[asterisk-users] Best way to get 60+ analogue extensions.
Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally fits and copes well with multiple cards, and the choice of Digium vs Sangoma or something else. I can see the Digium AEX2400 with 24 lines, physically they are all very deep, if I had 3 of these in a server it would seem straight forward assuming the motherboard doesn't haven't anything get in the way Equally the Digium TDM2400P supports 24 lines and physically requires similar space The Sangoma A400 provides 24 ports but uses two slots, having 3 of these in a server looks like I need to pick the server carefully. I may need an ISDN PRA inbound but am working hard to have the inbound lines via SIP, but if I do that means at least 4 slots on this plan. I am just interested in any recommendations for server hardware and card combinations that are currently in use. Also if anyone has provided call data out to the RMS system ( http://www.rms-global.com/Our-Products/RMS-Hotel/ ) I would be keen to hear how it worked. Thanks very much Cheers Duncan ___ -- 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] automatic call bridging when destination is available feature
2009/3/14 Vieri rentor...@yahoo.com --- On Sat, 3/14/09, Olivier oza-4...@myamail.com wrote: If I understand correctly, you're suggesting to implement the h priority instructions (or a hangup macro) to: 1) run a deadagi or a system() script to see if someone has left a request (eg. in astdb) to call-back-when-avail 2) create a call file with, say: Channel: SIP/102 and Context: internal Extension: 101 Priority: 1 Is that what you would do, more or less? yes, that's exactly what I meant. As I'm not too fluent in AEL scripting, I'm afraid I can't easily be more helpful but you eactly got what I suggested. What would think of that ? Would it fit for you ? Thanks for the feedback but there are some things that would not work right: 1) when 102 hangs up, the call file would try to reach 101 yes but it can be done the other way if think 102 is very likely to be busy ... but if 101 is busy, it may retry later (MaxRetries in call file) You can also set MaxRetries to 0 and strictly rely on the fact that extension 102 hangups ... and finally reach it and send 101 to the context where it would then Dial(SIP/102). However, inthe meantime, 102 could have received or made another call and be busy again (and that would irritate the 101 user). So ? When Telcos implement this service they limit retries in a 30mn time frame so that you're not called back at 2am, for intance. 2) 102 may not necessarily be on the phone when 101 first tries to contact. The 102 extension could simply be off-line/Unavailable/unregistered, so the 102 user would never hang up (thus the hangup macro would never be executed). Anyway, I was hoping Asterisk already had a trick up its sleeve for this ;-) but I guess I'll have to implement a custom event listener and try to bridge extensions. Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best way to get 60+ analogue extensions.
Duncan Turnbull wrote: Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally fits and copes well with multiple cards, and the choice of Digium vs Sangoma or something else. You have several options here, however due to the power requirements, I wouldn't recommend you use either the Sangoma or Digium analogue cards here - providing ring voltage to that many extensions is likely to over-tax the power supply in the server. I'd either be looking at three channel banks (3 24 channel channel banks would give you a total of 72 analogue channels) or two Xorcom Astribanks which would likewise give you up to 64 channels. The Astribanks are probably a cheaper way to go since they connect to your server via USB rather than T1/E1 ports. However, I haven't had any experience with multiple Astribanks connected to the same server, so there may be issues there that I'm not aware of. Channel banks are certainly the proven and reliable technology, but will be significantly more expensive since they connect to your Asterisk server via T1/E1 links. I may need an ISDN PRA inbound but am working hard to have the inbound lines via SIP, but if I do that means at least 4 slots on this plan. You'd need to be very sure of the bandwidth and quality of connection to your VoIP provider to go with SIP for more than half a dozen channels. This kind of connection can easily be far more expensive than a traditional T1/E1 line, so I wouldn't be pushing so hard for SIP. If you were to use channel banks, you would most likely end up with a four port T1/E1 card and would only be using three of those channels, leaving a spare one for an incoming T1/E1 line. If you were to use Astribanks, you would have plenty of space in the server to include a T1/E1 card. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best way to get 60+ analogue extensions.
Hi, Xorcom make what you are looking for. Steve On 3/15/09, Duncan Turnbull dun...@e-simple.co.nz wrote: Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally fits and copes well with multiple cards, and the choice of Digium vs Sangoma or something else. I can see the Digium AEX2400 with 24 lines, physically they are all very deep, if I had 3 of these in a server it would seem straight forward assuming the motherboard doesn't haven't anything get in the way Equally the Digium TDM2400P supports 24 lines and physically requires similar space The Sangoma A400 provides 24 ports but uses two slots, having 3 of these in a server looks like I need to pick the server carefully. I may need an ISDN PRA inbound but am working hard to have the inbound lines via SIP, but if I do that means at least 4 slots on this plan. I am just interested in any recommendations for server hardware and card combinations that are currently in use. Also if anyone has provided call data out to the RMS system ( http://www.rms-global.com/Our-Products/RMS-Hotel/ ) I would be keen to hear how it worked. Thanks very much Cheers Duncan ___ -- 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 -- Sent from my mobile device ___ -- 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] X-Asterisk-HangupCause - how to disable this?
