Re: [asterisk-users] PRI can receive calls but cannot dial out
--- On Fri, 12/7/12, Steve Totaro stot...@totarotechnologies.com wrote: Why don't your span numbers match? 1-4 but you have 3-6 in your .conf. What do you mean? I have the following: span=3,1,0,CCS,AMI span=4,2,0,CCS,AMI span=5,3,0,CCS,AMI span=6,4,0,CCS,AMI The first parameter is the port number (3-6). The second parameter is Timing (1-4). Is it mandatory to begin the port numbering with 1? Or does it simply have to be sequential? Anyway, I set the span port numbers from 3 to 6 because I based myself on the output of dahdi_scan which was the following: # dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM400P REV I Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV I location=PCI Bus 00 Slot 04 basechan=1 totchans=4 irq=18 type=analog port=1,FXO port=2,FXO port=3,FXO port=4,FXO [2] active=yes alarms=OK description=Wildcard TDM2400P name=WCTDM/0 manufacturer=Digium devicetype=Wildcard TDM2400P location=PCI Bus 00 Slot 05 basechan=5 totchans=24 irq=20 type=analog port=5,FXO port=6,FXO port=7,FXO port=8,FXO port=9,FXO port=10,FXO port=11,FXO port=12,FXO port=13,none port=14,none port=15,none port=16,none port=17,none port=18,none port=19,none port=20,none port=21,none port=22,none port=23,none port=24,none port=25,none port=26,none port=27,none port=28,none [3] active=yes alarms=OK description=B4XXP (PCI) Card 0 Span 1 name=B4/0/1 manufacturer=Digium devicetype=Wildcard B410P location=PCI Bus 00 Slot 06 basechan=29 totchans=3 irq=23 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI,HDB3 framing_opts=ESF,D4,CCS,CRC4 coding=AMI framing=CCS [4] active=yes alarms=OK description=B4XXP (PCI) Card 0 Span 2 name=B4/0/2 manufacturer=Digium devicetype=Wildcard B410P location=PCI Bus 00 Slot 06 basechan=32 totchans=3 irq=23 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI,HDB3 framing_opts=ESF,D4,CCS,CRC4 coding=AMI framing=CCS [5] active=yes alarms=OK description=B4XXP (PCI) Card 0 Span 3 name=B4/0/3 manufacturer=Digium devicetype=Wildcard B410P location=PCI Bus 00 Slot 06 basechan=35 totchans=3 irq=23 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI,HDB3 framing_opts=ESF,D4,CCS,CRC4 coding=AMI framing=CCS [6] active=yes alarms=RED description=B4XXP (PCI) Card 0 Span 4 name=B4/0/4 manufacturer=Digium devicetype=Wildcard B410P location=PCI Bus 00 Slot 06 basechan=38 totchans=3 irq=23 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI,HDB3 framing_opts=ESF,D4,CCS,CRC4 coding=AMI framing=CCS I assumed I should use as port numbers the values within square brackets above. Still, I'm wondering why outgoing calls don't work (dial/g2 in my example) if I disconnect the cable from: span=3,1,0,CCS,AMI and leave all the others connected. Vieri -- _ -- 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] BLF and call-limit in 1.8
- Original Message - From: Matthew Jordan mjor...@digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 08, 2012 12:43 AM Subject: Re: [asterisk-users] BLF and call-limit in 1.8 Thanks for your reply. I just tested creating a peer in sip.conf and that works as it should. In addition, the value of call-limit in sip show peer peer is correct (the same as in sip.conf, which in this case is 4). -Pan When you enable call-limit globally for all peers, it sets the call-limit for all peers to INT_MAX. Hence why each device can accept quite a few calls. The setting you're actually toggling is 'callcounter'. When its enabled (boolean true), it sets call-limit to INT_MAX. If not present, then the call-limit value is used. This is why every setting but '1' set the call-limit to 0. Since you're using Realtime, there me a number of issues at play. Do you have this problem with a peer defined in sip.conf? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
Re: [asterisk-users] ODBC Connection Problem
Am 10.12.2012 06:37, schrieb Chandrakant Solanki: Hi All, OS : CentOS 5 64bit OS Machine Asterisk: 1.8.13.0 ODBC Packages: unixODBC-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2 unixODBC-devel-2.2.11-7.1 res_odbc.conf [telco-ops] enabled = yes dsn = telco-ops username = dba password = c3podb@2012 pre-connect = yes sanitysql = select 1 idlecheck = 15 ;isolation = repeatable_read pooling = yes limit = 3600 connect_timeout = 10 negative_connection_cache = 30 Above is my installation package and configuration file (res_odbc.conf), when I try to execute odbc show all it always gives below output. *CLI odbc show all ODBC DSN Settings - Name: telco-ops DSN:telco-ops Last connection attempt: 1970-01-01 00:00:00 Pooled: Yes Limit: 3600 Connections in use: 1 - Connection 1: connected When Insert/Update/Select query will be executed, it can't update last connection attempt field. In result, ODBC stuck after few minutes, and in this case I also need to restart asterisk, because I can't type any command, it can't give any command's output. Also updated asterisk with 10.9.0, but same result. Please show us /etc/odbc.ini too. -- _ -- 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] BLF and call-limit in 1.8
