[asterisk-users] High utilization with SIP registration
Greetings all, I have a problem with a PBX that I manage. The system has 2 AVM Fritz boards connected to two BRI ISDN services using chan_capi in addition to several SIP trunks going out to Internet based providers for call termination via the Internet. They experience problems when the Internet connection goes down. Obviously the SIP trunks are lost. However the strange thing is that calls are dropped on the capi channels as well during these Internet outages. One of the engineers that I work with felt that the problem was due to Asterisk persistantly trying to re register the SIP services and was using up too much CPU in the process. In fact he was able to workaround the problem temporarily by commenting out the SIP registration in sip.conf, which would confirm his theory. I suppose my question is. Has anyone else seen this sort of behaviour before? Is there any SIP settings that we should be including to try to slow down the SIP registration so that it doesn't use up too many system resources? This message was sent using MyMail ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp (foip)
lee,Thanks for the feedback.in most diagrams explaining t38, it shows, the sending fax machine connecting to a pots before connecting to a gateway,then the internet. but if i've read and understood correctly, the sending end can use an ATA with t38 support instead of a pots. in that case, where does the packetization of the t30 data happen? at the ATA? level i presume?http://www.answers.com/topic/t-30-protocol-figure-01-jpgalso, can you recommened a good asterisk compatible ATA adapter with t38 support? i believe cisco has one.Thanks in advance.Lee Howard [EMAIL PROTECTED] wrote: Christopher Corn wrote: May I ask, from your own personal experience. is it not necessaritly worth (the headaches) of investing mytime into setting up SPANDSP into my asterisk system, but rather invest it into going to a company, like packet8 that offers t38 conversion?I am not really in a position to tell you what something will be worth to you - especially when I've not even used that something myself. I know and use spandsp as a library, with IAXmodem and HylaFAX, but I do not have any experience with spandsp in txfax/rxfax applications or in its new T.38 gatewaying. I suspect that I'll eventually get into spandsp's T.38 aspects, but without that I've only had a limited amount of hands-on exposure to T.38 applications in the form of t38modem and Cisco gateways (which experience was somewhat disenchanting - mostly because of the gateway T.30 processing).If you have a T.38 fax machine or if you have a T.38-capable ATA connected to a fax machine and you do not have your own PSTN lines then I would suspect that it would be worthwhile to use T.38 pass-through on Asterisk 1.4 or OpenPBX in conjunction with a T.38-supporting FoIP provider. (Because otherwise you don't have any straight-forward, reliable means for faxing from your internal fax machines.) what does the future of faxing lean towards? before entering an era when all fax machines run the t38 protocol. will we see more t38 termination services or faxing through g711?T.38 is the end-all solution for faxing over IP networks. So I suspect that you will see the pervasiveness of T.38 implementations increase along with the pervasiveness of VoIP in general. That said, VoIP has its own fair share of problems that keeps it from being capable of replacing PSTN circuits entirely, and so as long as those problems are not generally resolvable for your average business or service provider then you'll continue to also see more of the same, traditional, modem-ing fax machines. So I strongly suspect that you'll see more of T.38, but I don't think that the PSTN (and traditional fax machines with it) is going away any time soon. from what i've read, using a service that does t38 termination, seems to be where i should go.I would say that it entirely depends upon whether or not you have PSTN lines yourself. If you do, then I would take whatever efforts you can to avoid the additional points of T.30 processing/relaying (therefore avoiding T.38 gatewaying). But if you do not have PSTN lines, then take whatever efforts you can to properly implement T.38 to your FoIP provider who will gateway for you.Lee.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Display message on voip phone...hint?
Hi, use AOC. See here: http://www.snom.com/wiki/index.php/FAQs#Q:_How_to_show_billing_information_on_the_phone_display.3F Regards, Sven On Friday 22 September 2006 17:31, Ale wrote: Hi all, Can anyone help me... i need to display the cost of a call during a conversation on a sip or iax phone. I see on voip-info that some snom phone support sendtext application, but i don't know how to update the message with the cost on the phone during the conversation. Every suggestion is apreciated. Thx, Bye Bye Ale ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- snom technology AG Gradestraße 46 D-12347 Berlin Sven Fischer fax +49 30 39833111 PSTN/ENUM +49 30 39833434 mailto:[EMAIL PROTECTED] http://www.snom.com -- -- --- See our Docs, FAQs, etc at: http://snom.com/wiki --- snom technology AG Gradestraße 46 D-12347 Berlin Sven Fischer fax +49 30 39833111 mailto:[EMAIL PROTECTED] http://www.snom.com --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp (foip)
linksys spa3102 or 2100 are known to work. Grandstream also should do it with recent firmware. Don't be fooled by what is written on the box, lot of ata's out there claim t.38. (while the firmware doesnt contain anything related to t.38) Zoa Christopher Corn wrote: lee, Thanks for the feedback. in most diagrams explaining t38, it shows, the sending fax machine connecting to a pots before connecting to a gateway,then the internet. but if i've read and understood correctly, the sending end can use an ATA with t38 support instead of a pots. in that case, where does the packetization of the t30 data happen? at the ATA? level i presume? http://www.answers.com/topic/t-30-protocol-figure-01-jpg also, can you recommened a good asterisk compatible ATA adapter with t38 support? i believe cisco has one. Thanks in advance. */Lee Howard [EMAIL PROTECTED]/* wrote: Christopher Corn wrote: May I ask, from your own personal experience. is it not necessaritly worth (the headaches) of investing mytime into setting up SPANDSP into my asterisk system, but rather invest it into going to a company, like packet8 that offers t38 conversion? I am not really in a position to tell you what something will be worth to you - especially when I've not even used that something myself. I know and use spandsp as a library, with IAXmodem and HylaFAX, but I do not have any experience with spandsp in txfax/rxfax applications or in its new T.38 gatewaying. I suspect that I'll eventually get into spandsp's T.38 aspects, but without that I've only had a limited amount of hands-on exposure to T.38 applications in the form of t38modem and Cisco gateways (which experience was somewhat disenchanting - mostly because of the gateway T.30 processing). If you have a T.38 fax machine or if you have a T.38-capable ATA connected to a fax machine and you do not have your own PSTN lines then I would suspect that it would be worthwhile to use T.38 pass-through on Asterisk 1.4 or OpenPBX in conjunction with a T.38-supporting FoIP provider. (Because otherwise you don't have any straight-forward, reliable means for faxing from your internal fax machines.) what does the future of faxing lean towards? before entering an era when all fax machines run the t38 protocol. will we see more t38 termination services or faxing through g711? T.38 is the end-all solution for faxing over IP networks. So I suspect that you will see the pervasiveness of T.38 implementations increase along with the pervasiveness of VoIP in general. That said, VoIP has its own fair share of problems that keeps it from being capable of replacing PSTN circuits entirely, and so as long as those problems are not generally resolvable for your average business or service provider then you'll continue to also see more of the same, traditional, modem-ing fax machines. So I strongly suspect that you'll see more of T.38, but I don't think that the PSTN (and traditional fax machines with it) is going away any time soon. from what i've read, using a service that does t38 termination, seems to be where i should go. I would say that it entirely depends upon whether or not you have PSTN lines yourself. If you do, then I would take whatever efforts you can to avoid the additional points of T.30 processing/relaying (therefore avoiding T.38 gatewaying). But if you do not have PSTN lines, then take whatever efforts you can to properly implement T.38 to your FoIP provider who will gateway for you. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Dual core
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I have a few dual core that I have installed Asterisk on without any issues. Hi Bill! Sure you don't have any issues, but do you take any advantage of dual core processor? Why would I pay for something if I can't profit from it? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Dual core
Asterisk is very happy on dual core. It greatly reduces load. We just put a Pentium-D in poduction last week and it is working verry well. We have a Core 2 Duo on order that we should be putting in production next week. MATT--- Hi Matt! Thank you for this information. Can you please tell me if you weight Asterisk, does it divide that job on both processors or it's only one that does the job? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Dual core
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... My home Asterisk server is running dual proc dual core zeon 3ghz, seems happy, no crashes that I didn't bring about myself. ;) mpg123 does occasionally hang a pid at 100% now and then, but it does that on single proc/single core systems too. Hi Nick! You should use native MOH. Than you won't have that problems with mpg123. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with zaptel 1.4b2 and X101P Wildcard
On Sat, Sep 23, 2006 at 09:22:50PM +0200, Morten Isaksen wrote: Hi! I was trying to upgrade my Asterisk 1.2.1 at home (not the [EMAIL PROTECTED] dist) to 1.4b2 but ran into problems with zaptel. The OS is Fedora Core 3. When I start zaptel it fails with this error: [EMAIL PROTECTED] zaptel-1.4.0-beta1]# service zaptel start Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: tor2 wct4xxp wct1xxp wcfxo wctdmRunning ztcfg: ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) [FAILED] [EMAIL PROTECTED] zaptel-1.4.0-beta1]# ztcfg -vvv Notice: Configuration file is /etc/zaptel.conf line 235: Unable to read Zaptel version information. Zaptel Version: $êþP¦0 Echo Canceller: Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) What exactly is channel 1? Maybe you got the wrong number? cat /proc/zaptel/* -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Very high ping times from 7960 phones
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times 150ms. Every single one. I've scoured the settings on the 7960's and have looked and looked for why this might be the case. Cisco ata's (186) on the same network report ~ 10 ms. An xlite softphone reports ~ 5ms regardless of what computer it's installed on. I have two 7960 phones with 7.4 firmware and sip show peers tells me that response time is 70 and 72 ms. Hope this helps. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom MWI not turning off when message picked up.
