[asterisk-users] Asterisk 1.8 - BRI D Channel going up and down every few seconds
Dear all, I have the following challenge using Asterisk 1.8, using a Digium B410P card on BRI (The Netherlands, KPN ISDN) . DAHDI is running, dahdi_tools indicates OK on my span and light on back of card is green. However, in Asterisk i get the following warnings every few seconds: [Jan 2 20:33:17] == Primary D-Channel on span 1 up [Jan 2 20:33:26] == Primary D-Channel on span 1 down [Jan 2 20:33:26] WARNING[1671]: sig_pri.c:1095 pri_find_dchan: Span 1: D-channel is down! [Jan 2 20:33:27] == Primary D-Channel on span 1 up [Jan 2 20:33:37] == Primary D-Channel on span 1 down [Jan 2 20:33:37] WARNING[1671]: sig_pri.c:1095 pri_find_dchan: Span 1: D-channel is down! [Jan 2 20:33:38] == Primary D-Channel on span 1 up [Jan 2 20:33:47] == Primary D-Channel on span 1 down [Jan 2 20:33:47] WARNING[1671]: sig_pri.c:1095 pri_find_dchan: Span 1: D-channel is down! [Jan 2 20:33:48] == Primary D-Channel on span 1 up Result is I cannot dial out or in into Asterisk. The asterisk module states the D Channel is going up and down every few seconds. Some googling on Asterisk, BRI, and this warning indicates that there might be an issue with Aterisk 1.8.x and euro BRI, version of LibPRI etc. has anybody experienced these problems on BRI? Any suggestions with regards to these warnings are welcome! Kind regards, Marco Mooijekind. -- _ -- 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] Problem installing B410P BRI card for asterisk
Dear all, I know this is more a Digium hardware than an Asterisk issue. Already posted a question at Digium, however also like to see whether anyone in the Asterisk community has encountered the following situation: I installed a Digium B410P BRI PCI card on my new asterisk server, following the steps specified in the manual. I can see the PCI card is available using the lspci command: ... 04:00.0 Ethernet controller [0200]: Intel Corporation 82574L Gigabit Network Connection [8086:10d3] 05:00.0 Ethernet controller [0200]: Intel Corporation 82574L Gigabit Network Connection [8086:10d3] 08:00.0 PCI bridge [0604]: ASPEED Technology, Inc. AST1150 PCI-to-PCI Bridge [1a03:1150] (rev 02) 09:00.0 VGA compatible controller [0300]: ASPEED Technology, Inc. ASPEED Graphics Family [1a03:2000] (rev 10) 0a:01.0 ISDN controller [0204]: Digium, Inc. Wildcard B410 quad-BRI card [d161:b410] (rev 01) ... I specified the following in my system.conf in /etc/dahdi: loadzone = nl defaultzone = nl span = 1,1,0,ccs,ami bchan = 1,2 hardhdlc = 3 I loaded the driver using sudo modprobe wcb4xxp. Next I ran dahdi_cfg -vv which returns: DAHDI Tools Version - 2.5.0.2 DAHDI Version: 2.5.0.2 Echo Canceller(s): HWEC Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: none) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: none) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) 3 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) I'm in doubt about the DAHDI_SPANCONFIG failed on span 1: No such device or address (6). Next, if i execute sudo dmesg as specified by the manual it returns a huge trace: [ 376.082907] Wrote 0x0 to register 0x1ab but got back 0x4 [ 376.594754] Wrote 0x0 to register 0x1ab but got back 0x4 [ 377.106605] Wrote 0x0 to register 0x1ab but got back 0x4 [ 377.618423] Wrote 0x0 to register 0x1ab but got back 0x4 [ 378.130266] Wrote 0x0 to register 0x1ab but got back 0x4 [ 378.642088] Wrote 0x0 to register 0x1ab but got back 0x4 [ 1202.812870] show_signal_msg: 21 callbacks suppressed [ 1202.812876] dahdi_tool[1277]: segfault at 3fc378fa0 ip 004021ac sp 7fff131dd930 error 4 in dahdi_tool[40+3000] And a lot of Wrote 0x0 to register 0x1ab but got back 0x4 statements. If i run dahdi_tools it fails with a segmentation fault. Any suggestions are appreciated! Kind regards, Marco Mooijekind. -- _ -- 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] Dialing problem with Polycom phones after SIP update
Dear all, I'm using serveral Polycom 335 and 650 phones on Asterisk 1.8. All worked well. After applying the new Polycom UC 4.0.1 software update to the phones I notice the following: When dialing an extension, either on- or off hook, the phone immediately displays SIP URL:... This does not allow me to enter a regular numeric extension. The Polycom admin manual states that the phone displays the SIP URL input message if the phone is not registered. This is strange since i do see the phones registering themselves in the Asterisk verbose logging. Anyone experiencing this problem , any tips! Thanks in advance! Marco Mooijekind. -- _ -- 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] Dialing problem with Polycom phones after SIP update
