Re: [asterisk-users] Filtering duplicate RTP packets
Replying to myself Its fixed now Checking timestamps is optional according to RFC so asterisk is not doing it. Anyway, I made a patch and tested it and its working. Thanks. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Boris Bakchiev Sent: Wednesday, 5 December 2007 01:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Filtering duplicate RTP packets Hi, I have a SIP provider who sometimes sends duplicate RTP packets to me. Sent RTP packet to 10.55.20.201:17440 (type 08, seq 008536, ts 4846560, len 000160) Got RTP packet from10.55.20.201:17440 (type 08, seq 051978, ts 3647104992, len 000160) Got RTP packet from10.55.20.201:17440 (type 08, seq 051979, ts 3647104992, len 000160) Sent RTP packet to 10.55.20.201:17440 (type 08, seq 008537, ts 4846720, len 000160) Got RTP packet from10.55.20.201:17440 (type 08, seq 051980, ts 3647105152, len 000160) Got RTP packet from10.55.20.201:17440 (type 08, seq 051981, ts 3647105152, len 000160) Unfortunately they're unable to filter it because this is the way they receive it themselves. Can this be fixed within asterisk with a custom patch? Kind regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Filtering duplicate RTP packets
Hi, I have a SIP provider who sometimes sends duplicate RTP packets to me. Sent RTP packet to 10.55.20.201:17440 (type 08, seq 008536, ts 4846560, len 000160) Got RTP packet from10.55.20.201:17440 (type 08, seq 051978, ts 3647104992, len 000160) Got RTP packet from10.55.20.201:17440 (type 08, seq 051979, ts 3647104992, len 000160) Sent RTP packet to 10.55.20.201:17440 (type 08, seq 008537, ts 4846720, len 000160) Got RTP packet from10.55.20.201:17440 (type 08, seq 051980, ts 3647105152, len 000160) Got RTP packet from10.55.20.201:17440 (type 08, seq 051981, ts 3647105152, len 000160) Unfortunately they're unable to filter it because this is the way they receive it themselves. Can this be fixed within asterisk with a custom patch? Kind regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: The High Performance Echo Canceller (HPEC)
Hi Tony, Its a dual core system and combined CPU usage was 2%. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Thursday, 22 February 2007 12:07 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: The High Performance Echo Canceller (HPEC) In article <[EMAIL PROTECTED]>, Boris Bakchiev <[EMAIL PROTECTED]> wrote: > Hi, > > Has anyone noticed degraded voice quality with HPEC? > I have a client running TE4XX card who configured HPEC for couple of > channels with echocancel=1024. > > Whenever HPEC is used you get a background static in voice. > When HPEC is not used everything is crystal clear. > > What could cause this static? Try using a utility like "top" to see what the CPU loading is with HPEC. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The High Performance Echo Canceller (HPEC)
Hi, Has anyone noticed degraded voice quality with HPEC? I have a client running TE4XX card who configured HPEC for couple of channels with echocancel=1024. Whenever HPEC is used you get a background static in voice. When HPEC is not used everything is crystal clear. What could cause this static? regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Contacts for Chan_gsm_bt maintainer?
Anyone knows how to contact maintainers of Chan_gsm_bt? They http://changsmbt.free.fr/ site has no contact details. I believe I found the issue why it does not initiate SCO links properly.. It looks to be a timing issue. It sends additional AT commands without waiting for the responses for previous commands. The specification is HFP 1 & 1.5 shows that sequence of commands need to be sent when responses are received for previous commands. While Nokia's are more tolerant (and clearly respond much quicker) LG's for example completely screw up and do not setup SCO link if you send commands at will without processing responses.. What would it take to rewrite the part of the init code? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE420P/TE415P?
Can the TE406P card's VPM module be swapped for the new revision with Octasic chipset? > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Kevin P. Fleming > Sent: Sunday, June 25, 2006 8:08 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] TE420P/TE415P? > > - C F <[EMAIL PROTECTED]> wrote: > > I like the TC400P card, how many T1s will that take? or is it just a > > Daughter card on the TE4xx ? How many channels can it transcode? > > Neither. It's a separate device, entirely unrelated to any TDM cards > (which means it can be used for any type of channel, not just TDM). > > The final specs for the number of channels are not yet determined, but we > expect to do at least 100 channels of G.729 and/or G.723.1 per board. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAX + Digium + SpanDSP
Hi, We do J We use iaxmodem+hylafax combo on TE406P card. Around 4K of faxes were received without any problems (some faxes are over 80 pages long!) It is working really well! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pibix Sent: Friday, 16 June 2006 13:59 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] FAX + Digium + SpanDSP Hi, Anyone using SpanDSP with Digium TDM o TE cards to receive and email Faxes? Thanks, Javier Ergas R. Director General de Tecnología Pibix Telefonía IP http://www.pibix.cl ___ --Bandwidth 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] Will 200KB/s drive access be OK for voicemailstorage?
Its slow :) It will give you some delays but it will not be noticeable (most voice files are 5-100kb, so it should be ok... But writing to them.. Not sure.. It should be ok as well I'm guessing as kernel will provide some caching (since you have G and not GS it has less ram, so maybe chaching is not an option) Best would be to get WL-500G. It has a USB port so you can plug a USB memory stick into it. It will be faster and will give you more storage cheaper then SD. > I'm using a Linksys WRT54G router and it works great but it only has > 4MB of storage. One of my only options is to modify the router to > accept SD cards (eg. 512MB) but the access time is only around 200 > KB/s. > > Will this be fast enough to store voicemails? ...or do I need > another (faster) storage 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] Hard drive write cache
> The cold hard truth is that if Asterisk cannot achieve 99.999% uptime > without becoming much more expensive that a traditional PBX then it is not > a > viable alternative. Even elcheapo Key systems are rated for five nines. > That is what the telco world requires unless your just using Asterisk in > your basement as a hobby or as a one man company. Well, you can pretty much guarantee 100% software uptime with asterisk. The main causes of crashes of the working system are users. If it works... don't touch it, do not logon to it... forget about it. Create a minimalistic root system with busybox, have everything on CF on IDE adapter, user UPS with shutdown to protect the CF (as they're prone to failures on power loss) and you have yourself a VERY stable system. You can use JFFS2 on block device to reduce the wear on CF but you will not need it if you're not writing anything on CF (or have 2 CF's and md them together.) I have never seen PBX with guarantee of 99.999%. None of the manufacturers will commit to that unless it is a highly redundant system, but by then it's not elcheapo. About fanless PC's.. A stock standard intel fan would lust longer then you think unless its located in dusty and damp place. I have a p3 that's been running for 5 years non-stop and it was still going strong.. Half the capacitors started to leak on the motherboard but the fan was still spinning. :) Now that's reliability! > Redundant Servers is moving into the realm of non-competitive with > Traditional PBX IMHO. > More or less true. Any 100-200 extension highly redundant PBX system will costs you more or less the same money. ___ --Bandwidth 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] Hard drive write cache
These days you don't have to worry much about your write cache unless you're running application where once single byte changed will affect whole file. Look at it this way, the only corruption will occur is whatever the files were open by asterisk at the time of the crash. And only up to the point where the file was last open. As far as I know asterisk does not keep cdr or log files open so you would loose only the data that was written at the time of the power failure. Any journaling file system (ext3, resierfs, xfs, etc) will easily handle any power failure event. Your files will not be corrupt but could miss some of the data. At the most you will loose 10-50 cdr entries written to you log files. If you post CDR to a remote SQL database then you asterisk install and linux is more or less static and will not be affected by the power failure. What you need to do is minimise the writes to hard disk's: 1 - Send syslog to remote server and do not do ANY syslogs Or keep the circular buffer in memory if you have plenty of it. 2 - Send CDR's to SQL server (or log to ramdisk and send to remote server every few minutes via SSH) 3 - Do not record any calls (or do that somewhere else) 4 - Stop any services that write/read data on regular intervals. If you have no writes you have nothing to worry about during power failure and journaling file system will take care of the rest. Keep your partition size really small so that fsck will not take much time. You have to be realistic, you cannot achieve 99.999% uptime. That's 5 minutes per year downtime. You will have more or less 100% until your first hardware failure. Even if you have all the hardware components pre-purchased it will still take you 2-12 hours to detect, diagnose and fix the fault if you lucky. So your 5 minuets If the business is demanding 99.999% then it should be prepared to invest into the hardware. I would recommend a cluster or even better a fault tolerant server. Those are expensive but you can pretty much rule out the hardware failure and swap all of the failed components while the system is running (cpu, memory, hdd, etc). Look at Stratus or NEC FT servers if you need hardware redundancy. They're expensive but will give you the hardware reliability you need. Or get a traditional PABX :) > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of shadowym > Sent: Tuesday, 13 June 2006 10:34 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Hard drive write cache > > > I am looking at ways to harden my asterisk install to prevent computer > related issues from happening. I am concerned about about disk write > cache. > That seems to be a major source of hard drive corruption on power failure. > Hard Drive corruption is simply unacceptable for the 99.999% uptime > requirements of my Asterisk install that needs to be as reliable as a > proprietary PBX. > > Of course I will be using redundant power supplies, raid 1 and use a UPS. > None of those things mean much if the power cords accidentally get pulled > from the back of the server. Unlikely as it may be I have to consider ALL > possibilities. > > So is disabling the write cache a good way to reduce the risk of hard > drive > corruption for an Asterisk server? I am not too concerned about the > reduced > performance/lifetime of hardrives with write cache disabled since Asterisk > is not a very write intensive environment. Even with lot's of voicemail > going on. > > Any other recommendations/links for increasing the reliability of Asterisk > servers? > ___ > --Bandwidth 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] PABX Setup