It seems I only assumed a call-limit value of 1 in the DB would make call waiting not work. I tested it now, and because that sets the value in Asterisk to INT_MAX, a call-limit value of 1 in the DB does allow for call waiting. The same value in sip.conf does not. -Pan - Original Message - From: Pan B. Christensen p...@ibidium.no To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 10, 2012 11:54 AM Subject: Re: [asterisk-users] BLF and call-limit in 1.8 Thanks for your reply. I just tested creating a peer in sip.conf and that works as it should. In addition, the value of call-limit in sip show peer peer is correct (the same as in sip.conf, which in this case is 4). -Pan -- _ -- 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] ODBC Connection Problem
/etc/odbc.ini [telco-ops] Description = Asterisk realtime and other FUNC_ODBC access Driver = MySQL Server = 172.18.100.18 Socket = /var/lib/mysql/data3306/mysql.sock User= dba Password= c3podb@2012 Database= mytelcoexample Port= 3306 Option = 3 On Mon, Dec 10, 2012 at 4:34 PM, Thorsten Göllner t...@ovm-group.com wrote: Am 10.12.2012 06:37, schrieb Chandrakant Solanki: Hi All, OS : CentOS 5 64bit OS Machine Asterisk: 1.8.13.0 ODBC Packages: unixODBC-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2 unixODBC-devel-2.2.11-7.1 res_odbc.conf [telco-ops] enabled = yes dsn = telco-ops username = dba password = c3podb@2012 pre-connect = yes sanitysql = select 1 idlecheck = 15 ;isolation = repeatable_read pooling = yes limit = 3600 connect_timeout = 10 negative_connection_cache = 30 Above is my installation package and configuration file (res_odbc.conf), when I try to execute odbc show all it always gives below output. *CLI odbc show all ODBC DSN Settings - Name: telco-ops DSN:telco-ops Last connection attempt: 1970-01-01 00:00:00 Pooled: Yes Limit: 3600 Connections in use: 1 - Connection 1: connected When Insert/Update/Select query will be executed, it can't update last connection attempt field. In result, ODBC stuck after few minutes, and in this case I also need to restart asterisk, because I can't type any command, it can't give any command's output. Also updated asterisk with 10.9.0, but same result. Please show us /etc/odbc.ini too. -- Regards, Chandrakant Solanki -- _ -- 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] BLF and call-limit in 1.8
I finally found the real culprit. The call-limit DB field was mapped to both call-limit and callcounter in the view asterisk uses. The latter is what caused the strange behaviour. Removed both and everything works as expected now. -Pan - Original Message - From: Pan B. Christensen p...@ibidium.no To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 10, 2012 12:13 PM Subject: Re: [asterisk-users] BLF and call-limit in 1.8 It seems I only assumed a call-limit value of 1 in the DB would make call waiting not work. I tested it now, and because that sets the value in Asterisk to INT_MAX, a call-limit value of 1 in the DB does allow for call waiting. The same value in sip.conf does not. -Pan -- _ -- 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] Is there an issue with 11.0.2 and registration
I am running 11.0.2 from 1.4.43 previous. When I start up and do a sip show peers all devices are on and show an IP Address. After some time sip show peers shows two devices of my 12 as (Unspecified). I never had an issue with 1.4.43. Is there some issue with 11.0.2 and registration? Jerry -- _ -- 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] deadagi on 11 and 1.4
How can extensions.conf be changed to work with both Asterisk 11 and 1.4.X such that 1.4.X calls deadagi and 11 just calls agi as deadagi is no more. Thanks, jerry -- _ -- 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] deadagi on 11 and 1.4
Put a GLOBAL in extensions.conf with the version and use GOTOIF to run AGI/DEADAGI dependent on it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Monday, December 10, 2012 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] deadagi on 11 and 1.4 How can extensions.conf be changed to work with both Asterisk 11 and 1.4.X such that 1.4.X calls deadagi and 11 just calls agi as deadagi is no more. Thanks, jerry -- _ -- 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] callerid not received from dahdi
From the last time you sent this to the list, here's the response from Richard
Mudgett rmudg...@digium.com...
my scenario is below
analog phone (10 to 99)-- pbx--(77)asterisk
jitsi(2000)
i have analog telephone interface numbered 77 attached with asterisk
and
other sip user is 2000 on jitsi.
