Re: [Asterisk-Users] Convert you FXS port to FXO cheap
Title: Message Hi, I am the retailer. You can also purchase whole sale from me. This device is manufactured by my company. Thanks Dave - Original Message - From: Abdul Hakeem To: [EMAIL PROTECTED] Sent: Sunday, March 23, 2003 1:28 PM Subject: RE: [Asterisk-Users] Convert you FXS port to FXO cheap Hi, Could you pass on the details of the retailer/manufacturer ? Cheers, Abdul Hakeem -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: Sunday, March 23, 2003 8:49 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Convert you FXS port to FXO cheap If you have an FXS port and would like to attach a PSTN analog line to it this device would do the job by converting the FXS port to FXO. Its a small external device. Works well with VOIP FXS and other FXS interfaces. Interface: 2 RJ11 Jacks (one for the FXS port and one for the PSTN outlet.) Cost: $35.00 with USPS regular mail included. Power: 9 to 20V dc power supply (Not included) Thanks Dave
Re: [Asterisk-Users] Convert you FXS port to FXO cheap
Title: Message Ray, The name of the device is FXS to FXO converter. You need a device for each port. Thanks Dave - Original Message - From: Ray Dzek To: [EMAIL PROTECTED] Sent: Sunday, March 23, 2003 2:47 PM Subject: Re: [Asterisk-Users] Convert you FXS port to FXO cheap Well the obvious question is... Is this something that is supported in *? So for instance, if I have an 8 port MGCP VOIP box, can I "convert" a couple of those to FXO and have * utilize them? - Original Message - From: Dave George To: [EMAIL PROTECTED] Sent: Sunday, March 23, 2003 2:04 PM Subject: Re: [Asterisk-Users] Convert you FXS port to FXO cheap Hi, I am the retailer. You can also purchase whole sale from me. This device is manufactured by my company. Thanks Dave - Original Message - From: Abdul Hakeem To: [EMAIL PROTECTED] Sent: Sunday, March 23, 2003 1:28 PM Subject: RE: [Asterisk-Users] Convert you FXS port to FXO cheap Hi, Could you pass on the details of the retailer/manufacturer ? Cheers, Abdul Hakeem -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: Sunday, March 23, 2003 8:49 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Convert you FXS port to FXO cheap If you have an FXS port and would like to attach a PSTN analog line to it this device would do the job by converting the FXS port to FXO. Its a small external device. Works well with VOIP FXS and other FXS interfaces. Interface: 2 RJ11 Jacks (one for the FXS port and one for the PSTN outlet.) Cost: $35.00 with USPS regular mail included. Power: 9 to 20V dc power supply (Not included) Thanks Dave
[asterisk-users] sip attacks
My asterisk server is getting bogged down every 5 minutes. My ping time is going from 60ms to 800 ms and the call quality is bad. I have fail2ban running and I am using iptables. I have two ip connections to the box. How can I tell if the poor performance is due to sip attacks? I don't see any reg attempts in my asterisk cli. I use to get frequent attacks but fail2ban seems to be taking care of that. See how ping time gets worst in a short space of time and server performance at the time: 64 bytes from 4.2.2.1: icmp_seq=6 ttl=55 time=87.8 ms 64 bytes from 4.2.2.1: icmp_seq=7 ttl=55 time=99.8 ms 64 bytes from 4.2.2.1: icmp_seq=8 ttl=55 time=107 ms 64 bytes from 4.2.2.1: icmp_seq=9 ttl=55 time=115 ms 64 bytes from 4.2.2.1: icmp_seq=10 ttl=55 time=120 ms 64 bytes from 4.2.2.1: icmp_seq=11 ttl=55 time=122 ms 64 bytes from 4.2.2.1: icmp_seq=12 ttl=55 time=123 ms 64 bytes from 4.2.2.1: icmp_seq=13 ttl=55 time=126 ms 64 bytes from 4.2.2.1: icmp_seq=14 ttl=55 time=122 ms 64 bytes from 4.2.2.1: icmp_seq=15 ttl=55 time=142 ms 64 bytes from 4.2.2.1: icmp_seq=16 ttl=55 time=142 ms 64 bytes from 4.2.2.1: icmp_seq=17 ttl=55 time=137 ms 64 bytes from 4.2.2.1: icmp_seq=18 ttl=55 time=186 ms 64 bytes from 4.2.2.1: icmp_seq=19 ttl=55 time=255 ms 64 bytes from 4.2.2.1: icmp_seq=20 ttl=55 time=310 ms 64 bytes from 4.2.2.1: icmp_seq=21 ttl=55 time=387 ms 64 bytes from 4.2.2.1: icmp_seq=22 ttl=55 time=445 ms 64 bytes from 4.2.2.1: icmp_seq=23 ttl=55 time=514 ms 64 bytes from 4.2.2.1: icmp_seq=24 ttl=55 time=583 ms 64 bytes from 4.2.2.1: icmp_seq=25 ttl=55 time=650 ms 64 bytes from 4.2.2.1: icmp_seq=26 ttl=55 time=715 ms 64 bytes from 4.2.2.1: icmp_seq=27 ttl=55 time=783 ms 64 bytes from 4.2.2.1: icmp_seq=28 ttl=55 time=821 ms 64 bytes from 4.2.2.1: icmp_seq=29 ttl=55 time=810 ms 64 bytes from 4.2.2.1: icmp_seq=30 ttl=55 time=832 ms 64 bytes from 4.2.2.1: icmp_seq=31 ttl=55 time=812 ms 64 bytes from 4.2.2.1: icmp_seq=32 ttl=55 time=821 ms 64 bytes from 4.2.2.1: icmp_seq=33 ttl=55 time=826 ms 64 bytes from 4.2.2.1: icmp_seq=34 ttl=55 time=815 ms 64 bytes from 4.2.2.1: icmp_seq=35 ttl=55 time=821 ms 64 bytes from 4.2.2.1: icmp_seq=36 ttl=55 time=824 ms top - 19:02:38 up 4 days, 11:26, 4 users, load average: 0.36, 0.75, 0.82 Mem: 4051312k total, 1062964k used, 2988348k free, 167004k buffers Swap: 6094840k total,0k used, 6094840k free, 680144k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 4245 root 15 0 791m 86m 10m S 39.6 2.2 1192:32 asterisk 18280 root 15 0 3812 600 516 S 2.0 0.0 0:59.00 pppoe 2582 root 15 0 5912 628 504 S 0.3 0.0 2:02.19 syslogd 18978 root 15 0 12744 1096 812 R 0.3 0.0 0:00.02 top 1 root 15 0 10352 700 588 S 0.0 0.0 0:01.14 init 2 root RT -5 000 S 0.0 0.0 0:00.01 migration/0 3 root 34 19 000 S 0.0 0.0 0:31.90 ksoftirqd/0 4 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/0 5 root RT -5 000 S 0.0 0.0 0:00.01 