[asterisk-users] Unable to set TOS to 184?
I don't understand this message: [2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos: Unable to set TOS to 184 From what I have read the reason is asterisk can't set TOS if not running in root. Mine is running as asterisk. I found one post that says to run at boot: #!/bin/bash /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 4569 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 1:2 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 5060 -j DSCP --set-dscp-class ef Does this make sense? Is this the only method to end ths warning? Thanks, Bart___ -- 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] sched_settime: Request to schedule in the past?!?!
Does anybody know what this message means? [2009-10-17 08:11:59] DEBUG[9598]: sched.c:204 sched_settime: Request to schedule in the past?!?! I've tried all I can find on google with no change It seems to happen when call is at a READ() or Background() - What the caller hear is small delays after a press during a message. Still using asterisk 1.4.22.1 because it seems to have the best DTMF detection during a playback Thanks, Bart___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Method to use SOX inside a Dialplan
I'm trying create a feature that allows a callers to add more speech to his recording. I think this can be done inside a dialplan, but I can't find an example of how to do this. Basically,after he records the primary message, a menu would play asking if he wants to append to this message. If yes, then he would record a temp file with the additional message and when done, I want SOX to add the temp message to the primary message making it one larger message. Would you mind showing me an example of how to run SOX inside the dialplan? Thanks, Bart___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF problem during read()
I have a simple dialplan. Using the read cmd, I ask caller for his passcode. If the caller is calling from an plain old analog phone, his dtmf is not heard until the prompt stops playing. SIP phones work correctly. I've trird everything I found searching the net. I've tried all the dtmfmode. I'm using 1.4.26 Currently my vitelity sip account is setup: insecure=very canreinvite=no host=xx.xx.xx.xx qualify=yes dtmfmode=rfc2833 disallow=all allow=ulaw rfc2833compensate=yes I need to trouble shoot this furthur. I read I can enable rtp debug IP but I can't find any output. Thanks, Bart___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Method to downgrade asterisk
I currently have asterisk-1.4.26.2 installed and working. It was sugguested I try asterisk-1.4.25 to see if it fixes my SIP dtmf problems. What is the method to downgrade? Do I just do in the asterisk-1.4.25 folder: make clean ./configure make install Or do I need to 'make clean' in the asterisk-1.4.26.2 first then move to the asterisk-1.4.25 folder and do ./configure make install? Thanks, Bart___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Appending two voice files
Does anyone know how I can append to different user recorded voice files within a dial plan? For example Asterisk ask caller a question and records the answer, then ask another question record the answer to the end of the first answer - so when it's played back, all the answers are in one playback. TIA Bart___ --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] Freepbx changes dont reflect in asterisk
With FreePBX you can not modify certain conf files - many are overwritten at reload Bart Pedro Silva wrote: Hello, From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS. ___ --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] AEL2 Confusion
I just downloaded and installed asterisk-1.2.13 Reading http://www.voip-info.org/wiki/view/Asterisk+AEL2 says I should be using AEL2, but what's downloaded is pbx_ael not pbx_ael2 as expected. Where and how do I get current release of AEL2 - Is there some 'How To' somewhere? TIA Bart ___ --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] Match Chat Author?
I stumble on this URL that is Chat Line script written by Steven L. Edwards called 'Match Chat' here: http://bugs.digium.com/file_download.php?file_id=11080type=bug But I can't seem to find any additional info on Author or Applications - I was wondering if you might know more about either? Bart ___ --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] 2 CPU's, Only 1 taking IRQ's
I'm trying to solve a echo problem... The system is Centos 2.6.9-34.0.2.ELsmp (SMP) CentOS release 4.3 (Final). And the Box is Dual Intel Xeon CPU 2.80GHz with 2 GB memory. It appears that CPU1 in not taking any interrupts - What steps do I need to do bring up CPU1 and share IRQ requests for a Linux noob? Bart cat /proc/interrupts CPU0 CPU1 0: 70759112 0IO-APIC-edge timer 1:137 0IO-APIC-edge i8042 2: 0 0 XT-PIC cascade 8: 1 0IO-APIC-edge rtc 12: 2564 0IO-APIC-edge i8042 14: 173957 0IO-APIC-edge ide0 15: 635817 0IO-APIC-edge ide1 153: 0 0 IO-APIC-level uhci_hcd 161:1437910 0 IO-APIC-level eth0 169: 0 0 IO-APIC-level uhci_hcd 177: 0 0 IO-APIC-level uhci_hcd 185: 70717012 0 IO-APIC-level wct4xxp 193: 70716921 0 IO-APIC-level wct4xxp NMI: 0 0 LOC: 70763311 70763320 ERR: 0 MIS: 0 Distro Name ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's
Hmm, this must not be installed: # locate irqbalance # /usr/src/kernels/2.6.9-34.0.2.EL-smp-i686/include/config/irqbalance.h How do I install this? Bart Álvaro Palma wrote: It appears that CPU1 in not taking any interrupts - What steps do I need to do bring up CPU1 and share IRQ requests for a Linux noob? Run the IRQ balance daemon (/usr/sbin/irqbalance). It's part of the kernel-utils RPM. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I'm I wrong - No 3-way calling for Single line sets?
It appears the only way to cause a 3-way call (or a screened transfer) is by using conference - nasty This mean SLT would need to transfer to conference than add second party, then add themselves. I've searched and I can't find anything that works in asterisk like the Telco method or am I blind? Bart ___ --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] Caller ID has extra digits to strip
I was hoping someone might have the answer to this: As an update: even though you can modify the Caller ID in extensions.conf for call handling and use CallerID(number) in you script, asterisk does not honor the modified number in CDR and VoiceMail.- I need to fix the number at answer. Bart Bart Fisher wrote: === About 70% of the time, my Local DID provider sends me ANI II digits (see http://www.nanpa.com/number_resource_info/ani_ii_assignments.html) where there will be an extra 2 digits added to the Caller ID - For example 62714222 where '62' = Cell Phone for example.. The problem is, I have not found a way to remove these digits before it's used by Asterisk in CDR and Voice Mail with any Asterisk script or command. What can I do to strip these digits from Caller ID before answering the call so CDR and Voice Mail Caller ID announcement show correct number? TIA Bart ___ --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] Caller ID has extra digits to strip
Well, currently I do not use them, but I hate to give it up either - nice to be able to ID the call type and it might be useful some day. Now it's a pain to deal with. What you get is the same here from GBX, but with MCI, it the same with ANI II added to ANI Bart Steve Totaro wrote: Do you use the AnI II digits for anything? If not, call the telco and tell them to just send ten digits. When I used to have some T1s with UCN, they sent the ANI II digits in a separate field specifically for that. I could see them in PRI debug on the console. Now with Global Crossing, I get *ANI*DNIS* which is pretty funky. Thanks, Steve Bart Fisher wrote: I was hoping someone might have the answer to this: As an update: even though you can modify the Caller ID in extensions.conf for call handling and use CallerID(number) in you script, asterisk does not honor the modified number in CDR and VoiceMail.- I need to fix the number at answer. Bart Bart Fisher wrote: === About 70% of the time, my Local DID provider sends me ANI II digits (see http://www.nanpa.com/number_resource_info/ani_ii_assignments.html) where there will be an extra 2 digits added to the Caller ID - For example 62714222 where '62' = Cell Phone for example.. The problem is, I have not found a way to remove these digits before it's used by Asterisk in CDR and Voice Mail with any Asterisk script or command. What can I do to strip these digits from Caller ID before answering the call so CDR and Voice Mail Caller ID announcement show correct number? TIA Bart ___ --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] Caller ID has extra digits to strip
About 70% of the time, my Local DID provider sends me ANI II digits (see http://www.nanpa.com/number_resource_info/ani_ii_assignments.html) where there will be an extra 2 digits added to the Caller ID - For example 62714222 where '62' = Cell Phone for example.. The problem is, I have not found a way to remove these digits before it's used by Asterisk in CDR and Voice Mail with any Asterisk script or command. What can I do to strip these digits from Caller ID before answering the call so CDR and Voice Mail Caller ID announcement show correct number? TIA Bart ___ --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] em wink, TE110P, * answers too soon
