[asterisk-users] kernel: dahdi: Master changed to B4/0/x
Hi, I did replace an old asterisk box by a new shiny one (2 BRI ports used on a quad port card - BeroNet PCI Express) I noticed a message in the logs that puzzles me: May 11 19:10:02 kernel: dahdi: Master changed to B4/0/1 May 11 19:10:08 kernel: wcb4xxp :05:04.0: PCI INT A disabled May 11 19:10:08 kernel: wcb4xxp :05:04.0: Driver unloaded. May 11 19:10:08 kernel: dahdi: Telephony Interface Unloaded May 11 19:10:09 kernel: dahdi: Telephony Interface Registered on major 196 May 11 19:10:09 kernel: dahdi: Version: 2.3.0.1 May 11 19:10:09 kernel: wcb4xxp :05:04.0: probe called for b4xx... May 11 19:10:09 kernel: wcb4xxp :05:04.0: PCI INT A - GSI 32 (level, low) - IRQ 32 May 11 19:10:09 kernel: wcb4xxp :05:04.0: Identified BeroNet BN4S0 (controller rev 1) at 0001bc00, IRQ 32 May 11 19:10:09 kernel: wcb4xxp :05:04.0: NOTE: hardware echo cancellation has been disabled May 11 19:10:09 kernel: wcb4xxp :05:04.0: Port 1: TE mode May 11 19:10:09 kernel: wcb4xxp :05:04.0: Port 2: TE mode May 11 19:10:09 kernel: wcb4xxp :05:04.0: Port 3: TE mode May 11 19:10:09 kernel: wcb4xxp :05:04.0: Port 4: TE mode May 11 19:10:09 kernel: wcb4xxp :05:04.0: Did not do the highestorder stuff May 11 19:10:09 kernel: dahdi_echocan_oslec: Registered echo canceler 'OSLEC' May 11 19:10:09 kernel: dahdi: Registered tone zone 2 (France) May 11 19:10:09 kernel: wcb4xxp :05:04.0: new card sync source: port 1 May 11 19:10:30 kernel: dahdi: Master changed to B4/0/2 May 11 19:10:36 kernel: dahdi: Master changed to B4/0/1 May 11 19:10:56 kernel: dahdi: Master changed to B4/0/2 May 11 19:11:02 kernel: dahdi: Master changed to B4/0/1 May 11 19:11:12 kernel: dahdi: Master changed to B4/0/2 May 11 19:12:07 kernel: dahdi: Master changed to B4/0/1 May 11 19:12:17 kernel: dahdi: Master changed to B4/0/2 May 11 19:12:19 kernel: dahdi: Master changed to B4/0/1 May 11 19:12:40 kernel: dahdi: Master changed to B4/0/2 May 11 19:12:42 kernel: dahdi: Master changed to B4/0/1 When those messages appear a RED alarm is triggered on the corresponding span. Hardware: 05:04.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) Modules: dahdi_echocan_oslec 1386 8 echo3354 1 dahdi_echocan_oslec wcb4xxp74747 6 dahdi 193086 14 dahdi_echocan_oslec,wcb4xxp version:2.3.0.1 alias: dahdi_dummy asterisk version: Asterisk 1.6.2.9-2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Channel dumps: May 11 19:18:19 kernel: dahdi: Dump of DAHDI Channel 1 (B4/0/1/1,1,1): May 11 19:18:19 kernel: May 11 19:18:19 kernel: dahdi: flags: 501 hex, writechunk: 88013b74008c, readchunk: 88013b7400a4 May 11 19:18:19 kernel: dahdi: rxgain: a0276170, txgain: a0276170, gainalloc: 0 May 11 19:18:19 kernel: dahdi: span: 88013b740128, sig: 80 hex, sigcap: 10080 hex May 11 19:18:19 kernel: dahdi: inreadbuf: -1, outreadbuf: 0, inwritebuf: 0, outwritebuf: -1 May 11 19:18:19 kernel: dahdi: blocksize: 160, numbufs: 4, txbufpolicy: 0, txbufpolicy: 0 May 11 19:18:19 kernel: dahdi: txdisable: 0, rxdisable: 0, iomask: 0 May 11 19:18:19 kernel: dahdi: curzone: 88013cce, tonezone: 2, curtone: (null), tonep: 0 May 11 19:18:19 kernel: dahdi: digitmode: 0, txdialbuf: , dialing: 0, aftdialtimer: 0, cadpos. 