Hi, Is there any way to tell Asterisk not to generate additional headers like: X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 I can't find any relevant option in sip.conf file :-( Thanks for help. Chris ___ -- 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] X-Asterisk-HangupCause - how to disable this?
2009/3/15 Chris Maciejewski ch...@wima.co.uk Hi, Is there any way to tell Asterisk not to generate additional headers like: X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 I can't find any relevant option in sip.conf file :-( For curiosity's sake, what are the troubling consequences of having those headers included ? Thanks for help. Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using PRI_CAUSE to change SIP INVITE rejection response code
I am trying to change the SIP response to an incoming call and according to the docs and a quick scan of the chan_sip.c code one is supposed to be able to set the PRI_CAUSE variable and invoke Hangup but regardless of the value in PRI_CAUSE the SIP rejection is always 603. What is wrong here? Asterisk 1.4.23.1 exten = _NXXNXX,n,Set(PRI_CAUSE=17) exten = _NXXNXX,n,Hangup -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium B410P: misdn v1 or misdn v2 or dahdi + asterisk 1.6 ?
2009/3/11 Tzafrir Cohen tzafrir.co...@xorcom.com On Wed, Mar 11, 2009 at 02:56:58PM -0500, Kevin P. Fleming wrote: Vieri wrote: - use the latest release of misdn v1 - upgrade to the latest stable kernel and use the built-in misdn v2 There is no support for mISDN v2 in Asterisk to my knowledge. It is available in a separate, out of tree module that is developed alongside mISDNv2 (chan_lcr). This is interesting to know as I've never heard of Linux Call Router before (http://www.linux-call-router.de/). LCR targets a different hardware cards set (one port Cologne PCI and USB) from the one Dahdi (and its enhancements) targets, B410P belonging to both. IIRC, though, hfcmulti is not yet supported by mISDNv2 . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Dahdi Error
Got this in the log, with no calls active. Is it a problem with my isdn line, or * ? [Mar 15 11:36:18] ERROR[29161]: chan_dahdi.c:8735 dahdi_pri_error: ACK received for '0' outside of window of '39' to '40', restarting [Mar 15 11:36:18] == Primary D-Channel on span 1 down [Mar 15 11:36:18] WARNING[29161]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] Using PRI_CAUSE to change SIP INVITE rejection response code
On 15 Mar 2009, at 11:30, George Pajari wrote: I am trying to change the SIP response to an incoming call and according to the docs and a quick scan of the chan_sip.c code one is supposed to be able to set the PRI_CAUSE variable and invoke Hangup but regardless of the value in PRI_CAUSE the SIP rejection is always 603. What is wrong here? Asterisk 1.4.23.1 exten = _NXXNXX,n,Set(PRI_CAUSE=17) exten = _NXXNXX,n,Hangup Hangup(17) ___ -- 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] Too many notify events causing Asterisk crash?
Hi, We've implemented a 'page-all' function for some of our customers, and we've noticed that on occasion the page-all will cause asterisk to crash (safe_asterisk then restarts it again). The particular customer has about 20 phones, and also has 5 Linksys 932 to monitor the state of these extensions. I'm not sure whether it is the page-all that causes the crash, or the subsequent NOTIFY storm to all of the 932s that are monitoring those extensions (since the page-all causes them all to go to In-Use and then Idle). Here's a log snippet of right before the crash (previous crashes look very similar). Also, this is a production box, so there's no way at this time I can recompile with debugging symbols. The Asterisk version is 1.4.18.1. Thanks. [Mar 14 12:56:06] VERBOSE[30141] logger.c: == Spawn extension (ext-paging, PAGExx6295, 8) exited non-zero on 'Local/pagexx6...@ext-paging-ef0f,2' [Mar 14 12:56:08] VERBOSE[30193] logger.c: == Connect attempt from '127.0.0.1' unable to authenticate [Mar 14 12:56:13] NOTICE[8327] chan_sip.c: Registration from 'sip:1...@warp2biz.com' failed for '71.119.123.229' - Wrong password [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6297[ext-local] new state Idle for Notify User xx6324 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6297[ext-local] new state Idle for Notify User xx6293 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6297[ext-local] new state Idle for Notify User xx6311 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6297[ext-local] new state Idle for Notify User xx6315 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6297[ext-local] new state Idle for Notify User xx6297 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6329[ext-local] new state Idle for Notify User xx6324 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6329[ext-local] new state Idle for Notify User xx6293 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6329[ext-local] new state Idle for Notify User xx6311 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6329[ext-local] new state Idle for Notify User xx6315 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6329[ext-local] new state Idle for Notify User xx6297 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6295[ext-local] new state Idle for Notify User xx6324 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6295[ext-local] new state Idle for Notify User xx6293 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6295[ext-local] new state Idle for Notify User xx6311 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6295[ext-local] new state Idle for Notify User xx6315 