Well you say too much and not enough about the problem or configuration So, I assume the DID's are on Ports 1 - 24 T1?. If asterisk is missing the first digit, then I'll bet the DID T1 from Telco is set to immediate on their side, not wink - Because dialing should NOT start until after the wink from asterisk - Try changing Telco T1 to immediate start and test. Bart Steve Linabery wrote: Hi, I've been googling all over the place and have read the relevant articles in the Digium knowledge base. I have tried all the suggestions I found in the K.B. Spent some time on the asterisk irc, tweaking some parameters as people thereon thought would be helpful, but to no avail. I am trying to set up * on an em wink trunk currently attached to an Avaya Merlin Magix system. The provider of the T1 is McLeodUSA; our location is St Paul MN USA. I am in the process of getting more specific timing information from their tech support, but it takes days. I can call into the * PBX from my cell phone just fine. I can call between the two grandstream phones I bought for testing just fine. Here's the problem. When a call comes into *, * attempts to route it to an extension prematurely. For example, if the DTMF digits coming from upstream are '538', * tries to send the call to extn '53'. I still receive the '8', but too late. Here's a snip from /var/log/asterisk/messages where the incoming DID digits are '535': Aug 7 22:30:00 DEBUG[31492] chan_zap.c: Monitor doohicky got event Ring/Answered on channel 1 Aug 7 22:30:00 DEBUG[31478] devicestate.c: Changing state for Zap/1 - state 2 (In use) Aug 7 22:30:00 VERBOSE[31493] logger.c: Asterisk Ready. -- Starting simple switch on 'Zap/1-1' Aug 7 22:30:00 DEBUG[31494] app_queue.c: Device 'Zap/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Aug 7 22:30:01 DEBUG[31493] chan_zap.c: DTMF digit: 5 on Zap/1-1 Aug 7 22:30:01 DEBUG[31493] chan_zap.c: DTMF digit: 3 on Zap/1-1 Aug 7 22:30:01 DEBUG[31493] chan_zap.c: Enabled echo cancellation on channel 1 Aug 7 22:30:01 VERBOSE[31493] logger.c: == Unknown extension '53' in context 'demo' requested Aug 7 22:30:04 DEBUG[31493] channel.c: Set channel Zap/1-1 to write format gsm Aug 7 22:30:04 DEBUG[31493] channel.c: Scheduling timer at 160 sample intervals Aug 7 22:30:04 VERBOSE[31493] logger.c: -- Playing 'ss-noservice' (language 'en') Aug 7 22:30:04 DEBUG[31493] chan_zap.c: DTMF digit: 5 on Zap/1-1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Exception on 20, channel 1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Got event On hook(1) on channel 1 (index 0) Aug 7 22:30:07 DEBUG[31493] chan_zap.c: disabled echo cancellation on channel 1 Aug 7 22:30:07 DEBUG[31493] channel.c: Scheduling timer at 0 sample intervals Aug 7 22:30:07 DEBUG[31493] channel.c: Hanging up channel 'Zap/1-1' Aug 7 22:30:07 DEBUG[31493] chan_zap.c: zt_hangup(Zap/1-1) Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Hangup: channel: 1 index = 0, normal = 20, callwait = -1, thirdcall = -1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: disabled echo cancellation on channel 1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Updated conferencing on 1, with 0 conference users Aug 7 22:30:07 VERBOSE[31493] logger.c: -- Hungup 'Zap/1-1' Aug 7 22:30:07 DEBUG[31478] devicestate.c: Changing state for Zap/1 - state 0 (Unknown) Aug 7 22:30:07 DEBUG[31495] app_queue.c: Device 'Zap/1' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. Here are some settings from /etc/asterisk/zapata.conf: [trunkgroups] [channels] wink=300 rxwink=300 start=3000 context=default switchtype=national toneduration=100 usecallerid=no cidsignalling=dtmf hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no switchtype = national context = demo signalling = em_w group = 1 channel = 1-20 It has occurred to me that I could just set immediate=yes, read the incoming DTMF digits into a variable, and route to the appropriate extension. That seems more fragile to me since we could someday (when I'm not here) start getting more than 3 digits (caller id, for example). Plus I'd like to make it work the way it's *supposed* to. Any help/suggestions are appreciated! Cheers, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange CLI Output
While the console is in monitor mode (asterisk -r) I see duplicate messages from Asterisk one after the other - But not when I connect via SSH. Example: -- Starting simple switch on Zap/65-1 -- Starting simple switch on Zap/65-1 What could cause this? Thanks Bart ___ --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] Hotels...
LOL - PMS = Property Management System Bart C F wrote: Interesting you said PMS? here is the definition: http://en.wikipedia.org/wiki/PMS On 8/7/06, Brian Vincent (C) [EMAIL PROTECTED] wrote: Ideally you'd get billing to work by integrating directly with the property management software. Most of the big PMS systems, such as SMS, LMS, and FRS, have custom serial drivers written for them that interface with the PBX and related systems. The PMS software is responsible for activating long distance on the phones, adding/removing voicemail boxes, and collecting billing records. It may also do really complicated things like suiting. I don't think you're going to be able to get any of that. For that reason alone, are you sure Asterisk is the right solution? Maybe a little Mitel system would work with their software? Okay, now assuming you have a nice GUI to rebuild mailboxes, you'll have to decide if it's worth restricting/unrestricting phones for long distance. Keep in mind housekeeping staff may like to make international phone calls. It may be easier to just sell calling cards and open up the lines for free local calls (usually local calls incur a surcharge of $1 - $1.50.) Are there conference facilities involved? Do you need special pricing for provisioning lines there? I like the idea of using those Audiocodes boxes, but will fax services work with them? In theory I think they do, but we've had problems passing data over them. Can the Audiocodes boxes drive message waiting lamps? I can't remember, but you'll need that. Wake up calls? Asterisk supports it (Trixbox has a nice implementation), so be sure to test that. Is this a multiproperty hotel and will you need to support 911 to different buildings? --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Monday, August 07, 2006 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hotels... I have to bid on a hotel contract, but there are some things I don't know how to do -- but clearly Asterisk has been used by hotels before, so I figure someone on here must have some answers: 1) While the majority of the phones will be SIP, there will be a couple hundred analogs (due to wiring logistics); what should I terminate them into? 2) Phone activation at check-in/phone de-activation and billing at check-out. Are there GUI tools for this, or should I write my own back/front end? 3) Anything else that those familiar with hotels have bumped into that might not be obvious at the outset? Thanks! -Ken Ken, Long time no see on the list welcome back. 1) The best thing would be is to get a channel bank. Xorcom has one that I believe works over USB though never tried it so I cant comment on it. 2)I dont think there is any software out there for hotels per say but there has been talk about working some of the open source billing programs out there in to a custom app. The only reason why I would go for writing your own is A)You have more control. You can build it for your own custom needs for the ground up. B)People have asked about it before. While I dont know the market size I am sure that you can resell it once you are done. Dovid ___ --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 __ Confidentiality Warning: This message and any attachments are intended only for the use of the intended recipient(s), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by return e-mail, and delete this message and any attachments from your system. Thank you. __ ___ --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:
[asterisk-users] Run a script at certain CLI writes
I'm trying to detect when a T1 goes to Yellow or Red alarm. I noticed these events will be displayed on the CLI. What I'd like to do is cause an email to be sent when from a script on these events, but somehow I would need to capture the CLI outputs to detect messages Message are: wct4xxp: Setting yellow alarm on span 1 wct4xxp: Clearing yellow alarm on span 1 Any clues? Bart ___ --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] Run a script at certain CLI writes
Oh, good idea - the messages do appear there - I'll check it out Thanks Joey McDonald wrote: Have you looked to see if they're being logged to /var/log/asterisk/full ? That would be much easier to detect. --joey On 8/3/06, *Bart Fisher* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm trying to detect when a T1 goes to Yellow or Red alarm. I noticed these events will be displayed on the CLI. What I'd like to do is cause an email to be sent when from a script on these events, but somehow I would need to capture the CLI outputs to detect messages Message are: wct4xxp: Setting yellow alarm on span 1 wct4xxp: Clearing yellow alarm on span 1 Any clues? Bart ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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] [ANN] - Coder Needed for Patch
I've posted on GAF (Free Lance Site) a request for bids for modifications to Asterisk PBX source. If you are interest in bidding on this, please view it at http://www.getafreelancer.com/projects/78138.html Thanks you for your time. Bart Fisher [EMAIL PROTECTED] ___ --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] IAX over two T1 connections bad quality
Replace IAX with SIP - It solved my problems with several providers including FWD and Teliax Bart Jerry Geis wrote: Help please. I have two systems on the net. one in indiana and one in georgia. connected with IAX. local SIP phones in each office (10 each) are cisco and running sip. TDM04B card in each location has 4 local lines. Incoming calls to each location sound fine always. The problem is dialing between offices the call quality is BAD. Both offices are connected to the net with T1 lines. all data. All phones are setup ulaw 64bit. The IAX connection between the boxes is ulaw 64 bit. I tried skype between the two offices and talked for 15 minutes and had no issue. The machine CPU usage is running 92-97% idle most of the time. Running asterisk 1.2.9.1 and zaptel 1.2.6. There are switches in the mix that have voice traffic having priority. How do I determine what is the issue here? Why is the call quality bad and where is it that I can tweek. Jerry ___ --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 __ NOD32 1.1647 (20060706) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How do you recompile individual source modules?
How do you recompile individual source modules? I need to make a small change (addition) to chan_zap.c. I read somewhere you can recompile individual module source without the need to recompile the entire asterisk sources each time at change is made. Can someone tell this 'C' noob how to do this? TIA Bart ___ --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] How do you recompile individual source modules?