migration/1 6 root 34 19 000 S 0.0 0.0 0:08.43 ksoftirqd/1 7 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/1 8 root RT -5 000 S 0.0 0.0 0:00.13 migration/2 9 root 34 19 000 S 0.0 0.0 2:40.56 ksoftirqd/2 10 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/2 11 root RT -5 000 S 0.0 0.0 0:00.05 migration/3 12 root 34 19 000 S 0.0 0.0 0:44.56 ksoftirqd/3 13 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/3 14 root 10 -5 000 S 0.0 0.0 0:00.02 events/0 15 root 10 -5 000 S 0.0 0.0 0:00.00 events/1 16 root 10 -5 000 S 0.0 0.0 0:00.00 events/2 17 root 10 -5 000 S 0.0 0.0 0:00.00 events/3 18 root 10 -5 000 S 0.0 0.0 0:00.00 khelper 55 root 10 -5 000 S 0.0 0.0 0:00.00 kthread 62 root 10 -5 000 S 0.0 0.0 0:00.07 kblockd/0 63 root 10 -5 000 S 0.0 0.0 0:00.01 kblockd/1 64 root 10 -5 000 S 0.0 0.0 0:00.00 kblockd/2 65 root 10 -5 000 S 0.0 0.0 0:00.00 kblockd/3 66 root 17 -5 000 S 0.0 0.0 0:00.00 kacpid 166 root 17 -5 000 S 0.0 0.0 0:00.00 cqueue/0 167 root 18 -5 000 S 0.0 0.0 0:00.00 cqueue/1 Dave -- _ -- 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] TE410P hardware problems
TE410P card down. I have three (3) TE410P in one machine running asterisk with SS7. My problems started last week when one of my cards started switching to E1 every time after reboot. I set the following in dahdi.conf and that solve the problem. /etc/modprobe.d/ options wct4xxp t1e1override=0x00 Now all 4 ports on that card is down with Red Alarm. I tried rebooting the machine and restarting dahdi with no luck. The other two cards are working fine. I put a loop plug the ports and same problem. dahdi_scan [9] active=yes alarms=RED/LFA/LMFA description=T4XXP (PCI) Card 2 Span 1 name=TE4/2/1 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=193 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [10] active=yes alarms=RED/LFA/LMFA description=T4XXP (PCI) Card 2 Span 2 name=TE4/2/2 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=217 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [11] active=yes alarms=RED/LFA/LMFA description=T4XXP (PCI) Card 2 Span 3 name=TE4/2/3 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=241 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [12] active=yes alarms=RED/LFA/LMFA description=T4XXP (PCI) Card 2 Span 4 name=TE4/2/4 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=265 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF lspci 09:04.0 Communication controller: Digium, Inc. Wildcard TE410P/TE412P quad-span T1/E1/J1 card 3.3V (rev 02) 0a:02.0 Communication controller: Digium, Inc. Wildcard TE410P/TE412P quad-span T1/E1/J1 card 3.3V (rev 02) 0a:03.0 Communication controller: Digium, Inc. Wildcard TE410P/TE412P quad-span T1/E1/J1 card 3.3V (rev 02) Thanks Dave -- _ -- 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] TE410P hardware problems
Hi, We opened the server an checked that the cards were seated correctly and they are. I will have the tech completely remove them tomorrow and try again. I will post the results. Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Tuesday, August 02, 2011 10:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TE410P hardware problems On Tue, Aug 02, 2011 at 09:22:46PM -0400, Dave George wrote: > TE410P card down. > > I have three (3) TE410P in one machine running asterisk with SS7. > > My problems started last week when one of my cards started switching > to E1 every time after reboot. I set the following in dahdi.conf and > that solve the problem. > > /etc/modprobe.d/ > options wct4xxp t1e1override=0x00 > > Now all 4 ports on that card is down with Red Alarm. I tried > rebooting the machine and restarting dahdi with no luck. The other > two cards are working fine. I put a loop plug the ports and same problem. Any strange output in dmesg? Best guess is either the card has failed or it has started to unseat from it's PCI slot. Normally when it starts to unseat you'll see "Version Synchronization Errors" in dmesg when trying to load the card. -- 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 -- _ -- 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] TE410P hardware problems
dmesg: wct4xxp :0a:03.0: SPAN 9: Primary Sync Source wct4xxp :0a:03.0: Span 2 configured for ESF/B8ZS wct4xxp :0a:03.0: SPAN 10: Primary Sync Source wct4xxp :0a:03.0: All spans in alarm : No validspan to source RCLK from wct4xxp :0a:03.0: RCLK source set to span 1 wct4xxp :0a:03.0: Span 3 configured for ESF/B8ZS wct4xxp :0a:03.0: SPAN 11: Primary Sync Source wct4xxp :0a:03.0: Span 4 configured for ESF/B8ZS wct4xxp :0a:03.0: SPAN 12: Primary Sync Source wct4xxp :0a:03.0: All spans in alarm : No validspan to source RCLK from wct4xxp :0a:03.0: RCLK source set to span 1 system.conf span=1,1,0,esf,b8zs bchan=2-24 mtp2=1 span=2,1,0,esf,b8zs bchan=26-48 mtp2=25 span=3,1,0,esf,b8zs bchan=49-72 span=4,1,0,esf,b8zs bchan=73-96 span=5,1,0,esf,b8zs bchan=97-120 span=6,1,0,esf,b8zs bchan=121-144 span=7,1,0,esf,b8zs bchan=145-168 span=8,1,0,esf,b8zs bchan=169-192 span=9,1,0,esf,b8zs bchan=193-216 span=10,1,0,esf,b8zs bchan=217-240 span=11,1,0,esf,b8zs bchan=241-264 span=12,1,0,esf,b8zs bchan=265-288 Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Tuesday, August 02, 2011 10:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TE410P hardware problems On Tue, Aug 02, 2011 at 09:22:46PM -0400, Dave George wrote: > TE410P card down. > > I have three (3) TE410P in one machine running asterisk with SS7. > > My problems started last week when one of my cards started switching > to E1 every time after reboot. I set the following in dahdi.conf and > that solve the problem. > > /etc/modprobe.d/ > options wct4xxp t1e1override=0x00 > > Now all 4 ports on that card is down with Red Alarm. I tried > rebooting the machine and restarting dahdi with no luck. The other > two cards are working fine. I put a loop plug the ports and same problem. Any strange output in dmesg? Best guess is either the card has failed or it has started to unseat from it's PCI slot. Normally when it starts to unseat you'll see "Version Synchronization Errors" in dmesg when trying to load the card. -- 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 -- _ -- 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] TE410P hardware problems