Samsung PABX? Its TEPRI probably configured in overlap mode so you need to configure asterisk span that is connected to PABX to overlap mode as well. When user selects the outside line in overlap mode PABX connects to asterisk and then sends the digits to it as the user presses the key's. If overlap mode is not configured in asterisk switch is not started by asterisk and it just thinks that empty dial string was sent to it. Just use: overlapdial=yes in your zapata.conf Make sure you have exten => s,1,Busy() exten => s,2,Hangup in your 'samsungincoming' context so that users get a busy signal when they didn't enter any digits in allotted time otherwise you'll get a hanging channel in Samsung. We use that setup with OfficeServ 500 and it works really well. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sahil Gupta Sent: Tuesday, 6 June 2006 21:40 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] PABX Setup Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil Gupta VoiceValley ___ --Bandwidth 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] PCI-X PRI hardware
HI, Does anyone know if there is a PCI-X 4 port PRI cards available on the market? If so, have anyone used it and how reliable they were? Any help is appreciated... ___ --Bandwidth 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: call monitoring and indications / beeps
HI Ben,
Make following context in your extensions.conf
[notifycallrec]
exten => tone,1,Answer
exten => tone,2,Answer
exten => tone,3,Playtones(!950/50,0)
exten => tone,4,Wait(10)
exten => tone,5,Goto(3)
exten => h,1,StopPlaytones
Then you can call it with:
exten => _X.,1,Dial(Zap/r0/${EXTEN},30,G(notifycallrec^tone^1))
Obviously use the right technology in your dial string but make sure you
keep the G option
You can play around with Wait and Playtones if you wish.
This will make both callee and caller phones beep with tiny beep.
This works fine for me
Regards
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben
Dinnerville
Sent: Monday, 22 May 2006 16:25
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: call monitoring and indications / beeps
Nudge?
Ben Dinnerville wrote:
> Hi All,
>
> Is it possible to configure asterisk to play a beep at a regular
> interval when a conversation is being recorded / monitored?
>
> There are a number of ways indicating to a user that a conversation is
> being recorded, one is to play an announcement, another accepted way
is
> to play these beeps at a regular interval (15 / 30 seconds or similar)
> however i cannot seem to find a way to get them to play when
monitoring
> a call - any ideas?
>
> Cheers,
>
> Ben
>
> ___
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>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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RE: [Asterisk-Users] GXP-2000 w/ 1.1.0.11 firmware
I had the same problem! You have in your PXXX in your configs that 1.1.0.11 does not support. Took me an hour to go through my configs and the web page to find what PXXX in my configs unset the phone :) Once its done, the phone will be accept the configs with no problems. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Ringwald Sent: Wednesday, 17 May 2006 10:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] GXP-2000 w/ 1.1.0.11 firmware I had provisioning via tftp working on this phone. I have verified that after the firmware upgrade, it contacts the tftp server and downloads the cfgMACADDR file, and the ring/etc files successfully. Unfortunately, changes made to the config file don't make it to the phone (SIP account info/server info, etc). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium cards, so disappointing !
Opened pseudo zap interface, measuring accuracy... 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% --- Results after 111 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.999015 Server Specs: Asus P5WD2 Premium Pentium D 830 (Dual Core) Corsair DDR2-6400 2GB RAM (4 pices) 2xSATA2 RAID (linux software mirroring) TE406P (not TE411P as I stated before) Running debian with non-debian kernel (stock standard 2.6.15.4, email if you want .config ) Some anomalies have been observed during the testing of the server before implementing it into production. 1 – The server performed MUCH better with software RAID one then hardware, not so mention it was easier to setup. 2 – DDR2-6400 improved some of the benchmarks over DDR2-5200. My understanding that all samples that come in and out if Digium card are copied to user space so faster ram should be of benefit to the system. The system has not been restarted from December. Only asterisk was upgraded 3-4 times since December. Before unloading zaptel drivers we checked for IRQ misses with zttool (before each unload/load of drivers) and since December we had none. The system is now running realtime (mysql on the same machine), iaxmodem+hylafax combo for receiving faxes. I must say, spending just a little extra to get good hardware pays off in the long run. If you have any questions, email. Boris > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Anton Krall > Sent: Friday, 21 April 2006 14:27 > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] Digium cards, so disappointing ! > > Can you send the output of zttest ? Whats your average and what kind of > hardware are you using? > > That will give people pointers of what to use/expect. > > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium cards, so disappointing !
Our production asterisk server has TE411P and we route close to 50-70K of calls per month through its ports. We have NEVER EVER had any issues with faxing (close to 3k/month) with faxes connected on one of the spans of the card. Moreover, we have had quite a success receiving the faxes with iaxmodem+hylafax thanks to Lee Howard that we're now gradually switching the fax machines to iaxmodem+hylafax combo. Faxes are sensitive to timing and configuration settings of your asterisk. Once your system is "tuned to perfection" you should have no problems faxing at all despite the official stance from Digium. > issues). Then we switched to a TE411P for the hardware echo > cancellation. Now we want to receive fax (< 20/day) on it and > guess what ? Since April 2006 (again a few months after we bought > our brand new card), "officially, fax communications is not > supported with Digium cards" ( http://www.voip-info.org/wiki-Asterisk+fax ___ --Bandwidth 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: update - 512 Simultaneous Callswith DigitalRecording
> Are there any advantages/disadvantages to using tmpfs as opposed to the > following method: Matt, Its simple. To quote the docs, "tmpfs lives entirely in the kernel's caches" It will shrink and grow to accommodate the files that currently on the filesystem. So if you allocate 10GB for your /tmp but only use 500MB it will only use 500MB of RAM. Think of the time your server run out space on your RAM drive... With tmpfs you would still be ok for few weeks (provided you allocated enough space). :) This will also benefit whole system if /tmp is located on tmpfs, not that a stable, production asterisk system would actually use /tmp much (if at all). In short it gives you the benefits of LARGE RAM disk without allocating all that memory beforehand and you don't have to format anything during startup. For further info look at the following link to tmpfs.txt from kernel's docs http://www.kernelhq.cc/browse-view.py?fv_nr=232372 ___ --Bandwidth 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] te110p and interrupts
Is this dual CPU/Core or just P4 with HT enabled? If it is P4, I would recommend to disable HT. Try changing PCI slots for one of the cards (if you have spare PCI slots). > CPU0 CPU1 > 0: 17697848 17714488IO-APIC-edge timer ___ --Bandwidth 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: update - 512 Simultaneous Calls with DigitalRecording
The simplest solution and the one already implemented in linux, tmpfs. It would be best to allocate 4-8GB to tmpfs on /tmp and let the kernel do the work it was designed to do. And you would not be limited to PCI bus speeds. The DDR2800 is about 12GB/sec. Some would say "overheads, etc, etc". Agreed, even at 95% loss (doubtful) you still get higher badwitch then PCI bus/hard rive could do :) Asterisk can be directed to save files to tmp and them you can move the files to remote server with least possible priority. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luki Sent: Monday, 10 April 2006 18:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: update - 512 Simultaneous Calls with DigitalRecording > Has anyone seen these solid state "Drives" from gigabyte yet? - > http://www.pcper.com/article.php?aid=224&type=expert&pid=3 Interesting device. Looks like the burst throughput is right on par with good drives, but you have better sustained throughput and obviously near zero latency. But what truly is the advantage compared to having 4 GB (dedicated) RAM in the machine and making a RAM disk with it? You need the RAM either way and that ought to be at least as fast as this card on a 33 MHz PCI bus. You loose the "non-volatile" advantage but that's about it, no? --Luki ___ --Bandwidth 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] Iaxmodem speed limit?
