[asterisk-users] PRI Problem
I am having a problem with a new PRI turn-up on dahdi 2.5.0 and asterisk 1.8.5 that I have not seen before. The PRI is setup as B8ZS, ESF and the span shows up and ok. This PRI is merely a crossover T1 going into an old DC0 class 5 switch. I am getting the following errors over and over again [Aug 16 10:26:10] NOTICE[8002]: chan_dahdi.c:3043 my_handle_dchan_exception: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 [Aug 16 10:26:10] NOTICE[8002]: chan_dahdi.c:3043 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 1 I am also showing CRC4 errors on span as well. # asterisk -rx dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO T2XXP (PCI) Card 0 Span 1OK 1 0 2938622 ESF B8ZS 0 db (CSU)/0-133 feet (DSX-1) I am leaning towards a misconfiguration on the span on the switch side but here is my setup. Can anyone point me in the right direction? # dahdi_hardware pci::06:08.0 wct4xxp+ d161:1220 Wildcard TE220 (5th Gen) /etc/dahdi/system.conf # Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) B8ZS/ESF ClockSource span=1,1,0,esf,b8zs # termtype: unknown bchan=1-23 dchan=24 echocanceller=mg2,1-23 /etc/asterisk/chan_dahdi [channels] language=en context=Incoming-Pri switchtype=dms100 signalling=pri_cpe group=1 channel = 1-23 Thanks, Eric Merkel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Problem
On Tue, Aug 16, 2011 at 10:48 AM, Shaun Ruffell sruff...@digium.com wrote: On Tue, Aug 16, 2011 at 10:31:54AM -0400, Eric Merkel wrote: I am having a problem with a new PRI turn-up on dahdi 2.5.0 and asterisk 1.8.5 that I have not seen before. The PRI is setup as B8ZS, ESF and the span shows up and ok. This PRI is merely a crossover T1 going into an old DC0 class 5 switch. I'm not familiar with this switch however... I am getting the following errors over and over again [Aug 16 10:26:10] NOTICE[8002]: chan_dahdi.c:3043 my_handle_dchan_exception: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 [Aug 16 10:26:10] NOTICE[8002]: chan_dahdi.c:3043 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 1 Do these errors start right away or does it take a little bit of time before they start appearing? The error pretty much right away. I am also showing CRC4 errors on span as well. # asterisk -rx dahdi show status Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO T2XXP (PCI) Card 0 Span 1 OK 1 0 2938622 ESF B8ZS 0 db (CSU)/0-133 feet (DSX-1) I am leaning towards a misconfiguration on the span on the switch side but here is my setup. Can anyone point me in the right direction? # dahdi_hardware pci::06:08.0 wct4xxp+ d161:1220 Wildcard TE220 (5th Gen) /etc/dahdi/system.conf # Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) B8ZS/ESF ClockSource span=1,1,0,esf,b8zs # termtype: unknown bchan=1-23 dchan=24 echocanceller=mg2,1-23 You have this configured for the switch to provide you timing. Is the switch really expecting to provide timing to the TE220? I guess I was just assuming the switch would provide the timing as the telco normally provides this. Would you recommend just turning the timing off? /etc/asterisk/chan_dahdi [channels] language=en context=Incoming-Pri switchtype=dms100 signalling=pri_cpe group=1 channel = 1-23 Thanks, Eric Merkel -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- Thanks, Eric -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 4 Port FXO interface
I am looking to build a small PBX for an office that has 3 incoming analog lines and less than 10 extensions. For the Asterisk server I am going to use a small form factor PC with no-PCI slots so the FXO interface needs to be either FXO-SIP or USB. Can anyone make suggestions? I am looking at an AudioCodes MP114 FXO or possibly two Sangoma U100's but don't have experience with either. = Eric Merkel ejmerkel.li...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Xorcom Experience
I am looking at using for a Xorcom channel bank to provide some FXO FXS ports. Can anyone tell me how well they work with asterisk and your good/bad experiences with their products? Also, how well do they work with faxes and modems? Thanks! -- = Eric Merkel ejmerkel.li...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pri CLI command not available
I am in the process of trying to terminate a PRI into a new * server. The server has an old T100P T1/PRI card in it. I have compiled the following on Centos 5.4. dahdi-linux-complete-2.2.1+2.2.1 libpri-1.4.10.2 asterisk-1.4.29 Everything seems to have compiled fine. DAHDI reports Found a Wildcard: Digium Wildcard T100P T1/PRI on bootup. Dahdi_tool shows that the span is up and active with no alarms however the phone company is not seeing the trunkgroup going into service. I was wanting to take a look at the PRI debugs but for some reason the CLI pri option is not available. I libpri compiled without any issues prior to compiling asterisk. What would cause the pri debug commands to not be available in the CLI? = Eric Merkel ejmerkel.li...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pri CLI command not available