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6295[ext-local] new state Idle for Notify User xx6297 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6317[ext-local] new state Idle for Notify User xx6324 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6317[ext-local] new state Idle for Notify User xx6293 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6317[ext-local] new state Idle for Notify User xx6311 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6317[ext-local] new state Idle for Notify User xx6315 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6317[ext-local] new state Idle for Notify User xx6297 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6328[ext-local] new state Idle for Notify User xx6324 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6328[ext-local] new state Idle for Notify User xx6293 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6328[ext-local] new state Idle for Notify User xx6315 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6328[ext-local] new state Idle for Notify User xx6311 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6328[ext-local] new state Idle for Notify User xx6297 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6314[ext-local] new state Idle for Notify User xx6324 [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6314[ext-local] new state Idle for Notify User xx6293 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6314[ext-local] new state Idle for Notify User xx6311 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed xx6314[ext-local] new state Idle for Notify User xx6315 (queued) [Mar 14 12:56:18] VERBOSE[8301] logger.c: Extension Changed
Re: [asterisk-users] Best way to get 60+ analogue extensions.
On Sun, 15 Mar 2009, Duncan Turnbull wrote: Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally fits and copes well with multiple cards, and the choice of Digium vs Sangoma or something else. I can see the Digium AEX2400 with 24 lines, physically they are all very deep, if I had 3 of these in a server it would seem straight forward assuming the motherboard doesn't haven't anything get in the way Equally the Digium TDM2400P supports 24 lines and physically requires similar space The Sangoma A400 provides 24 ports but uses two slots, having 3 of these in a server looks like I need to pick the server carefully. I may need an ISDN PRA inbound but am working hard to have the inbound lines via SIP, but if I do that means at least 4 slots on this plan. I am just interested in any recommendations for server hardware and card combinations that are currently in use. Also if anyone has provided call data out to the RMS system ( http://www.rms-global.com/Our-Products/RMS-Hotel/ ) I would be keen to hear how it worked. I have done several hotels using Audicodes MP-124 gateways. No need for expensive T1 interfaces for channel banks - these boxes do SIP over ethernet. They run about $1300 for 24 ports, which is very cost effective, and I can vouch for their stability and voice quality over decent network links... We are actually working on a property managment system for hotels that we hope to release at the end of the summer. It may go open source. We have found that many hotels are used to the text interface / dumb terminals prevalent in the industry, and don't want to give up that speed for web based solutions, which is all you see in the marketplace these days. Our system actually runs on the asterisk server, provides both a web (mainly for reports) and text (ncurses) interface to give them the best of both worlds, in addition to tightly integrating the phone system with the operation of the property. Sorry to go off topic there ;) j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X-Asterisk-HangupCause - how to disable this?
For curiosity's sake, what are the troubling consequences of having those headers included ? My PSTN termination provider sometimes replies with SIP/2.0 513 Message too big to my BYEs with additional headers included. Just wanted to check if this is the reason, or maybe it is related to something else. 2009/3/15 Olivier oza-4...@myamail.com: 2009/3/15 Chris Maciejewski ch...@wima.co.uk Hi, Is there any way to tell Asterisk not to generate additional headers like: X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 I can't find any relevant option in sip.conf file :-( For curiosity's sake, what are the troubling consequences of having those headers included ? Thanks for help. Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No hardware timing source found in /proc/dahdi
Hello all, Ok it is Sunday afternoon and I am going crazy. I have been running in circles so long that I can't think straight. As an example, I sent this message to the wrong address the first try, AAAGGH. I have Asterisk 1.6.0.6 and DAHDI Tools Version - 2.1.0.2, DAHDI Version: 2.1.0.4, OpenSuSE 10.3 x86_64, tdm422 at the end of installing dahdi-linux and dahdi-tools I get: install -D dahdi.init /etc/init.d/dahdi /sbin/chkconfig --add dahdi dahdi 0:off 1:off 2:on 3:on 4:on 5:on 6:off DAHDI has been configured. If you have any DAHDI hardware it is now recommended you edit /etc/dahdi/modules in order to load support for only the DAHDI hardware installed in this system. By default support for all DAHDI hardware is loaded at DAHDI start. I think that the DAHDI hardware you have on your system is: pci::03:08.0 wctdm- e159:0001 Wildcard TDM400P REV E/F so it is seeing the card /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=3,4 echocanceller=mg2,1-4 channels=1-4 /etc/dahdi/init.conf: MODULES=$MODULES wctdm /etc/dahdi/modules # Digium TDM400P: up to 4 analog ports wctdm # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: modprobe wctdm No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: /usr/sbin/dahdi_cfg # /usr/sbin/dahdi_cfg - DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 4 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) *CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO DAHDI_DUMMY/1 (source: RTC) 1UNCONFI 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) I am really hoping that I am just missing something stupid. Anyone have any suggestions? TIA JohnM ___ -- 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] 428 Loop Detected
Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqb...@improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106 dtmfmode=rfc2833 iqb...@improvise:/etc/asterisk$ cat extensions.conf [default] exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,Playback(tt-monkeys) exten = s,4,Hangup [from-internal] include = default [phone1] [from-pstn] ;include = default exten = s,1,Dial(SIP/ph...@phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup [line1] So my home land line is going to the FXO port and my home phone is hanging off of FXS port. Here are the contexts for my fxo/fxs card improvise*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault 1from-internal default 2from-internal default 3from-pstn default 4from-pstn default I want to call from my cell and make my home phone ring and if I dont pickup in 10 secs I want the call go to my voicemail. But I am getting a loop detect. The debug output is attached. What am I doing wrong? -- Asif Iqbal PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? out.2 Description: Binary data ___ -- 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] No hardware timing source found in /proc/dahdi
John Millican wrote: # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: modprobe wctdm What is the output of the 'dmesg' command at this point? No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: /usr/sbin/dahdi_cfg If the dmesg shows that the driver found the card and there were not any conflicts, and dahdi_dummy is still loaded, this could be the result of an open reference to the old /proc/dahdi directory. i.e., you can force this to happen if you 'modprobe dahdi cd /proc/dahdi modprobe -r dahdi /etc/init.d/dahdi start' Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6022 fax: +1 256-428-6062 Check us out at: www.digium.com www.asterisk.org ___ -- 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] No hardware timing source found in /proc/dahdi
Shaun Ruffell wrote: John Millican wrote: # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: modprobe wctdm What is the output of the 'dmesg' command at this point? No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: /usr/sbin/dahdi_cfg If the dmesg shows that the driver found the card and there were not any conflicts, and dahdi_dummy is still loaded, this could be the result of an open reference to the old /proc/dahdi directory. i.e., you can force this to happen if you 'modprobe dahdi cd /proc/dahdi modprobe -r dahdi /etc/init.d/dahdi start' Cheers, Shaun All I see in dmesg is: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.4 dahdi_dummy: RTC rate is 1024 -- JohnM ___ -- 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] No hardware timing source found in /proc/dahdi
John Millican wrote: Shaun Ruffell wrote: John Millican wrote: # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: modprobe wctdm What is the output of the 'dmesg' command at this point? All I see in dmesg is: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.4 dahdi_dummy: RTC rate is 1024 Odds are there is another driver in your system then that is attaching to the tdm400 before the wctdm driver. I've seen where the hisax driver attaches first. Does 'lsmod | grep hisax' show that hisax is loaded? You can see what drivers may be configured for a board by looking at the /lib/modules/`uname -r`/modules.pcimap file. The tdm400 uses a vendorid of 0xe159 and a device id 0x0001. Search for any driver in the pcimap file that indicates support for that driver, and add it to the /etc/modprobe.d/blacklist file. Something like: blacklist hisax blacklist hisax_fcpcipnp Cheers, Shaun ___ -- 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] 428 Loop Detected
On Sun, Mar 15, 2009 at 6:28 PM, Asif Iqbal vad...@gmail.com wrote: Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqb...@improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106 dtmfmode=rfc2833 iqb...@improvise:/etc/asterisk$ cat extensions.conf [default] exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,Playback(tt-monkeys) exten = s,4,Hangup [from-internal] include = default [phone1] [from-pstn] ;include = default exten = s,1,Dial(SIP/ph...@phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup [line1] So my home land line is going to the FXO port and my home phone is hanging off of FXS port. Here are the contexts for my fxo/fxs card improvise*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudo default default 1 from-internal default 2 from-internal default 3 from-pstn default 4 from-pstn default I want to call from my cell and make my home phone ring and if I dont pickup in 10 secs I want the call go to my voicemail. But I am getting a loop detect. The debug output is attached. What am I doing wrong? -- Asif Iqbal PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? ___ Am I missing something or is your setup dahdi/zaptel only? What is the SIP stuff for? You are doing all TDM, no VoIP from what I gather. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best way to get 60+ analogue extensions.