I opened the jumpers on the card putting them in T1 mode and it worked. I had them set to T1 using the options in the dahdi.conf file under /etc/modprobe.d/ That worked well for over a year until it started acting up. Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Tuesday, August 02, 2011 11:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TE410P hardware problems If it doesn't go green when you put a hard loopback on the port, then contact Digium support. > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Dave George > Sent: Tuesday, August 02, 2011 10:52 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] TE410P hardware problems > > dmesg: > > wct4xxp :0a:03.0: SPAN 9: Primary Sync Source wct4xxp > :0a:03.0: Span 2 configured for ESF/B8ZS wct4xxp :0a:03.0: > SPAN 10: Primary Sync Source wct4xxp :0a:03.0: All spans in alarm > : No validspan to source RCLK from wct4xxp :0a:03.0: RCLK source > set to span 1 wct4xxp :0a:03.0: Span 3 configured for ESF/B8ZS > wct4xxp :0a:03.0: SPAN 11: Primary Sync Source wct4xxp > :0a:03.0: Span 4 configured for ESF/B8ZS wct4xxp :0a:03.0: > SPAN 12: Primary Sync Source wct4xxp :0a:03.0: All spans in alarm > : No validspan to source RCLK from wct4xxp :0a:03.0: RCLK source > set to span 1 -- _ -- 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
[asterisk-users] No IVR audio. Jump in RTP sequence number
My users dial *120 get to an IVR menu that plays their balance and then
ask them for a voucher. Ater the balance is played and the request for
the voucher is played the user don't hear any other audio from the
asterisk box. I can see the asterisk server playing the files to ask
for the voucher again but the user cannot hear any thing.
Has any one seens this issue with IVRs. I notice a change in RTP
sequence when voucher is being requested again.
sip debug
<--- SIP read from UDP:x.x.x.x:5060 --->
INVITE sip:*120@a.b.c.d SIP/2.0
Via: SIP/2.0/UDP x.x.x.x;rport;branch=z9hG4bK2vr8B7r60vS7j
Max-Forwards: 70
From: "14735201326" ;tag=0K219XHeF7K2j
To:
Call-ID: 00ddbda6-d9b1-122f-e7a7-00259025b560
CSeq: 24716447 INVITE
Contact:
User-Agent: Wireless Call Manager
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 231
Remote-Party-ID: "14735201326"
;party=calling;screen=yes;privacy=off
v=0
o=wCM 1330087502 1330087503 IN IP4 x.x.x.x
s=wCM
c=IN IP4 x.x.x.x
t=0 0
m=audio 17520 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<->
--- (16 headers 11 lines) ---
Sending to x.x.x.x:5060 (no NAT)
Using INVITE request as basis request -
00ddbda6-d9b1-122f-e7a7-00259025b560
Found peer 'STARMG1' for '14735201326' from x.x.x.x:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port x.x.x.x:17520
Looking for *120 in spicemobile (domain a.b.c.d)
list_route: hop:
<--- Transmitting (no NAT) to x.x.x.x:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
x.x.x.x;branch=z9hG4bK2vr8B7r60vS7j;received=x.x.x.x;rport=5060
From: "14735201326" ;tag=0K219XHeF7K2j
To:
Call-ID: 00ddbda6-d9b1-122f-e7a7-00259025b560
CSeq: 24716447 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0
<>
<--- SIP read from UDP:x.x.x.x:5060 --->
INVITE sip:*120@a.b.c.d SIP/2.0
Via: SIP/2.0/UDP x.x.x.x;rport;branch=z9hG4bK2vr8B7r60vS7j
Max-Forwards: 70
From: "14735201326" ;tag=0K219XHeF7K2j
To:
Call-ID: 00ddbda6-d9b1-122f-e7a7-00259025b560
CSeq: 24716447 INVITE
Contact:
User-Agent: Wireless Call Manager
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 231
Remote-Party-ID: "14735201326"
;party=calling;screen=yes;privacy=off
v=0
o=wCM 1330087502 1330087503 IN IP4 x.x.x.x
s=wCM
c=IN IP4 x.x.x.x
t=0 0
m=audio 17520 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<->
--- (16 headers 11 lines) ---
Ignoring this INVITE request
<--- Transmitting (no NAT) to x.x.x.x:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
x.x.x.x;branch=z9hG4bK2vr8B7r60vS7j;received=x.x.x.x;rport=5060
From: "14735201326" ;tag=0K219XHeF7K2j
To:
Call-ID: 00ddbda6-d9b1-122f-e7a7-00259025b560
CSeq: 24716447 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0
<>
-- Executing [*120@spicemobile:1] AGI("SIP/STARMG1-03c0",
"a2billing.php,6,voucher") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
AGI Tx >> agi_request: a2billing.php
AGI Tx >> agi_channel: SIP/STARMG1-03c0
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1330130582.960
AGI Tx >> agi_version: 1.8.7.1
AGI Tx >> agi_callerid: 14735201326
AGI Tx >> agi_calleridname: 14735201326
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: *120
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: spicemobile
AGI Tx >> agi_extension: *120
AGI Tx >> agi_priority: 1
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >> agi_threadid: 1118284096
AGI Tx >> agi_arg_1: 6
AGI Tx >> agi_arg_2: voucher
AGI Tx >>
AGI Rx << GET VARIABLE IDCONF