I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will come then i can typed 2000# and call go to 2000 named
user
in asterisk.
Now my problem is when i am calling from 10 to 99 (any number) this
number
should display to sip 2000's user. But its not showing to user. Its
shows
asterisk@my_asterisk_server_ip.
my config. as follow
extension.conf
exten = s,1,Goto(phrase-menu,s,1)
[phrase-menu]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten = s,4,Wait(2)
exten = s,5,Set(CALLERID(num,CID)=${CALLERID})
Remove the CID option. It does nothing in this case because
it does not apply. The CID option here only applies to reading
not writing. Please re-read the documentation for CALLERID().
exten = s,6,Dial(SIP/${PHRASEID},40,tT)
exten = h,1,Hangup()
and in chan_dahdi.conf
; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
cidsignalling=dtmf
cidstart=polarity
callerid=asreceived
rxgain=0.0
txgain=0.0
;FXO Modules
group=1
echocancel=yes
signalling=fxs_ks
context=default
channel=1-20
#include dahdi-channels.conf
From your description, the link between the pbx and (77)asterisk
is analog. Analog can only pass caller id information in one
direction. It looks like you have it setup to pass caller id
from the pbx to (77)asterisk. Is the pbx even sending caller id?
Is it sending it in the form you have configured in Asterisk?
(dtmf, polarity start, dtmfcidlevel=???)
On Sun, Dec 9, 2012 at 11:42 PM, Harish Mandowara
asteriskhelp2...@gmail.com wrote:
my scenario is below
analog phone (10 to 99)-- pbx--(77)asterisk jitsi(2000)
i have analog telephone interface numbered 77 attached with asterisk and
other sip user is 2000 on jitsi.
I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will come then i can typed 2000# and call go to 2000 named user
in asterisk.
Now my problem is when i am calling from 10 to 99 (any number) this number
should display to sip 2000's user. But its not showing to user. Its
showsasterisk@my_asterisk_server_ip
https://webmail.cdac.in/twig/index.php?s[mailbox]=mail%2Fsent-mails[mailGroup]=%2As[mail_startmsg]=1s[sortby]=dates[sortbyway]=1s[delete-return]=msgviews[mailtree]=0%7Cc[f]=mailc[a]=composeform[to]=asterisk@my_asterisk_server_ip.
my config. as follow
extension.conf
exten = s,1,Goto(phrase-menu,s,1)
[phrase-menu]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten = s,4,Wait(2)
exten = s,5,Set(CALLERID(num,CID)=${CALLERID})
exten = s,6,Dial(SIP/${PHRASEID},40,tT)
exten = h,1,Hangup()
and in chan_dahdi.conf
; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
cidsignalling=dtmf
cidstart=polarity
callerid=asreceived
rxgain=0.0
txgain=0.0
;FXO Modules
group=1
echocancel=yes
signalling=fxs_ks
context=default
channel=1-20
#include dahdi-channels.conf
any help
thanks..
--
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
--
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Re: [asterisk-users] ODBC Connection Problem
On Mon, Dec 10, 2012 at 5:23 AM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Password= c3podb@2012 In case you didn't realize you were sending this out publicly to a publicly archived and searchable list, you might want to change that password now. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Is there an issue with 11.0.2 and registration
On Mon, Dec 10, 2012 at 8:24 AM, Jerry Geis ge...@pagestation.com wrote: When I start up and do a sip show peers all devices are on and show an IP Address. After some time sip show peers shows two devices of my 12 as (Unspecified). When you say two, is it two every time? The same two? Is there something different about the two that show this behavior? There isn't enough information in your message. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Is there an issue with 11.0.2 and registration
When you say two, is it two every time? The same two? Is there something different about the two that show this behavior? There isn't enough information in your message. Yes it is the same two devices every time. I have the server running 11.0.2 , I have 8 asterisk devices (1.4.43), I have two polycom phones. They all seem fine. Then I have two other devices (IPSpeakers) that run fine on 1.4.43 and some time after inititally starting 11.0.2 they change from showing the IP address in sip show peers to showing unspecified. They work in the beginning until such time they show unspecified. Then if I stop asterisk 11.0.2 again, and restart it they start working again for some time. Jerry -- _ -- 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] Is there an issue with 11.0.2 and registration
Sounds like a registration timeout issue. What does the sip.conf entry look like for these? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Monday, December 10, 2012 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is there an issue with 11.0.2 and registration When you say two, is it two every time? The same two? Is there something different about the two that show this behavior? There isn't enough information in your message. Yes it is the same two devices every time. I have the server running 11.0.2 , I have 8 asterisk devices (1.4.43), I have two polycom phones. They all seem fine. Then I have two other devices (IPSpeakers) that run fine on 1.4.43 and some time after inititally starting 11.0.2 they change from showing the IP address in sip show peers to showing unspecified. They work in the beginning until such time they show unspecified. Then if I stop asterisk 11.0.2 again, and restart it they start working again for some time. Jerry -- _ -- 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] Is there an issue with 11.0.2 and registration