Hi, I've recently got a Snom 300 phone. When I set it up and a VM was left the light flashed as excpected. I can use the soft button with the Tick sign on it to go straight to voicemail, all fine so far. However once the message is picked up and listened to, the light still flashes? I'm using Asterisk 1.2.9.1 (Trixbox) and here is the phone details:- Application-Version:snom300-SIP 6.0.3 Rootfs-Version: snom300 jffs2 v3.36 Firmware-URL: Production Information: Mac:00041325244C;Version:Standard;Hardware:snom300 (MB V3.2_A11);Date: 04.04.06;Copyright(C) snom technology AG Any clues as to why the light is not being told to stop flashing. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Very high ping times from 7960 phones
On 09:42, Mon 25 Sep 06, Tomislav Par?ina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times 150ms. Every single one. I've scoured the settings on the 7960's and have looked and looked for why this might be the case. Cisco ata's (186) on the same network report ~ 10 ms. An xlite softphone reports ~ 5ms regardless of what computer it's installed on. I have the same here. All between 150 and 250 ms. The phones do work perfectly, only the time in sip show peers is higher then any other phone/device. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ougoing calls problem
Hi to all. I've registred my Asterisk 1.2.12.1 to a VoIP Service Provider and I've some problem with outgoing calls: there is a big delay for bidirectional audio flow. By traces, I've observed that several 200 OK SIP messages are sent by my SIP Provider until ACK is riceved. Maybe the 200 OK messages sequence freezes Asterisk introducing delay for biderctional audio flow. Can anyone tell me if there is some option to set in order to manage sip messages time or similar? Moreover..when I attempt to make an outgoing call with option canreinvite=yes, Asterisk notifies the follow message? Sep 20 14:13:42 WARNING[2373]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x819b240', 10 retries! Can anyone tell me what it does mean and how to fix it? Thanks, -- * (o ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Very high ping times from 7960 phones
I can confirm the same. It doesnt mean the audio will be delayed, the phone is just slow with replying to the sip messages. Zoa Michiel van Baak wrote: On 09:42, Mon 25 Sep 06, Tomislav Par?ina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times 150ms. Every single one. I've scoured the settings on the 7960's and have looked and looked for why this might be the case. Cisco ata's (186) on the same network report ~ 10 ms. An xlite softphone reports ~ 5ms regardless of what computer it's installed on. I have the same here. All between 150 and 250 ms. The phones do work perfectly, only the time in sip show peers is higher then any other phone/device. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Very high ping times from 7960 phones
Michiel van Baak wrote: On 09:42, Mon 25 Sep 06, Tomislav Par?ina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times 150ms. Every single one. I've scoured the settings on the 7960's and have looked and looked for why this might be the case. Cisco ata's (186) on the same network report ~ 10 ms. An xlite softphone reports ~ 5ms regardless of what computer it's installed on. I have the same here. All between 150 and 250 ms. The phones do work perfectly, only the time in sip show peers is higher then any other phone/device. That is a classic (and, AFAIK innocuous) behavior of the original Cisco ATA-186 ATAs as well. Nobody was ever able to explain why they are that way, but it seems to normal behavior. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's
Hi, On Centos IRQBalance should already be available. You should be able to run 'setup' from a console/terminal, go to System Services enable irqbalance. It will then be enabled on boot. To start it without re-booting, use service irqbalance start If it's already marked as enabled in the services list, the problem is elsewhere. Hope this helps, Robert Jenkins. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bart Fisher Sent: 25 September 2006 02:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's Hmm, this must not be installed: # locate irqbalance # /usr/src/kernels/2.6.9-34.0.2.EL-smp-i686/include/config/irqbalance.h How do I install this? Bart Álvaro Palma wrote: It appears that CPU1 in not taking any interrupts - What steps do I need to do bring up CPU1 and share IRQ requests for a Linux noob? Run the IRQ balance daemon (/usr/sbin/irqbalance). It's part of the kernel-utils RPM. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's
I'm not sure irqbalance is a good idea. (although i'm not familiar with it, its sounds like it balances it all the time, not just spreads it and leaves it). Maybe its best to do it manually ?. have a look at something i wrote ages ago: http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html and search for _Put your networkcard and pri card on a different CPU Zoa. _ Robert Jenkins wrote: Hi, On Centos IRQBalance should already be available. You should be able to run 'setup' from a console/terminal, go to System Services enable irqbalance. It will then be enabled on boot. To start it without re-booting, use service irqbalance start If it's already marked as enabled in the services list, the problem is elsewhere. Hope this helps, Robert Jenkins. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bart Fisher Sent: 25 September 2006 02:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's Hmm, this must not be installed: # locate irqbalance # /usr/src/kernels/2.6.9-34.0.2.EL-smp-i686/include/config/irqbalance.h How do I install this? Bart Álvaro Palma wrote: It appears that CPU1 in not taking any interrupts - What steps do I need to do bring up CPU1 and share IRQ requests for a Linux noob? Run the IRQ balance daemon (/usr/sbin/irqbalance). It's part of the kernel-utils RPM. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Very high ping times from 7960 phones
On 04:11, Mon 25 Sep 06, Brian Capouch wrote: That is a classic (and, AFAIK innocuous) behavior of the original Cisco ATA-186 ATAs as well. Nobody was ever able to explain why they are that way, but it seems to normal behavior. It really is something in the SIP image of the 7960. I have one at home that runs SCCP with chan_sccp and that one is only 4 ms away. The sip image on the 7960 is slow overall. I find the menu's and directories slow as well. But besides that the audio and stuff works great on the SIP image so I dont think it's actually a problem (unless you set 'qualify=100' in sip.conf. Now I think about it, there's one issue with the audio. If you put a call on hold on the 7960 SIP and unhold them, it sometimes takes 2 or 3 seconds before the audiostream is connected again. This is the only issue I can find in our ticket system. My personal favourite is the sccp image, but lots of ppl report crashes and stuff with chan_sccp (some of them I can confirm) so that's why not a lot of ppl will use it. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AgentCallbacklogin in Asterisk1.4 beta2
hi,I try The asterisk1.4 beta2, it said Callback mode (AgentCallbackLogin) is now deprecated, and let us use dialplan logic.I checked the docs/queues-with-callback-members.txt file, this example is complex, i can't understand it.I want use dialplan logic for AgentCallbackLogin only. Anybody have some easy explain example? ThanksLi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail greeting
Hi, When I use Voicemail function, there is a default system greeting before voicemail recording. Is it possible to change that greeting? How? unplug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970 - DTMF
In sip.conf for one friend (Cisco 7970 phone) I have define this dtmfmode=inband And in xml.conf of that phone I have preferredCodecnone/preferredCodec dtmfAvtPayload101/dtmfAvtPayload dtmfDbLevel3/dtmfDbLevel dtmfOutofBandnone/dtmfOutofBand But DTMF doesn't work for that phone. Phone establishes call using g711 alaw codec. How should I configure phone and sip.conf to make DTMF work? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: notransfer local channel on redirect
On Fri, 22 Sep 2006 17:51:10 -0400 Juan Pablo Abuyeres [EMAIL PROTECTED] wrote: Wow... I am looking for exactly the same feature. Did you find out how to do it??? Not yet, but maybe our chances rise now as we are already two... regards christian On Thu, 2006-09-21 at 18:41 +0200, Christian Benke wrote: 2006/9/21, Benko [EMAIL PROTECTED]: notransfer-option(\n) on redirected calls? sorry, it is called no release quote:(the n stands for no release) so is there a way to tell asterisk to not release a local channel on a redirect so the billsec and duration is written to it? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A Strange doubt and problem
Hi Friends,I got a strange doubt and problem. Is there any problem for SIP protocol, if we configure Intercom (SIP), VoIP (SIP) and PSTN in a single server (that may be Asterisk or Trixbox).My First Experience:Initially I configured Asterisk in a system and created SIP extensions and VoIP with Teliax. It worked very fine for a few days. After that, I configured PSTN (Digium 04B and 20B) and making outgoing and receive incoming calls. After that, SIP protocol was down and unable to make calls to USA using Teliax using SIP. So, I configured IAX2 Teliax account and its working fine now. Why SIP protocol was down?My Second Experience:I have installed Trixbox ISO image in a system and configured as mentioned above. Now, I have faced the same problem. After configuring PSTN only, SIP protocol was dead. What may be the reason?Error Message: When I am making call to USA using Teliax service, it is telling that "All circutis are busy. Please try call later. Thank you".Please share your feelings and experiences. Looking forward to your response. Thank you.Regards,Chandra. All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Snom MWI not turning off when message picked up.
On 9/25/06, Mike Dent [EMAIL PROTECTED] wrote: Hi, I've recently got a Snom 300 phone. When I set it up and a VM was left the light flashed as excpected. I can use the soft button with the Tick sign on it to go straight to voicemail, all fine so far. However once the message is picked up and listened to, the light still flashes? I'm using Asterisk 1.2.9.1 (Trixbox) and here is the phone details:- Application-Version:snom300-SIP 6.0.3 Rootfs-Version: snom300 jffs2 v3.36 Firmware-URL: Production Information: Mac:00041325244C;Version:Standard;Hardware:snom300 (MB V3.2_A11);Date: 04.04.06;Copyright(C) snom technology AG Any clues as to why the light is not being told to stop flashing. Mike Having rebooted the phone it seems to have fixed things. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Line Pickup Problem
Dear Users I setup asterisk in my home and i want to just using asterisk for outgoing call from internet to my PSTN line but when i connect fxo port to phone line asterisk pickup the line first , i want to asterisk wait to somebody pickup the line if line not picked up after proper time the asterisk do that which setting i must to do in extentsion.conf or zapata.conf? Regards Mohsen Basirat ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call back
First hi
I am trying to make a call back system
I am using callme.php(click to call) to write the file at
/var/spool/asterisk/outgoing
And at the incoming context ,I match the
incoming did as follow
exten = 009613504768,4,system(elinks -dump http://127.0.0.1/click/callme.php?number=SIP/1234channel=${CHANNEL})
at callme,php I hangup the call by
system(asterisk -r -x 'soft hangup $channel' );
my troubles is I that know that its a silly
idea to make a call back system in this way,
2nd I dont hear a ringback or hangup
tone.
Please if you know a better way to that that please dont
hesitate to inform me .
Regards
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Re: [asterisk-users] Re: Very high ping times from 7960 phones
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times 150ms. Every single one. I've scoured the settings on the 7960's and have looked and looked for why this might be the case. Cisco ata's (186) on the same network report ~ 10 ms. An xlite softphone reports ~ 5ms regardless of what computer it's installed on. I have two 7960 phones with 7.4 firmware and sip show peers tells me that response time is 70 and 72 ms. Hope this helps. I can't tell you why either, but a ping from a linux command line shows sub-millisecond response (phone and asterisk on same lan segment), while the qualify response time is around 79 milliseconds. Just taking a pure guess (without doing any packet sniffing) is the qualify method sends a sip packet to the phone and waits for a response. It is entirely possible that qualify ping might involve multiple packet interactions. Also, the qualify ping must essentially pass through all of the asterisk code, IP stack, etc, on both ends. That value would be greater then a simple icmp ping. There are no settings in the cisco phones that would impact this. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 - DTMF
Tomislav Parčina wrote: In sip.conf for one friend (Cisco 7970 phone) I have define this dtmfmode=inband And in xml.conf of that phone I have preferredCodecnone/preferredCodec dtmfAvtPayload101/dtmfAvtPayload dtmfDbLevel3/dtmfDbLevel dtmfOutofBandnone/dtmfOutofBand But DTMF doesn't work for that phone. Phone establishes call using g711 alaw codec. How should I configure phone and sip.conf to make DTMF work? In asterisk sip.conf, use dtmfmode=rfc2833 for that extension, and in the SIPDefault.cnf boot file for the cisco, include: dtmf_inband: 1 dtmf_outofband: avt dtmf_db_level: 3 (you'll need to translate the above 7960 parameters into the 7970 xml parameters since I don't have a 7970 to play with.) Taking a wild-ass guess, you might be able to get by simply using the dtmfmode=rfc2833 parameter in asterisk without touching the phone. Try it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail greeting
unplug wrote: Hi, When I use Voicemail function, there is a default system greeting before voicemail recording. Is it possible to change that greeting? How? Call into voicemail as though you were going to listen to your messages, and press 0 for Mailbox Options. Then press 3 to record your name. You might want to go through each of the various voicemail options to see what else you might be missing. There are more options. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Comments on new system plan.