Thanks very much Rob Stephen The channel banks look good. I am not sure if they are easily availble in NZ but we can get some in I am sure. Xorom make very positive comments about their astribanks and that you can have multiple channel banks on a server so they look pretty good (if they are honest). I can't tell the manufacturer of the other channel banks you were referring to. In Wellington, NZ, PRAs are pretty expensive and a 25Mbit/sec symmetrical fibre connection to a SIP provider is a better deal. On some of my other customers we have 15 SIP lines without issue using G711 and consuming about 80-100k per line if that. But I take the point so will revisit it in the design. Another reason for SIP is the Telepermited options available are limited over here, so to connect you really want to have an approved device in case you have any issues. But with SIP via a Provider you abstract that layer which is cleaner. If we need to have one E1 then having more for the Astribanks sounds fine. Cheers Duncan Rob Hillis wrote: Duncan Turnbull wrote: Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally fits and copes well with multiple cards, and the choice of Digium vs Sangoma or something else. You have several options here, however due to the power requirements, I wouldn't recommend you use either the Sangoma or Digium analogue cards here - providing ring voltage to that many extensions is likely to over-tax the power supply in the server. I'd either be looking at three channel banks (3 24 channel channel banks would give you a total of 72 analogue channels) or two Xorcom Astribanks which would likewise give you up to 64 channels. The Astribanks are probably a cheaper way to go since they connect to your server via USB rather than T1/E1 ports. However, I haven't had any experience with multiple Astribanks connected to the same server, so there may be issues there that I'm not aware of. Channel banks are certainly the proven and reliable technology, but will be significantly more expensive since they connect to your Asterisk server via T1/E1 links. I may need an ISDN PRA inbound but am working hard to have the inbound lines via SIP, but if I do that means at least 4 slots on this plan. You'd need to be very sure of the bandwidth and quality of connection to your VoIP provider to go with SIP for more than half a dozen channels. This kind of connection can easily be far more expensive than a traditional T1/E1 line, so I wouldn't be pushing so hard for SIP. If you were to use channel banks, you would most likely end up with a four port T1/E1 card and would only be using three of those channels, leaving a spare one for an incoming T1/E1 line. If you were to use Astribanks, you would have plenty of space in the server to include a T1/E1 card. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 428 Loop Detected
On Sun, Mar 15, 2009 at 8:28 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: On Sun, Mar 15, 2009 at 6:28 PM, Asif Iqbal vad...@gmail.com wrote: Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqb...@improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106 dtmfmode=rfc2833 iqb...@improvise:/etc/asterisk$ cat extensions.conf [default] exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,Playback(tt-monkeys) exten = s,4,Hangup [from-internal] include = default [phone1] [from-pstn] ;include = default exten = s,1,Dial(SIP/ph...@phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup [line1] So my home land line is going to the FXO port and my home phone is hanging off of FXS port. Here are the contexts for my fxo/fxs card improvise*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudo default default 1 from-internal default 2 from-internal default 3 from-pstn default 4 from-pstn default I want to call from my cell and make my home phone ring and if I dont pickup in 10 secs I want the call go to my voicemail. But I am getting a loop detect. The debug output is attached. What am I doing wrong? -- Asif Iqbal PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? ___ Am I missing something or is your setup dahdi/zaptel only? What is the SIP stuff for? You are doing all TDM, no VoIP from what I gather. I am probably missing something, being a newbie. I have a 4 port fxs/fxo (2/2) card. My land line is going to one of the FXO port and my home phone is connected to one of the FXS port. I want to be able to call my phone number from external phone (cell phone) and have my home phone ring. And if I do not pick up the phone in 10 secs I want the voicemail to pickup the call. I do have a dialtone when pick up my phone that is attached to the FXS port of my asterisk server -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 -- Asif Iqbal PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? ___ -- 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] 428 Loop Detected