Thanks you were exactly right. I had a problem in my chan_dadhi.conf file. Basically, I had the channels defined before the signaling and it wouldn't load. It did not show any errors that I could see on startup and there were no messages in the /var/log/asterisk/messages but when doing a load chan_dahdi.so from the command line showed me the problem. Thanks again! = Eric Merkel ejmerkel.li...@gmail.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony Francis - Handy Networks LLC Sent: 2010-01-21 15:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pri CLI command not available This is often caused by the dahdi module not loading, check /var/log/asterisk/messages for the reason, or better yet, from the cli load the module manually and see the error in real time. If I had to guess I would say it is a configuration error. Thank you and have a nice day, Anthony Francis From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Merkel (Mail Lists) Sent: Thursday, January 21, 2010 1:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] pri CLI command not available I am in the process of trying to terminate a PRI into a new * server. The server has an old T100P T1/PRI card in it. I have compiled the following on Centos 5.4. dahdi-linux-complete-2.2.1+2.2.1 libpri-1.4.10.2 asterisk-1.4.29 Everything seems to have compiled fine. DAHDI reports Found a Wildcard: Digium Wildcard T100P T1/PRI on bootup. Dahdi_tool shows that the span is up and active with no alarms however the phone company is not seeing the trunkgroup going into service. I was wanting to take a look at the PRI debugs but for some reason the CLI pri option is not available. I libpri compiled without any issues prior to compiling asterisk. What would cause the pri debug commands to not be available in the CLI? = Eric Merkel ejmerkel.li...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7960 Queue Issue
On 11/4/07, Nick Brown [EMAIL PROTECTED] wrote: Morning All, Quick question that has me stumped. Have a queue with several members (Statically defined in queues.conf at this stage for testing) who use Cisco 7960's. The queue is configured to use rrmemory and generally this works correctly. However if a member is already on a call their phone will still ring (The 7960 can show multiple incoming calls for one line). I really don't want members who are on calls to get more calls. Especially when we start logging out members who don't answer. Asterisk shows; -- Called 1014 -- SIP/1014-08f2e4d0 is ringing -- Local/[EMAIL PROTECTED];1 is ringing -- Nobody picked up in 15000 ms Short of disabling the feature to show multiple incoming calls on the 7960's (Which I don't know if it can be done anyway), has anyone got any suggestions? Yes, you can turn off this in the phone. Go into call preferences on the phone and turn off call waiting. Not optimal but can be done. -Eric ___ --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] Weird Error When upgrading 7960G to 8.2
I would guess from the error message that you have the filename of the firmware mispelled in the OS79XX.txt file or it's not in your tftp directory. It could also be bad permissions on the file. -Eric On 5/17/06, Ben Blakely [EMAIL PROTECTED] wrote: Hello All, I just picked up 10 new 7960G's and having a hardtime upgrading the firmware on them. We already have 30 or 40 of these phones in production. Typically when I get a new phone, I just plug it into the voice vlan and it auto grades to the firmware in OS79XX.txt. Now whats happening is im getting a weird error during the reading of that file. Here's the output from the TFTP logs. May 17 16:49:31 asterisk in.tftpd[31683]: sending NAK (1, File not found) to 192.168.10.38 May 17 16:49:40 asterisk in.tftpd[31684]: RRQ from 192.168.10.34 filename OS79XX.TXT May 17 16:49:40 asterisk in.tftpd[31684]: sending NAK (4, Request not null-terminated) to 192.168.10.34 May 17 16:49:40 asterisk in.tftpd[31685]: RRQ from 192.168.10.34 filename SEP001647051680.cnf.xml May 17 16:49:40 asterisk in.tftpd[31685]: sending NAK (1, File not found) to 192.168.10.34 Any ideas? ___ --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] Linksys PAP2-NA
I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that they heard from Cisco that these units were not due out until Dec. Did Cisco/Linksys pull these units off the shelves? -- Eric Merkel MetaLINK Technologies, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco AS5350 Gateway Intergration
I am beginning a project to integrate * with a Cisco AS5350 gateway for inbound/outbound termination. If anyone has experience with this, what channel type would you recommend? H.323, SIP or MGCP? I've scoured the archives to see what channel type would be the most stable but haven't found a definitive answer. Also, if anyone has dealt with this setup before and would be willing to share an example config (cisco *), that would be much appreciated! Thanks. -- Eric Merkel MetaLINK Technologies, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
