]) wrote At 17:32 4/11/2008, John covici wrote: Hi. One of my clients has an old Mitel SX 200 with a separate computer doing the voicemail and auto attendant and integrated via a COV card which is in his case an ISA card! Is it an ActiveVoice system? We would all like to migrate to asterisk, but as a first step, can asterisk integrate into the Mitel, so it can serve as auto attendant and the voicemail for the extensions? We've got a client with the exact same setup. They have suffered long enough with this dinosaur. They are in the process of going with an all-Asterisk system. You would probably make more money trying an intermediate step using the SX-200 and Asterisk, but it would be obviously more costly for them as well as prolong their misery. It's your call, but I would recommend getting away from 30 year old technology as fast as you can run. The ActiveVoice system is a cantankerous 20 year old system in itself. You have just received 2 cents worth of advice for FREE! If this is successful we could gradually migrate extensions, particularly if we could get the Mitel to talk to asterisk via one of its t1 cards. Any assistance or experience along these lines would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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 ___ -- 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] Is Asterisk really good??
On 4/10/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Thu, Apr 10, 2008 at 05:49:26PM -0400, Al Baker wrote: Please share more about this. What/How are you clustering the boxes ? Is this all VOIP or TDMF front and VOIP for agents in back ? What kind of Boxes ? What O/S What tools are you using to monitor this big-azz mother ? What, Matt? You haven't already talked about this here? :-) My new job is Matt Florell's old job, where VICIdial got started. I haven't counted the boxes lately, but I think there are 14 with quad-T cards in them, separate boxes for MySQL and Apache. Our architecture is FXS T-1 channel banks for the agent phones, usually 1 + 3 IXC spans per box, though we turned up a box a couple weeks ago with 3 channel banks, and no spans. All TDM; the only VoIP is the IAX trunks hauling load-balance calls around. And just the usual VICIdial tools, mostly, though I'm fixin to deploy either Big Sister or Nagios. Of course I have talked about it here, 3 years ago:) I even gave a presentation about it at Astricon in 2005: http://eflo.net/presentations/Astricon2005_matt_florell_PDF.pdf It is a bit dated(as are some of the servers there) but it is a good description of how that system was originally set up. 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] bandwidth required for Asterisk running on T1
Additionally Mark, a Channelized (also called Integrated) T1 offers 24 channels for voice/data, but after bit robbing (for signalling, etc) you only get around 56kbps per channel. ISDN PRI over T1 has 23 b-channels of voice/data and one d-channel for signalling, etc. PRI is preferred and most common. And of course, ISDN PRI over E1 gets 30 channels of voice/data and 2 channels for signalling. Jared Smith wrote: On Fri, 2008-04-11 at 01:18 -0700, mark morreny wrote: The T1 is 32 x 64Kbps channels ; Codec is GSM. That's incorrect... a T1 is 24 channels, and each channel is 64kbps. There are also a few extra bits for framing, which adds up to 1.544 megabits per second in each direction. The audio comes across a T1 as G.711 (not GSM as stated above), and on a T1 it's usually using ulaw companding. An E1 is 32 channels, and each channel is the same 64kbps. This adds up to 2.048 megabits per second. Again, the audio is in G.711 format, but alaw companding is typically used on an E1. -- Jared Smith Community Relations Manager 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] Is Asterisk really good??
wow! That was cool! thanks for the pdf. es Matt Florell wrote: On 4/10/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Thu, Apr 10, 2008 at 05:49:26PM -0400, Al Baker wrote: Please share more about this. What/How are you "clustering" the boxes ? Is this all VOIP or TDMF front and VOIP for agents in back ? What kind of Boxes ? What O/S What tools are you using to monitor this big-azz mother ? What, Matt? You haven't already talked about this here? :-) My new job is Matt Florell's old job, where VICIdial got started. I haven't counted the boxes lately, but I think there are 14 with quad-T cards in them, separate boxes for MySQL and Apache. Our architecture is FXS T-1 channel banks for the agent phones, usually 1 + 3 IXC spans per box, though we turned up a box a couple weeks ago with 3 channel banks, and no spans. All TDM; the only VoIP is the IAX trunks hauling load-balance calls around. And just the usual VICIdial tools, mostly, though I'm fixin to deploy either Big Sister or Nagios. Of course I have talked about it here, 3 years ago:) I even gave a presentation about it at Astricon in 2005: http://eflo.net/presentations/Astricon2005_matt_florell_PDF.pdf It is a bit dated(as are some of the servers there) but it is a good description of how that system was originally set up. 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 ___ -- 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] Need some input for Quad T1 and channel banks.