I Guess you can edit the following line in your hylafax config file for your iaxmodems. Class1RMQueryCmd: "!24,48,72,96" Put exclamation in front of 96 (as it is done with 24) and it should disable the receive with that speed. > > Is there a way to limit the speed of Hylafax to 7200 bits/s to get > more > reliable results? ___ --Bandwidth 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 and Hylafax, on the same box
That's not entirely correct :) > Fax and voice on the same DID is not possible when using a second > application like hylafax. Because how should the two applications decide > which one accepts the call? With the help of iaxmodem (which works really well) its easily done! Just detect the incoming call is fax and the route it to iaxmodem on fax extension. ___ --Bandwidth 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: Asterisk in production as a fax server, anyone?
Most of the problems like these for me are gone since I started using iaxmodem+hylafax combination. Hylafax has ECM capability which just tells the other side to resend the affected frames (not the whole page). With the latest 4.2.5.5 hylafax I even have color support :) Not that I probably needed. Give it a try. Regards > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Craig Guy > Sent: Saturday, 1 July 2006 09:57 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Re: Asterisk in production as a fax > server,anyone? > > In practice I've found that the fax receiving process is sensitive to CPU > load. If the load jumps too high you will see half page fax pages or > black > streaky pages mixed with perfectly good pages in a multipage fax. Things > that can cause this include running agi scripts or rendering your tiff to > another format on your * server. > > I render my faxes on the * server, however received tiffs are queued so as > to render them one at a time. If you get page problems you could try > rendering them on a dedicated server. > ___ --Bandwidth 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] Reload astdb?
> This doesn't answer the original question - why do you need to reload it? > I'll give you an example. An Active<->Active asterisk cluster. In the event one of the servers dies, the other server can take over without loosing registrations. Since most of the SIP clients know how to use DNS failovers its up to asterisk to do its part :) I don't understand why cat we use realtime for it? For sites that need to chare registrations (for whatever the valid or non-valid reasons anyone could think of) they should be able to use realtime architecture instead of astdb. Mind you that only sites that know what they doing will utilise that, so I don't think it will create a major support headache. Allowing users to select realtime or astdb is another step close to a reliable "carrier grade" asterisk active-active cluster :) This is just my 0.01c :) ___ --Bandwidth 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] 5,000 concurrent calls system rollout question
>The main sever is still connected via IP, correct? >Does not matter if you use * for media gateways or an APX8000 - the only >trunking options to get to the main box are IP based. Are seriously going to tell me that a quad xeon/opteron would not handle traffic from 4xGIG cards?? :) ___ --Bandwidth 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] 5,000 concurrent calls system rollout question
>I guess I just assumed that that the connection to asterisk would have >to be IP since it is absolutely impossible to connect ~208 T1s directly >to a single asterisk server. You would have to use an external media >gateway. I am not aware of any 200x T1 or 8x T3 cards for asterisk :) Not necessarily. Granted that you will not be able to have have that many T1's on one system but if the load is spread across multiple "Asterisk Media converters" you should be able to do anything and scale your system much better. Lets consider for example the following scenario: -- -- | * Media | | * Media | |server | |server | | 2x TE406P| | 2x TE406P| -- -- || --- | Main * | | server | | | --- || -- -- | * Media | | * Media | |server | |server | | 2x TE406P| | 2x TE406P| -- -- This will let you serve 192 channels per media server. Media servers will only need to convert PRI<->IP so a cheap DIY Dual Core Xeon MP with 4MB cache would be more then enough to process/compress 196 channels in/out of 2 TE406P's. Also media servers do not need much RAM, hard drives and can run from flash cards. My preference would be convert all the traffic coming out of media servers to G.729 and IAX2 trunk it to main server. IAX2 trunking will save you MANY interrupts and will improve your bandwidth utilisation between Media and Main servers. With this setup you can run Media and Main servers on private gigabit network which would be more then enough to handle IAX2 trunked G.729 traffic from media servers. Network redundancy can easily be achieved between Media and Main servers by adding NIC's to each and using many known techniques (bonding, routing, VRRP, etc, etc). The Main Asterisk server can be setup with load balancing/failover. Media servers will need to be aware of this. The good thing in the setup like this is that its easy to scale up when needed, you're not exposed of loosing all of your T1's if one of media servers fail, you can easily add more T1's in your setup. The Main server would need a quad gigabit card (intel is a good choice) and since it would not be hampered by Zaptel traffic and it would not need to do any transcoding (except for odd voicemail usage, that could be send to another server) you could use 2xDual Core Xeons. A separate dual port (for redundancy) gigabit card would be used to serve SIP clients. We're working with one of the ISP's on testing and perhaps implementing this setup for them. This setup is considerably cheaper then $1M proposed Cisco setup and can be made as reliable as Cisco solution is. Please don't get me wrong, if I'd have $1M-$5M to spare would go for Cisco. But most of us don't have that much money and if we would, we would never be reading any messages on asterisk-users. Asterisk can be made as reliable and scales as good if not better then any Cisco solution and the fraction of the cost. Now imagine all of this with the new DS3000P in media servers! All hail Asterisk! :) ___ --Bandwidth 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] Urgent: Unable To Execute after updating from SVN
Delete /usr/lib/asterisk/modules/app_md5.so and update your dialplan if you use MD5. It is now done in functions. /usr/lib/asterisk/modules/app_md5.so is a leftover from your previous installation. [app_md5.so]Jan 29 02:49:10 WARNING[32424]: loader.c:326 __load_resource: /usr/lib/asterisk/modules/app_md5.so: undefined symbol: option_priority_jumping Jan 29 02:49:10 WARNING[32424]: loader.c:555 load_modules: Loading module app_md5.so failed! ___ --Bandwidth 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] Win up to $2000 for Asterisk EnterpriseReferences!
Why not? Digium works hard in hardware & software department. It constantly improves its hardware offering. The software arm has been busier then ever! Million bug fixes, MANY MANY improvements, roadmap (at least from what I can see from contributing developers in SVN) is amazing. Asterisk and Digium have great feature together. Admittedly almost all had problems in one place or another but most of it is "user/config" problem. I would not have invested in Digium's hardware and taken up asterisk if I were not confident that Asterisk can "cut it". If you take a look at general Digium & Asterisk are success stories in itself! I'd volunteer for sure but my little installation probably a drop in the ocean compare to the ones I hear and read about and Australia is not exactly has competitive market for that. :) In fact I share the same view about any company that supports Asterisk community. Even for Digium competitors (who have the same dedication as Digium as well) Give it a chance, lets not forget that Digium spends great time, effort and expense getting Asterisk to where it is now. I don't know of many hardware manufacturers that do the same thing. Regards >I was going to bite my tongue on my response to this, but keeping quiet is >driving me nuts. >If this is a legit post... >In short, this irritates the heck out of me. Maybe if Digum supplied some >documentation for less than $175/hr, then there might be a few success >stories. The lack of any official documentation in my opinion is limiting >the success of Asterisk. I seem to spend most of my Asterisk time >researching people's personal heresay about how to get stuff to work. Often .the personal heresay is just someone else's heresay cut and pasted. >Why the heck should anyone help Digium with good press in this instance? ___ --Bandwidth 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] Echo cancellation over satellite link
In software asterisk can support more than that, no? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Tuesday, 6 December 2005 17:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Echo cancellation over satellite link On 12/6/05, funny guy <[EMAIL PROTECTED]> wrote: > Hi, > > Just wondering, is the echo canceller in the TE411P capable of cancelling > the echo caused by the delay over satellite link (i.e. approx 400 ms delay)? > > Does anyone have any success story to share? > > I'm kinda stuck with a really2 annoying echo... adjusting the gain didn't > help... and what should my zapata.conf look like for effective echo > cancellation? > > Thanks in advance ^_^ > No. Neither Digium nor Sangoma I believe are putting in hardware cans that would support a 400ms+ tail. I think the most you're going to get is 128ms. -- 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 ___ --Bandwidth 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] h323 vs oh323
No, max we used is 30 channels. But according to voip-info its faster protocol because it offloads media processing to asterisk (which is a better choice I think) and only looks after H323 call setup. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 11:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] h323 vs oh323 Ok.. how many channels are you using? More than 100? Maybe it can be good only for 10 or 20 simultaneous connections... Isamar On Tue, 6 Dec 2005, Boris Bakchiev wrote: > I like the chan_ooh323. > I like the idea of selfcontained H323 channel that doesn't rely external > libraries, often with specific versions that conflict with something ___ --Bandwidth 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] h323 vs oh323
I like the chan_ooh323. I like the idea of selfcontained H323 channel that doesn't rely external libraries, often with specific versions that conflict with something else. OOH323 works "right out of box" and since we started using it to interconnect Asterisk to Samsung OfficeServ 500 we had no problems whatsoever. regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 08:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] h323 vs oh323 Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: > Hello, > > Would you please share your experience regarding h323 and oh323 in asterisk. > I am confused to choose one. > > Thanks, > > > -- > You don't have any choice, you already made it before you came here.___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth 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] TE210P & Linux SMP
Hi Kris, I have TE406P (same as your but quad span) working on 2.6.13 with pre-empt. I had it working fine with 2.6.14 but I could not switch card's IRQ from CPU0 to CPU1 on the 2.6.14 On 2.6.13 CPU1 is handling IRQ's only for TE406P (with occasional timer IRQ's sneaking in). I suggest that you get the latest source for zaptel from SVN repository. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kris Amy Sent: Wednesday, 30 November 2005 19:31 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TE210P & Linux SMP Hi, Does anyone have this card(specifically the wct4xxp driver) working under linux and running a SMP kernel? I'm running it in a dual p4 xeon box and when I compile the kernel for SMP and then recompile libpri/zaptel the module doesn't behave correctly(doesn't pick up the pri's). In addition the lights on the back do the following (when no cable is plugged in):- No module - alternate red really fast Module under UP - alternate slow red Module under SMP - Blank I have tried the following kernels:- 2.4.29, 2.4.32, 2.6.14. I would really like to see it working correctly under 2.6 in SMP (with pre-empt etc). Otherwise half of this machine is kinda useless. ___ --Bandwidth 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] Recommended PCI latency time?