If I understand, I cd to asterisk source folder and run make - it take card of rest? Also, when/why should you use astxs? Bart Russell Bryant wrote: - Bart Fisher [EMAIL PROTECTED] wrote: I need to make a small change (addition) to chan_zap.c. I read somewhere you can recompile individual module source without the need to recompile the entire asterisk sources each time at change is made. Can someone tell this 'C' noob how to do this? If you're working in the same Asterisk source tree that you compiled and installed on the machine, then when you run make again, only the files you have modified will be recompiled. That is just a feature of the build system. There is also a utility called astxs in the contrib/scripts/ directory of the source tree that allows you to directly compile a single module. $ cd /usr/src/asterisk $ contrib/scripts/astxs channels/chan_zap.c ___ --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] Urgent source code changes needed
I need someone to patch what I believe to be a simple change to chan_zap.c - I know if I attempt I'll screw it up :) Whom would you approach for doing this? - My requests have received a 'blank stare' from Free Lance sites and I'm running out of time on this install. If you know someone or could handle this yourself, please contact me at [EMAIL PROTECTED] Thanks Bart ___ --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] I need help patching source
I'm trying to provide dial tone on EM Wink type trunks. I found where in source, 'chan_zap.c' where I believe the code needs to be added. Basically I believe I can copy parts used for PRI in to EM and EM Wink signal types. However with my attempts, it fails to compile at chan_zap. And I'm not sure how to proceed now. It seems there should be a way to only recompile 'chan_zap.so' without doing a full recompile - Any help would be appreciated for this novice. Thanks, Bart ___ --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] Dial Tone + EM
Maybe one of you can help me with this: We have T1's that come from both MCI and Global Crossing as uses channelized (24 Ports per T) with inband (DTMF) ANI and DNIS delivery (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk Server. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail correctly (Dialogic D240-SC-T1) - without issues. I guess you recognize these are NOT PRI T1's - but old style DS1. However, when the external voice mail system begins to dial out, it grabs the port waits for the Wink and expects dial tone to be returned afterwards - Hearing none, it just sits there until the time out and gives up. My thinking is there should be an EM signaling type that CAN provide dial tone. - A quick scan of the source (chan_zap.c), it appears there is no such provisions for DT for any of the EM types. To me it appears to be a simple patch, but I'm sure I would screw it up if I attempt this myself, not being a programmer. And if by chance I would get it working, the next update would also need that patch. I'm hoping I can find someone on the list that is willing to add a new EM method with a DT provision and make it available to the release sources Thanks Bart = Zaptel.conf # Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 AMI/D4 RED span=1,0,0,d4,ami em=1-24; = seems like my only choice (em) # Span 2: TE4/0/2 TE410P (PCI) Card 0 Span 2 AMI/D4 RED span=2,0,0,d4,ami em=25-48 ; = seems like my only choice (em) Zapata.conf: ; Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 ; This is attached to CUST 3 VMS System ; signalling =em_w ; = might be wrong choice (see below for others) context=default group = 1 channel = 1-24 ; Span 2: TE4/0/3 TE410P (PCI) Card 0 Span 3 ; This T1 is WorldCom Local 714 DID's ; signalling =em_w ; = might be wrong choice (see below for others) context=from-did group = 3 channel = 25-48 Anybody have a clue for me TIA Bart ___ --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] T1 + EM
Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system correctly (Dialogic D240-SC-T1) - with no issues with this part. I guess you can recognize there are NOT PRI T1's - but old DS1 However, when the external voice mail system begins to dial out, it grabs the port waits for Wink, but expects dial tone afterwards - so it sits there waiting until the time-out and gives up. My thinking is there should be an EM singnalling that provides dial tone. - Any clues? I could not find a good description of the different types and their operation. = Zaptel.conf # Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 AMI/D4 RED span=1,0,0,d4,ami em=1-24; = seems like my only choice (em) # Span 2: TE4/0/2 TE410P (PCI) Card 0 Span 2 AMI/D4 RED span=2,0,0,d4,ami em=25-48 ; = seems like my only choice (em) Zapata.conf: ; Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 ; This is attached to CUST 3 VMS System ; signalling =em_w ; = might be wrong choice (see below for others) context=default group = 1 channel = 1-24 ; Span 2: TE4/0/3 TE410P (PCI) Card 0 Span 3 ; This T1 is WorldCom Local 714 DID's ; signalling =em_w ; = might be wrong choice (see below for others) context=from-did group = 3 channel = 25-48 = I've searched for alternatives signalling types and found these. But nothing was found as to exactly what happens during inbound or outbound calls - (is dial tone provided or not) a.. em_w: E M Wink Start a.. featd: Feature Group D (The fake, Adtran style, DTMF) a.. featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through a Tandem Access point a.. fgccama Feature Group C-CAMA (DP DNIS, MF ANI) a.. fgccamamf Feature Group C-CAMA MF (MF DNIS, MF ANI) a.. featdmf: Feature Group D (The real thing, MF (domestic, US)) a.. featb: Feature Group B (MF (domestic, US)) a.. sf: SF (Inband Tone) Signalling a.. sf_w: SF Wink a.. sf_featd: SF Feature Group D (The fake, Adtran style, DTMF) a.. sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US)) a.. sf_featb: SF Feature Group B (MF (domestic, US)) Anybody have a clue for me TIA Bart ___ --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] EM + Dial tone
Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system correctly (Dialogic D240-SC-T1) - with no issues with this part. I guess you can recognize there are NOT PRI T1's - but old DS1 However, when the external voice mail system begins to dial out, it grabs the port waits for Wink, but expects dial tone afterwards - so it sits there waiting until the time-out and gives up. My thinking is there should be an EM singnalling that provides dial tone. - Any clues? I could not find a good description of the different types and their operation. = Zaptel.conf # Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 AMI/D4 RED span=1,0,0,d4,ami em=1-24; = seems like my only choice (em) # Span 2: TE4/0/2 TE410P (PCI) Card 0 Span 2 AMI/D4 RED span=2,0,0,d4,ami em=25-48 ; = seems like my only choice (em) Zapata.conf: ; Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 ; This is attached to CUST 3 VMS System ; signalling =em_w ; = might be wrong choice (see below for others) context=default group = 1 channel = 1-24 ; Span 2: TE4/0/3 TE410P (PCI) Card 0 Span 3 ; This T1 is WorldCom Local 714 DID's ; signalling =em_w ; = might be wrong choice (see below for others) context=from-did group = 3 channel = 25-48 = I've searched for alternatives signalling types and found these. But nothing was found as to exactly what happens during inbound or outbound calls - (is dial tone provided or not) a.. em_w: E M Wink Start a.. featd: Feature Group D (The fake, Adtran style, DTMF) a.. featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through a Tandem Access point a.. fgccama Feature Group C-CAMA (DP DNIS, MF ANI) a.. fgccamamf Feature Group C-CAMA MF (MF DNIS, MF ANI) a.. featdmf: Feature Group D (The real thing, MF (domestic, US)) a.. featb: Feature Group B (MF (domestic, US)) a.. sf: SF (Inband Tone) Signalling a.. sf_w: SF Wink a.. sf_featd: SF Feature Group D (The fake, Adtran style, DTMF) a.. sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US)) a.. sf_featb: SF Feature Group B (MF (domestic, US)) Anybody have a clue for me TIA Bart ___ --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] How many TE405 ...
The best I've done is 2 - the thirdcard will not start properly everytime - so I gave up - Forget about trying 4 cards, never happen Bart - Original Message - From: Ard To: asterisk-users@lists.digium.com Sent: Monday, June 05, 2006 2:29 PM Subject: [Asterisk-Users] How many TE405 ... Hi, Is it possible to use 4 TE405 boards in one server ? It mean, to have 16 E1s on just one server. Can somebody tell me how many boards is itpossible to have on one server ? Thanks, ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADIT 600 = Asterisk Help
I've been reading the Google searches trying to understand how to tie together Adit 600 to Asterisk to provide 2 way service. I'm about blind from reading. I assume, the answer is using MGCP between the boxes. However, the examples I found don't really explain fully enough to know how to modify examples to work for me. I'll have in the ADIT with T1's. There is a CMG and FXS card installed - later I'd like to add a FXO card. The goal would be able to route calls to and from ADIT from the T1's to Asterisk and route some Asterisk extensions to the FXS card. If you have done this, would you mind posting or sending me your mgcp.conf with some remarks explaining how why and a your CMG config? Thanks for the time Bart ___ --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] What about T400 T1 cards?
Can anyone clue me in about these T400 T1 cards I see advertised? I hear they are Digium Clones. Is there some reason to avoid these? How do they compare to TE410P's for example. Bart ___ --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] What about T400 T1 cards?
I'd love to see. Can you provide me your Google search parameters? I end up getting a lot of motorcycle data Bart - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, May 23, 2006 8:38 AM Subject: Re: [Asterisk-Users] What about T400 T1 cards? On Tuesday 23 May 2006 10:48, Bart Fisher wrote: Can anyone clue me in about these T400 T1 cards I see advertised? I hear they are Digium Clones. Is there some reason to avoid these? How do they compare to TE410P's for example. Google for the performance data on the TE410. They have some pretty graphs that benchmark against the older TE400, which don't bus master. This was the single biggest change between the TE400/TE410. The other is the expansion interface for the voice processing module. I also believe that the TE410 has significant improvements for PCI compatibility. -A. ___ --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] Slash Tone at pstn cut-though?