I am probably missing something, being a newbie. I have a 4 port fxs/fxo (2/2) card. My land line is going to one of the FXO port and my home phone is connected to one of the FXS port. I want to be able to call my phone number from external phone (cell phone) and have my home phone ring. And if I do not pick up the phone in 10 secs I want the voicemail to pickup the call. I do have a dialtone when pick up my phone that is attached to the FXS port of my asterisk server A printout from the CLI would be helpful - but I think you have your contexts crossed over. (call from outside hitting internal, instead of from-pstn) PaulH ___ -- 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] 428 Loop Detected
Hi, problem is that you are saying that phone in sip.conf is at the same ip address of your asterisk box so you are dialing into a loop to your self asterisk box [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 what you need is: [phone] type=friend context=phone1 secret=g00dpazzwerd dtmfmode=rfc2833 host=dynamic ;configuring your codecs (i don't know what else you have configured, just preventing audio for you) disallow=all allow=ulaw allow=alaw allow=gsm Dial sip/phone is enough too.. [from-pstn] ;include = default exten = s,1,Dial(SIP/phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup hope it helps. don't forget to asterisk reload on cli. Looking forward to hearing from you. cheers -- Marco Mouta On Sun, Mar 15, 2009 at 10:28 PM, Asif Iqbal vad...@gmail.com wrote: Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqb...@improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106 dtmfmode=rfc2833 iqb...@improvise:/etc/asterisk$ cat extensions.conf [default] exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,Playback(tt-monkeys) exten = s,4,Hangup [from-internal] include = default [phone1] [from-pstn] ;include = default exten = s,1,Dial(SIP/ph...@phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup [line1] So my home land line is going to the FXO port and my home phone is hanging off of FXS port. Here are the contexts for my fxo/fxs card improvise*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudo default default 1 from-internal default 2 from-internal default 3 from-pstn default 4 from-pstn default I want to call from my cell and make my home phone ring and if I dont pickup in 10 secs I want the call go to my voicemail. But I am getting a loop detect. The debug output is attached. What am I doing wrong? -- Asif Iqbal PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 428 Loop Detected
Hello Asif, I have experienced 'loop detected' when the peer where I want to send the calls to, and the asterisk Box have both the same IP address (That would make a loop). Could you please verify? Regards, Asif Iqbal wrote: Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqb...@improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106 dtmfmode=rfc2833 iqb...@improvise:/etc/asterisk$ cat extensions.conf [default] exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,Playback(tt-monkeys) exten = s,4,Hangup [from-internal] include = default [phone1] [from-pstn] ;include = default exten = s,1,Dial(SIP/ph...@phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup [line1] So my home land line is going to the FXO port and my home phone is hanging off of FXS port. Here are the contexts for my fxo/fxs card improvise*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault 1from-internal default 2from-internal default 3from-pstn default 4from-pstn default I want to call from my cell and make my home phone ring and if I dont pickup in 10 secs I want the call go to my voicemail. But I am getting a loop detect. The debug output is attached. What am I doing wrong? ___ -- 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 -- Jose P. Espinal http://www.eSlackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 ReceiveFAX problem
hi,all i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX, link to a E1 (DE410P) using dahdi this can receive the fax from E1 successfully, but i see many error message in the log like this: [Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded file descriptor. when i receive a 5 pages fax, i will see this error message over 200 lines. it seems the channel.c try to call ast_read(), read some bytes from the channel but there is nothing ... whether it's a loop to check data on the channel ? and many times when reciving tax , the E1 card will down , all the channel get red alarm... [Mar 16 09:49:19] DEBUG[20928] chan_dahdi.c: Monitor doohicky got event Alarm on channel 2 [Mar 16 09:49:19] WARNING[20928] chan_dahdi.c: Detected alarm on channel 2: Recovering i then try asterisk 1.4.23.2 and agx-ast-addon , when using spandsp 0.0.5 and spandsp 0.0.6, like above , sometimes all E1 channel get red alarm when reciving fax but use spandsp 0.0.4 get no error... some one can tell me what version of asterisk and spandsp is the best version for fax??? thanks a lot.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 ReceiveFAX problem