Paging: You can also use the server's audio card for paging, if it is close to the main amp. Beware of Bogen or Valcom as they mainly make FXO paging interfaces for trunk lines. Viking makes the paging toys you would want to look at. ABE: ABE is the supported version of Asterisk, same code but someone to call on the phone. I am sure that they can help you with your card. Depending on your comfort level you can select ABE or Public. No real draw backs, one benefit is that buying from Digium will keep the developers in free pizza land for longer. :-) POTS: While more exspensive a PRI T1 would better suit you. Yes it cost more, but it has some advantages. A channel bank is a little handier than an ATA, you just plug in and test to your hearts content. Just did a setup of this size, company was quoted 60k+ from the phone company so don't be afraid to setup PRI, Channelbanks for future testing or use as you will fall way under budget. On 9/22/06, Dave Fullerton [EMAIL PROTECTED] wrote: Greetings I'm in the process of planning my first production system and wondered if those with some experience would let me know if I'm doing anything stupid or have some suggestions. This is going to be used in a manufacturing facility with about 22 phones. About 10 of which are office staff. I'm not going to implement call recording, meetme, or queues or anything fancy at this point. I'll be using Polycom 601's and 501's for the office staff and 301's for the plant phones. I've already had a few phones set up in my office to test with and I've got what I need for provisioning figured out. I'll have all the phones set to canreinvite=yes and use the transfer functions of the phone. Voice mail will be provided for office staff. Since I don't have that many phones and everything will be on the LAN I'm just going to stick with ulaw for the codec. The planned server will be an HP Proliant ML110 G3 with a 3Ghz Pentium 630 processor, 1GB of RAM, and two 80GB SATA hard drives in a RAID1 (linux software raid) configuration. I'm planning on using the on-board gigabit network controller. I'll have about 8 POTS lines (no caller id or call waiting) connected to the system. I'm planning on using a Sangoma Remora A20004D (8 FXO with on-board echo canceler). Echo is actually my biggest fear of the whole project. There won't be any faxes coming through the server. For the few analog phones that may be used I'll be using some SPA-3000's I already have on hand for FXS ports. We will have need for overhead paging eventually. This is one area I'm a little unsure of. My current off-the-cuff plan is to use a Budgetone phone with the headset jack plugged into the amp and set to auto-answer. (Saw this on the wiki). I've looked at some of the other devices on the wiki but I'm not sure how to implement them. Any advice would be appreciated. I'm also trying to decide whether I want to use Asterisk Business Edition or stick with the downloaded version. Money really isn't a big issue but I'm not sure what the pros and cons are. I know I would get a hardened version thats not likely to have many bugs and support from Digium, but I'm not sure what version of asterisk it is or what features are in the 1.2 branch that aren't in ABE or vice-versa. I'm assuming ABE is in binary form, will it even work with Sangoma hardware, is it distro sensitive? (I was going to call Digium but ran out of time this week). I think that covers it. If anyone has some tips or constructive criticism I would appreciate hearing it. Thanks! -Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp (foip)
Lee Howard wrote: On Sun, Sep 24, 2006 at 01:58:21PM -0700, Christopher Corn wrote: A couple of faxing methods im confused about. The pass through method, sending fax data over G711 codec versus Relay method, t30 to t38 conversion Can someone explain to me why the pass through method doesn't require t30 to t38 conversion ( or does it do it?)? i believe the conversion to t38 is so that it can be routed through a packet network and then back to t30 so that the fax machine can understand. why is it that if you use a pass through method, and your still passing through a packet network, you dont need to convert to t38 and t30? Be careful about your wording. People here generally refer to pass through as T.38 pass-through and not G.711 pass-through. I think that if you understood how faxing works you would see that your questions here don't really make sense. In traditional PSTN faxing you have a total of two endpoints performing T.30 protocol. In a simplified form, the sender takes scanner image data and modulates it (into an audio waveform) and then passes that audio over the PSTN to the receiver which demodulates it (takes the audio and turns it into data again). As long as the demodulated data is identical to the original data, then everything should be okay... for the most part. However, if you start to consider audio corruption on the PSTN, then that's where difficulties start to ensue. If you have some audio, modulated data, and then you compress it or fracture it or otherwise corrupt it, then there's no possible way that the demodulator is going to be able to come up with the original data. Now introduce VoIP telephony... where a small amount of audio corruption (jitter) is anticipated on the UDP channel... and mix it with faxing and hopefully you can see how it just doesn't work well. VoIP is packetized audio passed over an IP network. Packetized audio is nothing new. ISDN circuits have had it for a long time now. Those circuits are digital - meaning the audio waveform is digitized at 8000 Hz... so the audio is represented with bytes and are packetized into frames. Those traditional digital circuits are designed to prevent any loss of that data. VoIP works similarly, except that the medium is lossy UDP/IP networking. ISDN doesn't packetize voice. ISDN is a strict circuit switched TDM system. Since VoIP works on *IP* networks, and since IP networks already handle data communication very well, there really is no reason to perform the modulation or the demodulation - just send the raw data through. So that's basically the punchline of T.38... it's fax protocol without the traditional modems involved. Then you have FoIP. However, these days the world is a hybrid of VoIP and PSTN environments (mostly PSTN still), and thus anyone using T.38 will need to have a gateway involved somewhere along the call path that can do that traditional modulation/demodulation. That is what the T.38 gateway is. If a T.38 relay does not act as a gateway (i.e. no modulators) then it performs only T.38 pass-through - meaning it only is useful for situations where calls are end-to-end T.38 or where an external FoIP service provider is used. Because of the way things work T.38 gateways will not only need to have traditional modems (hard or soft) but will also need to perform T.30. So when faxing with T.38 and the call is not end-to-end T.38 then you have at least three points along the call path performing T.30 (versus the traditional scenario of just two). So, to answer your questions... Why does using G.711 not require T.38? Because from the viewpoint that the question was given, G.711 and T.38 are competing approaches. T.38 was designed to replace G.711. You can packetize G.711 audio just fine without converting it to anything else. So when faxing with G.711 T.38 is not involved because its basically mimicking the old-style traditional PSTN faxing, except that the audio is passing over a different (less-reliable) medium. So the reason that T.38 exists is because UDP/IP is lossy and is not therefore reliable for the purposes of faxing with G.711 unless the communication can be guaranteed to be nearly lossless. For those that work on lossy channels, G.711 will just not work reliably. Lossless channels are only a part of it. If you look at http://www.soft-switch.org/foip-with-real-atas.html you will see examples of other problems that happen with a wide range of ATAs. Once that have FAX support modes, yet cann't possibly ever work with FAX. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail greeting
Hi, I am experiencing some similar difficulties with voicemail. I have an IP phone on extension 101 and I do not know how to dial in to access the voicemail options. When I dial 101 I have tried pressiong * and 0 but I do not get to the mail box menu. Can someone please help? -- Original message -- From: Rich Adamson [EMAIL PROTECTED] unplug wrote: Hi, When I use Voicemail function, there is a default system greeting before voicemail recording. Is it possible to change that greeting? How? Call into voicemail as though you were going to listen to your messages, and press "0" for Mailbox Options. Then press "3" to record your name. You might want to go through each of the various voicemail options to see what else you might be missing. There are more options. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.di gium.c om/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Trunk with Alcatel 4200 PABX
Hi guys,
I'm tryng to estabilish a trunk with an Alcatel 4200 Pabx. In asterisk i'm using a Digium TE110P as E1.
The span is ok with green led, but when pabx make calls to asterisk, i received this error:
asterisk*CLI
!! Unexpected Channel selection 3
-- Accepting call from '3069' to '30818559' on channel 1/31, span 1
-- Executing Dial(Zap/31-1, SIP/[EMAIL PROTECTED]|20|Tt) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/fp-33133000-09fdfa90 is ringing
!! Unexpected Channel selection 3
-- SIP/fp-33133000-09fdfa90 answered Zap/31-1
!! No channel map, no channel, and no ds1? What am I supposed to identify?
!! Unable to add IE 'Channel Identification'
== Spawn extension (default, 30818559, 1) exited non-zero on 'Zap/31-1'
-- Hungup 'Zap/31-1'
Sep 23 20:13:25 WARNING[3765]: chan_zap.c:8788 pri_dchannel: Hangup requested on unconfigured channel 255/255 span 1
Sep 23 20:13:29 WARNING[3765]: chan_zap.c:8788 pri_dchannel: Hangup requested on unconfigured channel 255/255 span 1
My configuration files is:
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
/etc/asterisk/zapata.conf
trunkgroup = 1,16
spanmap = 1,1,1
language=uk
context=default
switchtype=euroisdn
signalling=pri_net
group=1
callgroup=1
pickupgroup=1
immediate=no
echocancel=yes
channel = 1-15,17-31
/etc/asterisk/extensions.conf
# SIP - Alcatel
exten= 331330XX,1,Dial(Zap/g1/${EXTEN})
exten= 331330XX,2,Hangup
# Alcatel - SIP
exten= _,1,Dial(SIP/[EMAIL PROTECTED],20,Tt) #
exten= _,2,Hangup
What can be hrong in this configuration ???
Thanks.