0 May 11 19:18:19 kernel: dahdi: confna: 0, confn: 0, confmode: 0, confmute: 0 May 11 19:18:19 kernel: dahdi: ec: (null), deflaw: 0, xlaw: a0257d50 May 11 19:18:19 kernel: dahdi: itimer: 0, otimer: 0, ringdebtimer: 0 May 11 19:18:19 kernel: May 11 19:18:21 kernel: dahdi: Dump of DAHDI Channel 2 (B4/0/1/2,2,2): May 11 19:18:21 kernel: May 11 19:18:21 kernel: dahdi: flags: 501 hex, writechunk: 88013b740094, readchunk: 88013b7400ac May 11 19:18:21 kernel: dahdi: rxgain: a0276170, txgain: a0276170, gainalloc: 0 May 11 19:18:21 kernel: dahdi: span: 88013b740128, sig: 80 hex, sigcap: 10080 hex May 11 19:18:21 kernel: dahdi: inreadbuf: -1, outreadbuf: 0, inwritebuf: 0, outwritebuf: -1 May 11 19:18:21 kernel: dahdi: blocksize: 160, numbufs: 4, txbufpolicy: 0, txbufpolicy: 0 May 11 19:18:21 kernel: dahdi: txdisable: 0, rxdisable: 0, iomask: 0 May 11 19:18:21 kernel: dahdi: curzone: 88013cce, tonezone: 2, curtone: (null), tonep: 0 May 11 19:18:21 kernel: dahdi: digitmode: 0, txdialbuf: , dialing: 0, aftdialtimer: 0, cadpos. 0 May 11 19:18:21 kernel: dahdi: confna: 0, confn: 0, confmode: 0, confmute: 0 May 11 19:18:21 kernel: dahdi: ec: (null), deflaw: 0, xlaw: a0257d50 May 11 19:18:21 kernel: dahdi: itimer: 0, otimer: 0, ringdebtimer: 0 Do you guys have a clue about it ? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] Limit total length of calls to a specifig SIP peer
On 08/06/2010 19:19, Steve Edwards wrote: The ONLY way (how's that for humble) to do this in a reliable and robust method is to use a real database. Personally, I like MySQL and I prefer to do database work in an AGI in a compiled language like C. Maintaining the accumulated duration in a global variable will fail if you need to restart Asterisk at any time. A global variable will also fail if you have more than 1 call finish at the same time. Parsing log files is guaranteed to be a resource pig and still has race conditions. Hi, I'm gonna follow your advice and store the CDR in a PostgreSQL database. It will allow to easily plug an AGI script to it. Thanks Laurent -- _ -- 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] Limit total length of calls to a specifig SIP peer
Hi, I'm currently using a cheap SIP provider for outbound calls. I do have 6 channels to them. In their terms of service there is the following limit: The total duration of calls during one single day should not exceed 24 hours or we do have the right to terminate the contract...blah blah What is the best way to use this provider as long as we are below let's say 22h in a single day ? Thanks Laurent -- _ -- 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] Limit total length of calls to a specifig SIP peer
On 08/06/2010 15:21, Danny Nicholas wrote:
My .02 - I would set up a context for dialing with this provider that counts
minutes and stops the dial with a message once you get to 1321 minutes (22
hours).
Exten = _8X.,1,noop(call using Cheap sip)
Exten = _8X.,2,macro(call_out,${EXTEN:1})
Exten = _8X.,3,hangup
[macro-call_out]
Exten = s,1,Gotoif($[${GLOBAL(SECUSED)} 79200]?4)
Exten = s,n,playback(out-of-minutes)
Exten = s,n,hangup
Exten = s,n,dial(SIP/${ARG1}...)