I'm looking for a method to signal an insideextension (asterisk extension with external dialing appl.) with a DTMF "A" tone to indicate when Asterisk has completed dialing and the voice path has beencut-though on a ZAP T1 Trunk. If this can be done, I'd also like to know if there is a method to know when the called party has answered from the "A" and "B" Bits state changes on the Zap T1 Port. Thanks Bart ___ --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] EM and Dial tone
I'm a bit confused about how to handle this. I have Asterisk sitting in the middle between a Qwest Long Distance T1 (Voice T1, D4, SF, AMI) and an external voice mail PC using a Dialogic D/240SC-T1 card. The Qwest T1 originally was connected to the Dialogic card directly. The signaling was set to EM Wink Start because Dialogic used this as its default settings, so it just worked without fiddling. Before Asterisk: Incoming Qwest calls would wink the Dialogic card and then send DTMF to the Dialogic after it winked back. Outgoing calls from Dialogic would come off-hook, wait for wink. At this point Qwest would send dial tone. The Dialogic has call supervision and wait for dial tone enabled. With the Asterisk in the middle, Incoming from Qwest are directed to Dialogic and are answered correctly. The problem is when the dialogic card wants to dial out, we only get the wink Asterisk - no dial tone. The dialogic reports this as a failure and hangs-up If I remove the Dial Tone Detection option on the Dialogic and add pauses before the dial string, all works OK. My question is how can I emulate the Qwest functionality and provide a dial tone after the wink? TIA Bart My zaptel.conf: # Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 AMI/D4 RED span=1,0,0,d4,ami em=1-24 # Span 2: TE4/0/2 TE410P (PCI) Card 0 Span 2 AMI/D4 RED span=2,1,0,d4,ami em=25-48 My zapata.conf ; Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 ; This T1 is attached to in-house VM System ; signalling =em_w context=from-internal group = 1 channel = 1-24 ; Span 2: TE4/0/2 TE410P (PCI) Card 0 Span 2 ; This T1 is attached to Qwest LD ; signalling =em_w context=from-pstn group = 2 channel = 25-48 ___ --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] TE410P = Dialogic D/240SC-T1
I'm trying to connect an Asterisk T1 port to a Dialogic card. The Dialogic side is an external VMS. I setup for ISDN-PRI between systems and have green lights on both card/ports. Zttool shows connection is good also. However, when I tryattempt terminate or originate a call to either system, nothing appears of CLI. How can I monitor the "D" channel? zaptel.conf: # Span 8: TE4/1/4 "TE410P (PCI) Card 1 Span 4" ISDN/PRI RED span=8,0,0,esf,b8zsbchan=169-191dchan=192 zapata.conf: ;; Span 8: TE4/0/4 "TE410P (PCI) Card 1 Span 4" ; ;switchtype=nationalsignalling=pri_net ; Dialogic set for CPE;nsf = megacom pridialplan=unknownechocancel=yesoverlapdial=noimmediate=noechocancelwhenbridged=yesechotraining=yesrxgain=0.0callprogress=no group=8context=from-internalchannel = 169-191 Any ideas I could try? Bart ___ --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: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover
In these cases, the Transmit and Receive pairs are in different binders. Thus electrically isolating by virtue of how the binders are wrapped with each other and how the pairs are twisted within the binders. Bart - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 24, 2006 9:27 AM Subject: Re: [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover Andrew Kohlsmith wrote: Insulation (especially such thin insulation) does not prevent crosstalk. Distance, shielding and tighter twists do. Ever looked at the underground cable in the street outside your building? If it's more than 20 years old, it's probably paper-insulated gel-filled cable, with an _extremely_ thin amount of insulation between the conductors and _zero_ insulation between the pairs. T1s seem to work just fine on it, unless it's very old or they try to put more than 6-8 spans in a single 100-pair bundle :-( ___ --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] ANNOUNCE: Asterisk Jobs and Consulting Site
Yeah, I've had a project listed at asteriskhelpdesk.com for over a month and it still has 0 bids. I wouldn't waste my time redesigning the pages, they won't come... Bart - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, April 19, 2006 9:58 PM Subject: Re: [Asterisk-Users] ANNOUNCE: Asterisk Jobs and Consulting Site On Thursday 20 April 2006 00:13, Matt Gibson wrote: I would like to announce the availability of a new site dedicated to finding and creating jobs in the Asterisk VOIP field. I've created this site, after noticing there are no sites dedicated to providing quality job postings and hiring abilities to people in the field. You mean like http://www.asteriskhelpdesk.com? I'm sure there are a couple of others too I'm missing. -A. ___ --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] Voicemail() - Reading exit or return results
Here my script:
exten = 230,1,Answer exten
= 230,2,NoOpexten =
230,3,Voicemail(u${EXTEN})exten = 230,4,NoOp(Need results
from VoiceMail() above - should be non-zero and provided to ARG3)
exten = 230,5,NoOpexten =
230,6,GoToIf($[${ARG3} = 0]?s|8) exten =
230,7,system(/var/lib/asterisk/agi-bin/230.php|${EXTEN}) ; exten
= 230,8,Hangup
I need to know how to read and use 'exit' results
into ARG3 from voicemail() so the script will continue past line 6 or
not
Thanks
Bart
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Re: [Asterisk-Users] On site installtion Tech. wanted
Maybe I could help. Located in Buena Park Bart - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, March 25, 2006 2:06 PM Subject: [Asterisk-Users] On site installtion Tech. wanted Looking for a Tech. that could install and configure Asterisk systems in and out of California per job basis? Mark Voice international 714-279-0204 ext 102 ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: ADIT 600 Manual needed
If you would be willing to make available for download the Adit 600 Install / Configuration manual for this unit I would gladly PayPal you for your time and troubles... TIA Bart [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with two cards Digium
I'll guess the TE410P is being loaded first - Try swapping entries in zaptel.conf and zapata.conf Bart - Original Message - From: Bartosz Supczinski [EMAIL PROTECTED] To: Asterisk-Users asterisk-users@lists.digium.com; Asterisk-Dev asterisk-dev@lists.digium.com Sent: Tuesday, February 28, 2006 6:57 PM Subject: [Asterisk-Users] Problem with two cards Digium Hello, I`ve got a problem which I can`t deal with. I own 2 cards - TDM2400P and TE410P. I`ve put them into a HP Proliant DL380 G4 server, compiled the drivers according to the manual. Unfortunetly there are both cards channels are configured in zaptel.conf file the first module (in this file) sends an error. For configuration: span = 1, 1, 0, ccs, hdb3, crc4 fxsks = 21-24 bchan = 25-39, 41-55 dchan = 40 root# modprobe zaptel root# modprobe wctdm24xxp -- ZT_CHANCONFIG failed on channel 25: No such device or address (6) FATAL: Error running install command for wctdm24xxp For configuration: span = 1, 1, 0, ccs, hdb3, crc4 bchan = 1-15, 17-31 dchan = 16 fxsks = 145-148 root# modprobe zaptel root# modprobe wct4xxp -- ZT_CHANCONFIG failed on channel 145: No such device or address (6) FATAL: Error running install command for wct4xxp If the configuration applies only a single card the modules are loaded correctly. The analog card is equiped with one FXO module, which is placed in the last joint. I`ve disabled hyperthreading in my kernel and in BIOS, interrupts are not shared. Besides I`ve put the cards in other slots, switched on and off ACPI and other functions in BIOS as well as in my kernel. Maybe the problem is in the drivers? I`ve attached some information which might be useful. http://www.dir.pl/~supczinskib/logs.tgz -- With best regards Bartosz Supczinski IT Manager DIR Konstytucji 3 Maja 2 86-300 Grudziadz POLAND www.dir.pl t.: +48 (56) 6440100 f.: +48 (56) 6440111 m.: +48 (504) 019040 ___ --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] Answering Service Add-on?
Anybody seen some client/server asterisk add-on script for "live" answering services to provide call handling and message taking from an Operator? Bart ___ --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] Kudzu and Zaptel Cards
Redhat has a 'Hardware Discovery Utility' called Kudzu. When I change cards, kudzu pops up and ask to remove/config the card. Most of the time kudzu has trouble recognizing the Digium Zaptel cards and calls them something wrong, like calling the TDM card a network card. I'm having a devil of a time getting 3 TE410Pcards to come up with all green lights. For example one or two cards full green, and the other has one red and yellow. Swap cards give me some other form of workingness. My questionare: 1) How necessary is Kudzu? 2) Should it ran at all? 3) If I choose ignore or disable kudzu, will it stop the zaptel cards from being detected or working? Any hints will be welcome TIA Bart ___ --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] T1 to T1 dialout problem
I need a few minutes of time to work out a dial out problem. I'm willing to pay for your time. What I have is a system that connect 2 external VMS systems to one of two Telco T1's. Mainly the Telco T1's route inbound calls to one of the two external VM systems depending on the DNIS. This parts works correctly. These are connected using TE410P cards using standard em wink start, D4 T1's. The problem is, one External VM systems needs to be able to dial out to one of two Telco T1's. I tried to setup a context that will allow this but it's not working. I'll get congestion and something about context. What should happen: 1. VMS comes off hook and hears dial tone from asterisk. (Problem 1 - EM don't provide dial tone, maybe could play fake one in Background?) 2. VMS dials the telephone number (10 digits), pauses for 2 second, then send a 4 digit billing account code. (A tone comes from Telco when ready for code) 3. Asterisk then routes the call to a ZAP trunk group 7 for all area codes except 714 or 800 or group 3 for 714 800 - Pauses and then sends 4 digit account code. 4. After dialed party answers, the VM dials additional digits to system that was called and asterisk should ignore these 5. Then the VM terminates the call. Fairly simple - huh? I can get you SSH access and will detail more what the problem is when I hear from you Bart [EMAIL PROTECTED] ___ --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] Motherboard Selection Assistance
I need some help selecting a motherboard. I'm using 3 TE410P's and 1 TDM card in this system. I followed Digium suggestion and purchased a ASUS NRL-LS533 board. For the life of me, I cannot get all 4 cards to work in this environment. Unless the Digium cards are no good, then I assume it's the motherboard compatibility issue. My question for the group: Does anyone have 3 or more TE410P's in one box? If so, which motherboard worked for you? TIA Bart ___ --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] Help with this
I'm trying to get this to work, but it always goes to step 4 - there
something I don't understand about LEN with GotoIf:
exten = _,1,NoOp,${CALLERIDNUM} ; CID as
received
exten = _,2,GotoIf([LEN(${CALLERIDNUM}) = 10]?4:3) ; if CID length =
10 then do nothing
exten = _,3,SetCallerID(${CALLERIDNUM:2}) ; Remove
the first two digits
exten = _,4,NoOp,${CALLERIDNUM} ; CID
after fix
exten = _,5,goto(ext-did,${EXTEN},1)
TIA
Bart
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[Asterisk-Users] Problem with CallerIDNum
I've been jacking with this for a while but don't
understand all thatI'm reading...