Hello, On 10.12.2012 18:30, Jerry Geis wrote: When you say two, is it two every time? The same two? Is there something different about the two that show this behavior? There isn't enough information in your message. Yes it is the same two devices every time. Try pedantic=no in sip.conf. Also, enable a SIP debug on the peers, check if anything out of the ordinary appears. -- Cristi -- _ -- 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] ODBC Connection Problem
On 10 Dec 2012, at 16:13, Christopher Harrington wrote: On Mon, Dec 10, 2012 at 5:23 AM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Password= c3podb@2012 In case you didn't realize you were sending this out publicly to a publicly archived and searchable list, you might want to change that password now. Hostname address is RFC1918, he'll probably be ok ;) S-- _ -- 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] ODBC Connection Problem
On Mon, Dec 10, 2012 at 10:52 AM, Steven Howes steve-li...@geekinter.netwrote: On 10 Dec 2012, at 16:13, Christopher Harrington wrote: Hostname address is RFC1918, he'll probably be ok ;) Private subnet or not, that's a social engineering and recon target. If all it takes is a Google search for this guy's name and password, that's dangerous. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Partial authentication possible?
I have a non-standard SIP client that I am trying to integrate with an Asterisk 10 server. This client requires that it register with the Asterisk server and that this registration not be authenticated. When a call is passed from Asterisk to the SIP client, the client does require Asterisk to authenticate. Is it possible to configure Asterisk to not request authentication on the registration but to respond to authentication challenges on the invite? I am not able to make any configuration changes to this non-standard SIP client. John -- _ -- 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] IAX2 over OpenVPN connection.... working but
Here's where I am baffled and I am hoping someone with intricate knowledge of this implementation may be able to explain it to me. What we had to do to get this working was to set the host= parameter to the respective endpoint IP's of the VPN tunnel, 172.10.1.1 in my case, and 172.10.1.2 in his case. Calls flow normally now and we cannot understand how or why. I would have assumed with a destination of either LAN as defined by the routing table it would have left out on the OpenVPN connection by default, and what's even more strange is that IAX is the only protocol that does not appear to function as intended. My guess is asterisk is replying using the tunnel ip address which your original box won't accept unless you actually sent to that address. Thats what I see on our remote openvpn tunnels. If you want to know whats going on use tcpdump to check packets through the tunnel. Yes, I've seen this same problem. It has two possible solutions. The reason for the problem is this: IAX2 (the Asterisk implementation, at least) depends on the source address in the UDP packet it receives, to know which connection the packet is part of. When it talks to a peer, it expects to see the packets arrive from the peer with a source address which matches what it understands the peer's address to be. Packets arriving from unknown addresses, are simply dropped on the floor (considered to be misrouted, misconfigured, or forged, I think). Normally, the Asterisk IAX2 implementation does not bind itself to a single network interface. It will receive UDP packets to the IAX2 port, arriving from any interface. And, when it sends an IAX2 UDP packet, it simply sends it out through the socket which is bound to the any interface port. Because the socket isn't bound to a specific interface, it doesn't have a specific IP address associated with it. The Linux kernel chooses an IP address to put into the UDP packet source field, and the one it chooses is the IP address of the interface on which it is transmitting the packet. In the scenario that's being described here, an address result mismatches. Each system is transmitting UDP packets *to* the primary or official or public interface on its peer... and these packets are being transmitted by the Linux kernel on the OpenVPN interface, and are being given the system's OpenVPN tunnel endpoint address. In each case, when the packet arrives at the peer, the Asterisk IAX2 stack receives the packets, finds that it has no known peer at the tunnel IP address and no IAX2 session set up for this address, and discards the packet. There are, I believe, two solutions which don't involve modifying the IAX2 code in Asterisk. Both work equally well, as far as I know. One approach is the one you've taken - tell each system to talk to its peer's OpenVPN tunnel endpoint address, rather than to the primary address. This eliminates the IP address mismatch. This approach works fine if both systems are connected only via this OpenVPN tunnel, and always have the same OpenVPN addresses. The other approach is to configure each system to bind its IAX2 port *only* to one IP interface (usually the public one), to ensure that each peer knows how to reach its peer's public IP address (either directly, or via a route though the OpenVPN tunnel), and to tell each system to speak IAX2 to its peer's public IP address. In this case, since the Asterisk socket is bound to a specific interface, all packets sent through that socket will have the bound interface's IP address in its source field, and (once again) the address mismatch is eliminated. This second approach is preferable for road warrior configurations in which you might sometimes be using the OpenVPN tunnel, and sometimes not (e.g. a laptop or tablet IAX2 client which is sometimes on the corporate LAN and sometimes out on the Internet using OpenVPN). -- _ -- 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] Partial authentication possible?