I just wanted to check one more thing, system is connected to PSTN via SIP trunk ( No echo) , and terminates to customer analog phone's via Adit 600 fxs. I do not see any need for echo cancellation in this setup. There is no far end hybrid source, Any other thoughts? On Thu, Apr 3, 2008 at 8:18 AM, Darren Wright [EMAIL PROTECTED] wrote: I've used Adit600's almost exclusively for my installs. All have worked great for me. -D From: [EMAIL PROTECTED] on behalf of Steve Totaro Sent: Thu 4/3/2008 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Need some input for Quad T1 and channel banks. Just Google Quintum Tenor AX. Well worth the money. Thanks, Steve Totaro On Mon, Mar 31, 2008 at 10:03 PM, Al lists [EMAIL PROTECTED] wrote: Im guessing T1cas not PRI,just because its giving 24 fxs per T1. Steve, what are my options for SIP to fxs? thank you! On 3/31/08, Doug Lytle [EMAIL PROTECTED] wrote: Don Pobanz wrote: Doug Lytle wrote on Monday, March 31, 2008 5:40 PM This does not sound right. If it is 2 PRIs then it should be 46 channels I may have the terminology incorrect. I don't have a D channel, so I guess this would be called a T1 then? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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 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 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 was sent from D2 Technology, 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] question about queue
Do you mean autofill works in 1.4.x? But it doesn't work even I set it. On Fri, Apr 11, 2008 at 11:07 AM, BJ Weschke [EMAIL PROTECTED] wrote: Rilawich Ango wrote: Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Asterisk and the Mitel SX 200 integration
Jorge is correct you will not get the information need via FXO/FXS unless you program the Mitel to do DTMF inband. It is possible but a cludge of a fix at best. We have successfully integrated several Mitel SX200 and SX2000 switches via the PRI (preferred) or T1 using EM_Wink (works but you have delays while waiting for the winks. (wink, wink :-) ). The Mitel is rock-solid and depending on the size of the install a fork-lift replacement may not be desirable. I would start by replacing the VM (ActiveVoice) with and Asterisk box, you can give them unified messaging as well as a stable and current platform ( I have seen the Octel COV card catch on fire!!) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jorge Mendoza Sent: Friday, April 11, 2008 8:32 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Doug Subject: Re: [asterisk-users] Asterisk and the Mitel SX 200 integration I second Doug advice. Migrate to Asterisk asap. We have several Asterisk auto attendant integrated with Mitel, even billing using the Mitel's smdr. But voicemail is different. The COV card emulate a SS4 phone and receive information needed for a voice mail system. With FXO/FXS ports is not possible receive such information. Jorge Mendoza John covici wrote: Yep, you guessed it, an activvoice system. Anyway to make Asterisk act like that for a while? Thanks. on Friday 04/11/2008 Doug([EMAIL PROTECTED]) wrote At 17:32 4/11/2008, John covici wrote: Hi. One of my clients has an old Mitel SX 200 with a separate computer doing the voicemail and auto attendant and integrated via a COV card which is in his case an ISA card! Is it an ActiveVoice system? We would all like to migrate to asterisk, but as a first step, can asterisk integrate into the Mitel, so it can serve as auto attendant and the voicemail for the extensions? We've got a client with the exact same setup. They have suffered long enough with this dinosaur. They are in the process of going with an all-Asterisk system. You would probably make more money trying an intermediate step using the SX-200 and Asterisk, but it would be obviously more costly for them as well as prolong their misery. It's your call, but I would recommend getting away from 30 year old technology as fast as you can run. The ActiveVoice system is a cantankerous 20 year old system in itself. You have just received 2 cents worth of advice for FREE! If this is successful we could gradually migrate extensions, particularly if we could get the Mitel to talk to asterisk via one of its t1 cards. Any assistance or experience along these lines would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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 ___ -- 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
[asterisk-users] X100M never goes on-hook state
Hi guys, I've been experiencing a very strange issue with my Digium Card TDM400 as of this week. It has two FXS and two FXO. The FXO modules (both of them) never goes on-hook after hanging up in Asterisk. It had worked perfectly well for over four years. I put an ammeter in series with the line and the card, and immediately after plugging the connector to the card, I got 26mA in the circuit and a dial tone from the carrier, where it should be zero amper (on-hook state). If a Dial() something, it works perfectly. I can Hangup() the call, freeing the channel in Asterisk, but the hardware keeps off-hook forever, locking the line. If I Dial() again, Asterisk opens the line, sends the DTMFs normally, but it doesn't work since the carrier thinks I'm still holding the first call. It behaves exactly the same way with another analog line. If I plug either of the lines and my other Digium card (TDM2400), it works ok. The same with my Brazilian DigiVoice FXO card. Ok, you all might say: your card is damaged, throw it away. Ok, I could do it, but now comes the funny part: If I put an DSL filter in series with