Re: [asterisk-users] Telco with multipe SIP servers
On 02/02/14 14:42, Markus Reschke wrote: Hi! My telco is Deutsche Telekom and they got about 30 SIP servers right now. Currently I've set up a template for incoming calls in sip.conf and added each SIP server by it's IP address like this: [DTAG-in-1](DTAG-in-template) host=217.0.16.103 ... [DTAG-in-30](DTAG-in-template) host=217.0.20.99 I've done that to improve security and to be able to assign all calls coming in via Deutsche Telekom to a dedicated dialplan context. Unfortunately this approach is not scalable and it's a PITA to maintain a list of server IP addresses since Deutsche Telekom will get more SIP servers in the future. They've started to migrate the classic POTS/ISDN network to VoIP, the goal is get it done by 2016. Customers with DSL get VoIP directly, i.e. they need SIP phones or a SIP PBX, and customers with a phone line only are converted by the MSAN. And they don't provide an official list of the SIP servers :-( By some reverse engineering I found out that all SIP servers are within a specific subnet. Is there any way to match peers by subnet(s) instead of FQDNs or single IP addresses? If not, it would be a feature really needed to be able to cope with telcos running multiple or tons of SIP servers. cu, Markus You could consider making use of opensips. We use it for inbound sip connections and its fairly easy to get it to perform a database lookup against a connecting IP address and pull out a record and pass that onto Asterisk using a custom header. Asterisk can then trust connections from opensips and you can read in the custom header and have the dialplan decide what to do based upon the value. -- _ -- 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] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote:
.
.
.
[sipcall.ch]
type=peer
insecure=invite
defaultuser=123456789
fromuser=123456789
fromdomain=voipdomain.com
secret=gueswhat
host=voipdomain.com
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=123456789
add
directmedia=no
setvar=FAXOPT(gateway)=no
change
insecure=port,invite
[fax-rx]
exten => receive,1,NoOp( FAX RECEIVE )
exten => receive,n,Set(GLOBAL(FAXCOUNT)=$["${GLOBAL(FAXCOUNT)}" + "1"])
exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
Do you want to keep your received faxes or is it OK to overwrite them
the next time asterisk is re-started!?
udptl.conf
[general]
udptlstart=4000
udptlend=4999
udptlfecentries = 3
udptlfecspan = 3
use_even_ports = no
You may want to change
use_even_ports=yes
You will need to restart Asterisk for this change.
Some other suggestion if the above doesn't help are;
faxdetect=cng
t38pt_udptl=no
Larry.
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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Hi, changing
faxdetect=cng
and
t38pt_udptl=no
helped making it work.
Thanks
Am 03.02.2014 11:57, schrieb Larry Moore:
On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote:
.
.
.
[sipcall.ch]
type=peer
insecure=invite
defaultuser=123456789
fromuser=123456789
fromdomain=voipdomain.com
secret=gueswhat
host=voipdomain.com
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=123456789
add
directmedia=no
setvar=FAXOPT(gateway)=no
change
insecure=port,invite
[fax-rx]
exten => receive,1,NoOp( FAX RECEIVE )
exten => receive,n,Set(GLOBAL(FAXCOUNT)=$["${GLOBAL(FAXCOUNT)}" + "1"])
exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
Do you want to keep your received faxes or is it OK to overwrite them
the next time asterisk is re-started!?
udptl.conf
[general]
udptlstart=4000
udptlend=4999
udptlfecentries = 3
udptlfecspan = 3
use_even_ports = no
You may want to change
use_even_ports=yes
You will need to restart Asterisk for this change.
Some other suggestion if the above doesn't help are;
faxdetect=cng
t38pt_udptl=no
Larry.