Exten = s,n,Set(GLOBAL(SECUSED)=${GLOBAL(SECUSED)} + ${DIALEDTIME})
Exten = s,n,hangup
Exten = h,1,Set(GLOBAL(SECUSED)=${GLOBAL(SECUSED)} + ${DIALEDTIME})
By using DIALEDTIME instead of ANSWEREDTIME, you reduce the possibility of
going over on your minutes, but you could squeeze in an extra 2-20 seconds
per call using AT vs DT.
I'm using the following:
exten = 0145381068,1,NoOp()
exten = 0145381068,2,Gotoif($[${GLOBAL(SECUSED)}10]?4)
exten = 0145381068,3,hangup
exten = 0145381068,4,Dial(SIP/1...@sipoperator)
exten = 0145381068,5,Set(GLOBAL(SECUSED)=${GLOBAL(SECUSED)} +
${DIALEDTIME})
exten = 0145381068,6,hangup
I place a test call of 20 secs, hangup, and I can place another call
again through the same provider.
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Re: [asterisk-users] beroNet BN4S0e PCI Express ISDN Card with chan_dahdi
On 31/01/2010 04:35, Tzafrir Cohen wrote: Yes, please see https://issues.asterisk.org/view.php?id=16493 Basically the driver needs minimal fixing. Probably just to add the PCI ID to the list. Hi, Thanks Gonna have a look at it. -- _ -- 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] beroNet BN4S0e PCI Express ISDN Card with chan_dahdi
Hi, I'm currently trying to get a BN4S0e (which is basically a BN4S0 with a PCIe connector) working with dahdi. The module is loading properly but the card is not detected by the module. Is support on dahdi planned for this card ? In the meantime i'm gonna use mISDN with this card. Thanks Laurent -- _ -- 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] Unable to receive faxes
Laurent CARON wrote: I'm experiencing a quite strange behavior while trying to receive faxes through Asterisk (either directly through app_rxfax or with spandsp + hylafax). Hi, I should have mentionned (known?) that the telco is using G729 compression. Obviously FAXes will never get through with such a compression. Laurent ___ -- 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] Unable to receive faxes
Hi,
I'm experiencing a quite strange behavior while trying to receive faxes
through Asterisk (either directly through app_rxfax or with spandsp +
hylafax).
Config:
HFC quad BRI card (3 T0 connected to the card)
Asterisk 1.4.21
asterisk-app-fax 0.0.20070624-2
hylafax 2:4.4.4-10.1
libpri 1.4.2
libspandsp3 0.0.4pre16
/etc/zaptel.conf
loadzone=fr
defaultzone=fr
span=1,0,3,ccs,ami
span=2,0,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
bchan=1,2
dchan=3
bchan=4,5
dchan=6
bchan=7,8
dchan=9
/etc/asterisk/zapata.conf
[channels]
language=fr
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan = unknown
prilocaldialplan = dynamic
nationalprefix = 0
internationalprefix = 00
echocancel=yes
rxgain=1.1
txgain=1.1
group = 1
context=zaptel-in
channel = 1-2
channel = 4-5
channel = 7-8
/etc/asterisk/extensions.conf:
exten = 0258,1,Answer()
exten = 0258,n,Set(FAXFILE=/var/spool/asterisk/fax/fax-0258.tif)
exten = 0258,n,rxfax(${FAXFILE})
exten = 0258,n,NoOp()
exten = 0258,n,Hangup
Here is the asterisk cli output:
-- Accepting voice call from '' to '0258' on channel 0/1, span 1
-- Executing [0...@zaptel-in:1] NoOp(Zap/1-1, ) in new stack
-- Executing [0...@zaptel-in:2] Set(Zap/1-1, CALLERID(num)=) in
new stack
-- Executing [0...@zaptel-in:3] Goto(Zap/1-1,
default|0491140258|1) in new stack
-- Goto (default,0258,1)ff o
-- Executing [0...@default:1] Goto(Zap/1-1, 0258|1) in new stack
-- Goto (default,0258,1)
-- Executing [0...@default:1] Answer(Zap/1-1, ) in new stack
-- Executing [0...@default:2] Set(Zap/1-1,
FAXFILE=/var/spool/asterisk/fax/fax-0258.tif) in new stack
-- Executing [0...@default:3] RxFAX(Zap/1-1,
/var/spool/asterisk/fax/fax-0258.tif) in new stack
-- Channel 0/1, span 1 got hangup request, cause 16
-- Hungup 'Zap/1-1' channel
The resulting tiff file is nearly empty.