the line and the card, IT WORKS PERFECTLY!!! The filter imposes 25 ohms over the circuit. Maybe that's causing the card to work. When I put the filter and the ammeter in series, I get zero amper when on-hook and 26 mA when off-hook, that's the expected behaviour. I'm not an expert in electricity, so I really don't know why the card is behaving that way. What does that resistance make for the card to start working ok? I know the DSL filter isn't only a resistor. Maybe it has another electrical component that's helping more than the resistor. Just a guess. Tomorrow I'll buy a 30-ohm resistor, take the DSL filter off, and test the card only with the resistor, to check it out. In order to isolate the problem even more, I plugged the FXO port in one FXS port. Immediately after plugging it, Asterisk announced at the console that someone went off-hook at the FXS port. So, it's not really a carrier issue. The FXS port is perfectly -48V on-hook, and about 20 mA in the circuit when off-hook, closer than the carrier to the standard values. Any clue is welcome. -- MARLON DUTRA Propus GnuPG ID: 0x3E2060AC pgp.mit.edu http://www.propus.com.br/ http://hackers.propus.com.br/~marlon/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and the Mitel SX 200 integration
OK, this is exactly what I would like to do, can you either write me on or off list for further details. This would be the first baby step toward the 20th Century!! on Friday 04/11/2008 Alexander Lopez([EMAIL PROTECTED]) wrote Jorge is correct you will not get the information need via FXO/FXS unless you program the Mitel to do DTMF inband. It is possible but a cludge of a fix at best. We have successfully integrated several Mitel SX200 and SX2000 switches via the PRI (preferred) or T1 using EM_Wink (works but you have delays while waiting for the winks. (wink, wink :-) ). The Mitel is rock-solid and depending on the size of the install a fork-lift replacement may not be desirable. I would start by replacing the VM (ActiveVoice) with and Asterisk box, you can give them unified messaging as well as a stable and current platform ( I have seen the Octel COV card catch on fire!!) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jorge Mendoza Sent: Friday, April 11, 2008 8:32 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Doug Subject: Re: [asterisk-users] Asterisk and the Mitel SX 200 integration I second Doug advice. Migrate to Asterisk asap. We have several Asterisk auto attendant integrated with Mitel, even billing using the Mitel's smdr. But voicemail is different. The COV card emulate a SS4 phone and receive information needed for a voice mail system. With FXO/FXS ports is not possible receive such information. Jorge Mendoza John covici wrote: Yep, you guessed it, an activvoice system. Anyway to make Asterisk act like that for a while? Thanks. on Friday 04/11/2008 Doug([EMAIL PROTECTED]) wrote At 17:32 4/11/2008, John covici wrote: Hi. One of my clients has an old Mitel SX 200 with a separate computer doing the voicemail and auto attendant and integrated via a COV card which is in his case an ISA card! Is it an ActiveVoice system? We would all like to migrate to asterisk, but as a first step, can asterisk integrate into the Mitel, so it can serve as auto attendant and the voicemail for the extensions? We've got a client with the exact same setup. They have suffered long enough with this dinosaur. They are in the process of going with an all-Asterisk system. You would probably make more money trying an intermediate step using the SX-200 and Asterisk, but it would be obviously more costly for them as well as prolong their misery. It's your call, but I would recommend getting away from 30 year old technology as fast as you can run. The ActiveVoice system is a cantankerous 20 year old system in itself. You have just received 2 cents worth of advice for FREE! If this is successful we could gradually migrate extensions, particularly if we could get the Mitel to talk to asterisk via one of its t1 cards. Any assistance or experience along these lines would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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 ___ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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] NAT issue with Fortinet Firewall
FYI, I have probably 10 Fortinet units with multiple SIP phones behind each and all of the phones work flawlessly. As long as the Fortinet is ver 3.0 or newer, it does NAT so that you don't need to have nat=yes on *. No pinholes or static nat or anything, it just works. As a side note, I probably have 20+ Cisco PIX's with the same setup and they work flawlessly too. I've seen a lot of people saying fixup sip breaks phones, but not that I have seen. I just let the PIX do nat and it works fine. Carlos Chavez wrote: I have a customer with a Fortinet Firewall that is having stability issues with Asterisk and SIP endpoints (PAP2T) outside his network. The first issue I see is that Asterisk sees all phones as the IP address of the Fortinet. Since the parameter localnet defines the local network and that address falls in that range, how will Asterisk treat the endpoints? I have nat=yes for all phones and canreinvite=no as well. The externip parameter is set to the outside public IP address. Still we have calls with one way audio. This is the first setup with a firewall that rewrites the IP address of the endpoint so I do not know how that is affecting the packet flow. On my other servers I can always see the public IP of the endpoint. ___ -- 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