AGI Tx >> 200 result=0
AGI R
[asterisk-users] Loss of RTP stream during DTMF collection
I am using asterisk for voice mail. During DTMF collection Asterisk stop sending any RTP Packets. The gap between two consecutive packets are 4 seconds, which is huge enough to screw up the jitter buffer. When ever asterisk stops to receive DTMF, the RTP stream is cut and we loose audio. I don't have this issue when calling from a SIP phone. I only have this issue when calling from one media gateway to the asterisk box. Any suggestions welcome. Can I play some file in the back while collecting DTMF? Dave -- _ -- 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] Loss of RTP stream during DTMF collection
Hi Kevin, I have two asterisk boxes with the same issues. Box 1: asterisk ver 1.4.21.2 Box 2: Asterisk 1.8.7.1 setup: CDMA Phone <> CDMA Media Gateway WCM Asterisk voice mail The calls are SIP Based. DTMF collection is when the user is entering a password for voice mail access or voucher to recharge their account. voice mail: user is prompted for a password. After password is entered I can see asterisk playing the voice mail but no audio is heard on the phone. Other scenario user dials into a voucher menu (Asterisk2billing) and is prompted for a voucher. No audio after the voucher is entered. The CDMA guys did a trace on their end and this is what they explained is happening: The voicemail problem is due to the time stamp jump on the RTP steam sending WCM to BSC. There are about 5 seconds gap between two consecutive RTP packets. It was caused by Asterisk not sending any RTP packet to WCM. How can I enable the option to allow asterisk to maintain the RTP stream during DTMF collection? Thanks, Dave > Original Message > Subject: Re: [asterisk-users] Loss of RTP stream during DTMF collection > From: "Kevin P. Fleming" > Date: Fri, May 25, 2012 5:38 pm > To: asterisk-users@lists.digium.com > > > On 05/25/2012 04:30 PM, Dave George wrote: > > I am using asterisk for voice mail. During DTMF collection Asterisk > > stop sending any RTP Packets. The gap between two consecutive packets > > are 4 seconds, which is huge enough to screw up the jitter buffer. When > > ever asterisk stops to receive DTMF, the RTP stream is cut and we loose > > audio. > > > > I don't have this issue when calling from a SIP phone. I only have this > > issue when calling from one media gateway to the asterisk box. > > > > Any suggestions welcome. Can I play some file in the back while > > collecting DTMF? > > You are missing quite a lot of crucial information required for anyone > to help you. First, what version of Asterisk are you using? Second, what > type of channel is being used to connect to Asterisk? You mention it > works from a SIP phone, but not from a media gateway.. is that gateway > also using SIP, or something else? What does 'during DTMF collection' > mean? Do you mean after a prompt has been played and the voicemail > application is waiting for input, or is this during prompt playback, or > something else? > > Quite some time ago Asterisk was changed to ensure that silence would be > sent while an application was running and waiting for input from the > caller; if your version is older than this, then that could explain what > you are seeing. That's just a mildly-educated guess though. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > 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 -- _ -- 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] dahdi timing source multiple cards
I have a box with 12 T1s (4 Te410P cards). The PSTN provider is reporting slips and ask me to update the clock source. I have my system.conf set as the following but when I run dahdi_scan only the ports on Card 1 are showing up with syncsrc=1 system.conf : span=1,1,0,esf,b8zs bchan=2-24 mtp2=1 span=2,2,0,esf,b8zs bchan=26-48 mtp2=25 span=3,3,0,esf,b8zs bchan=49-72 span=4,4,0,esf,b8zs bchan=73-96 span=5,5,0,esf,b8zs bchan=97-120 span=6,6,0,esf,b8zs bchan=121-144 span=7,7,0,esf,b8zs bchan=145-168 span=8,8,0,esf,b8zs bchan=169-192 span=9,9,0,esf,b8zs bchan=193-216 span=10,10,0,esf,b8zs bchan=217-240 span=11,11,0,esf,b8zs bchan=241-264 span=12,12,0,esf,b8zs bchan=265-288 dahdi_scan : [1] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 1 name=TE4/0/1 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=1 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [2] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 2 name=TE4/0/2 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=25 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [3] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 3 name=TE4/0/3 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=49 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [4] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 4 name=TE4/0/4 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=73 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [5] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 1 name=TE4/1/1 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=97 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [6] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 2 name=TE4/1/2 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=121 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [7] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 3 name=TE4/1/3 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=145 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [8] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 