-- Frederico Madeira[EMAIL PROTECTED]
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Re: [asterisk-users] AGI Errors
You can put fastagi-mapping.properties in the root dir of the classes of your project. []'s, Edmilson Santana Unitech Tecnologia de Informação (http://www.unitech.com.br/) [EMAIL PROTECTED] wrote: We try to work with asterisk-java and FastAGI (for our diploma). We did everything like on asterisk-java tutorial But still 2 errors appear: ...the server starts up correctly and we now make call to the agi-extension... 15.09.2006 17:41:51 net.sf.asterisk.util.impl.JavaLoggingLog error SEVERE: Unable to create AGIScript instance of type HelloAgiScript 15.09.2006 17:41:51 net.sf.asterisk.util.impl.JavaLoggingLog error SEVERE: No script configured for URL 'agi://localhost/hello.agi' (script 'hello.agi') What could that be? One thing on asterisk-java tutorial we are not sure: ...called fastagi-mapping.properties that must be on the classpath... Which classpath hast o be defined? Thanks for your help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail greeting - How to access vociemail
On Mon, 25 Sep 2006 [EMAIL PROTECTED] wrote: Hi, I am experiencing some similar difficulties with voicemail. I have an IP phone on extension 101 and I do not know how to dial in to access the voicemail options. When I dial 101 I have tried pressiong * and 0 but I do not get to the mail box menu. Can someone please help? You are the one who would define what to dial or how to enter your voicemail account. In your /etc/asterisk/extensions.conf file you could add a line such as the following in the correct place. The correct place means in a context [somename] that matches a location your extension 101 would see. exten = 1234,1,VoiceMailMain() With that line in the correct place, you could now dial 1234 to enter voicemail. If you have installed Asterisk from a package like TrixBox, then mention that in your message. The simple answer would be *98 is the default. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] progress problems from SIP to PRI
Hi! I have problems when bridging from SIP to PRI. As soon as the setup message is sent, Asterisk replies with 183 to the sender. Although there is nor PROGRESS message received, the 183 is sent as the SIP channel received a voice frame and thus activates early media. I wonder why Asterisk reads from the PRI although there was no PROGRESS message received yet. I want to get rid of this 183 - it should be sent only when a PROGRESS is received on the PRI. Can this be configured somehow? thanks klaus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dual core
I believe asterisk for the most part is single threaded, you will get some advantages by having other system processes use the extra Processor/Core, but I don't think asterisk will use alot of the other CPU. On 9/25/06, Tomislav Parčina [EMAIL PROTECTED] wrote: Asterisk is very happy on dual core. It greatly reduces load. We just put a Pentium-D in poduction last week and it is working verry well. We have a Core 2 Duo on order that we should be putting in production next week. MATT---Hi Matt!Thank you for this information. Can you please tell me if you weight Asterisk, does it divide that job on both processors or it's only one that does the job? --Tomislav ParčinaLama Computers SplitStinice 12, 21000 SplitTel.: +385(21)495148Mob.: +385(91)1212148SIP: [EMAIL PROTECTED]e-mail: tparcina#lama.hr http://www.lama.hr___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztcfg / X100P question
Hi, folks. I've got an X100P Wildcard here. I get an odd error when running ZTCFG on it. === pbx1:~# asterisk -V Asterisk SVN-branch-1.2-r43509 pbx1:~# lsmod Module Size Used by wcfxo 13184 0 zaptel202148 1 wcfxo crc_ccitt 2208 1 zaptel pbx1:~# dmesg | grep -i zap Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-branch-1.2-r1468 Echo Canceller: KB1 pbx1:~# ztcfg -vv Notice: Configuration file is /etc/zaptel.conf line 10: Cannot get number of tones for channel 1 line 10: Cannot init tones for channel 1 2 error(s) detected === I've run google on the errors, but all I turn up are Asterisk source code hunks that really don't explain to me what *triggers* that error. Could someone suggest to me what the issue could be? -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Cisco 7970 - DTMF
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... In asterisk sip.conf, use dtmfmode=rfc2833 for that extension, and in the SIPDefault.cnf boot file for the cisco, include: dtmf_inband: 1 dtmf_outofband: avt dtmf_db_level: 3 (you'll need to translate the above 7960 parameters into the 7970 xml parameters since I don't have a 7970 to play with.) Taking a wild-ass guess, you might be able to get by simply using the dtmfmode=rfc2833 parameter in asterisk without touching the phone. Try it. Hi Rich! dtmfmode=rfc2833 in sip.conf with dtmfAvtPayload101/dtmfAvtPayload dtmfDbLevel3/dtmfDbLevel dtmfOutofBandavt/dtmfOutofBand In sepmac.cnf.xml works well. Thank you! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dual core
For what we do with Asterisk(lots of meetme and Zap - IAX2) It does spread the load across both cores. In our initial comparisons for equal call traffic, the P4-D had half or the average loadavg for a 6 hour time period of the P4 of the same speed. MATT--- On 9/25/06, Tomislav Parčina [EMAIL PROTECTED] wrote: Asterisk is very happy on dual core. It greatly reduces load. We just put a Pentium-D in poduction last week and it is working verry well. We have a Core 2 Duo on order that we should be putting in production next week. MATT--- Hi Matt! Thank you for this information. Can you please tell me if you weight Asterisk, does it divide that job on both processors or it's only one that does the job? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp (foip)
On Mon, Sep 25, 2006 at 08:14:27PM +0800, Steve Underwood wrote: Now introduce VoIP telephony... where a small amount of audio corruption (jitter) is anticipated on the UDP channel... and mix it with faxing and hopefully you can see how it just doesn't work well. VoIP is packetized audio passed over an IP network. Packetized audio is nothing new. ISDN circuits have had it for a long time now. Those circuits are digital - meaning the audio waveform is digitized at 8000 Hz... so the audio is represented with bytes and are packetized into frames. Those traditional digital circuits are designed to prevent any loss of that data. VoIP works similarly, except that the medium is lossy UDP/IP networking. ISDN doesn't packetize voice. ISDN is a strict circuit switched TDM system. He didn't say anything about compressed, Steve; yeah, ISDN frames the bytes it sends. It sends them isochronously, certainly, so jitter is less of a problem by a couple orders of magnitude or mode, but they're still sent in packets. Just not *IP* packets. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Missing sound in spanish from 1.4 beta2
On Sun, Sep 24, 2006 at 11:00:41PM -0500, Jason Parker wrote:
- Jay R. Ashworth [EMAIL PROTECTED] wrote:
I will assume that you are a native speaker; I'm not equipped to
evaluate whether ... well, anyway. Anyone know where those prompts
actually *came* from? :-)
The Spanish language core-sounds came from Allison Smith. She is the
same person who does the English language sounds (You can get English
and Spanish language prompts from Allison Smith, or French language
prompts from June Wallack {who does the French language core-sounds
prompts}, via http://www.digium.com/en/products/voice/ - they both do
very good work).
Well, with all due respect to Allison, Jason, apparently someone for
whom Spanish *is* a primary language disagrees on that point. :-)
Cheers,
-- jra
--
Jay R. Ashworth[EMAIL PROTECTED]
Designer Baylink RFC 2100
Ashworth AssociatesThe Things I Think'87 e24
St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274
That's women for you; you divorce them, and 10 years later,
they stop having sex with you. -- Jennifer Crusie; _Fast_Women_
___
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Re: [asterisk-users] rxfax, spandsp and lack of ecm
Hi Steve, On 9/14/06, Steve Davies [EMAIL PROTECTED] wrote: On 9/14/06, Steve Underwood [EMAIL PROTECTED] wrote: Steve Davies wrote: [snip] This looks pretty good I have to say - The ECM seems as if it may be a little intolerant... On a fax machine where I got 100% success in the past with 0.0.2, I am now getting result (60) Disconnected after permitted retries. on about every 4th page. Is the ECM tolerance level tuneable in spandsp, or is this hard-defined in the standard? Is it just a matter of changing: #define MAX_MESSAGE_TRIES 3 Your problem probably has nothing to do with tolerance. If an exchange doesn't succeed after 3 tries, it is unlikely to ever succeed. You are probably hitting a bug. It is new code. :-) Can you enable debug with |debug on the command line to rxfax/txfax, and send me the log? I've been following the snapshot changes, and trying to dig into my RxFax problem a little more - I think I have finally found a case which breaks things, and perhaps some useful logs... I added loads of extra debug statements to track the number of ECM retries that were occuring, and to log the retries value whenever it was set or changed. It turns out that the problem I see occurs both with ECM and with Non-ECM capable machines! s-retries is occasionally increased due to t4 timeouts, and I presume that the retries never succeed, as I have yet to see a fax send recover from this state (Perhaps the fax machines do not like the drop back from phase C to phase B?) The trace during the t4 timeout retries is: Sep 25 13:36:03 DEBUG[6532]: Tx: DIS with final frame tag Sep 25 13:36:03 DEBUG[6532]: Tx: ff 13 80 00 ce f8 c4 80 89 80 80 80 98 80 80 80 80 80 00 Sep 25 13:36:05 DEBUG[6532]: Send complete in phase T30_PHASE_B_TX, state 15 Sep 25 13:36:05 DEBUG[6532]: Send complete in phase T30_PHASE_B_TX, state 15 Sep 25 13:36:05 DEBUG[6532]: Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX Sep 25 13:36:05 DEBUG[6532]: FLOW FAX Set rx type 4 Sep 25 13:36:05 DEBUG[6532]: FLOW FAX Set tx type 0 Sep 25 13:36:05 DEBUG[6532]: HDLC carrier up in state 15 Sep 25 13:36:08 DEBUG[6532]: T4 timeout in phase T30_PHASE_B_RX, state 15 Sep 25 13:36:08 DEBUG[6532]: timer_t4_expired bumped retries count to 2 Sep 25 13:36:08 DEBUG[6532]: Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX Sep 25 13:36:08 DEBUG[6532]: FLOW FAX Set rx type 0 Sep 25 13:36:08 DEBUG[6532]: FLOW FAX Set tx type 4 Sep 25 13:36:08 DEBUG[6532]: DIS: This happens 4 times, and then the exchange is cancelled. It looks as thought this state is initiated when the following occurs at the start of a page: Non-ECM trace... Sep 25 12:59:32 DEBUG[4533]: Send complete in phase T30_PHASE_D_TX, state 20 Sep 25 12:59:32 DEBUG[4533]: Changing from state 20 to 10 Sep 25 12:59:32 DEBUG[4533]: Changing from phase T30_PHASE_D_TX to T30_PHASE_C_NON_ECM_RX Sep 25 12:59:32 DEBUG[4533]: FLOW FAX Set rx type 8 Sep 25 12:59:32 DEBUG[4533]: FLOW FAX Set tx type 0 Sep 25 12:59:32 DEBUG[4533]: Non-ECM carrier up in state 10 Sep 25 12:59:32 DEBUG[4533]: HDLC carrier up in state 10 Sep 25 12:59:33 DEBUG[4533]: Non-ECM carrier training failed in state 10 Sep 25 12:59:39 DEBUG[4533]: T2 timeout in phase T30_PHASE_C_NON_ECM_RX, state 10 Sep 25 12:59:39 DEBUG[4533]: Changing from phase T30_PHASE_C_NON_ECM_RX to T30_PHASE_B_TX Sep 25 12:59:39 DEBUG[4533]: FLOW FAX Set rx type 0 Sep 25 12:59:39 DEBUG[4533]: FLOW FAX Set tx type 4 Sep 25 12:59:39 DEBUG[4533]: Start receiving document Sep 25 12:59:39 DEBUG[4533]: Changing from state 10 to 15 Sep 25 12:59:39 DEBUG[4533]: DIS: Sep 25 12:59:39 DEBUG[4533]: ...0= Store and forward Internet fax (T.37): Not set Sep 25 12:59:39 DEBUG[4533]: .0..= Real-time Internet fax (T.38): Not set And a similar ECM enabled trace... Sep 25 14:25:11 DEBUG[6532]: Send complete in phase T30_PHASE_D_TX, state 13 Sep 25 14:25:11 DEBUG[6532]: Changing from state 13 to 10 Sep 25 14:25:11 DEBUG[6532]: Changing from phase T30_PHASE_D_TX to T30_PHASE_C_ECM_RX Sep 25 14:25:11 DEBUG[6532]: FLOW FAX Set rx type 8 Sep 25 14:25:11 DEBUG[6532]: FLOW FAX Set tx type 0 Sep 25 14:25:11 DEBUG[6532]: HDLC carrier up in state 10 Sep 25 14:25:11 DEBUG[6532]: HDLC carrier up in state 10 Sep 25 14:25:12 DEBUG[6532]: HDLC carrier training failed in state 10 Sep 25 14:25:18 DEBUG[6532]: T2 timeout in phase T30_PHASE_C_ECM_RX, state 10 Sep 25 14:25:18 DEBUG[6532]: Changing from phase T30_PHASE_C_ECM_RX to T30_PHASE_B_TX Sep 25 14:25:18 DEBUG[6532]: FLOW FAX Set rx type 0 Sep 25 14:25:18 DEBUG[6532]: FLOW FAX Set tx type 4 Sep 25 14:25:18 DEBUG[6532]: Start receiving document Sep 25 14:25:18 DEBUG[6532]: Changing from state 10 to 15 Sep 25 14:25:18 DEBUG[6532]: DIS: Sep 25 14:25:18 DEBUG[6532]: ...0= Store and forward Internet fax (T.37): Not set Sep 25 14:25:18 DEBUG[6532]: .0..= Real-time Internet fax (T.38): Not set Sometimes I can get 5 or 6 pages through before this occurs. this is on fax machines which appeared
Re: [asterisk-users] RE: Dual core
On Mon, Sep 25, 2006 at 09:04:41AM +0200, Tomislav Par?ina wrote: Sure you don't have any issues, but do you take any advantage of dual core processor? Why would I pay for something if I can't profit from it? Well, it would seem to me that with a little attention to processor affinity, you could run your Asterisk and DBMS code on one processor, and let the other one handle the device interrupts; ie: that sounds to me like a feature, rather than a bug... Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Very high ping times from 7960 phones
On Mon, Sep 25, 2006 at 04:11:33AM -0400, Brian Capouch wrote: I have the same here. All between 150 and 250 ms. The phones do work perfectly, only the time in sip show peers is higher then any other phone/device. That is a classic (and, AFAIK innocuous) behavior of the original Cisco ATA-186 ATAs as well. Nobody was ever able to explain why they are that way, but it seems to normal behavior. None of y'all hang out on NANOG. :-) Cisco has built routers for a living for 25 years: they always prioritize things like ping response lower than actually getting the work done. This is probably a symptom of that. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] progress problems from SIP to PRI
On 9/25/06, Klaus Darilion [EMAIL PROTECTED] wrote: Hi! I have problems when bridging from SIP to PRI. As soon as the setup message is sent, Asterisk replies with 183 to the sender. Although there is nor PROGRESS message received, the 183 is sent as the SIP channel received a voice frame and thus activates early media. I wonder why Asterisk reads from the PRI although there was no PROGRESS message received yet. I want to get rid of this 183 - it should be sent only when a PROGRESS is received on the PRI. Can this be configured somehow? From the SIP RFC: The 183 (Session Progress) response is used to convey information about the progress of the call that is not otherwise classified... My reading of this is that the 183 message you describe is perfectly legal and appropriate. The PRI has made progress by sending a SETUP request. What nature of problem does this cause you? Perhaps there is a better solution than stopping legal SIP messages being sent :) Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Opinions on Aastra 480i CT?