--
_
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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 7:15 PM, Jakob-Matthias Böttger wrote: Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Hmm, the fax will be received as an audio call rather than T.38, setting t38pt_udptl=no has turned off T.38. Do you know if your upstream provider supports T.38? Larry. -- _ -- 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] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data Am 03.02.2014 12:34, schrieb Larry Moore: On 3/02/2014 7:15 PM, Jakob-Matthias Böttger wrote: Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Hmm, the fax will be received as an audio call rather than T.38, setting t38pt_udptl=no has turned off T.38. Do you know if your upstream provider supports T.38? Larry. -- _ -- 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] call rejected because extension not found in context 'internal
Hi all, I want to two sip clients connect through Asterisk in local network for testing. My sip.conf file looks like this [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse localnet=192.168.1.0/255.255.255.0 [7001] type=friend host=dynamic secret=123abcd context=internal [7002] type=friend host=dynamic secret=456abcd context=internal Am using linphone as sip client and create account on linphone with user name 7001 and 7002 7001 is running on 192.168.2.15:5060 7002 is running on 192.168.2.45:5060 when i try to call from 7002 to 7001 i specified sip:7001@192.168.2.15 it working fine as i know ip adress i specified it as url. if i dnt know the ipadress how can i call to 7001? i try to call sip:7001@192.168.2.20 it through call rejected because extension not found in context 'internal, error. How can call to sip id with out knowning ipadress where it is runnning? Any modification required for sip.conf file? Thanks, Raghav -- _ -- 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] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. -- _ -- 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] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. after that i started udptl debug as well and now i'm getting lots of UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq 152, len 11) and in between [Feb 3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short and in the end [Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog '24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' with owner SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling destruction for 1 ms [Feb 3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535 generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7' failure, reason: 'fax session timed-out' (TIMEOUT) == Spawn extension (fax-rx, receive, 11) exited non-zero on 'SIP/sipcall.ch-0007' Thx, Jakob -- _ -- 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] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger: Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. after that i started udptl debug as well and now i'm getting lots of UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq 152, len 11) and in between [Feb 3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short and in the end [Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog '24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' with owner SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling destruction for 1 ms [Feb 3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535 generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7' failure, reason: 'fax session timed-out' (TIMEOUT) == Spawn extension (fax-rx, receive, 11) exited non-zero on 'SIP/sipcall.ch-0007' Thx, Jakob may do i have to open more ports then udp 1:2 (RTP), udp 4000:4999 (UDPTL) and tcp 5060,5061(SIP/TLS) -- _ -- 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 read IRQs and timing slips values
Follow up on this thread: Today, I could at last insert the TE220 board into an other machine in my lab. 1. For the moment, I plugged the card into an available PCIe x1 slot and checked IRQs with: # dmesg ... [ 42.969568] dahdi: Version: SVN-trunk-r10414 [ 43.281832] wct4xxp :04:08.