4 name=TE4/1/4 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=169 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [9] active=yes alarms=OK description=T4XXP (PCI) Card 2 Span 1 name=TE4/2/1 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=193 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [10] active=yes alarms=OK description=T4XXP (PCI) Card 2 Span 2 name=TE4/2/2 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=217 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [11] active=yes alarms=OK description=T4XXP (PCI) Card 2 Span 3 name=TE4/2/3 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=241 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [12] active=yes alarms=OK description=T4XXP (PCI) Card 2 Span 4 name=TE4/2/4 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=265 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [root@aislecom28502 dahdi]# Thanks, Dave
Re: [asterisk-users] dahdi timing source multiple cards
Thanks Matt. The suggestion helped. No more slip erros. Dave > Original Message > Subject: Re: [asterisk-users] dahdi timing source multiple cards > From: Matthew Fredrickson > Date: Fri, December 21, 2012 3:41 pm > To: asterisk-users@lists.digium.com > > > You must make sure that for each card, the timing parameter does not > exceed the number of spans on the card (unless you're using a timing > cable between cards). So you probably don't want to have anything above > a 4 for the timing parameter... I see below that you have 5-12 listed in > the timing parameter for the spans on the other cards. > > You probably want something more like this: > > span=1,1,0,esf,b8zs > span=2,2,0,esf,b8zs > span=3,3,0,esf,b8zs > span=4,4,0,esf,b8zs > span=5,1,0,esf,b8zs > span=6,2,0,esf,b8zs > span=7,3,0,esf,b8zs > span=8,4,0,esf,b8zs > span=9,1,0,esf,b8zs > span=10,2,0,esf,b8zs > span=11,3,0,esf,b8zs > span=12,4,0,esf,b8zs > > Hope that helps. > > Matthew Fredrickson > Digium, Inc. > > On 12/20/12 10:42 PM, Dave George wrote: > > I have a box with 12 T1s (4 Te410P cards). The PSTN provider is > > reporting slips and ask me to update the clock source. I have my > > system.conf set as the following but when I run dahdi_scan only the > > ports on Card 1 are showing up with syncsrc=1 > > > > system.conf : > > > > span=1,1,0,esf,b8zs > > > > bchan=2-24 > > > > mtp2=1 > > > > span=2,2,0,esf,b8zs > > > > bchan=26-48 > > > > mtp2=25 > > > > span=3,3,0,esf,b8zs > > > > bchan=49-72 > > > > span=4,4,0,esf,b8zs > > > > bchan=73-96 > > > > span=5,5,0,esf,b8zs > > > > bchan=97-120 > > > > span=6,6,0,esf,b8zs > > > > bchan=121-144 > > > > span=7,7,0,esf,b8zs > > > > bchan=145-168 > > > > span=8,8,0,esf,b8zs > > > > bchan=169-192 > > > > span=9,9,0,esf,b8zs > > > > bchan=193-216 > > > > span=10,10,0,esf,b8zs > > > > bchan=217-240 > > > > span=11,11,0,esf,b8zs > > > > bchan=241-264 > > > > span=12,12,0,esf,b8zs > > > > bchan=265-288 > > > > dahdi_scan : > > > > [1] > > > > active=yes > > > > alarms=OK > > > > description=T4XXP (PCI) Card 0 Span 1 > > > > name=TE4/0/1 > > > > manufacturer=Digium > > > > devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) > > > > location=Board ID Switch 0 > > > > basechan=1 > > > > totchans=24 > > > > irq=225 > > > > type=digital-T1 > > > > syncsrc=1 > > > > lbo=0 db (CSU)/0-133 feet (DSX-1) > > > > coding_opts=B8ZS,AMI > > > > framing_opts=ESF,D4 > > > > coding=B8ZS > > > > framing=ESF > > > > [2] > > > > active=yes > > > > alarms=OK > > > > description=T4XXP (PCI) Card 0 Span 2 > > > > name=TE4/0/2 > > > > manufacturer=Digium > > > > devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) > > > > location=Board ID Switch 0 > > > > basechan=25 > > > > totchans=24 > > > > irq=225 > > > > type=digital-T1 > > > > syncsrc=1 > > > > lbo=0 db (CSU)/0-133 feet (DSX-1) > > > > coding_opts=B8ZS,AMI > > > > framing_opts=ESF,D4 > > > > coding=B8ZS > > > > framing=ESF > > > > [3] > > > > active=yes > > > > alarms=OK > > > > description=T4XXP (PCI) Card 0 Span 3 > > > > name=TE4/0/3 > > > > manufacturer=Digium > > > > devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) > > > > location=Board ID Switch 0 > > > > basechan=49 > > > > totchans=24 > > > > irq=225 > > > > type=digital-T1 > > > > syncsrc=1 > > > > lbo=0 db (CSU)/0-133 feet (DSX-1) > > > > coding_opts=B8ZS,AMI > > > > framing_opts=ESF,D4 > > > > coding=B8ZS > > > > framing=ESF > > > > [4] > > > > active=yes > > > > alarms=OK > > > > description=T4XXP (PCI) Card 0 Span 4 > > > > name=TE4/0/4 > > > > manufacturer=Digium > > > > devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) > > > > location=Board ID Switch 0 > > > > basechan=73 > > > > totchans=24 > > > > irq=225 > > > > type=digital-T1 > &
[asterisk-users] setting up callerid
I am using Asterisk 1.6.2.5-0 running on ubuntu and I have a problem
passing called ID on calls to the PSTN
When I make a call to the PSTN the caller-Id is showing up as
IMSI310410381554227
I want the number set in the callerid field to show up.
My peer is setup as follows:
[IMSI310410381554227]
canreinvite=no
type=peer
context=openbts
callerid=473520
disallow=all
allow=gsm
host=dynamic
dtmfmode=info
I use the following in extensions.conf to dial:
exten => _45.,1,Dial(SIP/${ext...@ss72)
Thanks,
Dave
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Re: [asterisk-users] setting up callerid
Tried the following but no luck:
exten => _53.,1,Set(CALLERID(num)=473520)
exten => _53.,n,Dial(SIP/${ext...@ss74)
I am still passing IMSI310410381554227 as the CALLERID.