Hello Gord, the line icon is solid black, which should indicate the lines are registered. Marco. On Fri, Dec 16, 2011 at 10:24 PM, Gord Urquhart gord...@gmail.com wrote: Does the phone show the line as registered? The little phone icon on the display should be solid for a registered line and just a outline for a unregistered line. Using wireshark to watch the SIP traffic is a easy way to ensure the REGISTER signally is complete. On Fri, Dec 16, 2011 at 1:02 PM, Marco Mooijekind marco.mooijek...@gmail.com wrote: Dear all, I'm using serveral Polycom 335 and 650 phones on Asterisk 1.8. All worked well. After applying the new Polycom UC 4.0.1 software update to the phones I notice the following: When dialing an extension, either on- or off hook, the phone immediately displays SIP URL:... This does not allow me to enter a regular numeric extension. The Polycom admin manual states that the phone displays the SIP URL input message if the phone is not registered. This is strange since i do see the phones registering themselves in the Asterisk verbose logging. Anyone experiencing this problem , any tips! Thanks in advance! Marco Mooijekind. -- _ -- 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 -- _ -- 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 -- _ -- 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] How to program a 100ms delay between the ringing of queued calls w/ ringall
Maybe local channels will do the trick? They allow you to schedule delays between subsequent devices ringing. Not sure whether they work as queue members.. Marco. Op 5 dec. 2011 16:35 schreef Sammy Govind govoi...@gmail.com het volgende: Hi, I dont think that 2 Queue commands would help, also wrapup time is for an putting delay in an agent who just answered the call and hungup. I think for this purpose you may need to open up the source code for queue and put some delay in the relevant code. Regards, Sammy. On Mon, Dec 5, 2011 at 6:56 PM, Scott Gifford sgiff...@suspectclass.comwrote: On Tue, Nov 22, 2011 at 5:34 PM, Douglas Mortensen d...@impalanetworks.com wrote: Hello, ** ** Does anyone have any idea of how I can program a 100ms delay in between the ringing of 2 subsequent calls in a queue configured with a ringall strategy? Does the wrapuptime queue option do what you want? http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf -Scott. -- _ -- 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 -- _ -- 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 -- _ -- 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] Issue with Polycom SPIP650 and its sidecar
Maybe use a power supply instead of PoE, see if problem still occurs. Marco. Op 30 nov. 2011 18:46 schreef Olivier oza_4...@yahoo.fr het volgende: 2011/11/30 Mike l...@net-wall.com Hi Olivier, ** ** It if occurs only on the sidecar, I would imagine this is either a defective sidecar/Polycom phone, or a defective PoE switch not giving enough power. Changing PoE port would eliminate of confirm the PoE port being the issue, but I’m betting on a Polycom defect. ** ** Make sure the PoE port is configured (if it`s a smart switch) to send maximum power to the port, with a sidecar I think the phone requires 12W. This info is very interesting. I wouldn't be too surprised that a PoE switch not supplying its theorical 15W output on a long period. I'll try to use work around this possible cause by not using PoE. In any case, I'll report my findings here. ** ** Regards, ** ** Mike ** ** ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* Wednesday, November 30, 2011 10:27 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Issue with Polycom SPIP650 and its sidecar*** * ** ** Hello, On one location, I've got from time to time (let say one a week) the following issue : the phone SoundPoint 650 works ok (can call or answer, display and sound are ok), the sidecar looses its display : entries on sidecar's LCD screen are not displayed anymore, or names are truncated, or BLF are not shown or updated. I only have one SPIP650 on this system so I can't compare with others. What could be the root cause of this ? Regards -- _ -- 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 -- _ -- 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 -- _ -- 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