Hi, What would be a recommended PCI latency timing for server running TE406P card? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] high CPU usage when using -c
Hi > Might check to see how many mpg processes are running, or use top to > see if that's the culprit. If so, kill off the mpg that's doing it. I'm not running any mpg123 processes as I'm using native music on hold (raw files) It has something to do with the color option for the asterisk. If I don't use it, no CPU usage. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] high CPU usage when using -c
Hi, I have a "peculiar" problem with asterisk using 100% cpu (one of the thread just nails one of the CPU's on dual-code system). Asterisk is running chrooted and under its own username. If I alter the init script and add -c to PARAMS variable one of the CPU's is being hammered by asterisk. I remove -c and everything is fine. Has anyone else noticed that? Boris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] List of Motherboards or Servers that are testedok with Asterisk and Digium boards
I think someone needs to start some sort of wiki that everyone can enter the details of they systems J From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of vador loupe Sent: Thursday, 17 November 2005 09:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] List of Motherboards or Servers that are testedok with Asterisk and Digium boards I have TE110P runnig fine in: S5112G2NR > http://www.tyan.com/products/html/tomcati7210_spec.html > On 11/16/05, Robbie Hughes <[EMAIL PROTECTED]> wrote: I'm just bought a Dell SC430 (cheapest server they do) running a P4 2.8 with 512mb ram. I did the install today and it appears to work perfectly with a te110p for pri isdn and tdm400p with 4 modules for 4 fax machines. No call drops, pops, squeaks or anything over any channel in 4 hours of testing. zttest gives the following: # ./zttest Opened pseudo zap interface, measuring accuracy... 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% --- Results after 12 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.997965 i'm using [EMAIL PROTECTED] 1.5 > Date: Wed, 16 Nov 2005 16:48:03 +0200 > From: George Vagenas <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] List of Motherboards or Servers that are > tested ok with Asterisk and Digium boards > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi all, > > Can i find somewhere a list of tested motherboards or server that > works > fine with Asterisk and Digium boards? Digium has a page that mention > some models that are already known don't work fine with the Digium > cards, but i am looking for something more updated. Any clue? > > Thanks > George > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Linksys PAP2: supported codecs
You can do multiple g723 codecs on PAP2 though. > > Yeah, I can confirm that. I added more "allow" > statements for other codecs for that device as a > fallback. Either codec works great, just not at the > same time when calling each other. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF detection in TE406P ??
Hi, I’m getting a lot of false DTMF detections on my system. Following is a diagram of my system: PRI<->TE406P SPAN1<->TE406P SPAN3<->PABX Basically anyone talking to me with a higher pitch voice (Ladies) I get “beeps” all over the place. If I unplug PRI from Asterisk and plug it directly to PABX I do not get any “beeps” during conversation. I noticed that the latest wct4xxp sources allow disabling DTMF support in VPM modules. Will it help me in this situation and if disabled, will I still be able to call IVR systems from my PABX? Thanks in advance! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?
Get VoiceBlue VoIP GSM gateway. It works very well with asterisk. I have been using it for the last 4 month and its fantastic! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomasz Chmielewski Sent: Friday, October 28, 2005 10:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk? I was wondering if there is something like that on this Earth: Some of our users are "mobile users" - they are rarely in one place for longer than 15 minutes. They use mobile phones a lot. From our mobile operator we have an offer which allows us to call for ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone running zaptel's watchdog in production?
Hi, Is anyone running zaptel’s watchdog in production? Any adverse effects on using it? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo canceller on TE406 & Asterisk
Hi Kevin, Thanks for your reply. That probably what it was. :) Could echo cancellation on PBX conflict with VPM module and create the "warping babble" sound that my users are reporting? Do echocancelwhenbridged and echotraining do anything when VPM module is used? Should I be using them? Regards > > I was expecting that asterisk would disable its echo cancellation once > > it find on-board module. > > If you have 'echocancel=no' in zapata.conf, then there is no echo > cancellation, software or hardware. > > If you have 'echocancel=yes', then there is echo cancellation. If you > have hardware available, it will use it, otherwise it will be done in > software. > > I suspect you had the hardware echo canceller disabled without realizing > it. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo canceller on TE406 & Asterisk
Hi, I have TE406P (2nd gen card with echo cancellation on-board). We still notice quite often echo on our PBX that is connected to one of the spans on TE406P (with calls routers to PRI provider on another span). I've tried to experiment with the echo cancellation on asterisk. I enabled echo cancellation in Zapata.conf to see if I can improve the situation and users started reporting "warping bubble" (description I got from one of the users) sound on calls from PABX->Asterisk->PRI (and other way). I was expecting that asterisk would disable its echo cancellation once it find on-board module. The strange thing I noticed that after system reboot things are now better. Although I cannot say for sure because the system was ever rebooted 2 times. Can anyone shed some light on this? Has anyone had similar problems? Or point me into right direction for troubleshooting? Regards Boris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp frame slip tolerance.