Looks good, great price: http://www.aastratelecom.com/ipphones/pro_243.asp Anybody using these? How's the cordless? Does it play nice with * ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue failover and wrap time
I have a asterisk box with some queues for a call center and need help on two points: 1. I have a scenario where if a queue has no agents logged in, an inbound call should immediately failover to the failover destination for that queue. However, this does not seem to be working in that, even if no agents are logged in, the call goes into the queue. Is there a config option I'm missing (or did I misunderstand how the failover works?) 2. I have the wrap time set ideally for agents, but sometimes they want to pickup the next call in queue before the wrap time expires. Is there a way for agents to grab the next call? -Ahmed- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk to cell phone network
I would like to know if any of you have a cell phone like a pci card to install in one slot to my asterisk server?, I want to make a connection from my asterisk to the cellular network.Does anybody has a solution like this?Regards,Yrving Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Very high ping times from 7960 phones
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times 150ms. Every single one. I've scoured the settings on the 7960's and have looked and looked for why this might be the case. Cisco ata's (186) on the same network report ~ 10 ms. An xlite softphone reports ~ 5ms regardless of what computer it's installed on. I have two 7960 phones with 7.4 firmware and sip show peers tells me that response time is 70 and 72 ms. Hope this helps. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks everyone for all the info. I'm going to assume this is normal for this phone and doesn't adversely affect performance. I've installed these on several different asterisk systems (including 1.4) and it's all the same. Anyone running sip firmware 8.4 know if this is 'fixed' ? Again thanks for all the help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] REQUERIMIENTOS TE110P Y PANASONIC TDA620
SEÑORES DIGIUMLA PRESENTE ES PARA CONFIRMAR QUE REQUERIMIENTOS DE HARDWARE Y/O SOFTWARE SON NECESARIOS PARA CONECTAR UNA CENTRAL TELEFONICA PANASONIC TDA620 QUE TIENE INSTALADA UNA TARJETA SE5E18 (E1) CON UNA TARJETA TE110P. ATENTAMENTEDIEGO FERNANDO GÜIZA ARCE ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT: Opinions on Aastra 480i CT?
It's excellent home phone. I wouldn't use it in a business environment. No hold, no one-touch voicemail. However, it works great! /R -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Monday, September 25, 2006 10:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] OT: Opinions on Aastra 480i CT? Looks good, great price: http://www.aastratelecom.com/ipphones/pro_243.asp Anybody using these? How's the cordless? Does it play nice with * ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Spam? [asterisk-users] OT: Opinions on Aastra 480i CT?
I have this phone on my desk. It works very very well! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Monday, September 25, 2006 10:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Spam? [asterisk-users] OT: Opinions on Aastra 480i CT? Looks good, great price: http://www.aastratelecom.com/ipphones/pro_243.asp Anybody using these? How's the cordless? Does it play nice with * ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI Errors
i have all files in the same directory: c:\agi (asterisk-java-0.2.jar, fastagi-mapping.properties, HelloAgiScript.class and HelloAgiScript.java). My slasspath is also c:\agi Did you mean this? But i get still the following errors: if i start it with eclipse: ... INFO: Received connection. 25.09.2006 16:36:30 net.sf.asterisk.util.impl.JavaLoggingLog error Severe: Unable to create AGIScript instance of type HelloAgiScript 25.09.2006 16:36:30 net.sf.asterisk.util.impl.JavaLoggingLog error Severe: No script configured for URL 'agi://localhost.ch/hello.agi' (script 'hello.agi') if i start from the console another error occurs: INFO: Received connection. 25.09.2006 16:42:15 net.sf.asterisk.util.impl.JavaLoggingLog error Severe: Resource bundle 'fastagi-mapping' is missing. 25.09.2006 16:42:15 net.sf.asterisk.util.impl.JavaLoggingLog error Severe: No script configured for URL 'agi://localhost/hello.agi' (scri pt 'hello.agi') What could that be? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] progress problems from SIP to PRI
Hi Steve! The problem is following PSTN PSTN | | | | E1 E1 | | PBX1--E1--Asterisk1---SIP---Asterisk2--E1--PBX2 2 offices. Asterisk between the PBX and the PSTN. Calls between the offices are routed directly via SIP (toll bypass). SETUP--- --SETUPACK-- ---INVITE--- ---100-- SETUP--- ---183-- --PROGRESS-- --SETUPACK-- --CALLPROCEEDING-- --ALERTING ---180-- --CALLPROCEEDING-- --ALERTING A user at PBX1 call a user at PBX2. Asterisk 2 sends 183 immediately after the SETUP from Asterisk2 to PBX2. This causes Asterisk1 to generate a PROGRESS message. Thus PBX1 activates inband audio. But, as PBX2 does not generate inband audio, the users at PBX1 do not hear ringback. As you can see, the 183 causes Asterisk go signal PROGRESS with inband audio although there is no inband audio (if there would be inband audio PBX2 would send PROGRESS too). My reading of this is that the 183 message you describe is perfectly legal and appropriate. The PRI has made progress by sending a SETUP request. IMO that's not a real progress. And as you see from the signaling, at site 1 there is a PROGRESS message which is not at site 2. regards klaus Steve Davies wrote: On 9/25/06, Klaus Darilion [EMAIL PROTECTED] wrote: Hi! I have problems when bridging from SIP to PRI. As soon as the setup message is sent, Asterisk replies with 183 to the sender. Although there is nor PROGRESS message received, the 183 is sent as the SIP channel received a voice frame and thus activates early media. I wonder why Asterisk reads from the PRI although there was no PROGRESS message received yet. I want to get rid of this 183 - it should be sent only when a PROGRESS is received on the PRI. Can this be configured somehow? From the SIP RFC: The 183 (Session Progress) response is used to convey information about the progress of the call that is not otherwise classified... My reading of this is that the 183 message you describe is perfectly legal and appropriate. The PRI has made progress by sending a SETUP request. What nature of problem does this cause you? Perhaps there is a better solution than stopping legal SIP messages being sent :) Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue failover and wrap time
On 9/25/06, Michelle Dupuis [EMAIL PROTECTED] wrote: I have a asterisk box with some queues for a call center and need help on two points: 1. I have a scenario where if a queue has no agents logged in, an inbound call should immediately failover to the failover destination for that queue. However, this does not seem to be working in that, even if no agents are logged in, the call goes into the queue. Is there a config option I'm missing (or did I misunderstand how the failover works?) 2. I have the wrap time set ideally for agents, but sometimes they want to pickup the next call in queue before the wrap time expires. Is there a way for agents to grab the next call? 1) joinempty=strict 2) Not at the preseent time, no. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Very high ping times from 7960 phones
- Original Message - From: Steve Glaus [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, September 22, 2006 4:25 PM Subject: [asterisk-users] Very high ping times from 7960 phones I've asked this here before and never really got a response, so I'll try again :) I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times 150ms. Every single one. I've scoured the settings on the 7960's and have looked and looked for why this might be the case. Cisco ata's (186) on the same network report ~ 10 ms. An xlite softphone reports ~ 5ms regardless of what computer it's installed on. Does anyone have any idea what might be causing this? I thought that it might just be a 'reporting' issue but there is definite latency there when I do an echo test. I'm running cisco sip firmware 8.2 on all the phones. All my Cisco phones show less than 75ms except for one (mine of course). I do have a switch in my cube that I use for extra ports and that's the only real difference. Do you have anything plugged into the extra network port on the phone? What's in between your phone and the asterisk server? _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Opinions on Aastra 480i CT?
On Mon, Sep 25, 2006 at 08:25:10AM -0600, Colin Anderson wrote: Looks good, great price: http://www.aastratelecom.com/ipphones/pro_243.asp Anybody using these? How's the cordless? Does it play nice with * ? Well, anyone who thinks that a 4-p,4-c modular jack *has* an RJ designation makes me exceedingly nervous... (RJ - Registered Jack: a modular or Amphenol connector *and wiring pattern* specified in FCC Part 68. Handset connectors are not mentioned in that regulation.) Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Opinions on Aastra 480i CT?