My peer is setup as follows:
[IMSI310410381554227]
canreinvite=no
type=peer
context=openbts
callerid=473520
disallow=all
allow=gsm
host=dynamic
dtmfmode=info
Thanks,
Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten
Göllner
Sent: Monday, December 13, 2010 4:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] setting up callerid
Am 12.12.2010 20:49, schrieb dave george:
I am using Asterisk 1.6.2.5-0 running on ubuntu and I have a problem
passing called ID on calls to the PSTN
When I make a call to the PSTN the caller-Id is showing up as
IMSI310410381554227
I want the number set in the callerid field to show up.
My peer is setup as follows:
[IMSI310410381554227]
canreinvite=no
type=peer
context=openbts
callerid=473520
disallow=all
allow=gsm
host=dynamic
dtmfmode=info
I use the following in extensions.conf to dial:
exten => _45.,1,Dial(SIP/${ext...@ss72)
Thanks,
Dave
Take a look:
http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCallerID
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Re: [asterisk-users] setting up callerid
Tried the following but no luck:
exten => _53.,1,Set(CALLERID(num)=473520)
exten => _53.,n,Dial(SIP/${ext...@ss74)
I am still passing IMSI310410381554227 as the CALLERID.
My peer is setup as follows:
[IMSI310410381554227]
canreinvite=no
type=peer
context=openbts
callerid=473520
disallow=all
allow=gsm
host=dynamic
dtmfmode=info
Thanks,
Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten
Göllner
Sent: Monday, December 13, 2010 4:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] setting up callerid
Am 12.12.2010 20:49, schrieb dave george:
I am using Asterisk 1.6.2.5-0 running on ubuntu and I have a problem
passing called ID on calls to the PSTN
When I make a call to the PSTN the caller-Id is showing up as
IMSI310410381554227
I want the number set in the callerid field to show up.
My peer is setup as follows:
[IMSI310410381554227]
canreinvite=no
type=peer
context=openbts
callerid=473520
disallow=all
allow=gsm
host=dynamic
dtmfmode=info
I use the following in extensions.conf to dial:
exten => _45.,1,Dial(SIP/${ext...@ss72)
Thanks,
Dave
Take a look:
http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCallerID
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Re: [asterisk-users] setting up callerid
When I call from a mobile to mobile (both registered on OPENBTS) the correct
caller ID is passed. That is the callerid that I set in the callerid=
field.
When calling from openbts to the PSTN the config header is passed.
Thanks,
Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Friday, December 17, 2010 10:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] setting up callerid
Hi Dave,
On Thu, Dec 16, 2010 at 1:52 PM, dave george
wrote:
> Tried the following but no luck:
>
> exten => _53.,1,Set(CALLERID(num)=473520)
>
> exten => _53.,n,Dial(SIP/${ext...@ss74)
>
> I am still passing IMSI310410381554227 as the CALLERID.
>
> My peer is setup as follows:
>
> [IMSI310410381554227]
>
> canreinvite=no
>
> type=peer
>
> context=openbts
>
> callerid=473520
I see you are using OpenBTS. To my understanding, OpenBTS does not
support caller ID, so I don't think it can work.
But as I have the same issue as you, I'd be glad to be wrong ! :D Let me
know.
Regards
Axelle
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[asterisk-users] asterisk regiserted as a client check_auth: username mismatch on calls from client
I have my asterisk Server A registered as a client with another asterisk Server B. When I place a call from Server A to B I get the following: WARNING[834]: chan_sip.c:12673 check_auth: username mismatch, have , digest has NOTICE[834]: chan_sip.c:19961 handle_request_invite: Failed to authenticate device "0014734068436" ;tag=as0b50a2f9 I can call from server A to Server B with no problem My sip.conf (asterisk client A) register => openbts1:openbt...@x.x.x.x:5060 [SS72] type=peer canreinvite=no secret=openbts12 username=openbts1 host=x.x.x.x context=openbts fromuser=openbts1 dtmfmode=info fromdomain=x.x.x.x insecure=port,invite qualify=yes disallow=all allow=gsm sip.conf (asterisk Server B) [openbts1] type=peer username=openbts1 secret=openbts12 host=dynamic canreinvite=no qualify=yes context=local dtmfmode=rfc2833 nat=yes disallow=all allow=gsm any suggestions welcome. Dave -- _ -- 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] load balance with 2 wan connections
Need some advise or paid help on running asterisk on two WAN connection. I need load balancing and failover support. WAN: 1 DSL + 1 Cable ISP. Dave -- _ -- 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] load balance with 2 wan connections
Server will have two fix public ips. Dave > Original Message > Subject: Re: [asterisk-users] load balance with 2 wan connections > From: Alejandro Imass > Date: Sat, December 25, 2010 1:58 pm > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > On Sat, Dec 25, 2010 at 1:18 PM, dave george wrote: > > Need some advise or paid help on running asterisk on two WAN connection. �I > > need load balancing and failover support. > > > > WAN: 1 DSL + 1 Cable ISP. > > > > There are _many_ issues. First outgoing and incoming traffic is > completely different for what you want to do. > > Second SIP is hard enough to NAT and route with a single IP let alone > 2 or more and probably dynamic! > > Third, load balancing/fail-over is not a simple matter even doing by > hand with Linux or BSD, there will still be issues with static routing > and such. There are some cheap hw that may claim it does, but most > probably it will not be meant for VoIP, SIP or IAX. > > Depending on your budget and needs, if you need reliability and high > bandwidth, probably a better solution is to host your main pbx in a > reliable server on a fixed and public IP and then route the calls to a > local Asterisk using IAX and even SIP. If local bandwidth is limited > IAX is a better bet. By having a public box routing calls to local > box(es) on your private LAN, you could load balance with multiple > local Asterisk servers (easy balance by dialplan, for example). To > save on hardware, you could use virtualization or FreeBSD Jails for > example. Dunno how the telephony hw works with virtualization or jails > (yet, thoug I do have a single Asterisk running on a FBSD jail). > > Good luck, > Alejandro Imass > > > > > Dave > > > > > > -- > > _ > > -- 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
[asterisk-users] sip attack.. fail2ban not stopping attack