-rw-r--r-- 1 asterisk asterisk 8 2009-03-18 22:31
/var/spool/asterisk/fax/fax-0258.tif
Do anyone have a clue about this issue ?
What is cause 16 about ?
Thanks
Laurent
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[asterisk-users] Call Pickup (*8) / Attended forward and CallerID
Hi, Since we're moving from a legacy PABX that has been serving one of our customers for more than 15 years, we'd like this process to require no human habits change among the users. Software: Asterisk 1.4.22 Hardware: Polycom phones (mainly 430/601) Here are the problems: We did configure call groups, pickup groups, ... - When someone picks up a call from another person, the display of his phone only shows *8 and not the original CallerID. - When doing an attended transfer, the callerid of the original caller (A calls B, then B forwards to C = We want to show C the original callerid somewhere on his phone's screen). - When using the blind transfer feature, the CallerID is fine. I know this has already been discussed in 2006 (from digium's BTS), and would like to know if this situation did change, or not. Is it still considered as features ? Is it considered as bugs ? Will it be implemented in another way in some future release ? ...? Thanks Laurent ___ -- 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] TE410P alarms stay RED with 1.4.22
Bonjour Louis-David, Asterisk envoie-t-il le signal au boitier pour le failover ? Laurent Le 11 nov. 08 à 08:49, Louis-David Mitterrand [EMAIL PROTECTED] g a écrit : Hi, I tried upgrading from debian's 1.4.21.2 package to vanilla 1.4.22 but then my TE410P alarms stay RED and no zap channels can be created, even if they are correctly listed by zap show channels. I tried adding dahdichanname = no to asterisk.conf's [options] to no effect. Going back to 1.4.21.2 brings my alarms back to OK. This is with zaptel 1.4.12.1. -- http://www.critikart.net ___ -- 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] TE410P alarms stay RED with 1.4.22
Hi, I'm sorry i didn't check the recipient while replying. Sorry about the noise... Laurent Le 11 nov. 08 à 11:44, Laurent Caron [EMAIL PROTECTED] a écrit : Bonjour Louis-David, Asterisk envoie-t-il le signal au boitier pour le failover ? Laurent ___ -- 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] Random reboots on IP-601 after changing network topology
Hi, We did move an office from a remote building to another floor in our building, allowing us to directly hook the switch in that building to our core switch (GigE). In the past, the phones were on the same subnet as the * server. All the phones worked flawlessy for about two years. Since the move, the phones are on the network mysubnet.13.0/24 and the * server on mysubnet.0.0/24. Randomly my IP601 (430's are not affected by this bug) are rebooting. I've got the boot app log of the phones should it help to track the problem down. The phones are all hooked to a 3COM 4500 PWR switch. If i move the IP601 to the same VLAN as the * server, the phones are not rebooting anymore. Did someone already experience such a behavior ? Thanks Laurent 0803215603|copy |4|03|Upload of 'log/0004f2187952-app.log' FAILED on attempt 1 (addr 1 of 1) 0803215603|copy |4|03|Upload of 'log/0004f2187952-app.log' FAILED on attempt 1 (addr 1 of 1) 0803195742|so |*|03|-- Initial log entry -- 0803195742|so |*|03|Platform: Model=SoundPoint IP 601, Assembly=2345-11605-001 Rev=B 0803195742|so |*|03|Platform: MAC=0004f2187952, IP=mysubnet.13.195, Subnet Mask=255.255.255.0 0803195742|so |*|03|Platform: BootBlock=2.6.0 (11605_001) 30-Apr-05 12:50 0803195742|so |*|03|Platform: Bootrom=4.1.1.0232 29-Mar-08 16:39 0803195742|so |*|03|Application, main: Label=SIP, Version=3.0.3.0401 22-May-08 15:13 0803195742|so |*|03|Application, main: P/N=3150-11530-303 0803195742|ethf |*|03|Initial log entry. Current logging level 4 0803195742|so |5|03|utilCertificateInit failed. 