asterisk 1.4.23.2 and spandsp 0.0.4 get the same error nowbut less times than other version ... [Mar 16 10:12:50] DEBUG[23749]: chan_dahdi.c:7115 do_monitor: Monitor doohicky got event Alarm on channel 1 [Mar 16 10:12:50] DEBUG[23752]: chan_dahdi.c:4731 __dahdi_exception: Exception on 11, channel 4 [Mar 16 10:12:50] DEBUG[23752]: chan_dahdi.c:3828 dahdi_handle_event: Got event Alarm(4) on channel 4 (index 0) [Mar 16 10:12:50] NOTICE[23748]: chan_dahdi.c:8704 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Mar 16 10:12:50] WARNING[23749]: chan_dahdi.c:3787 handle_alarms: Detected alarm on channel 1: Red Alarm [Mar 16 10:12:50] DEBUG[23748]: chan_dahdi.c:8715 pri_dchannel: Got event HDLC Abort (6) on D-channel for span 1 [Mar 16 10:12:50] WARNING[23749]: chan_dahdi.c:1507 dahdi_disable_ec: Unable to disable echo cancellation on channel 1: Invalid argument [Mar 16 10:12:50] NOTICE[23748]: chan_dahdi.c:8704 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 [Mar 16 10:12:50] DEBUG[23749]: chan_dahdi.c:7115 do_monitor: Monitor doohicky got event Alarm on channel 2 [Mar 16 10:12:50] WARNING[23748]: chan_dahdi.c:2486 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! [Mar 16 10:12:50] DEBUG[23748]: chan_dahdi.c:8715 pri_dchannel: Got event Alarm (4) on D-channel for span 1 [Mar 16 10:12:50] WARNING[23749]: chan_dahdi.c:3787 handle_alarms: Detected alarm on channel 2: Red Alarm [Mar 16 10:12:50] WARNING[23749]: chan_dahdi.c:1507 dahdi_disable_ec: Unable to disable echo cancellation on channel 2: Invalid argument [Mar 16 10:12:50] DEBUG[23749]: chan_dahdi.c:7115 do_monitor: Monitor doohicky got event Alarm on channel 3 [Mar 16 10:12:50] WARNING[23749]: chan_dahdi.c:3787 handle_alarms: Detected alarm on channel 3: Red Alarm ... the chan_dahdi.conf : [channels] language=en context=default switchtype=euroisdn pridialplan=unknown prilocaldialplan=national signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=no echotraining=no relaxdtmf=no rxgain=0 txgain=0 group=1 callgroup=1 pickupgroup=1 immediate=no callerid=asreceived callprogress=no channel = 1-15,17-31 the /etc/dahdi/system.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 loadzone= cn defaultzone = cn versions i try asterisk 1.4.23.2 / 1.6.0.6 / 1.6.0.7rc1 libpri 1.4.9 spandsp 0.0.4pre15 / 0.0.5pre4 / 0.0.6pre3 / 0.0.6pre6 dahdi 2.1.0.4+2.1.0.2 在2009-03-16 09:57:18，MaxGao ss...@126.com 写道： hi,all i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX, link to a E1 (DE410P) using dahdi this can receive the fax from E1 successfully, but i see many error message in the log like this: [Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded file descriptor. when i receive a 5 pages fax, i will see this error message over 200 lines. it seems the channel.c try to call ast_read(), read some bytes from the channel but there is nothing ... whether it's a loop to check data on the channel ? and many times when reciving tax , the E1 card will down , all the channel get red alarm... [Mar 16 09:49:19] DEBUG[20928] chan_dahdi.c: Monitor doohicky got event Alarm on channel 2 [Mar 16 09:49:19] WARNING[20928] chan_dahdi.c: Detected alarm on channel 2: Recovering i then try asterisk 1.4.23.2 and agx-ast-addon , when using spandsp 0.0.5 and spandsp 0.0.6, like above , sometimes all E1 channel get red alarm when reciving fax but use spandsp 0.0.4 get no error... some one can tell me what version of asterisk and spandsp is the best version for fax??? thanks a lot. 网易邮箱，中国第一大电子邮件服务商___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 ReceiveFAX problem
On Sun, Mar 15, 2009 at 9:57 PM, MaxGao ss...@126.com wrote: and many times when reciving tax , the E1 card will down , all the channel get red alarm... [Mar 16 09:49:19] DEBUG[20928] chan_dahdi.c: Monitor doohicky got event Alarm on channel 2 [Mar 16 09:49:19] WARNING[20928] chan_dahdi.c: Detected alarm on channel 2: Recovering You shouldn't be getting red alarms on your lines. Maybe a wiring or voltage problem? Or maybe a dahdi soft config that isn't correct for the lines? Or whatever you're plugging into on the other end is having problems? Do you plug from the phone company straight into the DE410P? A fax is just a phone call with tones that represent data. If you were doing voice you would probably be having the same problem. Work on getting calls in without getting the red alarms. I don't think there's any link between SpanDSP versions and red alarms on your card. ___ -- 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] No hardware timing source found in /proc/dahdi
Shaun Ruffell wrote: John Millican wrote: Shaun Ruffell wrote: John Millican wrote: # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: modprobe wctdm What is the output of the 'dmesg' command at this point? All I see in dmesg is: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.4 dahdi_dummy: RTC rate is 1024 Odds are there is another driver in your system then that is attaching to the tdm400 before the wctdm driver. I've seen where the hisax driver attaches first. Does 'lsmod | grep hisax' show that hisax is loaded? You can see what drivers may be configured for a board by looking at the /lib/modules/`uname -r`/modules.pcimap file. The tdm400 uses a vendorid of 0xe159 and a device id 0x0001. Search for any driver in the pcimap file that indicates support for that driver, and add it to the /etc/modprobe.d/blacklist file. Something like: blacklist hisax blacklist hisax_fcpcipnp Cheers, Shaun Well, lsmod | grep hisax returns nothing plain lsmod: Module Size Used by dahdi_dummy22472 0 dahdi 215776 1 dahdi_dummy crc_ccitt 18944 1 dahdi af_packet 57100 2 snd_pcm_oss67456 0 