My server is being attached all day and fail2ban is not stopping the attack. I updated stamstamp to match fail2ban requirements. [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 handle_request_register: Registration from '"7002" ' failed for '38.108.40.94' - No matching peer found [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 handle_request_register: Registration from '"7002" ' failed for '38.108.40.94' - No matching peer found [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 handle_request_register: Registration from '"7002" ' failed for '38.108.40.94' - No matching peer found [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 handle_request_register: Registration from '"7002" ' failed for '38.108.40.94' - No matching peer found [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 handle_request_register: Registration from '"7002" ' failed for '38.108.40.94' - No matching peer found [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 handle_request_register: Registration from '"7002" ' failed for '38.108.40.94' - No matching peer found [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 handle_request_register: Registration from '"7002" ' failed for '38.108.40.94' - No matching peer found [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 handle_request_register: Registration from '"7002" ' failed for '38.108.40.94' - No matching peer found [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 handle_request_register: Registration from '"7002" ' failed for '38.108.40.94' - No matching peer found [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 handle_request_register: Registration from '"7002" -- _ -- 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] sip attack.. fail2ban not stopping attack
Yes we have that set in logger.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Ustinov Sent: Saturday, December 25, 2010 6:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip attack.. fail2ban not stopping attack Make sure you have dateformat=%F %T in logger.conf On Sun, Dec 26, 2010 at 1:04 AM, Dave George wrote: > My server is being attached all day and fail2ban is not stopping the > attack. I updated stamstamp to match fail2ban requirements. > > [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 > handle_request_register: Registration from '"7002" ' > failed for '38.108.40.94' - No matching peer found > [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 > handle_request_register: Registration from '"7002" ' > failed for '38.108.40.94' - No matching peer found > [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 > handle_request_register: Registration from '"7002" ' > failed for '38.108.40.94' - No matching peer found > [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 > handle_request_register: Registration from '"7002" ' > failed for '38.108.40.94' - No matching peer found > [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 > handle_request_register: Registration from '"7002" ' > failed for '38.108.40.94' - No matching peer found > [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 > handle_request_register: Registration from '"7002" ' > failed for '38.108.40.94' - No matching peer found > [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 > handle_request_register: Registration from '"7002" ' > failed for '38.108.40.94' - No matching peer found > [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 > handle_request_register: Registration from '"7002" ' > failed for '38.108.40.94' - No matching peer found > [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 > handle_request_register: Registration from '"7002" ' > failed for '38.108.40.94' - No matching peer found > [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 > handle_request_register: Registration from '"7002" > Dave > > > > -- > _ > -- 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] load balance with 2 wan connections
We have another gateway in the USA that will send traffic to both IPs. The US gateway will load balance the traffic to both IPs. This is not used for phones. It is used mainly for wholesale traffic. Asterisk is being used as an SS7 gateway. Each DSL limits us to about 16 calls. We are thinking of combining DSL + DSL + Cable ISP on the same box and have our USA box send traffic to all 3 IPS. Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood McGowan Sent: Monday, December 27, 2010 4:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] load balance with 2 wan connections The biggest issue with any solution to use two different providers for your IP service that will be used by your VOIP provider to deliver calls to your Asterisk server, is that each internet service will have a separate address. Therefore, for INBOUND calls, your VOIP provider will have to do the load balancing. For outbound calls, it won't be that hard as long as your provider allows you to send calls from both IP addresses. -- _ -- 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] sip attack.. fail2ban not stopping attack
jail.conf
[asterisk-iptables]
enabled = true
filter = asterisk
action = iptables-allports[name=ASTERISK, protocol=all]
sendmail-whois[name=ASTERISK, dest=root,
sender=fail2...@example.org]
logpath = /var/log/asterisk/messages
maxretry = 5
bantime = 259200
filter asterisk.conf
[INCLUDES]
# Read common prefixes. If any customizations available -- read them from
# common.local
#before = common.conf
[Definition]
#_daemon = asterisk
# Option: failregex
# Notes.: regex to match the password failures messages in the logfile. The
# host must be matched by a group named "host". The tag ""
can
# be used for standard IP/hostname matching and is only an alias
for
# (?:::f{4,6}:)?(?P\S+)
# Values: TEXT
#
failregex = NOTICE.* .*: Registration from '.*' failed for '' - Wrong
password
NOTICE.* .*: Registration from '.*' failed for '' - No
matching peer found
NOTICE.* .*: Registration from '.*' failed for '' -
Username/auth name mismatch
NOTICE.* .*: Registration from '.*' failed for '' - Device
does not match ACL
NOTICE.* failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' \(from \)
NOTICE.* .*: Host failed MD5 authentication for '.*' (.*)
NOTICE.* .*: Failed to authenticate user .*@.*
ignoreregex =
logger.conf
[general]
;
; Customize the display of debug message time stamps
; this example is the ISO 8601 date format (-mm-dd HH:MM:SS)
;
; see strftime(3) Linux manual for format specifiers. Note that there is
also
; a fractional second parameter which may be used in this field. Use %1q
; for tenths, %2q for hundredths, etc.
;
dateformat=%F %T ; ISO 8601 date format
;dateformat=%F %T.%3q ; with milliseconds
Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
Sent: Monday, December 27, 2010 5:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip attack.. fail2ban not stopping attack
On Sat, Dec 25, 2010 at 04:04:59PM -0700, Dave George wrote:
> My server is being attached all day and fail2ban is not stopping the
> attack. I updated stamstamp to match fail2ban requirements.
How about posting your fail2ban config?