0803195742|hw |*|03|Initial log entry. Current logging level 4 0803195742|ares |*|03|Initial log entry. Current logging level 4 0803195742|dns |*|03|Initial log entry. Current logging level 3 0803195742|cfg |*|03|Initial log entry. Current logging level 3 0803195742|cfg |3|03|RT|Checking DHCP option 160 type string 0803195742|cfg |3|03|RT|Runtime basic IP parameters updated. 0803195742|cfg |3|03|RT|Runtime provisioning server parameters updated. 0803195742|cfg |3|03|RT|Runtime SNTP parameters updated. 0803195742|dns |*|03|DNS resolver servers are 'mysubnet.0.3' 'mysubnet.0.2' 0803195742|dns |*|03|DNS resolver search domain is 'mydomain.com' 0803195742|log |*|03|Initial log entry. Current logging level 4 0803195742|so |4|03|[SoFontsC]: Font item (6)(1) is NULL. 0803195742|curl |*|03|Initial log entry. Current logging level 3 0803195742|utilm|*|03|Initial log entry. Current logging level 4 0803195742|copy |*|03|Initial log entry. Current logging level 3 0803195742|rtos |*|03|Initial log entry. Current logging level 4 0803195742|sec |*|03|Initial log entry. Current logging level 4 0803195742|cfg |3|03|Prm|Beginning to provision phone 0803195742|copy |3|03|'ftp://PlcmSpIp:[EMAIL PROTECTED]/2345-11605-001.bootrom.ld' from 'mysubnet.0.3' 0803195742|cfg |3|03|Prm|Image 2345-11605-001.bootrom.ld has not changed 0803195742|copy |3|03|buffered_write: transfer Terminated on entry. Return 0 0803195742|copy |3|03|Download of '2345-11605-001.bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 0803195742|cfg |3|03|Prm|Downloaded bootROM is identical to current version 4.1.1 0803195742|copy |3|03|'ftp://PlcmSpIp:[EMAIL PROTECTED]/0004f2187952.cfg' from 'mysubnet.0.3' 0803195742|copy |3|03|Download of '0004f2187952.cfg' succeeded on attempt 1 (addr 1 of 1) 0803195742|copy |3|03|'ftp://PlcmSpIp:[EMAIL PROTECTED]/2345-11605-001.sip.ld' from 'mysubnet.0.3' 0803195742|cfg |3|03|Prm|Image 2345-11605-001.sip.ld has not changed 0803195742|copy |3|03|buffered_write: transfer Terminated on entry. Return 0 0803195742|copy |3|03|Download of '2345-11605-001.sip.ld' succeeded on attempt 1 (addr 1 of 1) 0803195742|copy |3|03|'ftp://PlcmSpIp:[EMAIL PROTECTED]/phone-0156800677.cfg' from 'mysubnet.0.3' 0803195742|copy |3|03|Download of 'phone-0156800677.cfg' succeeded on attempt 1 (addr 1 of 1) 0803195742|copy |3|03|File /ffs0/phone-0156800677_cfg.zzz, is upto date 0803195742|copy |3|03|'ftp://PlcmSpIp:[EMAIL PROTECTED]/sip.cfg' from 'mysubnet.0.3' 0803195742|copy |3|03|Download of 'sip.cfg' succeeded on attempt 1 (addr 1 of 1) 0803195742|copy |3|03|File /ffs0/sip_cfg.zzz, is upto date 0803195742|cfg |3|03|Prm|Check of configuration files suceeded 0803195742|cfg |3|03|Prm|Phone successfully provisioned 0803195742|cfg |*|03|Prm|Configuration file phone-0156800677.cfg is from template phone1.cfg, revision 1.83.2.2 0803195742|cfg |*|03|Prm|Configuration file sip.cfg is from template sip.cfg, revision 1.273.2.69 0803195742|so |*|03|Configuration files: phone-0156800677.cfg,sip.cfg 0803195742|copy |3|03|'ftp://PlcmSpIp:[EMAIL PROTECTED]/0004f2187952-phone.cfg' from 'mysubnet.0.3' 0803195742|copy |4|03|Download of '0004f2187952-phone.cfg' FAILED on attempt 1 (addr 1 of 1) 0803195742|copy |4|03|Server 'mysubnet.0.3' said '0004f2187952-phone.cfg' is not present 0803195742|utilm|4|03|uBLFCompressed: File /ffs0/local/0004f2187952-phone_cfg.zzz doesn't