Consider using a sip proxy server such as OpenSIPS or Kamailio. Regards, Ali Pey On Mon, Dec 10, 2012 at 12:59 PM, John Gilbert jo...@motorolasolutions.comwrote: I have a non-standard SIP client that I am trying to integrate with an Asterisk 10 server. This client requires that it register with the Asterisk server and that this registration not be authenticated. When a call is passed from Asterisk to the SIP client, the client does require Asterisk to authenticate. Is it possible to configure Asterisk to not request authentication on the registration but to respond to authentication challenges on the invite? I am not able to make any configuration changes to this non-standard SIP client. John -- _ -- 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] Is there an issue with 11.0.2 and registration
Try pedantic=no in sip.conf. Also, enable a SIP debug on the peers, check if anything out of the ordinary appears. seems as though pedantic=no was the issue. they are staying online. further looking (which I seemed to miss) was in 1.4 pedantic as default no, in 11 default is yes. my sip.conf had no setting. Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with SIP trunk I've set up between two * boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf
#include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf
type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf
#include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf
type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include = aggregate
include = passthrough
exten = _7XXX,1,Dial(SIP/box2/${EXTEN})
exten = _7XXX,2,Hangup()
---
When I dial, all I get is (I'll attach the full dialog up to that point
from SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial(SIP/6110-08291cb0,
SIP/box2/7444) in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
---
Where am I goofing up? Any pointers?
Thanks!
-Ken
---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1
s=pjmedia
c=IN IP4 172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request -
nUiGauUpyxjNOJfcZog476ws.Art7jZS
--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=172.17.9.1;rport=55388
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=16883b72
Content-Length: 0
Scheduling destruction of SIP dialog 'nUiGauUpyxjNOJfcZog476ws.Art7jZS'
in 32000 ms (Method: INVITE)
Found user '6110'
--- SIP read from 172.17.9.1:55388 ---
ACK sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 ACK
Route: sip:172.17.0.17;transport=udp;lr
Content-Length: 0
-
--- (9 headers 0 lines) ---
--- SIP read from 172.17.9.1:55388 ---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24153 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Proxy-Authorization: Digest username=6110, realm=asterisk,
nonce=16883b72, uri=sip:7444@172.17.0.17,
response=b75389c5938b4f185b3d31bd4463abf3, algorithm=MD5
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1
s=pjmedia
c=IN IP4 172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
Re: [asterisk-users] Problem with SIP trunk I've set up between two * boxes.
Does each box show up in the others SIP SHOW PEERS?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include = aggregate
include = passthrough
exten = _7XXX,1,Dial(SIP/box2/${EXTEN}) exten = _7XXX,2,Hangup()
---
When I dial, all I get is (I'll attach the full dialog up to that point from
SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial(SIP/6110-08291cb0,
SIP/box2/7444) in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
---
Where am I goofing up? Any pointers?
Thanks!
-Ken
---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4 172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request - nUiGauUpyxjNOJfcZog476ws.Art7jZS
--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=1
72.17.9.1;rport=55388
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=16883b72
Content-Length: 0
Scheduling destruction of SIP dialog 'nUiGauUpyxjNOJfcZog476ws.Art7jZS'
in 32000 ms (Method: INVITE)
Found user '6110'
--- SIP read from 172.17.9.1:55388 --- ACK sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 ACK
Route: sip:172.17.0.17;transport=udp;lr
Content-Length: 0
-
--- (9 headers 0 lines) ---
--- SIP read from 172.17.9.1:55388 ---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24153 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Proxy-Authorization: Digest username=6110, realm=asterisk,
nonce=16883b72, uri=sip:7444@172.17.0.17,
response=b75389c5938b4f185b3d31bd4463abf3, algorithm=MD5
Content-Type: application/sdp
Re: [asterisk-users] deadagi on 11 and 1.4
On Mon, 10 Dec 2012, Jerry Geis wrote:
How can extensions.conf be changed to work with both Asterisk 11 and
1.4.X such that 1.4.X calls deadagi and 11 just calls agi as deadagi is
no more.
On Mon, 10 Dec 2012, Danny Nicholas wrote:
Put a GLOBAL in extensions.conf with the version and use GOTOIF to run
AGI/DEADAGI dependent on it.