The problem is sometimes I get ANI II digits from
the phone company. These will be two digits that prefix ANI- so some
callerid might arrive as only "00" or "007147391234", "00714", "714"
or normal"7147391234".
The prefix digits I get are
"00", "23", "61", "62", "63" - see http://www.nanpa.com/number_resource_info/ani_ii_assignments.htmlfor
info on ANI II digits.
I need ascriptdeals with thisby
normalizes the ANI as received at the beginning of the call.
What I would like to do is ( ANI =
${CALLERIDNUM} ):
if the ANI is a 10 digit number - do noting
if ANI is greater than 10 digits and the first two digits are one of these:
"00", "23", "61", "62", "63" or might be others not found
yet-Then strip the first two digits and make CALLERIDNUM =
corrected ANI
if ANI is less than 10 digit and the first two digits are one of these:
"00", "23", "61", "62", "63" ormight be others not found yet - then
strip these digits and make CALLERIDNUM = corrected ANI
I was wondering if you could show me an example of
how you would do this?
TIA
Bart
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Re: [Asterisk-Users] Zaptel T1 Timing Source
My understanding there should only be one timing source per TE410. You should use a REAL Telco T1 for a timing source. - Otherwise, do not choose any if for example all PBX T1's installed. The settings is only a priority level for asterisk to obtain the source. Example: 1 = use this source first choice, 2 = use this source if source 1 is down, and so on.. Bart - Original Message - From: Waldo Rubinstein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 09, 2005 9:12 AM Subject: [Asterisk-Users] Zaptel T1 Timing Source Hi guys, I have a question about the timing source parameter in zaptel.conf. I have 4 T1s coming into a TE410P. One T1 is with one carrier, who provides timing signal. The other 3 T1s are from a different carrier, all sharing the same timing signal. Based on this, I have in /etc/zaptel.conf something like: span=1,1,0,esf,b8zs em=1-24 span=2,1,0,esf,b8zs em=25-48 span=3,2,0,esf,b8zs em=49-72 span=4,2,0,esf,b8zs em=73-96 What I have done is set the timing source of the first T1 to be the primary source for itself. For the other three T1s, I set the second T1 to be the primary source for the group of 3 and the other two as secondary sources. Is this correct? The reason I ask is because every so often I hear people complaint about call drops. It doesn't happen to everyone, so I don't know if it has anything to do with time source selection and synchronization issues that may be affecting individual channels. After a report of a call drop, I check dmesg and I don't really see any errors. Sometimes I just see ... disable echo cancel... messages on specific channels, but that shouldn't be a reason to drop a call. Am I right? Any ideas? Thanks, Waldo ___ --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] Zaptel T1 Timing Source
I don't understand your question - so I'll tell you what to do: Select one timing source from one T1 that is a Telco connected T1 and make it 1 - example: most reliable T1 Source. If there is two Telco connected T1, select it as source 2 Now if timing source 1 goes down, timing source 2 will take over. Bart - Original Message - From: Waldo Rubinstein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 09, 2005 10:06 AM Subject: Re: [Asterisk-Users] Zaptel T1 Timing Source These are REAL Telco T1s and not connected to a PBX. Am I to assume that even if they are different providers the timing should be the same? That doesn't make a lot of sense to me. Thanks, Waldo On Nov 9, 2005, at 12:34 PM, Bart Fisher wrote: My understanding there should only be one timing source per TE410. You should use a REAL Telco T1 for a timing source. - Otherwise, do not choose any if for example all PBX T1's installed. The settings is only a priority level for asterisk to obtain the source. Example: 1 = use this source first choice, 2 = use this source if source 1 is down, and so on.. Bart - Original Message - From: Waldo Rubinstein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 09, 2005 9:12 AM Subject: [Asterisk-Users] Zaptel T1 Timing Source Hi guys, I have a question about the timing source parameter in zaptel.conf. I have 4 T1s coming into a TE410P. One T1 is with one carrier, who provides timing signal. The other 3 T1s are from a different carrier, all sharing the same timing signal. Based on this, I have in /etc/zaptel.conf something like: span=1,1,0,esf,b8zs em=1-24 span=2,1,0,esf,b8zs em=25-48 span=3,2,0,esf,b8zs em=49-72 span=4,2,0,esf,b8zs em=73-96 What I have done is set the timing source of the first T1 to be the primary source for itself. For the other three T1s, I set the second T1 to be the primary source for the group of 3 and the other two as secondary sources. Is this correct? The reason I ask is because every so often I hear people complaint about call drops. It doesn't happen to everyone, so I don't know if it has anything to do with time source selection and synchronization issues that may be affecting individual channels. After a report of a call drop, I check dmesg and I don't really see any errors. Sometimes I just see ... disable echo cancel... messages on specific channels, but that shouldn't be a reason to drop a call. Am I right? Any ideas? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dial during greeting to access another extension if busy or not available?
Is there some way to allow dialing on top of mailbox greeting during playback to allow caller to move to another extension, and not the operator? Using version 1.2.0 Bart ___ --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: Double DTMF with tdm card
Just wanted to let the group know this problem is fixed (for me). Mark log-on to my system and found a bug in chan_zap.c on Saturday night and made the correction - I believe the change is available for download by now at zaptel 1.0.9.2, or CVS Head. He stated that recent changes unmask the bug and the change will slightly improve TE410P performance Thanks for you help! Bart - Original Message - From: Bart Fisher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 6:20 PM Subject: Re: [Asterisk-Users] Re: Double DTMF with tdm card I just heard back from Mark. I volunteered my system to used for testing. From Mark: Generally, issues which involve Digium hardware should go through technical support, even if it's a newly introduced problem, because they can help narrow down the nature of the failure, what might have changed, etc. If you or a representative of this group want to fill this role instead, I'm happy to work with you, but I need the situation labbed up in an environment where the problem can be demonstrated, where I can remotely log in, and where I can edit, recompile, and test in real time (i.e. not on a production server). If you want to set all this up and contact me with login details and a number where I can see the problem occur, then when it's ready, I can work with you directly. Mark Bart - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 2:41 PM Subject: Re: [Asterisk-Users] Re: Double DTMF with tdm card If in fact it is the exact same issue, then I'd suggest creating a feature request to add disable dtmf detection after answer supervision and post it to the -dev list (which is what Kevin is suggesting now). You will need to be explain the wanted functionality in terms that non-telephone technical folks can understand. I'd suggest a zapata.conf configuration option that is something like ignore-dtmf-after-answersup with a default value of however it works today (=no). Think about that carefully as the option set to =yes will disable dtmf from interacting with your internal * ivr (assuming you have one). What you want is kind of related to a pass-thru connection and not necessarily for a connection terminating within *. There might be other ways to handle your objective. This same issue comes up in other cases where interaction with an external ivr is needed, some airlines automated systems, etc. I honestly believe the exact same thing should apply to iax2 incoming trunks as well. Not so sure about sip trunks. I'd agree with your statement relative to digium support being contacted, but if the boss-man suggests it, there might be an unstated reason for that. If properly worded (and with the supporting documentation that you heard the problem with a T1 analyzer), they might be able to help support the need for some kind of option. This is exactly what is happening... It's bad news... In my case the T1 is connected to a PBX Voice Mail. So, double dialing really messes up thing like when entering a passcode. Where passcode 1234 arrives as 11223344 - no good. This would always be an issue in cases where the call is Tandem thru Asterisk. In fact, I can't see any reason to repeat the digits when the signal is inband and/or Zap Bridged call. - And why was it changed from 1.0.9? Makes no sense. It seems an easy fix, maybe a digit time-out parameter or disable sending after answer supervision has been achieved. Given what you say, Digium Support won't be able to fix without code changes - I don't know what Mark is thinking here. Bart - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 1:17 PM Subject: Re: [Asterisk-Users] Re: Double DTMF with tdm card I might be able to shed a little light on this... Asterisk is constantly listening for dtmf tones on most channels. Its either listening for inband or rfc-out-of-band, depending upon how the attached device is defined and how asterisk def's for that device is defined. For pstn interfaces, the cards don't listen for any dtmf, but rather the zap sutff is listening. If a call is generated from some external source (coming into *), the dtmf will be inband once a channel is answered. For commercial telephone equipment, once a channel is answered, the telephone equipment no longer listens for dtmf (its simply passed inband). Not so with asterisk, and this point has been argued with Mark some time ago; asterisk still listens and trys to handle the dtmf, translating to rfc2833 as it thinks is necessary. So, it sounds like you have an answered T1 call where * is still
[Asterisk-Users] How does Nightly Downloads work at ftp://ftp.digium.com/pub/nightly
Maybe someone can explain how the Digium Nightly works. At ftp://ftp.digium.com/pub/nightly- As far as I can tell, there is a file posted regardless if any changes were made - True? The strange part is the file dates inside the "tar" - I would expect dates that were more current. For example: zaptel-2005-11-05.tar.bz1 the modified date show the latest file date is 04-21-05. Bart ___ --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] Double DTMF with tdm card