HI, How many frame slips would spandsp tolerate before faxing becomes impossible? Using ztclock my current system slips a frame every 60 seconds. Does each frame slip means a failed fax or will there be retransmission of the block/page that had the frame slip? Regards Boris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap Clocking - Frame Slips - tdm400p wcfxozttest cpu spikes spandsp
Hi, My output from TE406P is: 483328 samples in 60.415876 sec. (483327 sample intervals) 99.999794% 483328 samples in 60.415900 sec. (483328 sample intervals) 100.00% 483328 samples in 60.415872 sec. (483327 sample intervals) 99.999794% Estimate 8 frame slips every 483.328003 seconds. Does that mean I can run spandsp more or less reliably for faxing? Regards > 483328 samples in 60.413670 sec. (483310 sample intervals) 99.996277% > 483328 samples in 60.413665 sec. (483310 sample intervals) 99.996277% > 483328 samples in 60.413670 sec. (483310 sample intervals) 99.996277% > Estimate 8 frame slips every 26.851555 seconds. > > I see the above appears to be slightly better then the numbers posted > in your example. Running spandsp fails on the above system with > nothing else running on this system (no calls, no nothing). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium Cards in Australia
I'll vouch for them. :) Very nice people and service. Boris > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Callum McGillivray > Sent: Wednesday, 14 September 2005 12:42 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Digium Cards in Australia > > Hi Rudolf, > > Talk to Australian Technology Partnerships (www.atp.org.au). > > Cheers, > > Callum > > [EMAIL PROTECTED] wrote: > > >Hi, all > > > >Where can I get "Asterisk Developer's PCI Kit" in Australia? > >It is a TDM400P with 1FXS and 1 FXO module. I amight need an extra FXS > module as well. > > > >Thanks, > >Rudolf > > > >___ > >--Bandwidth and Colocation sponsored by Easynews.com -- > > > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE406p no interrupts
Well. That means pci_register_driver probably not ding what it supposed to do. In newer kernels pci_module_init should be replaced with pci_register_driver as pci_module_init doesn't it what it supposed to. How brave are you at getting a new kernel on your system? I'm currently running on 2.6.13 on 955X chipset and it works really well. At first I had all sorts of problems with interrupts but with couple of patches to wct4xxp all working just fine with close to 3-5K of calls per day. What is the model of the motherboard you have? See if you can force a particular IRQ on a slot where your TE406P is. Some motherboards do allow this, so you can assign IRQ bellow 15 to the card. That could help as well. For now, revert the changes back. If you can, try new kernel (in parallel) with the pci_register_driver. Regards > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jason Kim > Sent: Monday, 12 September 2005 11:28 > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] TE406p no interrupts > > I modified wct4xxp.c and installed it. > This is the message for 'modprobe wct4xxp' > > -- > FATAL: Error inserting wct4xxp > (/lib/modules/2.6.9-1.667smp/extra/wct4xxp.ko): No > such device > FATAL: Error running install command for wct4xxp > astpbx kernel: Oops: [1] SMP > astpbx kernel: CR2: a0362081 > > Regards, > Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE406p no interrupts
You should have just done this: rmmod wct4xxp rmmod zaptel modprobe wct4xxp It will do the same thing > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jason Kim > Sent: Monday, 12 September 2005 00:34 > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] TE406p no interrupts > > I modified wct4xxp.c and make clean; make linux26; > make install; reboot; > But the system is not rebooted. > Because the system is in remote office I will check it > next morning. > Could you let me know your linux version, * version > and motherboard? > > Thank you Boris. > > --- Boris Bakchiev <[EMAIL PROTECTED]> wrote: > > > Well. > > Try this please (but only if you're running on the > > latest sources). > > Open wct4xxp.c sources and search for > > pci_module_init > > Replace it with pci_register_driver > > So the line should read: > > res = pci_register_driver(&t4_driver); > > > > That allows you to get the card working on 2.6.13 in > > almost exactly the > > same setup as yours. > > > > One weird thing though. Do no use insmod > > ./wct4xxp.ko from zaptel > > directory as it will not work. Do a proper make > > install and then > > modprobe. > > > > > > This is just part of the fixes you might need to do. > > If you encounter a problem after span > > reconfiguration (ztcfg) let me > > know. > > > > If you get stuck.. let me know. > > > > Regards > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On > > Behalf Of Jason Kim > > Sent: Sunday, September 11, 2005 8:14 PM > > To: asterisk-users@lists.digium.com > > Subject: RE: [Asterisk-Users] TE406p no interrupts > > > > I'm using FC3. > > > > uname -a > > - > > Linux billitipcc 2.6.9-1.667smp #1 SMP Tue Nov 2 > > 15:09:11 EST 2004 x86_64 x86_64 x86_64 GNU/Linux > > ___ > > --Bandwidth and Colocation sponsored by Easynews.com > > -- > > > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > __ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE406p no interrupts
Well. Try this please (but only if you're running on the latest sources). Open wct4xxp.c sources and search for pci_module_init Replace it with pci_register_driver So the line should read: res = pci_register_driver(&t4_driver); That allows you to get the card working on 2.6.13 in almost exactly the same setup as yours. One weird thing though. Do no use insmod ./wct4xxp.ko from zaptel directory as it will not work. Do a proper make install and then modprobe. This is just part of the fixes you might need to do. If you encounter a problem after span reconfiguration (ztcfg) let me know. If you get stuck.. let me know. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Sunday, September 11, 2005 8:14 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] TE406p no interrupts I'm using FC3. uname -a - Linux billitipcc 2.6.9-1.667smp #1 SMP Tue Nov 2 15:09:11 EST 2004 x86_64 x86_64 x86_64 GNU/Linux ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE406p no interrupts
What kernel are you using? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Sunday, September 11, 2005 7:48 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TE406p no interrupts Hi, I've installed an TE406p, asterisk1.2 on tyan opteron board. After installation there is no interrupts from TE406p. Is this board stable? Should i change * version to 1.0.9? Regards, Jason __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI and Caller ID when immediate=yes
Hi,
I would like to utilise immediate=yes to monimise the delay that simle
switch introduces.
When I set this option, ${EXTEN} is not populated so I I’m
unable to do some prepocessing of calls.
Is there a way to populate or retreive EXTEN from a channel that’s
been setup with immediate=yes option?
Thanks
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[Asterisk-Users] Problem with PRI channels, restarted after every call.
Hi,
I got a problem with PRI that I’m not sure how to
solve.
Asterisk sits between PABX and PRI.
PRI is span 1 and PABX is span 2.
After every single call (no matter in what direction) I
get “pri_fixup_principle: Call specified, but not found?” and “pri_dchannel:
Hangup on bad channel” messages and the channel in question is restarted.
As far as I can see, all calls complete fine.
What could cause this? I’d appreciate any help given.
Here is the debug log.
-- Goto (PRI_NET_Out,96422241,1)
-- Executing Dial("Zap/92-1",
"Zap/r0/96422241|300|j") in new stack
-- Requested transfer capability: 0x00 -
SPEECH
-- Called r0/96422241
-- Zap/15-1 is proceeding passing it to
Zap/92-1
-- Zap/15-1 is ringing
-- Zap/14-1 is ringing
-- Zap/14-1 answered Zap/93-1
-- Attempting native bridge of Zap/93-1
and Zap/14-1
-- Zap/15-1 answered Zap/92-1
-- Attempting native bridge of Zap/92-1
and Zap/15-1
-- Channel 0/31, span 3 got hangup
request
-- Hungup 'Zap/14-1'
== Spawn extension (PRI_NET_Out, 97354333, 1) exited
non-zero on 'Zap/93-1'
-- Hungup 'Zap/93-1'
Sep 8 10:36:03 WARNING[13375]: chan_zap.c:7651
pri_fixup_principle: Call specified, but not found?
Sep 8 10:36:03 WARNING[13375]: chan_zap.c:7651
pri_fixup_principle: Call specified, but not found?
Sep 8 10:36:03 WARNING[13375]: chan_zap.c:8701 pri_dchannel: Hangup on bad channel
0/14 on span 1
Sep 8 10:36:07 WARNING[13375]: chan_zap.c:7651
pri_fixup_principle: Call specified, but not found?
Sep 8 10:36:07 WARNING[13375]: chan_zap.c:8701
pri_dchannel: Hangup on bad channel 0/14 on span 1
-- B-channel 0/14 restarted on span 1
Boris Bakchiev
Jildent Pty Ltd
Tel: + 61 3 8080 5898
Fax: +61 3 9811 4716
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[Asterisk-Users] Activate/Deactivate Hardware echo cancellation on TE406/TE411 when briging
Hi, How would one activate/deactivate hardware echo cancellation on the TE406 card? Can it be done per channel? I'm going to run TE406 in the following scenario: ISDN -> TE406 -> PABX I understand from Steve Underwood's site that echo cancellation is not good for faxes (and they do that themselves). So what I want to do and bypass echo cancellation for selected extensions before bridging the calls from span1 to span2. Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voiceblue and slow dialling
Hi Raph,
We have bought the units from the same supplier as you (Talk to Us).
All our calls take about 5 seconds before the mobile we're calling
starts ringing. Some calls take up to 7 seconds but I think it depends
on the carrier.
I believe we have even tested one of your units for Matt because he was
puzzled why it takes so long to call, sure enough it only took 5 seconds
to for my mobile to ring.
We're running latest CVS here so I cannot comment if you're running
something older. I suggest check your peer setup in sip.conf, maybe
initiate sip debug so you can see the Asterisk and VB exchanging SIP
messages.. Maybe you can stumble across something obvious.
VoiceBlue unit is excellent and is definitely much faster then any other
GSM diallers we have tried (Including Ericsson's and Tellular).
In fact none of the GSM diallers we have tested ever went bellow 7
seconds (when a remote mobiles rings) and usually been between 8-15
seconds.
Regards
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Raph Even-Chaim
> Sent: Wednesday, 24 August 2005 09:32
> To: 'asterisk-users@lists.digium.com'
> Subject: [Asterisk-Users] Voiceblue and slow dialling
>
> Hi,
>
> I have recently started a job as a system administrator, and as part
of my
> responsibilities I have to look after an asterisk system. Quite
impressed
> with it, but have one or two niggling issues. One of the last things
my
> predecessor here did, was install a VoiceBlue mobile gateway unit, and
> though it seems to work ok, nearl 20 seconds pass from dialling a
number
> to the call connecting, which is entirely too long. The supplier of
the
> voiceblue, reckons it shouldn't take any longer than 5 seconds.
> Watching the calls go through, it seems that the longest bit of time
is
> the Asterisk handover to the gateway, so I am wondering if anyone out
> there can help me sort this out.