I don't have experience using the 480i CT, only using the 9112i, so you should take what I say with a grain of salt. I have been nothing but impressed with this phone. In terms of being friendly with *, they dedicate a section of their manual to asterisk configuration, which makes things go quite smoothly (not that the configuration is particularly difficult: its a totally standard SIP setup). As for the No hold, no one-touch voicemail, this isn't strictly true. It has programmable soft-keys, and though I haven't experimented that extensively with them, they can be configured to dial a line (i.e. your voicemail), or park a call (i.e. hold). There's also other cool features, like the ability to write custom menus for the phone, that get called over HTTP. All in all, my 9112i has been pretty good (I had a few lockups, but none since I upgraded the firmware), and I would say its definitely worth buying one to see if it will work for your needs. --Brian On Mon, Sep 25, 2006 at 10:48:00AM -0400, Richard wrote: It's excellent home phone. I wouldn't use it in a business environment. No hold, no one-touch voicemail. However, it works great! /R -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Monday, September 25, 2006 10:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] OT: Opinions on Aastra 480i CT? Looks good, great price: http://www.aastratelecom.com/ipphones/pro_243.asp Anybody using these? How's the cordless? Does it play nice with * ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk to cell phone network
On 9/25/06, yrving rivas [EMAIL PROTECTED] wrote: I would like to know if any of you have a cell phone like a pci card to install in one slot to my asterisk server?, I want to make a connection from my asterisk to the cellular network. Does anybody has a solution like this? Regards, Yrving I've seen 'boxes' advertised on the biz list which do what you want. Basically they take a GSM SIM card and act as an FXO port if memory serves me. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT: Opinions on Aastra 480i CT?
It's excellent home phone. I wouldn't use it in a business environment. No hold, no one-touch voicemail. However, it works great! aw crap, that's a biggie but I think I can work around it, teach the user to dial *98 for voicemail, *700 for park and hash to transfer, currently the users dial feature-9-8-1 for voicemail right now so they are used to doing things the hard way. But a dedicated hold and transfer button would've been nice. The users' big requirement is inbound /outbound / missed call logging, how is that? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Opinions on Aastra 480i CT?
Colin Anderson wrote: Looks good, great price: http://www.aastratelecom.com/ipphones/pro_243.asp Anybody using these? How's the cordless? Does it play nice with * ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Colin: We have a few of these. So far, I really like them, but we don't have tons of usage. They work fine with *. The wireless works well, but I don't have battery life stats yet. The soft key programming is very flexible via a central server. Just my quick initial thoughts. Mike Clark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Opinions on Aastra 480i CT?
On 9/25/06, Colin Anderson [EMAIL PROTECTED] wrote: It's excellent home phone. I wouldn't use it in a business environment. No hold, no one-touch voicemail. However, it works great! aw crap, that's a biggie but I think I can work around it, teach the user to dial *98 for voicemail, *700 for park and hash to transfer, currently the users dial feature-9-8-1 for voicemail right now so they are used to doing things the hard way. But a dedicated hold and transfer button would've been nice. The users' big requirement is inbound /outbound / missed call logging, how is that? The 480i does have a hold button, and also a transfer and Conference button (so does the 9112i in fact...). It does not have a one-touch voicemail pickup, which is accessed with something like Services-2-1-Select, but they are quite nice phones overall. I have been unable to get hold of the 480i CT variant (Which I believe adds DECT to the phone) to test it so I do not know how much it differs from the 480i I agree that it is worth getting one to try it out for your needs. Cheers, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT: Opinions on Aastra 480i CT?
Thanks for the feedback. More questions: 1. How's the range on the wireless? 2. Is there a soft key that can be programmed on the wireless handset? 3. Can I make a soft key basically do anything, any keystroke? 4. How's the call log detail? -Original Message- From: Mike Clark [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Opinions on Aastra 480i CT? Colin Anderson wrote: Looks good, great price: http://www.aastratelecom.com/ipphones/pro_243.asp Anybody using these? How's the cordless? Does it play nice with * ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Colin: We have a few of these. So far, I really like them, but we don't have tons of usage. They work fine with *. The wireless works well, but I don't have battery life stats yet. The soft key programming is very flexible via a central server. Just my quick initial thoughts. Mike Clark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk to cell phone network
yrving rivas wrote: I would like to know if any of you have a cell phone like a pci card to install in one slot to my asterisk server?, I want to make a connection from my asterisk to the cellular network. Does anybody has a solution like this? Regards, Yrving http://www.techtopia.com.au/product_info.php/cPath/36_75/products_id/1147 -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco 7960 Double Natted
On the 7960 with a SIP image, Press the Settings button and go to option 4 SIP Configuration. Scroll down to line 24 NAT Enabled and set it to yes. Then set 25 NAT Address to the external IP address. This will need to be manually changed every time the phone's router pulls a new DHCP lease. In your sip.conf, make sure that you have nat=yes and qualify=yes. I have had double-NATed 7960s work with this setup, but you are at the mercy of the routers involved in performing the NAT. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop Sent: Sunday, September 24, 2006 5:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco 7960 Double Natted Hi All Yes I know double Nat is a problem But I have a Cisco 7960 which is remote from the * PBX ad connected via the Internet. Each side has NAT (1) Sometimes it will work often it won't. And when it decides to work is random Always (2) The Register side works fine. SIP SHOW PEERS has the phone listed with the correct IP address and an average Qualify time (121 ms) Always (3) You can make calls outbound with the Cisco phone through the * PBX Problem (4) You can not receive any calls (when not working correctly) (a) The Phone rings but not voice goes through (b) Sometimes get a 481 Call Leg Does Not Exist (c) Sometimes get a -- is circuit-busy (5) On a reload of the * box you will 95 % sure loose the connection if it was working ? (6) SIP 5060 - 5063 and RTP 1 - 25000 is open and port forwarded on both sides (7) All calls are VoIP and terminate or originate via a VoIP Provider Anybody got any ideas, I have tried everything Thanks All Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Running Multiple Instances of Asterisk
I'd like to know if anyone has sucessfully managed to run multiple instances of Asterisk on the same system. - Did you run each instance as a separate user? - Did you have any install or config problems? - It looks like the G729 codec registration utility doesn't work when files aren't installed in standard places. Did you have this problem? - How many instances could be run on a single Asterisk box? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Opinions on Aastra 480i CT?
On Mon, 25 Sep 2006 08:25:10 -0600, Colin Anderson wrote Looks good, great price: http://www.aastratelecom.com/ipphones/pro_243.asp Anybody using these? How's the cordless? Does it play nice with * ? Very good phone. The range of the cordless unit is not the greatest but enough to be used in an office environment. The speaker phone is good and you have all the regular functions like transfer and hold. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PBX TDA620 AND TE110P
SEÑORES DIGIUMLA PRESENTE ES PARA CONFIRMAR QUE REQUERIMIENTOS DE HARDWARE Y/O SOFTWARE SON NECESARIOS PARA CONECTAR UNA CENTRAL TELEFONICA PANASONIC TDA620 QUE TIENE INSTALADA UNA TARJETA (E1) SE5E18, NO SE SI SEA ESA LA REFERENCIA, CON UNA TARJETA DIGUIM TE110P. VI EN ALGUNOS FOROS QUE TIENE QUE TENER UN TIPO DE SEÑALIZACIÓN LA CENTRAL PARA QUE FUNCIONE CON LA TARJETA. MUCHA GRACIASATENTAMENTEDIEGO FERNANDO GÜIZA ARCE THE PRESENT IS FOR CONFIRMING THAT REQUIREMENTS OF HARDWARE AND/OR SOFTWARE ARE NECESSARY TO CONNECT A PBX PANASONIC TDA620 THAT IT HAS INSTALLED A CARD (E1) SE5E18, NOT IF SHE IS THAT THE REFERENCE, WITH A CARD DIGUIM TE110P. I SAW IN SOME FORUMS THAT IT HAS TO HAVE A TYPE OF SIGNALING THE PBX SO THAT IT WORKS WITH THE CARD. MANY THANKS ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi Servers
Lets say you have a cluster of, say, 10 Asterisk servers. After doing a local lookup to see if a number is available locally, in order to find out if the number is available on one of the other 9 servers, this peer has to query all 9 remaining peers. Is that true? Is there a way to have 'registration servers' that accept registrations from phones, and which somehow notify 'DUNDI servers' (two for redundancy) that the registration servers query? To terminate a call, a peer would only have to query the DUNDi servers, not every other peer. After looking at the config files, I can't imagine how this could work, or if it's even possible with DUNDi. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Got SIP response 415 Unacceptable Content-Type back from 192.168.1.209
Hi Folks, Has anyone seen these errors repeatedly in the CLI? Incoming call: Got SIP response 415 Unacceptable Content-Type back from 192.168.1.209 We're using GXP-2000s. TIA, Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Opinions on Aastra 480i CT?
At 07:25 AM 9/25/2006, you wrote: Anybody using these? How's the cordless? Does it play nice with * ? I have 3 of them here, we're very happy with them. The cordless is fine, about the range of my old Panasonic cordless. Sound quality is good and the speaker phone seems good. Plays fine with Asterisk. Even though it says many handsets per base, in reality you should probably limit it to 1 per base as the system can only handle 2 active voice streams at a time and annoyingly enough, only one of them can be G.729, something about the phone does not have enough processor power to encode 2 G.729 streams at a time. Easy to configure from a TFTP server. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 autoconf and /etc/asterisk directory
I just downloaded asterisk 1.4beta2, and did a: ./configure --prefix=/home/pbx/1.4 [11:[EMAIL PROTECTED](pbx1):asterisk-1.4.0-beta2]# ls /home/pbx/1.4 bin include lib sbin share What happened to etc? If I do a 'make samples', the default conf files get thrown in /etc/asterisk. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk to cell phone network
On 09:35, Mon 25 Sep 06, yrving rivas wrote: I would like to know if any of you have a cell phone like a pci card to install in one slot to my asterisk server?, I want to make a connection from my asterisk to the cellular network. Does anybody has a solution like this? 1/2/4 simslot pci card: http://www.junghanns.net/en/GSM-PCI_produkt.html If they are as stable as the quad/octo BRI cards they have it's a real winner. We have a couple of voiceblue GSM/SIP gateways in production and they work great as well. http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html Hope this helps. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi Servers
Hi Doug,On 9/25/06, Douglas Garstang [EMAIL PROTECTED] wrote: Lets say you have a cluster of, say, 10 Asterisk servers. After doing a local lookup to see if a number is available locally, in order to find out if the number is available on one of the other 9 servers, this peer has to query all 9 remaining peers. Is that true?Yes, or it could send one query to a server which in turned queried the other 9. Either way though, all 9 get queried unless the answer was cached. Caching is tricky with registrations as you don't want to cache a registration which hasn't been renewed. Is there a way to have 'registration servers' that accept registrations from phones, and which somehow notify 'DUNDI servers' (two for redundancy) that the registration servers query? To terminate a call, a peer would only have to query the DUNDi servers, not every other peer. After looking at the config files, I can't imagine how this could work, or if it's even possible with DUNDi. Yes, it is possible to push peer information as well as pull it. You could also, as you say, limit the number of registration servers (i.e. servers doing both the registration and DUNDi) and then only query to them. I'm sure the hybrid model you suggest would work as well although it'd need testing to see whether you got more performance out of splitting the DUNDI and registration roles or just adding more dual-purpose machines. SimonDoug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] can someone recommened a reliable, cheap t38 origination/termination provider
one that also offers support for it. thanks.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT: Opinions on Aastra 480i CT?