When do you know what version of Asterisk is executing your dialplan?
If you know at installation time, pass your dialplan through a
preprocessor (even something as simple as sed -e 's/@DEADAGI@/deadagi/g').
If you truly don't know until execution time (and I can't imagine this is
true), you could define a channel variable ('DEADAGI') to either 'deadagi'
or 'agi' once and then reference it in your dialplan as:
exten = *,n, execif(1,${MY-AGI},null-agi,--null)
At least you aren't evaluating your version condition throughout your
dialplan.
Curiously (and unfortunately) (at least in 1.2), you can't just define a
variable and then execute it as:
exten = *,n, ${MY-AGI}(null-agi,--null)
I'd advise the full preprocessor approach. The additional functionality it
provides will result in a easier to maintain system. I do it with all of
my Asterisk 'conf' files.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP trunk I've set up between two * boxes.
On 2012-12-10 16:16, Danny Nicholas wrote:
Does each box show up in the others SIP SHOW PEERS?
Yup -- each shows in the other's. Sorry I didn't mention that.
-Ken
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken
D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between
two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include = aggregate
include = passthrough
exten = _7XXX,1,Dial(SIP/box2/${EXTEN}) exten = _7XXX,2,Hangup()
---
When I dial, all I get is (I'll attach the full dialog up to that
point from
SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial(SIP/6110-08291cb0,
SIP/box2/7444) in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
---
Where am I goofing up? Any pointers?
Thanks!
-Ken
---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4
172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request -
nUiGauUpyxjNOJfcZog476ws.Art7jZS
--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=1
72.17.9.1;rport=55388
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=16883b72
Content-Length: 0
Scheduling destruction of SIP dialog
'nUiGauUpyxjNOJfcZog476ws.Art7jZS'
in 32000 ms (Method: INVITE)
Found user '6110'
--- SIP read from 172.17.9.1:55388 --- ACK sip:7444@172.17.0.17
SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 ACK
Route: sip:172.17.0.17;transport=udp;lr
Content-Length: 0
-
--- (9 headers 0 lines) ---
--- SIP read from 172.17.9.1:55388 ---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24153 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Proxy-Authorization: Digest username=6110,
Re: [asterisk-users] Problem with SIP trunk I've set up between two * boxes.
Looks like a connectivity issue, doesn't it?
IP of box2, 172.17.145.145, doesn't show up even once in the SIP dialogues.
What happens on box2 (asterisk -vvvr and tcpdump port 5060) in the
moment that you place a call through box1 to box2?
Also what's strange is that you are trying to call from box2 to box2?
Because local_SIP is the context on box2, and on box1 it's adhearsion.
The console message you pasted shows @local_SIP however, so it looks
like you are calling from box2 to box2?
Am 10.12.2012 22:53, schrieb Ken D'Ambrosio:
On 2012-12-10 16:16, Danny Nicholas wrote:
Does each box show up in the others SIP SHOW PEERS?
Yup -- each shows in the other's. Sorry I didn't mention that.
-Ken
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken
D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between
two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include = aggregate
include = passthrough
exten = _7XXX,1,Dial(SIP/box2/${EXTEN}) exten = _7XXX,2,Hangup()
---
When I dial, all I get is (I'll attach the full dialog up to that
point from
SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial(SIP/6110-08291cb0,
SIP/box2/7444) in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
---
Where am I goofing up? Any pointers?
Thanks!