>
> The setting in the extensions.conf is as follows (set to route 80% of
our
> mobile calls through the voiceblue, and 20% through our other mobile
> gateway)
>
>
> -
> [macro-mobrdial]
> include => local
> exten => s,1,AbsoluteTimeout,3600
> exten => s,2,Random(80:5)
> exten => s,3,Dial(IAX2/mobgwy/${ARG1})
> exten => s,4,Dial(Zap/g2/${ARG1})
> exten => s,5,Background(extension)
> exten => s,6,Dial(SIP/[EMAIL PROTECTED],,)
> exten => s,7,Dial(Zap/g2/${ARG1})
> exten => s,8,Hangup
> exten => s,105,Hangup
> exten => s,107,Hangup
>
>
>
> thanks
> Raph Even-Chaim
>
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RE: [Asterisk-Users] WARRNING REGARDING Support from 2n.cz !!!!
2n also asked to buy from local distributor, and I’m glad they asked. Works out that here in Australia distributor had to get the units tested and certified for us to be able to use them. I just checked the 2n’s site, all downloads are available without logon for VoiceBlue products.. What have you tried to download? Regards From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Florin Mandache Sent: Wednesday, August 24, 2005 7:04 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] WARRNING REGARDING Support from 2n.cz Importance: High Just a warring for everybody regarding this company. I emailed 2n.cz to buy 2 gateways to integrate 3g network and GSM with my asterisk box. They emailed me back to contact the local dealer for them. So I did and I bought the 3G analog gateway (bought from the local dealer) and BLUEVoice gateway(I bought it from a dealer in Serbia when I was there for a meeting) too. Having problems setting them up I was looking for the updates of the software (the software from the CD has HUGE bugs, and I even can’t make it talk with the gateway in first place!). On their site to access the files u need to have an account, so I signed up, and after 12 hours someone from 2n.cz contacted me telling me to contact the reseller for the updates (the reseller doesn’t have such section for updates AT ALL, or even news section regarding the updates!), so I have to keep look on the 2n.cz site, see if is any updates, and then to email the reseller and god know after how many days to receive the update! Also in the email I been told that IF I bought from them direct in the first place I was gain access to their support section (HELLO, I wanted to buy direct from them and they told me to buy from the distributor!!!). Checking their manual (from the cd) I found that they say:” ! Important ! The manufacturer constantly improves the firmware contained in the product. The ISP technology (In System Programming) used therein helps you load the latest version into the VoiceBlue gateway using a common PC anytime. For the latest firmware version including all necessary details see www.2n.cz and for instructions see the Firmware Upgrade section hereof. We recommend you to apply the latest version to avoid problems that have been eliminated and acquire new functions free of charge.” So catch 21 !!!??!! So looks like I’m going to have hell getting updates for this products (which looks like don’t even work properly!!!) So if you want hell support please go ahead and buy from 2n.cz! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Solved: Unable to load module for TE406P
It works out that name "Unified t4xxp/t2xxp driver" is not accepted anymore by 2.6.13 kernel. Need to remove "/" for it to load properly > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Boris Bakchiev > Sent: Monday, 15 August 2005 18:17 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Unable to load module for TE406P > > Hi, > > I'm unable to load wct4xxp module for TE406P card. > > I've compiled 2.6.13-rc6, got latest CVS (as of today) of zaptel, but > when I try to load the module I get this: > > kobject_register failed for Unified t4xxp/t2xxp driver (-13) > [kobject_register+53/73] kobject_register+0x35/0x49 > [bus_add_driver+62/153] bus_add_driver+0x3e/0x99 > [driver_register+55/58] driver_register+0x37/0x3a > [pci_register_driver+120/134] pci_register_driver+0x78/0x86 > [pg0+945848335/1069265920] t4_init+0xf/0x22 [wct4xxp] > [sys_init_module+199/462] sys_init_module+0xc7/0x1ce > [syscall_call+7/11] syscall_call+0x7/0xb > > Can anyone point me into right direction to solve this? > Thanks! > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to load module for TE406P
Hi, I'm unable to load wct4xxp module for TE406P card. I've compiled 2.6.13-rc6, got latest CVS (as of today) of zaptel, but when I try to load the module I get this: kobject_register failed for Unified t4xxp/t2xxp driver (-13) [kobject_register+53/73] kobject_register+0x35/0x49 [bus_add_driver+62/153] bus_add_driver+0x3e/0x99 [driver_register+55/58] driver_register+0x37/0x3a [pci_register_driver+120/134] pci_register_driver+0x78/0x86 [pg0+945848335/1069265920] t4_init+0xf/0x22 [wct4xxp] [sys_init_module+199/462] sys_init_module+0xc7/0x1ce [syscall_call+7/11] syscall_call+0x7/0xb Can anyone point me into right direction to solve this? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Failover question
The registry's are stored in DB. Just export your database with 'database show' Schedule it with cron to run every 5 minutes or so. You can do that with -rx command line switch for asterisk. Send the file across to other node and pipe it through awk/perl/cut or whatever you like and import it when you bring the other node up. You will have to stop and start asterisk I think. I think this should work :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohamed A. Gombolaty Sent: Thursday, June 30, 2005 10:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Failover question Dear All, I am using Linux-High Availability between two Asterisk servers, everything is fine but I do have one problem with this, When a server fails and the other assumes the ip address and start asterisk on server 2, the ip phone must re-register themselves again, otherwise the phones won't ring. Does anyone have Ideas of how to overcome this. Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: iax2 can't listen on virtual interface
He is using HA so I'm assuming he is running Master-Slave combo. That means HA will start asterisk on slave after taking over the IP and becoming a master. Until that time, asterisk does not need to be running on a slave so there should be no problems whatsoever. If he wants to run asterisk in Master-Master that is a different story but probably not what you want. Even then it is possible. When becoming a master just script HA to unload chan_iax, assume the virtual IP, substitute the bindip in iax.conf (sed will do just fine) and then load chan_iax backup again. All that can be done while asterisk is still running. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Thursday, June 16, 2005 5:19 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: iax2 can't listen on virtual interface In article <[EMAIL PROTECTED]>, Boris Bakchiev <[EMAIL PROTECTED]> wrote: > Yes you can. > > Just tell iax to bind to that virtual address in iax.conf I don't think that will work on the box that doesn't currently own that virtual address. I think the only way is to make sure the bind address is 0.0.0.0 If you've already done that and it still doesn't work, then I don't know, sorry. Cheers Tony > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Lance Grover > > Sent: Thursday, 16 June 2005 14:53 > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] iax2 can't listen on virtual interface > > > > Can anyone shed some light on this, I have two asterisk boxes using > > heartbeat for failover. Sip traffic works just fine with the virtual > > IP but IAX does not. For example on my servers one server has the > > following: > > > > eth0 = 192.168.1.95 > > eth0:0 = 192.168.1.2 > > > > the other server has: > > > > eth0 = 192.168.1.220 > > > > if the first "Master" server goes down the second server will take > > that virtal IP for it's eth0:0 but in either case the IAXY phones > > cannot connect to this floating virtual IP but can connect to either > > of the regular interfaces IPs. > > > > Please let me know if I am incorrect or if ther is something I can do. > > > > -- > > Thanks, > > > > Lance Grover > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > This message (and any associated files) is intended only for the use of the individual or > entity to which it is addressed and may contain information that is confidential, subject to > copyright or constitutes a trade secret. If you are not the intended recipient you are > hereby notified that any dissemination, copying or distribution of this message, or files > associated with this message, is strictly prohibited. If you have received this message in > error, please notify us immediately by replying to the message and deleting it from your > computer. Messages sent to and from us may be monitored... > > Internet communications cannot be guaranteed to be secured or error-free as information > could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain > viruses. Therefore, we do not accept responsibility for any errors or omissions that are > present in this message, or any attachment, that have arisen as a result of e-mail > transmission. If verification is required, please request a hard-copy version. Any views or > opinions presented are solely those of the author and do not necessarily represent those of > the company. > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iax2 can't listen on virtual interface