At 07:48 AM 9/25/2006, you wrote: It's excellent home phone. I wouldn't use it in a business environment. No hold, no one-touch voicemail. However, it works great! No Hold? Mine has a hold button and programming one touch voice mail would be no problem at all. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT: Opinions on Aastra 480i CT?
At 08:31 AM 9/25/2006, you wrote: aw crap, that's a biggie but I think I can work around it, teach the user to dial *98 for voicemail, *700 for park and hash to transfer, currently the It has a dedicated hold button and you can easily program dedicated Park and voice mail buttons. I've not done the voice mail because I'd rather use those buttons for dialing people and voice mail is only 4 button presses away. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Multiple Instances of Asterisk
Asterisk does not support this, as it already has features for multi-client configuration within a single Asterisk installation/process. Douglas Garstang wrote: I'd like to know if anyone has sucessfully managed to run multiple instances of Asterisk on the same system. - Did you run each instance as a separate user? - Did you have any install or config problems? - It looks like the G729 codec registration utility doesn't work when files aren't installed in standard places. Did you have this problem? - How many instances could be run on a single Asterisk box? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got SIP response 415 Unacceptable Content-Type back from 192.168.1.209
Bidirectional SIP trace usually helps in these situations.On 9/25/06, Mr. Jones [EMAIL PROTECTED] wrote: Hi Folks,Has anyone seen these errors repeatedly in the CLI?Incoming call: Got SIP response 415 Unacceptable Content-Type back from 192.168.1.209We're using GXP-2000s.TIA,Brian___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Anthony D Cennami ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Double Natted
Thanks for the input Yes I have nat=yes and qualify=yes I know in the SIPMacAddress.cnf file I have # NAT/Firewall Traversal nat_enable: 1 nat_received_processing: 1 nat_address: phone's public IP Address Do I still need to set it again in SIP Configuration ? Thanks all Barry Hughes, Sam wrote: On the 7960 with a SIP image, Press the Settings button and go to option 4 SIP Configuration. Scroll down to line 24 NAT Enabled and set it to yes. Then set 25 NAT Address to the external IP address. This will need to be manually changed every time the phone's router pulls a new DHCP lease. In your sip.conf, make sure that you have nat=yes and qualify=yes. I have had double-NATed 7960s work with this setup, but you are at the mercy of the routers involved in performing the NAT. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT: Opinions on Aastra 480i CT?
At 09:23 AM 9/25/2006, you wrote: 2. Is there a soft key that can be programmed on the wireless handset? Not really, there's a function key menu and you can set that up any way you want, but what you can assign to the functions is very limited. The cordless is very handy, but the functionality is limited and the interface is a bit awkward. It's usable and learnable, but it has to be learned, it's not obvious like the base. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Very high ping times from 7960 phones
All my Cisco phones show less than 75ms except for one (mine of course). I do have a switch in my cube that I use for extra ports and that's the only real difference. Do you have anything plugged into the extra network port on the phone? Yes, I have workstations plugged into the extra ports on some of the phones - Doesn't seem to make a difference What's in between your phone and the asterisk server? My asterisk server has 2 NICs . One with a public IP and one with an internal LAN IP. All the phones configure to the LAN IP so there's basically nothing between them. A 3com switch and that's it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Running Multiple Instances of Asterisk
-Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Asterisk does not support this, as it already has features for multi-client configuration within a single Asterisk installation/process. Douglas Garstang wrote: I'd like to know if anyone has sucessfully managed to run multiple instances of Asterisk on the same system. - Did you run each instance as a separate user? - Did you have any install or config problems? - It looks like the G729 codec registration utility doesn't work when files aren't installed in standard places. Did you have this problem? - How many instances could be run on a single Asterisk box? What do you mean 'does not support'? How easy do you think the management of the configuration files is going to be if your trying to host several dozen companies on the one Asterisk instance? Sure, you can split things into contexts, but just try and imagine how complex the management is going to become when several companies comprise the same file space. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Very high ping times from 7960 phones
My asterisk server has 2 NICs . One with a public IP and one with an internal LAN IP. All the phones configure to the LAN IP so there's basically nothing between them. A 3com switch and that's it. basically nothing is wrong. I have a 3com switch in front of the one phone that reports the large time. Now I'm thinking it has something specifically to do with 3com switches and these phones. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Multiple Instances of Asterisk
Doug, Why do you want to do this to begin with? I think the best solution is to use the realtime stuff, and build your own management tools, which would allow you to do this (you could drastically cut the complexity with the right tools). Even if you could run them together, how would you put everything on the appropriate ports? How would you deal with multiple instances accessing hardware? I'm not convinced that maintaining the config files, binaries and other components of multiple asterisk's is easier than just building better tools to configure one. You could also try User-Mode-Linux or something like that. --Brian On Mon, Sep 25, 2006 at 12:28:30PM -0600, Douglas Garstang wrote: -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Asterisk does not support this, as it already has features for multi-client configuration within a single Asterisk installation/process. Douglas Garstang wrote: I'd like to know if anyone has sucessfully managed to run multiple instances of Asterisk on the same system. - Did you run each instance as a separate user? - Did you have any install or config problems? - It looks like the G729 codec registration utility doesn't work when files aren't installed in standard places. Did you have this problem? - How many instances could be run on a single Asterisk box? What do you mean 'does not support'? How easy do you think the management of the configuration files is going to be if your trying to host several dozen companies on the one Asterisk instance? Sure, you can split things into contexts, but just try and imagine how complex the management is going to become when several companies comprise the same file space. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] channel.c: Nobody there, continuing...
I'm seeing channel.c: Nobody there, continuing... in the asterisk full.log. This error is repeated 20+ times per second when it occurs. I thought this problem was specific to one PBX that performs call recording on all the call queues, but after disabling all call recording, the error persists, although less often. The system was hanging badly requiring daily reboots, however since disabling call recording, the system has stabilized. I've since noticed this behavior on another less loaded system. The asterisk versions are 1.2.11 and 1.2.9.1 respectively, and both are running Trixbox. Other systems running older versions of Asterisk, some with AMP/FreePBX don't seem to exhibit this problem. At this point I'm not sure if this is specific to Trixbox, or a problem with later versions of Asterisk. Google turns up very little regarding this error, and the few bugs listed at bugs.digium.com appear to be unrelated. Anyone seen this issue and know what is causing it? Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Running Multiple Instances of Asterisk
-Original Message- From: Brian Rogan [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Doug, Why do you want to do this to begin with? I think the best solution is Because we are trying to build a hosted IPT solution, not an enterprise solution. to use the realtime stuff, and build your own management tools, which would allow you to do this (you could drastically cut the complexity with the right tools). Even if you could run them together, how would you put everything on the appropriate ports? How would you deal with multiple instances accessing hardware? Realtime is resource intensive, requiring many queries to perform simple lookups. We can easily create multiple virtual IP address, and since each virtual IP address can bind to port 5060, each phone can register with domain.com:5060 without a problem. We don't need multiple instances to access hardware as this is a SIP only solution. Our PSTN access is via external Audiocodes gateways, not via Digium T1 cards. The dial plan was not able to handle the complexity we needed (for example the MySQL() application command could not do nested queries), and so right now, we have a 2000 line python script and several very complex MySQL stored procedures in order to fulfull our requirements. I'm not convinced that maintaining the config files, binaries and other components of multiple asterisk's is easier than just building better tools to configure one. I am. I look at our configuration which is currently for one customer, and there's already several dozen contexts in order to cover a lot of complexity. Multiply that by a couple of hundred, and I won't want to be administering it! You could also try User-Mode-Linux or something like that. I was going to give v-servers a try. There's a guide at: http://www.telephreak.org/papers/vpa/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Multiple Instances of Asterisk
On Mon, Sep 25, 2006 at 12:28:30PM -0600, Douglas Garstang wrote: How easy do you think the management of the configuration files is going to be if your trying to host several dozen companies on the one Asterisk instance? Sure, you can split things into contexts, but just try and imagine how complex the management is going to become when several companies comprise the same file space. Assuming that all Asterisk config files support #include, and I believe they do, then it shouldn't be all *that* hard. Now, if you are trying to give each company direct control of their own configs, then yes, avoiding interactions will be harder. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Multiple Instances of Asterisk
On Mon, Sep 25, 2006 at 12:52:43PM -0600, Douglas Garstang wrote: -Original Message- From: Brian Rogan [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk to use the realtime stuff, and build your own management tools, which would allow you to do this (you could drastically cut the complexity with the right tools). Even if you could run them together, how would you put everything on the appropriate ports? How would you deal with multiple instances accessing hardware? Realtime is resource intensive, requiring many queries to perform simple lookups. Check out the static config option, which just loads everything to memory at startup (just like the config file method). http://www.voip-info.org/wiki-Asterisk+RealTime (Extconfig-Static Configs section) I'm not convinced that maintaining the config files, binaries and other components of multiple asterisk's is easier than just building better tools to configure one. I am. I look at our configuration which is currently for one customer, and there's already several dozen contexts in order to cover a lot of complexity. Multiply that by a couple of hundred, and I won't want to be administering it! That's one way to look at it. The flip side, is you just need to maintain the same complexity just a bunch of times. Either way, I wouldn't want to be administering it ;-), but with good configuration utilities, you shouldn't have to deal with this complexity at all: you should have utilities that maintain configuration for you, and if you're going to do this, realtime is by far the best way to go. I don't pretend to know what you want in your application, but It seems clear that YOU NEED GOOD TOOLS to manage it. If you build these though, I still don't see what you could do with multiple instances that you can't do with one. If you abstract away the dial plan with your tools, what does it matter that the underlying plan is a complicated mess. In any case, take that for what its worth. --Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] trixbox t38 pass through
I know asterisk 1.4 has t.38 pass through, but I don't think trixbox does. i run trixbox. looks like for now i will have to setup my fax machine to connect directly to my t38 provider. anyone know when trixbox may have this update?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Multiple Instances of Asterisk
Doug, I actually see this as a pretty logical way to solve the problem. Please keep us posted if you have any luck sorting out running multiple instances, or mail me off-list if no one else is interested. Thanks, On 9/25/06 1:52 PM, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Brian Rogan [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Doug, Why do you want to do this to begin with? I think the best solution is Because we are trying to build a hosted IPT solution, not an enterprise solution. to use the realtime stuff, and build your own management tools, which would allow you to do this (you could drastically cut the complexity with the right tools). Even if you could run them together, how would you put everything on the appropriate ports? How would you deal with multiple instances accessing hardware? Realtime is resource intensive, requiring many queries to perform simple lookups. We can easily create multiple virtual IP address, and since each virtual IP address can bind to port 5060, each phone can register with domain.com:5060 without a problem. We don't need multiple instances to access hardware as this is a SIP only solution. Our PSTN access is via external Audiocodes gateways, not via Digium T1 cards. The dial plan was not able to handle the complexity we needed (for example the MySQL() application command could not do nested queries), and so right now, we have a 2000 line python script and several very complex MySQL stored procedures in order to fulfull our requirements. I'm not convinced that maintaining the config files, binaries and other components of multiple asterisk's is easier than just building better tools to configure one. I am. I look at our configuration which is currently for one customer, and there's already several dozen contexts in order to cover a lot of complexity. Multiply that by a couple of hundred, and I won't want to be administering it! You could also try User-Mode-Linux or something like that. I was going to give v-servers a try. There's a guide at: http://www.telephreak.org/papers/vpa/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Texter ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Very high ping times from 7960 phones