My problem is slightly different as there is 2 T1 Ports involved - With a T1 test set I can clearly hear two tones sent quickly with each outside caller press. I assume one of the tones is the actual audio passing thru the connection and the other generated by the T1 card itself.If I make the same test with a TDM400 as input connection and the TE410P port as output connection, there is no double dialing. Same results if an inside extension is used as input connection. It only happens if it's a T1 to T1 Bridge... If it is a regenerated tone from the TE410, it seems there should be some option to stop listening for tone touch after connection has been established? Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: Eric ManxPower Wieling [EMAIL PROTECTED] Cc: Walt Reed [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 6:50 AM Subject: Re: [Asterisk-Users] Double DTMF with tdm card Note this is on external calls to external applications Not Asterisk DTMF detection. It's as though DTMF is distorted when going through a TDM fxs port, or that it's being caught (too late) and then retransmitted. Does * intercept outgoing dtmf? I haven't found good docs that tell exactly what relaxdtmf does. On Thu, Nov 03, 2005 at 08:01:03AM -0600, Eric ManxPower Wieling said: Did you try relaxdtmf=no Walt Reed wrote: Nope - I saw your posts on it though. Very frustrating. I've had to discontinue use of my TDM FXS ports until some solution is found. On Wed, Nov 02, 2005 at 10:18:47AM -0800, Bart Fisher said: Did you ever find a solution for this problem? I have it on latest Beta 2 Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, October 21, 2005 7:26 AM Subject: [Asterisk-Users] Double DTMF with tdm card I have a TDM22B (latest rev), Sipura 2000, and Cisco ATA 186. Running CVS HEAD from about a week ago. Calls made from a SIP device on either the cisco or sipura work fine. Call made from an analog phone hooked up to one of the FXS ports on the TDM22B sound fine, but any DTMF entered after the call is bridged to an outside number (like entering a PIN for a bank or external conference bridge) is frequently doubled. Entering 1234 may be recognized as 112344 for example. I ran fxotune, and played with the rx and tx gains a little, but have been unable to resolve the problem... * has no problem dialing outside numbers. It's just DTMf after the call is bridged between zap channels... Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Double DTMF with tdm card
OK, then... I posted on the Bugs Web Site and markster said: This is a technical support issue. Please pursue through Digium tech support ([EMAIL PROTECTED]) and contact me if you have any issues., Hmmm... So I have written support - still waiting for answer - If I hear anything I'll let you know Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: Bart Fisher [EMAIL PROTECTED] Cc: Walt Reed [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 9:57 AM Subject: Re: [Asterisk-Users] Double DTMF with tdm card Frankly, I think this may be happening to me too. It's still a zap to zap channel problem. On Thu, Nov 03, 2005 at 08:27:59AM -0800, Bart Fisher said: My problem is slightly different as there is 2 T1 Ports involved - With a T1 test set I can clearly hear two tones sent quickly with each outside caller press. I assume one of the tones is the actual audio passing thru the connection and the other generated by the T1 card itself.If I make the same test with a TDM400 as input connection and the TE410P port as output connection, there is no double dialing. Same results if an inside extension is used as input connection. It only happens if it's a T1 to T1 Bridge... If it is a regenerated tone from the TE410, it seems there should be some option to stop listening for tone touch after connection has been established? Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: Eric ManxPower Wieling [EMAIL PROTECTED] Cc: Walt Reed [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 6:50 AM Subject: Re: [Asterisk-Users] Double DTMF with tdm card Note this is on external calls to external applications Not Asterisk DTMF detection. It's as though DTMF is distorted when going through a TDM fxs port, or that it's being caught (too late) and then retransmitted. Does * intercept outgoing dtmf? I haven't found good docs that tell exactly what relaxdtmf does. On Thu, Nov 03, 2005 at 08:01:03AM -0600, Eric ManxPower Wieling said: Did you try relaxdtmf=no Walt Reed wrote: Nope - I saw your posts on it though. Very frustrating. I've had to discontinue use of my TDM FXS ports until some solution is found. On Wed, Nov 02, 2005 at 10:18:47AM -0800, Bart Fisher said: Did you ever find a solution for this problem? I have it on latest Beta 2 Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, October 21, 2005 7:26 AM Subject: [Asterisk-Users] Double DTMF with tdm card I have a TDM22B (latest rev), Sipura 2000, and Cisco ATA 186. Running CVS HEAD from about a week ago. Calls made from a SIP device on either the cisco or sipura work fine. Call made from an analog phone hooked up to one of the FXS ports on the TDM22B sound fine, but any DTMF entered after the call is bridged to an outside number (like entering a PIN for a bank or external conference bridge) is frequently doubled. Entering 1234 may be recognized as 112344 for example. I ran fxotune, and played with the rx and tx gains a little, but have been unable to resolve the problem... * has no problem dialing outside numbers. It's just DTMf after the call is bridged between zap channels... Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
Re: [Asterisk-Users] Re: Double DTMF with tdm card
Well, it seems so... Don't know how - It appeared at Beta upgrade for me Bart - Original Message - From: Steven [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 1:00 PM Subject: [Asterisk-Users] Re: Double DTMF with tdm card SO is he definitively saying that the asterisk software is not involved here? (listening or regenerating tones) -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Bart Fisher [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] OK, then... I posted on the Bugs Web Site and markster said: This is a technical support issue. Please pursue through Digium tech support ([EMAIL PROTECTED]) and contact me if you have any issues., Hmmm... So I have written support - still waiting for answer - If I hear anything I'll let you know Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: Bart Fisher [EMAIL PROTECTED] Cc: Walt Reed [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 9:57 AM Subject: Re: [Asterisk-Users] Double DTMF with tdm card Frankly, I think this may be happening to me too. It's still a zap to zap channel problem. On Thu, Nov 03, 2005 at 08:27:59AM -0800, Bart Fisher said: My problem is slightly different as there is 2 T1 Ports involved - With a T1 test set I can clearly hear two tones sent quickly with each outside caller press. I assume one of the tones is the actual audio passing thru the connection and the other generated by the T1 card itself.If I make the same test with a TDM400 as input connection and the TE410P port as output connection, there is no double dialing. Same results if an inside extension is used as input connection. It only happens if it's a T1 to T1 Bridge... If it is a regenerated tone from the TE410, it seems there should be some option to stop listening for tone touch after connection has been established? Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: Eric ManxPower Wieling [EMAIL PROTECTED] Cc: Walt Reed [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 6:50 AM Subject: Re: [Asterisk-Users] Double DTMF with tdm card Note this is on external calls to external applications Not Asterisk DTMF detection. It's as though DTMF is distorted when going through a TDM fxs port, or that it's being caught (too late) and then retransmitted. Does * intercept outgoing dtmf? I haven't found good docs that tell exactly what relaxdtmf does. On Thu, Nov 03, 2005 at 08:01:03AM -0600, Eric ManxPower Wieling said: Did you try relaxdtmf=no Walt Reed wrote: Nope - I saw your posts on it though. Very frustrating. I've had to discontinue use of my TDM FXS ports until some solution is found. On Wed, Nov 02, 2005 at 10:18:47AM -0800, Bart Fisher said: Did you ever find a solution for this problem? I have it on latest Beta 2 Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, October 21, 2005 7:26 AM Subject: [Asterisk-Users] Double DTMF with tdm card I have a TDM22B (latest rev), Sipura 2000, and Cisco ATA 186. Running CVS HEAD from about a week ago. Calls made from a SIP device on either the cisco or sipura work fine. Call made from an analog phone hooked up to one of the FXS ports on the TDM22B sound fine, but any DTMF entered after the call is bridged to an outside number (like entering a PIN for a bank or external conference bridge) is frequently doubled. Entering 1234 may be recognized as 112344 for example. I ran fxotune, and played with the rx and tx gains a little, but have been unable to resolve the problem... * has no problem dialing outside numbers. It's just DTMf after the call is bridged between zap channels... Any ideas? ___ --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
Re: [Asterisk-Users] Re: Double DTMF with tdm card
This is exactly what is happening... It's bad news... In my case the T1 is connected to a PBX Voice Mail. So, double dialing really messes up thing like when entering a passcode. Where passcode 1234 arrives as 11223344 - no good. This would always be an issue in cases where the call is Tandem thru Asterisk. In fact, I can't see any reason to repeat the digits when the signal is inband and/or Zap Bridged call. - And why was it changed from 1.0.9? Makes no sense. It seems an easy fix, maybe a digit time-out parameter or disable sending after answer supervision has been achieved. Given what you say, Digium Support won't be able to fix without code changes - I don't know what Mark is thinking here. Bart - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 1:17 PM Subject: Re: [Asterisk-Users] Re: Double DTMF with tdm card I might be able to shed a little light on this... Asterisk is constantly listening for dtmf tones on most channels. Its either listening for inband or rfc-out-of-band, depending upon how the attached device is defined and how asterisk def's for that device is defined. For pstn interfaces, the cards don't listen for any dtmf, but rather the zap sutff is listening. If a call is generated from some external source (coming into *), the dtmf will be inband once a channel is answered. For commercial telephone equipment, once a channel is answered, the telephone equipment no longer listens for dtmf (its simply passed inband). Not so with asterisk, and this point has been argued with Mark some time ago; asterisk still listens and trys to handle the dtmf, translating to rfc2833 as it thinks is necessary. So, it sounds like you have an answered T1 call where * is still trying to handle dtmf (regenerating it), AND, the dtmf is being passsed inband as well. If that is what you are seeing, then