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17 [ 43.281855] wct4xxp :04:08.0: 5th gen card with initial latency of 2 and 1 ms per IRQ [ 43.281861] wct4xxp :04:08.0: Firmware Version: c01a016d [ 43.284411] wct4xxp :04:08.0: FALC Framer Version: 3.1 [ 43.284420] IRQ 17/wct2xxp: IRQF_DISABLED is not guaranteed on shared IRQs [ 43.284465] wct4xxp :04:08.0: Found a Wildcard: Wildcard TE220 (5th Gen) [ 43.306504] wct4xxp :04:08.0: firmware: requesting dahdi-fw-oct6114-064.bin [ 43.568475] VPM450: echo cancellation for 64 channels [ 48.722467] wct4xxp :04:08.0: VPM450: hardware DTMF disabled. [ 48.722471] wct4xxp :04:08.0: VPM450: Present and operational servicing 2 span(s) [ 50.304512] wct4xxp :04:08.0: TE2XXP: Span 1 configured for CCS/HDB3 [ 50.304543] wct4xxp :04:08.0: RCLK source set to span 1 [ 50.304547] wct4xxp :04:08.0: Recovered timing mode, RCLK set to span 1 [ 50.304615] wct4xxp :04:08.0: SPAN 1: Primary Sync Source # cat /proc/interrupts CPU0 CPU1 0: 46 24 IO-APIC-edge timer 1: 0 2 IO-APIC-edge i8042 7: 1 0 IO-APIC-edge parport0 8: 0 1 IO-APIC-edge rtc0 9: 0 0 IO-APIC-fasteoi acpi 12: 0 6 IO-APIC-edge i8042 14: 0 0 IO-APIC-edge pata_atiixp 15: 0 0 IO-APIC-edge pata_atiixp 16: 1339 IO-APIC-fasteoi xhci_hcd:usb3, ohci_hcd:usb4, ohci_hcd:usb5, HDA Intel 17: 22167235841 IO-APIC-fasteoi ehci_hcd:usb1, wct2xxp ... Does it qualify as "incompatible with Dahdi because of shared IRQ" or can I get along and keep testing on this machine ? 2. I plugged a straight patch cord into one E1 port and the other cord's plug into the small socket provided with card. (I'm referring here to a female RJ45 socket mounted on a 1cm x 2cm PCB) The port light turns green and dahdi_tool says status is OK. At the same time, I'm seeing timing slips as with: # cat /proc/dahdi/1 Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS ClockSource Timing slips: 75 The rate is 3 slips/minute. Has this figure any meaning (I don't have any other card to compare with) ? 3. When I'm plugging the same E1 port into a Smartnode E1 gateway port (then using an E1 crossover cable), dahdi_tool tells status is OK but asterisk logs are full of lines such as (~ 20 occurences per second)): [2014-02-03 13:46:04] NOTICE[2076]: chan_dahdi.c:3099 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 1 Sometimes, a line like the one bellow slides between the above ones. [2014-02-03 13:50:24] NOTICE[2076]: chan_dahdi.c:3099 my_handle_dchan_exception: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 Thoughts ? 2014-01-15 Olivier : > I could replace the card this morning and the timing slips disappeared. > > Given Adrian's testimony, that doesn't mean the card is faulty but as the > card is now off service, I'm really eagger to investigate further. > > At the moment, I can't insert this card and test it again in my lab but > I'll certainly do it within a couple of hours (and report here). > Stay tuned ... > > > > 2014/1/14 Shaun Ruffell > >> On Mon, Jan 13, 2014 at 08:42:13PM -0500, Paul Belanger wrote: >> > > cat /proc/dahdi/2 >> > > Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2" (MASTER) HDB3/CCS >> > > Timing slips: 175319 >> > > >> > > 32 TE2/0/2/1 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 33 TE2/0/2/2 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 34 TE2/0/2/3 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 35 TE2/0/2/4 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 36 TE2/0/2/5 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 37 TE2/0/2/6 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 38 TE2/0/2/7 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 39 TE2/0/2/8 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 40 TE2/0/2/9 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 41 TE2/0/2/10 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 42 TE2/0/2/11 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 43 TE2/0/2/12 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 44 TE2/0/2/13 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 45 TE2/0/2/14 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 46 TE2/0/2/15 Clear (In use) (EC: VPMOCT064 - INACTIVE) >> > > 47 TE2/0/2/16 HDLCFCS (In use) (EC: VPMOCT064 - INACTIVE) >> > > 48 TE2/0/2/17 Clear (In use) (EC: VPMOCT064 - INACTIVE) >
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 8:42 PM, Jakob-Matthias Böttger wrote: Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger: Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. after that i started udptl debug as well and now i'm getting lots of UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq 152, len 11) and in between [Feb 3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short and in the end [Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog '24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' with owner SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling destruction for 1 ms [Feb 3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535 generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7' failure, reason: 'fax session timed-out' (TIMEOUT) == Spawn extension (fax-rx, receive, 11) exited non-zero on 'SIP/sipcall.ch-0007' Thx, Jakob may do i have to open more ports then udp 1:2 (RTP), udp 4000:4999 (UDPTL) and tcp 5060,5061(SIP/TLS) The T.38 connection will be attempted when ReceiveFax is called. The port number to use should be in the SDP information, yes, allow udp ports 4000-4999 in and out. If your firewall can be so configured you could set it to allow traffic in and out based upon the user ID Asterisk is running as, assuming it is using a unique unprivileged id. You may like to try the following to see if your SIP provider will initiate a T.38 re-invite. sip.conf [general] faxdetect=t38 [sipcall.ch] directmedia=no In extensions.conf change Wait(2) to Wait(5), if your VSP sends you a T.38 re-invite this will trigger the switch to the Fax extension. If this proves successful you can work on removing the Wait() from your dialplan as Asterisk will remain in the audio path and should successfully switch to the fax extension if extension 200 or 201 answer a call that happens to be a fax. Larry. -- _ -- 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] call rejected because extension not found in context 'internal
Howdy, Your sip.conf file looks fine for some testing, though I would recommend _not_ using an extension number to name a sip endpoint. Instead, name the sip endpoint something more descriptive of the device. [Linphone-01] [Linphone-02] for example. Then you'll want to configure extensions.conf to Dial() the sip endpoint whenever the extension is dialed. Justin Hester Digium, Inc. · Technical Trainer 445 Jan Davis Drive NW · Huntsville, AL 35806 · USA ph: +1 256 428 6238 Check us out at: http://digium.com · http://asterisk.org On Mon, Feb 3, 2014 at 5:45 AM, Raghav Goud wrote: > Hi all, > >I want to two sip clients connect through Asterisk in local network for > testing. My sip.conf file looks like this > > [general] > context=internal > allowguest=no > allowoverlap=no > bindport=5060 > bindaddr=0.0.0.0 > srvlookup=no > disallow=all > allow=ulaw > alwaysauthreject=yes > canreinvite=no > nat=yes > session-timers=refuse > localnet=192.168.1.0/255.255.255.0 > > [7001] > type=friend > host=dynamic > secret=123abcd > context=internal > > [7002] > type=friend > host=dynamic > secret=456abcd > context=internal > > > Am using linphone as sip client and create account on linphone with user > name 7001 and 7002 > 7001 is running on 192.168.2.15:5060 > 7002 is running on 192.168.2.45:5060 > > when i try to call from 7002 to 7001 i specified sip:7001@192.168.2.15 it > working fine as i know ip adress i specified it as url. if i dnt know the > ipadress how can i call to 7001? i try to call sip:7001@192.168.2.20 it > through call rejected because extension not found in context 'internal, > error. > > How can call to sip id with out knowning ipadress where it is runnning? > Any modification required for sip.conf file? > > Thanks, > Raghav > > > > > -- > _ > -- 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 read IRQs and timing slips values
On Mon, Feb 3, 2014 at 6:52 AM, Olivier wrote: > [ 42.969568] dahdi: Version: SVN-trunk-r10414 I would suggest updating DAHDI. We moved from svn to git years ago and that repository is not maintained. > 17: 22167235841 IO-APIC-fasteoi ehci_hcd:usb1, wct2xxp > Does it qualify as "incompatible with Dahdi because of shared IRQ" or can I > get along and keep testing on this machine ? This is fine. I would only worry about sharing irqs with hardware that would compete for interrupt time, like a network controller or disk controller. > At the same time, I'm seeing timing slips as with: > The rate is 3 slips/minute. > Has this figure any meaning (I don't have any other card to compare with) ? 3 slips a minute is too high. You'll likely hear audio artifacts like clicks or pops if you were listening to audio on the line. This may give you a bit more info on acceptable slip rates. http://www.youtube.com/watch?v=2UNfzEuv1uk#t=1578 > my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of > Thoughts ? HDLC aborts are a symptom of timing slips, typically. If you're legitimately recovering timing from the network properly, then I would look at what the upstream provider is doing. Your provider should be able to run long term loop/pattern tests to your smartjack to determine link quality issues. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com & www.asterisk.org -- _ -- 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