Mailing List wrote: My asterisk server has 2 NICs . One with a public IP and one with an internal LAN IP. All the phones configure to the LAN IP so there's basically nothing between them. A 3com switch and that's it. basically nothing is wrong. I have a 3com switch in front of the one phone that reports the large time. Now I'm thinking it has something specifically to do with 3com switches and these phones. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users That's a possibility but I took one of these phones home and pointed it at my own asterisk system and it reports ~ the same. What make and model 3com switch are you using? what does the phone in your office report? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Multiple Instances of Asterisk
Best of luck getting multiple instances of Asterisk to play nice when accessing Zap channels. James Texter wrote: Doug, I actually see this as a pretty logical way to solve the problem. Please keep us posted if you have any luck sorting out running multiple instances, or mail me off-list if no one else is interested. Thanks, On 9/25/06 1:52 PM, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Brian Rogan [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Doug, Why do you want to do this to begin with? I think the best solution is Because we are trying to build a hosted IPT solution, not an enterprise solution. to use the realtime stuff, and build your own management tools, which would allow you to do this (you could drastically cut the complexity with the right tools). Even if you could run them together, how would you put everything on the appropriate ports? How would you deal with multiple instances accessing hardware? Realtime is resource intensive, requiring many queries to perform simple lookups. We can easily create multiple virtual IP address, and since each virtual IP address can bind to port 5060, each phone can register with domain.com:5060 without a problem. We don't need multiple instances to access hardware as this is a SIP only solution. Our PSTN access is via external Audiocodes gateways, not via Digium T1 cards. The dial plan was not able to handle the complexity we needed (for example the MySQL() application command could not do nested queries), and so right now, we have a 2000 line python script and several very complex MySQL stored procedures in order to fulfull our requirements. I'm not convinced that maintaining the config files, binaries and other components of multiple asterisk's is easier than just building better tools to configure one. I am. I look at our configuration which is currently for one customer, and there's already several dozen contexts in order to cover a lot of complexity. Multiply that by a couple of hundred, and I won't want to be administering it! You could also try User-Mode-Linux or something like that. I was going to give v-servers a try. There's a guide at: http://www.telephreak.org/papers/vpa/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with zaptel 1.4b2 and X101P Wildcard
On 9/25/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] zaptel-1.4.0-beta1]# ztcfg -vvv Notice: Configuration file is /etc/zaptel.conf line 235: Unable to read Zaptel version information. Zaptel Version: $êþP¦0 Echo Canceller: Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25)What exactly is channel 1? Maybe you got the wrong number? cat /proc/zaptel/* [EMAIL PROTECTED] zaptel-1.4.0-beta1]# service zaptel startLoading zaptel framework: [ OK ]Waiting for zap to come online...OKLoading zaptel hardware modules:Running ztcfg: ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) [FAILED][EMAIL PROTECTED] zaptel-1.4.0-beta1]# cat /proc/zaptel/*Span 1: WCFXO/0 Wildcard X101P Board 1 1 WCFXO/0/0 The same configuration works perfect with zaptel 1.2.1-- Morten Isaksenhttp://www.misak.dk/blog/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Very high ping times from 7960 phones
On 14:31, Mon 25 Sep 06, Mailing List wrote: My asterisk server has 2 NICs . One with a public IP and one with an internal LAN IP. All the phones configure to the LAN IP so there's basically nothing between them. A 3com switch and that's it. basically nothing is wrong. I have a 3com switch in front of the one phone that reports the large time. Now I'm thinking it has something specifically to do with 3com switches and these phones. I can confirm this. We have one location where the phones are connected to HotBrick switches and the times are low there (5-20 ms range) All the other locations are running 3com switches and there the times are in the 150+ range -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Running Multiple Instances of Asterisk
you didn't listen. SIP only. Anyone can understand that multiple instances on the same machine can't touch the same hardware. I can see how this would be very easy - dedicate an IP to an instance, and it'll play nice. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Monday, September 25, 2006 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Best of luck getting multiple instances of Asterisk to play nice when accessing Zap channels. James Texter wrote: Doug, I actually see this as a pretty logical way to solve the problem. Please keep us posted if you have any luck sorting out running multiple instances, or mail me off-list if no one else is interested. Thanks, On 9/25/06 1:52 PM, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Brian Rogan [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Doug, Why do you want to do this to begin with? I think the best solution is Because we are trying to build a hosted IPT solution, not an enterprise solution. to use the realtime stuff, and build your own management tools, which would allow you to do this (you could drastically cut the complexity with the right tools). Even if you could run them together, how would you put everything on the appropriate ports? How would you deal with multiple instances accessing hardware? Realtime is resource intensive, requiring many queries to perform simple lookups. We can easily create multiple virtual IP address, and since each virtual IP address can bind to port 5060, each phone can register with domain.com:5060 without a problem. We don't need multiple instances to access hardware as this is a SIP only solution. Our PSTN access is via external Audiocodes gateways, not via Digium T1 cards. The dial plan was not able to handle the complexity we needed (for example the MySQL() application command could not do nested queries), and so right now, we have a 2000 line python script and several very complex MySQL stored procedures in order to fulfull our requirements. I'm not convinced that maintaining the config files, binaries and other components of multiple asterisk's is easier than just building better tools to configure one. I am. I look at our configuration which is currently for one customer, and there's already several dozen contexts in order to cover a lot of complexity. Multiply that by a couple of hundred, and I won't want to be administering it! You could also try User-Mode-Linux or something like that. I was going to give v-servers a try. There's a guide at: http://www.telephreak.org/papers/vpa/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400P vs Sangoma A200
Greetings List, I'm putting together a plan for a new Asterisk system and I'm trying to decided on an interface card to use. I was originally planning on using a Sangoma A200 but now I'm considering a Digium TDM2400P. The server is large enough to accommodate the full sized TDM and I'll be using 8 FXO channels so molex power connectors aren't an issue. The connector will be slightly more to deal with but not a biggie. Either card I get will have the on-board echo canceler. For the extra $150 for the TDM, not having to mess with two sets of drivers is pretty appealing. Anyone have experience with both cards to give advice one way or the other? (And in case anyone suggests I just go with a PRI, I can't. I'm stuck with POTS lines for now). Thanks -Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Running Multiple Instances of Asterisk
We aren't accessing ZAP channels. No Digium hardware is installed! -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Best of luck getting multiple instances of Asterisk to play nice when accessing Zap channels. James Texter wrote: Doug, I actually see this as a pretty logical way to solve the problem. Please keep us posted if you have any luck sorting out running multiple instances, or mail me off-list if no one else is interested. Thanks, On 9/25/06 1:52 PM, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Brian Rogan [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Doug, Why do you want to do this to begin with? I think the best solution is Because we are trying to build a hosted IPT solution, not an enterprise solution. to use the realtime stuff, and build your own management tools, which would allow you to do this (you could drastically cut the complexity with the right tools). Even if you could run them together, how would you put everything on the appropriate ports? How would you deal with multiple instances accessing hardware? Realtime is resource intensive, requiring many queries to perform simple lookups. We can easily create multiple virtual IP address, and since each virtual IP address can bind to port 5060, each phone can register with domain.com:5060 without a problem. We don't need multiple instances to access hardware as this is a SIP only solution. Our PSTN access is via external Audiocodes gateways, not via Digium T1 cards. The dial plan was not able to handle the complexity we needed (for example the MySQL() application command could not do nested queries), and so right now, we have a 2000 line python script and several very complex MySQL stored procedures in order to fulfull our requirements. I'm not convinced that maintaining the config files, binaries and other components of multiple asterisk's is easier than just building better tools to configure one. I am. I look at our configuration which is currently for one customer, and there's already several dozen contexts in order to cover a lot of complexity. Multiply that by a couple of hundred, and I won't want to be administering it! You could also try User-Mode-Linux or something like that. I was going to give v-servers a try. There's a guide at: http://www.telephreak.org/papers/vpa/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Multiple Instances of Asterisk
But if I segment my zap channels, that shouldn't be an issue, correct? I.e. Instance 1 = Port 1, Instance 2 = Port 2, etc. Of course, you are also assuming there is Zap channels, as I believe he is using a gateway, which takes that out of the equation. On 9/25/06 2:23 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Best of luck getting multiple instances of Asterisk to play nice when accessing Zap channels. James Texter wrote: Doug, I actually see this as a pretty logical way to solve the problem. Please keep us posted if you have any luck sorting out running multiple instances, or mail me off-list if no one else is interested. Thanks, On 9/25/06 1:52 PM, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Brian Rogan [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Doug, Why do you want to do this to begin with? I think the best solution is Because we are trying to build a hosted IPT solution, not an enterprise solution. to use the realtime stuff, and build your own management tools, which would allow you to do this (you could drastically cut the complexity with the right tools). Even if you could run them together, how would you put everything on the appropriate ports? How would you deal with multiple instances accessing hardware? Realtime is resource intensive, requiring many queries to perform simple lookups. We can easily create multiple virtual IP address, and since each virtual IP address can bind to port 5060, each phone can register with domain.com:5060 without a problem. We don't need multiple instances to access hardware as this is a SIP only solution. Our PSTN access is via external Audiocodes gateways, not via Digium T1 cards. The dial plan was not able to handle the complexity we needed (for example the MySQL() application command could not do nested queries), and so right now, we have a 2000 line python script and several very complex MySQL stored procedures in order to fulfull our requirements. I'm not convinced that maintaining the config files, binaries and other components of multiple asterisk's is easier than just building better tools to configure one. I am. I look at our configuration which is currently for one customer, and there's already several dozen contexts in order to cover a lot of complexity. Multiply that by a couple of hundred, and I won't want to be administering it! You could also try User-Mode-Linux or something like that. I was going to give v-servers a try. There's a guide at: http://www.telephreak.org/papers/vpa/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Texter ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to stream audio to external app for speech recognition and recognize dtmf in parallel ?
Hi, we're writting interface module for our speech recognition system. We would like to export stream of audio samples to external app, but to preserve dtmf recognition and dialplan progress. I wonder if recording application would be a good start for that (recording application obviously streams audio and makes recording out of it in parallel) We're also interested in best way to report speech recognition results back Best way would be to be able to call extension in dialplan, for instance : 1, DTMF 1 2, DTMF 2 support, spoken word support sales, spoken word sales -- but also putting results in variable would be probably fine Any advice how to develope such scenario ? What is the best module code to start with ? Any similar solutions ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