-Ken
---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4 172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request - nUiGauUpyxjNOJfcZog476ws.Art7jZS
--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=1
72.17.9.1;rport=55388
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=16883b72
Content-Length: 0
Scheduling destruction of SIP dialog 'nUiGauUpyxjNOJfcZog476ws.Art7jZS'
in 32000 ms (Method: INVITE)
Found user '6110'
--- SIP read from 172.17.9.1:55388 --- ACK sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 ACK
Route: sip:172.17.0.17;transport=udp;lr
Content-Length: 0
-
--- (9 headers 0 lines) ---
--- SIP read from 172.17.9.1:55388 ---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
[asterisk-users] asterisk 1.8.19.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.19.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.19.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Prevent resetting of NATted realtime peer address on reload. (Closes issue ASTERISK-18203. Reported by daren ferreira) * --- Do not use a FILE handle when doing SIP TCP reads. (Closes issue ASTERISK-20212. Reported by Phil Ciccone) * --- Fix execution of 'i' extension due to uninitialized variable. (Closes issue ASTERISK-20455. Reported by Richard Miller) * --- Ensure that the Queue application tracks busy members in off nominal situations (Closes issue ASTERISK-20623. Reported by Bryan Walters) * --- Properly extract the Body information of an EWS calendar item (Closes issue ASTERISK-19738. Reported by Dmitry Burilov) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.19.0 Thank you for your continued support of Asterisk! -- _ -- 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] Asterisk 10.11.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 10.11.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.11.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Prevent resetting of NATted realtime peer address on reload. (Closes issue ASTERISK-18203. Reported by daren ferreira) * --- Do not use a FILE handle when doing SIP TCP reads. (Closes issue ASTERISK-20212. Reported by Phil Ciccone) * --- Fix ConfBridge crash if no timing module loaded. (Closes issue ASTERISK-19448. Reported by feyfre) * --- confbridge: Fix a bug which made conferences not record with AMI/CLI commands (Closes issue ASTERISK-20601. Reported by Vilius) * --- Fix execution of 'i' extension due to uninitialized variable. (Closes issue ASTERISK-20455. Reported by Richard Miller) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.11.0 Thank you for your continued support of Asterisk! -- _ -- 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] Asterisk 11.1.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.1.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Fix execution of 'i' extension due to uninitialized variable. (Closes issue ASTERISK-20455. Reported by Richard Miller) * --- Prevent resetting of NATted realtime peer address on reload. (Closes issue ASTERISK-18203. Reported by daren ferreira) * --- Fix ConfBridge crash if no timing module loaded. (Closes issue ASTERISK-19448. Reported by feyfre) * --- Fix the Park 'r' option when a channel parks itself. (Closes issue ASTERISK-19382. Reported by James Stocks) * --- Fix an issue where outgoing calls would fail to establish audio due to ICE negotiation failures. (Closes issue ASTERISK-20554. Reported by mmichelson) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.1.0 Thank you for your continued support of Asterisk! -- _ -- 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] monitoring - hangup channel
How can I monitor channel that hangup? I'm using asterisk 1.8.15.1 and there are many times that nobody is using the line but when I run: asterisk -rx core show channels it show: Channel Location State Application(Data) SIP/pstn--00 (None) Up AppDial((Outgoing Line)) SIP/pstn-9998-00 7807586576@internal: Up Dial(SIP/97807807586576@pstn-4 2 active channels 1 active call How can I prevent it or monitor it? At this point I have to hang it up manually or I will not be able to restart the asterisk. -- Joseph -- _ -- 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] Problem with SIP trunk I've set up between two * boxes.
Hi, Ken
I have almost the same setup as yours: new
asterisk-SIP-Trixbox(Asterisk 1.4)---PRIpots
Here are my configs:
new box sip.conf:
[126]
directmedia=no
type=friend
host=trixbox_IP_addr
secret=my_secret
username=126 ;this is for outgoing calls from new asterisk via trixbox
fromuser=126 ;this is for outgoing calls from new asterisk via trixbox
context=default
disallow=all
allow=alaw
allow=ulaw
qualify=yes
qualifyfreq=60
nat=yes
pickupgroup=1
callgroup=1
trixbox
[126]
type=friend
secret=mysecret
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=
host=dynamic
dtmfmode=rfc2833
dial=SIP/126
context=from-internal
canreinvite=no
callgroup=
callerid=device 126
accountcode=
call-limit=50
New box's account (126) registers to the Trixbox so as to make incoming calls
from trixbox to new box possible.
The config in the new box implies that the trixbox require authorization in
calls from the new box (username and fromuser options are necessary for this).
Actually looking through the sip.conf in 1.8 asterisk I found that there are
auth option as well as remotesecret and remoteuser - but I can not
understand how they work in case if I need to authorise my outgoing calls
(probably sip.conf will be more logical in the future 12th version).
Hope this helps.
Dmitry Pavlenko
From: Ken D'Ambrosio k...@jots.org
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 11, 2012 3:53 AM
Subject: Re: [asterisk-users] Problem with SIP trunk I've set up between two *
boxes.
On 2012-12-10 16:16, Danny Nicholas wrote:
Does each box show up in the others SIP SHOW PEERS?
Yup -- each shows in the other's. Sorry I didn't mention that.
-Ken
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken
D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between
two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include = aggregate
include = passthrough
exten = _7XXX,1,Dial(SIP/box2/${EXTEN}) exten = _7XXX,2,Hangup()
---
When I dial, all I get is (I'll attach the full dialog up to that
point from
SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial(SIP/6110-08291cb0,
SIP/box2/7444) in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
---
Where am I goofing up? Any pointers?
Thanks!