Yes you can. Just tell iax to bind to that virtual address in iax.conf > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Lance Grover > Sent: Thursday, 16 June 2005 14:53 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] iax2 can't listen on virtual interface > > Can anyone shed some light on this, I have two asterisk boxes using > heartbeat for failover. Sip traffic works just fine with the virtual > IP but IAX does not. For example on my servers one server has the > following: > > eth0 = 192.168.1.95 > eth0:0 = 192.168.1.2 > > the other server has: > > eth0 = 192.168.1.220 > > if the first "Master" server goes down the second server will take > that virtal IP for it's eth0:0 but in either case the IAXY phones > cannot connect to this floating virtual IP but can connect to either > of the regular interfaces IPs. > > Please let me know if I am incorrect or if ther is something I can do. > > -- > Thanks, > > Lance Grover > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P Polarity reversal detection
Hi Can TDM400P detect polarity reversal on FXO module? We have C.O. lines that reverse polarity on Answer and release. Thank You This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G729 codec
Steve, Do you know if Digium's implementation has any of those features? I was not able to find any tech info about it. > There are Annexes up to I. The earlier versions are fixed point, reduced > complexity fixed point and floating point at 8kbps. These are all > compatible. Later annexes add (if memory serves me correctly) silence > suppression, assistance for packet loss concealment (some people say > G.729 includes PLC. It doesn't. What it includes is some features to > reduce how badly PLC works with it), the standard for bit packing, and > additional bit rates of 6.4kbps and 11.8kbps. > > Regards, > Steve This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Credit Card Machines
I had CC readers going over the internet (with pings over 80ms) connected to Linksys PAP2. It was only successful once every 3 attempts. I had 100% reliability when it was connected on LAN. Timing is an issue, if you doing everything on LAN it should not be a problem. Just make sure you use G.711 protocol. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Chris Coulthurst > Sent: Tuesday, 17 May 2005 19:12 > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [Asterisk-Users] Asterisk and Credit Card Machines > > I am planning to deploy an Asterisk server at a local restaurant and was > thinking: I hear a lot of troubles using fax machines with IP trunks. > What about using Credit Card readers? Same basic technology right? A > slow modem to negotiate the transaction. Does anyone have any caveats? > Suggestions? > > Incidentally, the credit cards might be on POTS lines with a Digium > TDM22B. Any concerns using this arrangement? > > > Chris Coulthurst > [EMAIL PROTECTED] > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice Quality
I would use g.729, and if this is an issue, GSM. Setup trunking between both IAX peers so that you can save a lot of bandwidth. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] > Sent: Wednesday, 4 May 2005 00:52 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Voice Quality > > Hello, > > I have setup two * servers and they are communicating using IAX. I'm > passing calls from SRV A (internet connection T1) to SRV B (internet > connection: 512). > > For some reasons I have an issue with the quality. The voice is a bit > scratchy. I have tried iLBC and SPEEX, but it didn't make any difference. > > Now, assuming that I have an issue with Bandwidth, what would be the best > way to configure my iax.conf. (A bit confused about jitterbuffer and tos) > > Here is my iax.conf @ location A: > > [general] > port=4569 > bandwidth=low > disallow=all > allow=ilbc > ;allow=ulaw > ;allow=speex > jitterbuffer=200 > jitterbuffer=yes > tos=lowdelay > > and iax.conf @ location B: > > [general] > port=4569 > bandwidth=low > disallow=all > allow=ilbc > ;allow=ulaw > ;allow=speex > jitterbuffer=200 > jitterbuffer=yes > tos=lowdelay > > [guest] > type=user > context=default > callerid="Guest IAX User" > disallow=all > allow=ilbc > > > Thanks guys > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Transcoding times
Most probably your server was busy starting up when asterisk loaded and calculated the table. Next time, just issue show translation recalc without after the server settles down. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili > Sent: Thursday, 28 April 2005 11:04 > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [Asterisk-Users] Transcoding times > > On what trascoding time depends on? > > I started server, run * and issued command show translations > -- > sipsrv1*CLI> show translation > Translation times between formats (in milliseconds) > Source Format (Rows) Destination Format(Columns) > > g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc >g723 - 3 2 2 2 2 1 358 - 61 > gsm53 - 2 2 2 2 1 358 - 61 >ulaw53 3 - 1 2 2 1 358 - 61 >alaw53 3 1 - 2 2 1 358 - 61 >g72653 3 2 2 - 2 1 358 - 61 > adpcm53 3 2 2 2 - 1 358 - 61 >slin52 2 1 1 1 1 - 257 - 60 > lpc1054 4 3 3 3 3 2 -59 - 62 >g72955 5 4 4 4 4 3 5 - - 63 > speex - - - - - - - - - - - >ilbc54 4 3 3 3 3 2 459 - - > -- > > If I restart * and issue the same command > > -- > *CLI> show translation > Translation times between formats (in milliseconds) > Source Format (Rows) Destination Format(Columns) > > g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc >g723 - 3 2 2 2 2 1 315 - 14 > gsm14 - 5 5 5 5 4 618 - 17 >ulaw11 3 - 1 2 2 1 315 - 14 >alaw11 3 1 - 2 2 1 315 - 14 >g72611 3 2 2 - 2 1 315 - 14 > adpcm11 3 2 2 2 - 1 315 - 14 >slin10 2 1 1 1 1 - 214 - 13 > lpc1012 4 3 3 3 3 2 -16 - 15 >g72913 5 4 4 4 4 3 5 - - 16 > speex - - - - - - - - - - - >ilbc12 4 3 3 3 3 2 416 - - > -- > > Why? > > Irakli > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400P and SCSI/SATA = * noise problems???
We're running asterisk on a pair of 1GB 12mb/s flash cards running on separate IDE channels. We've setup software RAID1 to protect ourselves from failures if any of the flash cards die. VoiceMail is stored on a small IDE that is dedicated just for this. It appears to work quite well. Although we don't have TDM's on our system. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Damian Funnell > Sent: Wednesday, 20 April 2005 05:30 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] TDM400P and SCSI/SATA = * noise problems??? > > Hi Tim, > > Thanks for your post, it's most insightful. It certainly puts a pretty > large dent in my confidence in the TDM for commercial use - imagine if > there was more than one TDM in a system (especially with a RAID adapter). > > Running a PABX without hardware RAID 0 is not an option for us, as we > don't want disk failure to result in the PABX dying, so I guess we are > going to have to research ways of retarding it somehow. > > Cheers, > D. > > FFF Managed Technology Ltd > 60 Cook St > P.O. 6368 Wellesley St > Auckland > t +64 9 356 2911 > f +64 9 358 9070 > m +64 21 415 297 > w www.fff.co.nz > > > > [EMAIL PROTECTED] wrote: > > >Yes. It has to do with latency and bus contention. > > > >I've run a TDM board in an IBM Netfinity 5600 server with an IBM > ServeRAID > >3L controller (SCSI-U2W). The big difference, though, is that the RAID > >controller was on its own PCI bus, and the TDM card was on its own PCI > >bus. > > > >With both controllers on the bus, you can have latency issues. For > >example, if the RAID controller sets up a DMA of a big chunk of disk, it > >owns the bus for that transfer. If an Ethernet packet is delayed by 50us > >during that time, nobody cares. But if the TDM card is delayed, it most > >certainly cares: especially as its generating 1000 interrupts a second! > > > >That's the problem with the TDM cards. They do *nothing* on the CPU > side. > > The CPU has to do *everything*, and it has to do it *immediately*. When > >you are using plain-jane IDE, you can tweak the kernel to put the IDE > >stuff at a low priority. But when you've got a fancy RAID controller, it > >tends to think it's the most important thing in the system. And as a > >rule, hard drive I/O usually *is* the most important I/O going on in a > >system. However, in this case, the TDM card trumps that. And Digium > >doesn't know how to tweak every last RAID driver in existence for > >low-priority operation--or even if it's possible. Hence, the > >recommendation for IDE. > >Yet they require PCI 2.2, which eliminates most Pentium III's and lower! > >:) > > > >I'm still in the midst of testing the TDM cards. So far, so good, in an > >EPIA-based solution and in the 5600. But I've been through at least half > >a dozen different systems before I've found these... > > > >Tim Massey > > > >___ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk timer on Digium's TDM cards?
Hi, Wondering if there is a timer provided on TDM cards? I don't have use for TE110P and it seems expensive just to get it for timer function. I do have ztdummy running but it is hovering on 99.975586% and I'm not sure if this is good enough or not. Any info is appreciated. This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dynamic Dialplan - Turn VM on/off?
Rod,
Here is my macro for this:
[macro-sipexten]
exten => a,1,VoicemailMain(${ARG1})
exten => a,2,Hangup()
exten => s,1,DBGet(NATIMEOUT=Features/${EXTEN}/NATIMEOUT)
exten => s,2,Dial(${ARG2},${NATIMEOUT})
exten => s,3,Goto(s-${DIALSTATUS},1)
exten => s,102,Goto(s,350)
exten => s,350,SetVar(NATIMEOUT=30)
exten => s,351,Goto(s,2)
As you can see it picks it up from DB with default being 30secs if no DB
entry exist.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Rod Bacon
> Sent: Monday, 18 April 2005 15:58
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Dynamic Dialplan - Turn VM on/off?