its the same design problem that was argued with Mark, and he's insistent the current operation is correct. I disagree, but I'm only one person. SO is he definitively saying that the asterisk software is not involved here? (listening or regenerating tones) -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Bart Fisher [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] OK, then... I posted on the Bugs Web Site and markster said: This is a technical support issue. Please pursue through Digium tech support ([EMAIL PROTECTED]) and contact me if you have any issues., Hmmm... So I have written support - still waiting for answer - If I hear anything I'll let you know Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: Bart Fisher [EMAIL PROTECTED] Cc: Walt Reed [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 9:57 AM Subject: Re: [Asterisk-Users] Double DTMF with tdm card Frankly, I think this may be happening to me too. It's still a zap to zap channel problem. On Thu, Nov 03, 2005 at 08:27:59AM -0800, Bart Fisher said: My problem is slightly different as there is 2 T1 Ports involved - With a T1 test set I can clearly hear two tones sent quickly with each outside caller press. I assume one of the tones is the actual audio passing thru the connection and the other generated by the T1 card itself.If I make the same test with a TDM400 as input connection and the TE410P port as output connection, there is no double dialing. Same results if an inside extension is used as input connection. It only happens if it's a T1 to T1 Bridge... If it is a regenerated tone from the TE410, it seems there should be some option to stop listening for tone touch after connection has been established? Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: Eric ManxPower Wieling [EMAIL PROTECTED] Cc: Walt Reed [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 6:50 AM Subject: Re: [Asterisk-Users] Double DTMF with tdm card Note this is on external calls to external applications Not Asterisk DTMF detection. It's as though DTMF is distorted when going through a TDM fxs port, or that it's being caught (too late) and then retransmitted. Does * intercept outgoing dtmf? I haven't found good docs that tell exactly what relaxdtmf does. On Thu, Nov 03, 2005 at 08:01:03AM -0600, Eric ManxPower Wieling said: Did you try relaxdtmf=no Walt Reed wrote: Nope - I saw your posts on it though. Very frustrating. I've had to discontinue use of my TDM FXS
Re: [Asterisk-Users] Re: Double DTMF with tdm card
. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Bart Fisher [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] OK, then... I posted on the Bugs Web Site and markster said: This is a technical support issue. Please pursue through Digium tech support ([EMAIL PROTECTED]) and contact me if you have any issues., Hmmm... So I have written support - still waiting for answer - If I hear anything I'll let you know Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: Bart Fisher [EMAIL PROTECTED] Cc: Walt Reed [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 9:57 AM Subject: Re: [Asterisk-Users] Double DTMF with tdm card Frankly, I think this may be happening to me too. It's still a zap to zap channel problem. On Thu, Nov 03, 2005 at 08:27:59AM -0800, Bart Fisher said: My problem is slightly different as there is 2 T1 Ports involved - With a T1 test set I can clearly hear two tones sent quickly with each outside caller press. I assume one of the tones is the actual audio passing thru the connection and the other generated by the T1 card itself. If I make the same test with a TDM400 as input connection and the TE410P port as output connection, there is no double dialing. Same results if an inside extension is used as input connection. It only happens if it's a T1 to T1 Bridge... If it is a regenerated tone from the TE410, it seems there should be some option to stop listening for tone touch after connection has been established? Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: Eric ManxPower Wieling [EMAIL PROTECTED] Cc: Walt Reed [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 6:50 AM Subject: Re: [Asterisk-Users] Double DTMF with tdm card Note this is on external calls to external applications Not Asterisk DTMF detection. It's as though DTMF is distorted when going through a TDM fxs port, or that it's being caught (too late) and then retransmitted. Does * intercept outgoing dtmf? I haven't found good docs that tell exactly what relaxdtmf does. On Thu, Nov 03, 2005 at 08:01:03AM -0600, Eric ManxPower Wieling said: Did you try relaxdtmf=no Walt Reed wrote: Nope - I saw your posts on it though. Very frustrating. I've had to discontinue use of my TDM FXS ports until some solution is found. On Wed, Nov 02, 2005 at 10:18:47AM -0800, Bart Fisher said: Did you ever find a solution for this problem? I have it on latest Beta 2 Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, October 21, 2005 7:26 AM Subject: [Asterisk-Users] Double DTMF with tdm card I have a TDM22B (latest rev), Sipura 2000, and Cisco ATA 186. Running CVS HEAD from about a week ago. Calls made from a SIP device on either the cisco or sipura work fine. Call made from an analog phone hooked up to one of the FXS ports on the TDM22B sound fine, but any DTMF entered after the call is bridged to an outside number (like entering a PIN for a bank or external conference bridge) is frequently doubled. Entering 1234 may be recognized as 112344 for example. I ran fxotune, and played with the rx and tx gains a little, but have been unable to resolve the problem... * has no problem dialing outside numbers. It's just DTMf after the call is bridged between zap channels... Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http
Re: [Asterisk-Users] Double DTMF with tdm card
Did you ever find a solution for this problem? I have it on latest Beta 2 Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, October 21, 2005 7:26 AM Subject: [Asterisk-Users] Double DTMF with tdm card I have a TDM22B (latest rev), Sipura 2000, and Cisco ATA 186. Running CVS HEAD from about a week ago. Calls made from a SIP device on either the cisco or sipura work fine. Call made from an analog phone hooked up to one of the FXS ports on the TDM22B sound fine, but any DTMF entered after the call is bridged to an outside number (like entering a PIN for a bank or external conference bridge) is frequently doubled. Entering 1234 may be recognized as 112344 for example. I ran fxotune, and played with the rx and tx gains a little, but have been unable to resolve the problem... * has no problem dialing outside numbers. It's just DTMf after the call is bridged between zap channels... Any ideas? ___ --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] Double DTMF sent on T1 to T1 Native Bridge
Bump - I'm stuck until I can find a solutions Please help - I'll try anything! Bart - Original Message - From: Bart Fisher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, November 01, 2005 5:37 PM Subject: Re: [Asterisk-Users] Double DTMF sent on T1 to T1 Native Bridge An update: If I dial with an internal single FXS phone or inbound to TDM400 (FXO) it works correctly. It appears that the Telco T1 is regenerating the DTMF as received at the same time the audio DTMF is past though the bridged connection. So, the effect is I hear two tones on Legacy PBX connection. - Make sense? This is a new problem since Asterisk 1.0.9, so I guess it's a bug? Seems there should be some way to make the Telco T1 stop listening and sending DTMF after connection Bart - Original Message - From: Bart Fisher [EMAIL PROTECTED] To: Asterisk-Users@lists.digium.com Sent: Tuesday, November 01, 2005 1:41 PM Subject: [Asterisk-Users] Double DTMF sent on T1 to T1 Native Bridge I have asterisk sitting in the middle with Telco on one side and Legacy PBX on the other using two T1 ports on a TE410P. I also have the latest Beta 2 installed. My problem is after a call is connected (port to port T1) and the outside user presses a touch tone, asterisk is repeating the digit. So if I press 1234 the PBX hears 11223344 - really messes up accessing the voice mail on PBX. If I dial into PBX from an internal phone it works correctly. This is a new problem since my upgrade from Asterisk 1.0.9, so I guess there is some keyword to disable this feature zapata? Bart ___ --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
[Asterisk-Users] Double DTMF sent on T1 to T1 Native Bridge
I have asterisk sitting in the middle with Telco on one side and Legacy PBX on the other using two T1 ports on a TE410P. I also have the latest Beta 2 installed. My problem is after a call is connected (port to port T1) and the outside user presses a touch tone, asterisk is repeating the digit. So if I press 1234 the PBX hears 11223344 - really messes up accessing the voice mail on PBX. If I dial into PBX from an internal phone it works correctly. This is a new problem since my upgrade from Asterisk 1.0.9, so I guess there is some keyword to disable this feature zapata? Bart ___ --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] Double DTMF sent on T1 to T1 Native Bridge
An update: If I dial with an internal single FXS phone or inbound to TDM400 (FXO) it works correctly. It appears that the Telco T1 is regenerating the DTMF as received at the same time the audio DTMF is past though the bridged connection. So, the effect is I hear two tones on Legacy PBX connection. - Make sense? This is a new problem since Asterisk 1.0.9, so I guess it's a bug? Seems there should be some way to make the Telco T1 stop listening and sending DTMF after connection Bart - Original Message - From: Bart Fisher [EMAIL PROTECTED] To: Asterisk-Users@lists.digium.com Sent: Tuesday, November 01, 2005 1:41 PM Subject: [Asterisk-Users] Double DTMF sent on T1 to T1 Native Bridge I have asterisk sitting in the middle with Telco on one side and Legacy PBX on the other using two T1 ports on a TE410P. I also have the latest Beta 2 installed. My problem is after a call is connected (port to port T1) and the outside user presses a touch tone, asterisk is repeating the digit. So if I press 1234 the PBX hears 11223344 - really messes up accessing the voice mail on PBX. If I dial into PBX from an internal phone it works correctly. This is a new problem since my upgrade from Asterisk 1.0.9, so I guess there is some keyword to disable this feature zapata? Bart ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Zaptel Versions Command?
Is there a command line for discovery of Asterisk and Zaptel Versions? Bart ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Zaptel Versions Command?
Thanks, but what I was really hoping for was something that could be used in
a script to report current revisions... me sad
Bart
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 31, 2005 12:33 PM
Subject: Re: [Asterisk-Users] Asterisk and Zaptel Versions Command?