[asterisk-users] SIP peer unrechable when using an aliased interface
Hi, I'm experiencing a strange behavior on one of my asterisk servers. When I make a sip connection (sip.conf) between the 2 boxes using the primary interface it works fine. When however use an aliased interface (ethX:Y) it fails. from sip show peers: ast-rem-vpn 192.168.3.253A 5060UNREACHABLE when I use the ethX interface directly, it works fine. Did anyone experience such a strange behavior ? Thanks Laurent ___ --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] Polycom phone registration problem
Hi, One of my users is in trouble with his polycom phone hooked to an asterisk server. The phone works fine for a few days, and then disappears from the registered sip peers in asterisk. The user is able to place outbound phone calls, but can't receive incoming calls until the network plug is unplugged/plugged. Working line XXYYZZAA24/XXYYZZAA24 10.50.5.186 D A 5060 OK (12 ms) Non working line (sip show peers) XXYYZZAA24/XXYYZZAA24 (Unspecified) D A 5060 OK (12 ms) Do you guys have any clue about this issue ? Thanks Laurent ___ --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] Polycom phone registration problem
Rob Schall wrote: In the logs, does that phone try to re-register itself, or does it just give up? If its not trying to re-register, you might want to look at the Expires, Register and Retry settings in the phone. Here is the config snippet: server voIpProt.server.1.address= voIpProt.server.1.port=5060 voIpProt.server.1.transport=DNSnaptr voIpProt.server.1.expires= voIpProt.server.1.expires.overlap= voIpProt.server.1.register=1 voIpProt.server.1.retryTimeOut=0 voIpProt.server.1.retryMaxCount=0 voIpProt.server.1.expires.lineSeize=30 voIpProt.server.1.lcs= voIpProt.server.2.transport=DNSnaptr voIpProt.server.dhcp.available= voIpProt.server.dhcp.option= voIpProt.server.dhcp.type=/ Thanks Laurent ___ --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] Polycom phone registration problem
Rob Schall wrote: In the logs, does that phone try to re-register itself, or does it just give up? The phone is giving up. Jun 7 14:29:36 NOTICE[22015] chan_sip.c: Auto-congesting SIP/XXYYZZAA24-08553940 Laurent ___ --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] Poor man's High Availability solution
On Thu, May 03, 2007 at 12:47:46AM +0200, Laurent Caron wrote: On Sun, Apr 29, 2007 at 09:06:53PM +0200, Clayton Milos wrote: Since a PRI is a physical connection as well as a logical one, if you can get the server to shut down when it has a problem you could put a 4-pole relay to change the PRI over to the other box. I think the ISDNGuard is more or less like a relay. Here is how I plan to set it up. Hook the Pri to the Net port of the ISDNGuard, hook the first port of each asterisk server's pri card to the 1st ast CPE and isdn CPE port, hook the second port of the pri card to the 2nd ast CPE and isdn CPE port, and finally plug the second Net port to the legacy PABX. Do you think this is the correct way of plugging two asterisk servers in front of a legacy pabx ? The attached file shows how it was supposed to be. attachment: ast.png___ --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] Poor man's High Availability solution