--
Daniel Tryba
--
_
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--
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[asterisk-users] no ring back 180 with SDP
I have a box (Genband) expecting the following: 100 trying 180 ringing with SDP Or 100 trying 183 with SDP And asterisk is sending: 100 trying 180 ringing 183 with SDP Any way to modify asterisk to send what he is expecting? Thanks, Dave -- _ -- 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] no ring back 180 with SDP
I did ask this question. However it's a big carrier using Genband and they don't care. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, June 11, 2010 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] no ring back 180 with SDP On Friday 11 June 2010 09:31:43 dave george wrote: > Any way to modify asterisk to send what he is expecting? Probably, but what you really should be asking is why the endpoint is not RFC-compliant. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) 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 -- _ -- 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] no ring back 180 with SDP
I tried no, yes and never in the sip profile for that carrier and it did not make a difference. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von Klitzing Sent: Friday, June 11, 2010 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] no ring back 180 with SDP Hi! > I did ask this question. However it's a big carrier using Genband and > they don't care. Look at "progressinband=" in sip.conf. http://www.voip-info.org/wiki/view/Asterisk+sip+progressinband Philipp -- _ -- 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] no ring back 180 with SDP
I set it under the sip profile for the box sending calls to asterisk. [BREKEKE] type=peer context=wholesale host=x.x.x.x nat=no canreinvite=no progressinband=yes dtmfmode=rfc2833 insecure=port disallow=all allow=g729 Thanks, Dave George -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von Klitzing Sent: Friday, June 11, 2010 5:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] no ring back 180 with SDP Hi! > I tried no, yes and never in the sip profile for that carrier and it did > not make a difference. > > Look at "progressinband=" in sip.conf. Just to make sure: Maybe you forgot the SIP RELOAD? Are you 100% sure inbound calls arrive with the peer that you set progressinband for? Verify this using SIP DEBUG. Sip peer matching can be quite confusing in Asterisk. Philipp -- _ -- 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
[asterisk-users] Faxes
We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI) cards. I am having trouble completing faxes. Carrier send calls to me using SIP. Any recommendation to have some success with Fax. We trying using T.38 pass through and using G711U codec. Asterisk Version 1.6.1.1 Thanks, Dave -- _ -- 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] Faxes
The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. The carrier sending the calls wants me to be able to pass faxes to physical fax machines on the PSTN. So far they are failing. We just want ot be able to pass faxes using g711u or t.38 pass through. Thanks, Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, September 03, 2010 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Faxes On Fri, Sep 3, 2010 at 10:49 AM, dave george wrote: > We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI) > cards. > > > > I am having trouble completing faxes. Carrier send calls to me using SIP. > Any recommendation to have some success with Fax. > > We trying using T.38 pass through and using G711U codec. > > > > Asterisk Version 1.6.1.1 > > > > Thanks, > > Dave > Dave, T.38 in some fashion. But you don't really explain your call flow or what you are trying to do. You say you have PSTN and then talk about SIP. Are you just trying to pass the calls to physical FAX machines, or a server to handle faxing? Elaborate a bit and I am sure someone can offer some advice. Thanks, Steve Totaro -- _ -- 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] Faxes
All my attempts are failing. Thanks Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Friday, September 03, 2010 1:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Faxes On Fri, Sep 3, 2010 at 11:50 AM, dave george wrote: > The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk > SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the > PSTN. You don't say the percentage that are failing. However, people who have worked with SIP on asterisk have been known to do: exten => s,1,Playback(silence/1) exten => s,n,Whatever(is_next) And I don't know why, but this seems to make things better. If you're doing an Answer and then a receive_Fax, try putting a playback silence in between and see if that helps anything. -- _ -- 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] Faxes
Thanks Kevin, I tried passing it over VOIP using g711U codecs with no success. I will try using the patches that you mentioned and post the results. Thanks, Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, September 03, 2010 2:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Faxes On 09/03/2010 10:50 AM, dave george wrote: > The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk > SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the > PSTN. > > The carrier sending the calls wants me to be able to pass faxes to physical > fax machines on the PSTN. So far they are failing. > > We just want ot be able to pass faxes using g711u or t.38 pass through. As I told you on the asterisk-ss7 list, you can't 'pass through' T.38, because the PSTN does not speak T.38. If one side of the call is SIP, and the other side is TDM, then you have only two choices: pass the call through in audio mode (FAX over VoIP), or act as a T.38 gateway (FAX over T.38). At this time, the only option without patching Asterisk is to pass the call through in audio mode, but there are many, many problems with doing FAX over VoIP (Steve Underwood's page on the soft-switch.org site explains them very well). There are patches in the issue tracker at issues.asterisk.org to add T.38 gateway functionality to various releases of Asterisk, and they work well for quite a few people. If you added that, you'd be able to act as a T.38 gateway, which would dramatically increase your chances of success. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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 -- _ -- 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] Registration attempts
I am getting several hundred registration attempts on my aserterisk per minute. I have fail2ban installed but it's not stopping the attempts. Any suggestions. Whatever they are using is changing the userid on each attempt. Latest IP: 209.172.57.219 Thanks, Dave -- _ -- 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] Registration attempts
I had to make a few minor edit in fail2ban to get it to work. I had to change the logger messages format for asterisk to: [general] dateformat=%F %T Thanks, Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Friday, September 17, 2010 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Registration attempts It means that fail2ban is not configured correctly on your machine. For me it works fine, and in fact lately these registration/hack attempts have gone up significantly, thanks to cloud computing I guess. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-17 5:28 PM, "dave george" wrote: I am getting several hundred registration attempts on my aserterisk per minute. I have fail2ban installed but it's not stopping the attempts. Any suggestions. Whatever they are using is changing the userid on each attempt. Latest IP: 209.172.57.219 Thanks, Dave -- _ -- 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
[asterisk-users] CDMA Media gateway EVRC codec
Hi, I want to use asterisk as a media gateway for a CDMA application. I need support for EVRC codec. Anyone know which cards support EVRC? Thanks, Thanks, Dave -- _ -- 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