snd_mixer_oss 34176 1 snd_pcm_oss snd_seq74992 0 snd_seq_device 25620 1 snd_seq vmnet 72992 3 parport_pc 58456 0 parport56588 1 parport_pc vmmon 158908 0 sunrpc198600 1 iptable_filter 19840 0 ip_tables 37848 1 iptable_filter ip6table_filter19584 0 ip6_tables 31944 1 ip6table_filter x_tables 37000 2 ip_tables,ip6_tables ipv6 372344 29 cpufreq_conservative24968 0 cpufreq_userspace 23680 0 cpufreq_powersave 18560 0 powernow_k831504 0 apparmor 58672 0 loop 36356 0 dm_mod 77152 0 ohci1394 51272 0 ieee1394 115800 1 ohci1394 i2c_nforce222784 0 snd_hda_intel 368804 0 i2c_core 43648 1 i2c_nforce2 snd_pcm 108680 2 snd_pcm_oss,snd_hda_intel snd_timer 42632 2 snd_seq,snd_pcm snd84984 7 snd_pcm_oss,snd_mixer_oss,snd_seq,snd_seq_device,snd_hda_intel,snd_pcm,snd_timer k8temp 22656 0 hwmon 20232 1 k8temp button 26400 0 usblp 30976 0 forcedeth 65416 0 rtc_cmos 25016 0 rtc_core 38156 1 rtc_cmos rtc_lib19968 1 rtc_core sr_mod 33444 0 cdrom 52392 1 sr_mod usb_storage 102816 0 soundcore 25360 1 snd snd_page_alloc 27280 2 snd_hda_intel,snd_pcm ide_core 165648 1 usb_storage sg 53304 0 usbhid 58160 0 hid43776 1 usbhid ff_memless 22536 1 usbhid sd_mod 45824 6 ohci_hcd 38020 0 ehci_hcd 50572 0 usbcore 155560 6 usblp,usb_storage,usbhid,ohci_hcd,ehci_hcd edd26760 0 ext3 156688 3 mbcache26248 1 ext3 jbd89192 1 ext3 fan22792 0 sata_nv38404 4 pata_amd 31876 0 libata164096 2 sata_nv,pata_amd scsi_mod 176536 5 sr_mod,usb_storage,sg,sd_mod,libata thermal34576 0 processor 59592 2 powernow_k8,thermal in /etc/modprobe.d/blacklist there is already: # ISDN modules are load from /lib/udev/isdn.sh snip a lot of unrelated blacklist hisax blacklist hisax_fcpcipnp blacklist hisax_st5481 less /lib/modules/`uname -r`/modules.pcimap | grep 0xe159 hisax0xe159 0x0002 0x 0x 0x 0x 0x0 hisax0xe159 0x0001 0x 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xa159 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xe159 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xb100 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xb1d9 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xb118 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xb119 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xa9fd 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xa8fd 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xa800 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xa801
Re: [asterisk-users] 428 Loop Detected
Again, if I am interpreting this correctly, he is not using SIP. A four port card 2fxo/2fxs means to me that he is not using SIP at all. If by card, you mean some kind of SIP gateway, then I misunderstood and the problem, but seeing DAHDI channels leads me to believe that SIP is not required and actually causing your problems. SIP is a protocol for VoIP, DAHDI/Zaptel is TDM (analog POTS in this case)... If you had a SIP device, it would be connected to the data network, not a phone line. Can you just plug your phone into a regular landline jack and get dialtone? If so, forget SIP for now. Comment out or delete all your sip.conf peers since you are not using SIP. Change your dialplan to not (Dial/SIP but (Dial/DAHDI/1,10) and the correct channel to your FXS port that the phone is connected to. Thanks, Steve Totaro On Sun, Mar 15, 2009 at 9:20 PM, Marco Mouta marco.mo...@gmail.com wrote: Hi, problem is that you are saying that phone in sip.conf is at the same ip address of your asterisk box so you are dialing into a loop to your self asterisk box [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 what you need is: [phone] type=friend context=phone1 secret=g00dpazzwerd dtmfmode=rfc2833 host=dynamic ;configuring your codecs (i don't know what else you have configured, just preventing audio for you) disallow=all allow=ulaw allow=alaw allow=gsm Dial sip/phone is enough too.. [from-pstn] ;include = default exten = s,1,Dial(SIP/phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup hope it helps. don't forget to asterisk reload on cli. Looking forward to hearing from you. cheers -- Marco Mouta On Sun, Mar 15, 2009 at 10:28 PM, Asif Iqbal vad...@gmail.com wrote: Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqb...@improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106 dtmfmode=rfc2833 iqb...@improvise:/etc/asterisk$ cat extensions.conf [default] exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,Playback(tt-monkeys) exten = s,4,Hangup [from-internal] include = default [phone1] [from-pstn] ;include = default exten = s,1,Dial(SIP/ph...@phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup [line1] So my home land line is going to the FXO port and my home phone is hanging off of FXS port. Here are the contexts for my fxo/fxs card improvise*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudo default default 1 from-internal default 2 from-internal default 3 from-pstn default 4 from-pstn default I want to call from my cell and make my home phone ring and if I dont pickup in 10 secs I want the call go to my voicemail. But I am getting a loop detect. The debug output is attached. What am I doing wrong? -- Asif Iqbal PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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