On Sun, Apr 29, 2007 at 09:06:53PM +0200, Clayton Milos wrote: Since a PRI is a physical connection as well as a logical one, if you can get the server to shut down when it has a problem you could put a 4-pole relay to change the PRI over to the other box. I think the ISDNGuard is more or less like a relay. Here is how I plan to set it up. Hook the Pri to the Net port of the ISDNGuard, hook the first port of each asterisk server's pri card to the 1st ast CPE and isdn CPE port, hook the second port of the pri card to the 2nd ast CPE and isdn CPE port, and finally plug the second Net port to the legacy PABX. |AST1--| | | T2 == |=Legacy PABX | | |AST2--| Net ports Ast CPE ports Isdn CPE ports 1 2 3 4 1 2 3 4 1 2 3 4 | | | | | | | | | | | | | | | | | | I L | | | | n e I O I O c g n u n u o a p t p t m c u p u p i y t u t u n t t g P A f f P B r r R X 1 o 1 o I s m s m t t A A p S p S o T o T r 1 r 2 t t t t o o o o f f l l A e A e S g S g T a T a 1 c 2 c y y Do you think this is the correct way of plugging two asterisk servers in front of a legacy pabx ? Thanks Laurent ___ --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] Poor man's High Availability solution
Tim Panton wrote: Here's what we do when consulting in this area: First decide what the maximum acceptable downtime is, and what the costs to the business of that downtime would be. Use that as the starting point for the HA design. Discuss with the telco what they can do. They can install a second PRI to route calls when the 1st line is down. Avoid complexity at _all_ costs. Since it is possible with BRI cards, i'm wondering if it could be done with PRI. No, BRI has a 'bus' topology, PRI is point-to-point. ok Taking a BRI from the same telco into a second asterisk box and having them redirect calls to it when the PRI is down could be pretty economical. In fact if you could persuade them to put the BRI in the same hunt group, but at a lower priority the switch over would be seamless at their end - all you would need to do is ensure the main asterisk powered itself off if it crashed, thus dropping the PRI. Since my telco would charge about $700~800/month, i'd rather get the ISDNGuard way ;) ___ --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] Poor man's High Availability solution
Hi, I'm wondering what the best option to obtain a high availability asterisk server is. I currently use a TE410P (4 x E1) card. I'm thinking of 2 different solutions: - 2 servers configured with Heartbeat + DRBD (drbd mainly for voicemail) and the E1 span plugged to the 2 servers (with a TE410P in each server). - 2 servers configures with Heartbeat + DRBD with the E1 span hooked to an ISDN guard connected to the main server and the backup one. Here comes the real question. Is it technically good to connect an E1 span to 2 cards at the same time (with only one accepting the calls). Since it is possible with BRI cards, i'm wondering if it could be done with PRI. Thanks Laurent ___ --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] Asterisk stops accepting calls
Hi, I've got a serious problems. I have an * box set up at a custommer office. * seems to work well until this message appears when i try to call from the outside to any number managed by * Jul 7 17:12:08 WARNING[8792]: chan_zap.c:9256 pri_dchannel: Ring requested on channel 0/1 already in use on span 2. Hanging up owner. Do anyone have a clue about it, as restarting * more often than not is not a really good solution? Thanks Laurent ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk consultants
chawki hammoud wrote: hi: i would greatly appreciate it if somebody can refer me to asterisk consultants. Hi, Here you are http://www.voip-info.org/wiki-Asterisk+consultants+Europe Laurent -- [EMAIL PROTECTED] ___ 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