>
> G'day. I've been working with * for some time now, but mostly from a
> enterprise perspective. I've just setup my own box at home and want to
> enable some more "home user" type functionality.
>
> Does anyone have a trick to allow the dynamic modification of the
> dialplan by users? I want the ability to switch voicemail on/off (or
at
> least alter the timeout).
>
> In essence, I want to simulate the act of manually turning an
answering
> machine on when you leave home (for my wife).
>
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
This message (and any associated files) is intended only for the use of the
individual or entity to which it is addressed and may contain information that
is confidential, subject to copyright or constitutes a trade secret. If you are
not the intended recipient you are hereby notified that any dissemination,
copying or distribution of this message, or files associated with this message,
is strictly prohibited. If you have received this message in error, please
notify us immediately by replying to the message and deleting it from your
computer. Messages sent to and from us may be monitored...
Internet communications cannot be guaranteed to be secured or error-free as
information could be intercepted, corrupted, lost, destroyed, arrive late or
incomplete, or contain viruses. Therefore, we do not accept responsibility for
any errors or omissions that are present in this message, or any attachment,
that have arisen as a result of e-mail transmission. If verification is
required, please request a hard-copy version. Any views or opinions presented
are solely those of the author and do not necessarily represent those of the
company.
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[Asterisk-Users] Digium G.729 vs. IPP G.729
Hi, Did anyone compare G.729 implementations (from Digium and the one based on IPP) on features, stability, quality and reliabilty? It would be intresting to know how they fair against each other. I could be wrong, but in my testing I did notice a bit more hiss on Digium’s codec thein IPP’s. Anyone? This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7940G SIP Conversion
I made the same mistake with my 7960 The content of 'OS79XX.TXT' should be P0S3-07-4-00 and not P003-07-4-00 Same goes for SIPdefault.cnf. After the change everything worked like magic > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Michael West > Sent: Wednesday, 13 April 2005 22:27 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Cisco 7940G SIP Conversion > > Hi, > > I have three Cisco 7940G phones that I'm trying to convert to SIP Image > P0S3-07-3-00 or P0S3-07-4-00. The phone I'm attempting right now has > App Load ID P00305000500. I'm running Cisco's TFTP (v1.1) on a Windows > XP platform. I have configured my DHCP server to hand out the correct > TFTP address as the phone confirms it knows where to find a TFTP server. > > In the Cisco TFTP status window, I'm receiving the following message > continuously: > > > Wed Apr 13 08:21:44 2005: Sending 'OS79XX.TXT' file to 192.168.1.30 in > binary mode# > > > I would expect it to attempt to load the image file next that is listed > in the OS79XX.txt file (P003-07-4-00, but it continues to load JUST the > OS79XX.TCT file continuously. > > Any ideas? > > Michael J. West > [EMAIL PROTECTED] > WESTMark Consulting, Inc. > 34 Wasilla Drive > Worcester, MA 01604-2411 > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for comments on robustness of SpanDSP / app-rxfax / mime-construct
Hi,
SpanDSP works great for me, even over SIP.
I did change the fax macro a bit to make it work better.
The macro records total number of faxes and pages received by the
extension.
Here is the example of 7644 fax extension.
1. Have this in your dialplan
exten => _7644,1,Macro(faxreceive)
2. Here is my macro-faxreceive
[macro-faxreceive]
exten => s,1,SetVar(SavedDATETIME=${DATETIME:4:4}-${DATETIME:2:2})
exten =>
s,n,DBGet(ReceivedFaxes=Features/${EXTEN}/${SavedDATETIME}/Faxes)
exten => s,n,SetVar(SavedCALLERID=${CALLERIDNUM})
exten => s,n,DBGet(TotalPages=Features/${EXTEN}/${SavedDATETIME}/Pages)
exten => s,n,SetVar(FAXFILE=/var/spool/asterisk/fax/in/${UNIQUEID}.tif)
exten => s,n,DBGet(EMAILADDR=Features/${EXTEN}/Email)
exten => s,n,NoOp
exten => s,n,DBGet(LOCALSTATIONID=Features/${EXTEN}/CSID)
exten => s,n,rxfax(${FAXFILE})
exten => s,n,SetVar(SavedREMOTESTATIONID=${REMOTESTATIONID})
exten => s,n,GotoIf($["${FAXPAGES}" >= "1"]?12:17)
exten => s,n,Math(ReceivedFaxes,${ReceivedFaxes}+1)
exten => s,n,Math(TotalPages,${TotalPages}+${FAXPAGES})
exten =>
s,n,DBput(Features/${EXTEN}/${SavedDATETIME}/Faxes=${ReceivedFaxes})
exten =>
s,n,DBput(Features/${EXTEN}/${SavedDATETIME}/Pages=${TotalPages})
exten => s,n,system(/etc/asterisk/batch/sendfax "${FAXFILE}"
"${EMAILADDR}" "${EXTEN}" "${FAXPAGES}" "${SavedCALLERID}"
exten => s,n,NoOp
exten => s,103,SetVar(ReceivedFaxes=0)
exten => s,104,Goto(3)
exten => s,105,SetVar(TotalPages=0)
exten => s,106,Goto(5)
exten => s,107,SetVar([EMAIL PROTECTED])
exten => s,108,Goto(8)
exten => s,109,SetVar(LOCALSTATIONID=${EXTEN})
exten => s,110,Goto(9)
3. Here is my sendfax script
#!/bin/sh
FAXFILE=$1
EMAILADDRESS=$2
RECIPIENT=$3
PAGES=$4
SENDER=$5
CSID=$6
DATES=`date '+%A, %e %B, %Y, %H:%M:%S'`
DATEF=`date +%d%m%Y-%H%M`
[ -z "$FAXFILE" ] || [ -z "$EMAILADDRESS" ] || [ -z "$RECIPIENT" ] &&
exit 0
[ -z "$SENDER" ] && [ -z "$CSID" ] && SENDER=Unknown
[ -z "$SENDER" ] && SENDER="$CSID"
[ -f "$FAXFILE" ] || exit 0
mkdir -p "/var/spool/asterisk/fax/in/$RECIPIENT"
tiff2pdf -z -p A4 -f -c "Virtual Fax" -t "Fax from \"$SENDER\" on
$DATES" -s "Fax from \"$SENDER\" on $DATES" -k "\"$
/etc/asterisk/batch/mime-construct --to "$EMAILADDRESS" --subject "Fax
received on $DATES" --string "You have received $PAGE
mv $FAXFILE "/var/spool/asterisk/fax/in/$RECIPIENT/$DATEF.tif"
This might not be efficent asterisk programming, but it works well.
I did modify mime-construct to send attachments propertly to Outlook.
If Asterisk guru's could improve it, please do so.
regards
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Colin Anderson
> Sent: Wednesday, 13 April 2005 05:57
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] Looking for comments on robustness of
SpanDSP /
> app-rxfax / mime-construct
>
> I've been testing spandsp with mime-construct on 1.0 - stable with the
> inbound fax number being routed as a DID on a PRI. While I have it
> working,
> and it emails me a PDF fine most of the time, I've noticed some issues
on
> receive:
>
> 1. It's a little bitchy on long faxes from analog machines; it just
seems
> to
> rx and rx and rx forever and never finish. Using a Class 1 USR
faxmodem
> with
> our fax software (Zetafax) it works 100%; only sometimes is it a
problem
> with a "plain old fax" that's long. Not using a Canon fax, it's a
Ricoh G4
> 33.6.
>
> 2. Sometimes I watch it spool the inbound file then hang up the Zap
> channel
> and mime-construct doesn't kick in, like it's bailing out of the
dialplan.
> So, the fax comes in ok, but mime-construct never gets around to doing
> it's
> thing. This was really a problem when I had the Asterisk server
pointing
> to
> a slow DNS server since I'm relaying the PDF's to an outside SMTP
server.
> Changing the DNS server to a local DNS server help greatly, but still,
> sometimes the PDF never gets mailed.
>
> I am using the ext-fax context provided in the latest AMP. I haven't
> changed
> anything in that context from the stock install.
>
> My questions:
>
> 1. Any comments on faxrecieve macro robustness in general?
> 2. Are there any recommended ways to set this up on a PRI beyond what
I've
> done?
> 3. Problem (2) I have above seems to be a timing problem in the
dialplan -
> would a Wait() or two help any?
>
> tia
> ___
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This message (and any associated files) is intended only for the use of the
individual or entity to which it is addressed and may contain information that
is confidential, subject to copyright or constitutes a trade secret. If you are
not the intended recipient you are hereby notified that any dissemination,
copying or distribution of this