Yeah, show versions in the CLI will give you the version of your asterisk
build
Also you can do the following in the CLI:
show version files filename
where filename is a valid file name.
As always in Linux you can press TAB to get a list of available
commands in the CLI, for example you can type:
show version files {TAB}
that will give you a list of all the files you can then type the file
you want. Or you could narrow it down like this:
show version files chan{TAB}
that will give you a list of all the avaiable files that start with
chan, you could also do just {TAB} to get a list of all the commands.
To get help you could type help command.
Hope this helps.
On 10/31/05, Bart Fisher [EMAIL PROTECTED] wrote:
Is there a command line for discovery of Asterisk and Zaptel Versions?
Bart
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[Asterisk-Users] I give up - Help with TE410P
I'm trying to install two TE410P's in one box. Would like to get 3 total. I can always get one card to work. If I install only one card, I will get green lights on all ports when loop back plugs installed - everything is perfect... If I install 2 cards, I'll get yellow alarm on span 2 and 6 and 7. Flashing Red alarm on span 1,3,4,5 and 8. There is no error messages that I can find. What is the correct procedure for installing these cards? Can you give me a step-by-step on how to install these cards?I've been working on this for a week and getting frustrated.\ TIA Bart Some info (not sure what else might be needed): # cat /proc/interrupts CPU0 0:3593335 XT-PIC timer 1:518 XT-PIC i8042 2: 0 XT-PIC cascade 5: 30 XT-PIC aic7xxx 7: 37562 XT-PIC eth0 8: 1 XT-PIC rtc 9:3257067 XT-PIC acpi, wctdm 10:3257002 XT-PIC wct4xxp 11:3260152 XT-PIC wct4xxp 14: 13296 XT-PIC ide0 NMI: 0 ERR: 0 # lspci -v 00:00.0 Host bridge: Broadcom GCNB-LE Host Bridge (rev 32) Flags: fast devsel 00:00.1 Host bridge: Broadcom GCNB-LE Host Bridge Flags: fast devsel 00:02.0 SCSI storage controller: Adaptec AIC-7892P U160/m (rev 02) Subsystem: Adaptec AIC-7892P U160/m Flags: bus master, 66Mhz, medium devsel, latency 32, IRQ 5 BIST result: 00 I/O ports at d800 [disabled] [size=256] Memory at fe00 (64-bit, non-prefetchable) [size=4K] Expansion ROM at febe [disabled] [size=128K] Capabilities: [dc] Power Management version 2 00:03.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5702X Gigabit Ethernet (rev 02) Subsystem: ASUSTeK Computer Inc.: Unknown device 80a9 Flags: bus master, 66Mhz, medium devsel, latency 64, IRQ 7 Memory at fd80 (64-bit, non-prefetchable) [size=64K] [virtual] Expansion ROM at febd [disabled] [size=64K] Capabilities: [40] PCI-X non-bridge device. Capabilities: [48] Power Management version 2 Capabilities: [50] Vital Product Data Capabilities: [58] Message Signalled Interrupts: 64bit+ Queue=0/3 Enable- 00:04.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b100:0001 Flags: bus master, medium devsel, latency 32, IRQ 9 I/O ports at d400 [size=256] Memory at fd00 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 00:05.0 Communication controller: Unknown device d161:0410 (rev 02) Flags: bus master, medium devsel, latency 32, IRQ 10 Memory at fc80 (32-bit, non-prefetchable) [size=128] 00:06.0 Communication controller: Unknown device d161:0410 (rev 02) Flags: bus master, medium devsel, latency 32, IRQ 11 Memory at fc00 (32-bit, non-prefetchable) [size=128] 00:09.0 VGA compatible controller: ATI Technologies Inc Rage XL (rev 27) (prog-if 00 [VGA]) Subsystem: ATI Technologies Inc Rage XL Flags: bus master, stepping, medium devsel, latency 32, IRQ 10 Memory at fb00 (32-bit, non-prefetchable) [size=16M] I/O ports at d000 [size=256] Memory at fa80 (32-bit, non-prefetchable) [size=4K] Expansion ROM at feba [disabled] [size=128K] Capabilities: [5c] Power Management version 2 00:0f.0 ISA bridge: Broadcom CSB6 South Bridge (rev a0) Subsystem: Broadcom: Unknown device 0201 Flags: bus master, medium devsel, latency 32 00:0f.1 IDE interface: Broadcom CSB6 RAID/IDE Controller (rev a0) (prog-if 8a [Master SecP PriP]) Subsystem: Broadcom: Unknown device 0212 Flags: bus master, medium devsel, latency 64 I/O ports at ignored I/O ports at ignored I/O ports at ignored I/O ports at ignored I/O ports at 8800 [size=16] 00:0f.3 Host bridge: Broadcom GCLE-2 Host Bridge Subsystem: Broadcom: Unknown device 0230 Flags: bus master, medium devsel, latency 0 # uname -a Linux asterisk1.local 2.6.9-22.0.1.EL #1 Thu Oct 27 12:26:11 CDT 2005 i686 i686 i386 GNU/Linux # cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 2 model name : Intel(R) Pentium(R) 4 CPU 2.80GHz stepping: 9 cpu MHz : 2799.826 cache size : 512 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe cid xtpr bogomips: 5521.40 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list
Re: [Asterisk-Users] I give up - Help with TE410P
Yep - that was easy part :) and these are T1 (D4, AMI, SF, and EM Wink) BTW Bart - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, October 29, 2005 3:09 PM Subject: Re: [Asterisk-Users] I give up - Help with TE410P On Saturday 29 October 2005 18:06, Bart Fisher wrote: I'm trying to install two TE410P's in one box. Would like to get 3 total. I can always get one card to work. You are adjusting the 'ident' rotary switch on the others, right? -A. ___ --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] I give up - Help with TE410P
Well, have you ever tried their support? They assume we are all dummies... A bunch of canned email messages to remind you to plug in the power cable. :) Ok, in a disparate act (and this might help someone body someday) I removed all the Digium card and emptied the zap*.conf files from the box and rebooted. I allowed Linux to remove the missing cards - this of course installs ztdummy. Next I shutdown and added all the cards at one time. - Booted and let Linux discover cards and allowed configuration. Copied back my zap*.conf files rebooted. This time it comes up 6 spans with green lights and 2 on first card with flashing red. I shutdown, and swap the two TE410P. Rebooted - all light green now. Since it's working, I'm done - but only go to show you these cards are flaky. Bart - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, October 29, 2005 3:35 PM Subject: Re: [Asterisk-Users] I give up - Help with TE410P On Saturday 29 October 2005 18:19, Bart Fisher wrote: Yep - that was easy part :) and these are T1 (D4, AMI, SF, and EM Wink) BTW Ok, well I'll go for the obvious question: have you contacted Digium technical assistance? You have paid for support within the price of the card. -A. ___ --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] I give up - Help with TE410P
Yep, it CentOS 4.0 (RH) - Kudzu - also seems to be the root of my problem. I later rebooted and now back to some ports working again. I'm using a Loop-Back plug to test with - no real T1 attached until I can fix this. Swapping card does not seem to follow issues. Maybe I'll give support another :) Bart - Original Message - From: Jason Walker [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, October 29, 2005 5:09 PM Subject: RE: [Asterisk-Users] I give up - Help with TE410P My 2 cents: If you are running kudzu on RH or FC, new and remove hardware should be detected...in most cases. I assume other distros have something similar...? If 2 of 8 T1s are not coming up - sounds like you may have a wiring issue. Can you swap cables from a bad circuit to a good circuit? Are all of the circuits the same configuration from the carrier? As far as support, Digium's email support has ALWAYS been helpful to me - from basic questions to systematic issues. They have always been helpful and responsive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Saturday, October 29, 2005 4:50 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] I give up - Help with TE410P On Saturday 29 October 2005 19:30, Bart Fisher wrote: Well, have you ever tried their support? They assume we are all dummies... A bunch of canned email messages to remind you to plug in the power cable. Actually my support from them has been great... Ok, in a disparate act (and this might help someone body someday) I removed all the Digium card and emptied the zap*.conf files from the box and rebooted. I allowed Linux to remove the missing cards - this of course installs ztdummy. allowed linux to remove the missing cards ?? what distro are you using? Next I shutdown and added all the cards at one time. - Booted and let Linux discover cards and allowed configuration. Copied back my zap*.conf files rebooted. This time it comes up 6 spans with green lights and 2 on first card with flashing red. I shutdown, and swap the two TE410P. Rebooted - all light green now. Again, what distro, what version of asterisk and whatnot? Is this [EMAIL PROTECTED] Since it's working, I'm done - but only go to show you these cards are flaky. It sounds like your system is what's flaky here... Linux doesn't need to remove the cards... Definitely something nonstandard from my point of view. I am glad it's working for you though. -A. ___ --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
[Asterisk-Users] EM to EM Dialing - TE410P
I have a TE410P with two real Telco T1's and the other 2 portsterminate into an in-house voice mail/IVR system. Calls arrive from Telco are routed to the appropriate in-house system based on the DID Digits. This part works perfectly. Now I what to allow the in-house VMS to dial though asterisk to the Telco T1's. The VMS complains there is no dial tone on its T1 and drops the attempt. I could disable dial tone detection, but rather not. Instead, I would like asterisk to provide a dial tone after the trunk has been seized. These T1's are all the same - D4, AMI, SF EM Wink Start. Any Ideas? Bart ___ 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