-Ken
---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4
172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request -
nUiGauUpyxjNOJfcZog476ws.Art7jZS
--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---
SIP/2.0 407 Proxy Authentication Required
Via:
[asterisk-users] date - outgoing call
When a call comes in asterisk records the date correctly but when I cake a call out I get only something like: Date: 60 here is an example: From: 7807560785 To: s Date: 2012-12-11 00:46:04 Status: ANSWERED From: 5 To: 4331235 Date: 60 Status: 4 -- Joseph -- _ -- 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] callerid not received from dahdi
Hi,
Thank you for your reply.
77 ext. number is connected with my asterisk. so any one want to talk with
jitsi(pc), they have to dial 77 then 2000#(jitsi sip user number).
my pbx is sending callerid. i can see on other analog phone display.
Yes pbx is sending callerid. When i dial any ext. number from jitsi. On the
recipient phone display shows 77 ext number.
i tried all combination from
https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India
but it does not work.
any help
On Mon, Dec 10, 2012 at 9:39 PM, Christopher Harrington ch...@acsdi.comwrote:
From the last time you sent this to the list, here's the response from Richard
Mudgett rmudg...@digium.com...
my scenario is below
analog phone (10 to 99)-- pbx--(77)asterisk
jitsi(2000)
i have analog telephone interface numbered 77 attached with asterisk
and
other sip user is 2000 on jitsi.
I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will come then i can typed 2000# and call go to 2000 named
user
in asterisk.
Now my problem is when i am calling from 10 to 99 (any number) this
number
should display to sip 2000's user. But its not showing to user. Its
shows
asterisk@my_asterisk_server_ip.
my config. as follow
extension.conf
exten = s,1,Goto(phrase-menu,s,1)
[phrase-menu]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten = s,4,Wait(2)
exten = s,5,Set(CALLERID(num,CID)=${CALLERID})
Remove the CID option. It does nothing in this case because
it does not apply. The CID option here only applies to reading
not writing. Please re-read the documentation for CALLERID().
exten = s,6,Dial(SIP/${PHRASEID},40,tT)
exten = h,1,Hangup()
and in chan_dahdi.conf
; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
cidsignalling=dtmf
cidstart=polarity
callerid=asreceived
rxgain=0.0
txgain=0.0
;FXO Modules
group=1
echocancel=yes
signalling=fxs_ks
context=default
channel=1-20
#include dahdi-channels.conf
From your description, the link between the pbx and (77)asterisk
is analog. Analog can only pass caller id information in one
direction. It looks like you have it setup to pass caller id
from the pbx to (77)asterisk. Is the pbx even sending caller id?
Is it sending it in the form you have configured in Asterisk?
(dtmf, polarity start, dtmfcidlevel=???)
On Sun, Dec 9, 2012 at 11:42 PM, Harish Mandowara
asteriskhelp2...@gmail.com wrote:
my scenario is below
analog phone (10 to 99)-- pbx--(77)asterisk jitsi(2000)
i have analog telephone interface numbered 77 attached with asterisk and
other sip user is 2000 on jitsi.
I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will come then i can typed 2000# and call go to 2000 named user
in asterisk.
Now my problem is when i am calling from 10 to 99 (any number) this number
should display to sip 2000's user. But its not showing to user. Its
showsasterisk@my_asterisk_server_ip
https://webmail.cdac.in/twig/index.php?s[mailbox]=mail%2Fsent-mails[mailGroup]=%2As[mail_startmsg]=1s[sortby]=dates[sortbyway]=1s[delete-return]=msgviews[mailtree]=0%7Cc[f]=mailc[a]=composeform[to]=asterisk@my_asterisk_server_ip.
my config. as follow
extension.conf
exten = s,1,Goto(phrase-menu,s,1)
[phrase-menu]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten = s,4,Wait(2)
exten = s,5,Set(CALLERID(num,CID)=${CALLERID})
exten = s,6,Dial(SIP/${PHRASEID},40,tT)
exten = h,1,Hangup()
and in chan_dahdi.conf
; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
cidsignalling=dtmf
cidstart=polarity
callerid=asreceived
rxgain=0.0
txgain=0.0
;FXO Modules
group=1
echocancel=yes
signalling=fxs_ks
context=default
channel=1-20
#include dahdi-channels.conf
any help
thanks..
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Re: [asterisk-users] date - outgoing call
On Mon, 10 Dec 2012, Joseph wrote: When a call comes in asterisk records the date correctly but when I cake a call out I get only something like: Date: 60 here is an example: From: 7807560785 To: s Date: 2012-12-11 00:46:04 Status: ANSWERED From: 5 To: 4331235 Date: 60 Status: 4 What version of Asterisk? Where are you seeing the data you display above? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] date - outgoing call
On 12/10/12 20:45, Steve Edwards wrote: On Mon, 10 Dec 2012, Joseph wrote: When a call comes in asterisk records the date correctly but when I cake a call out I get only something like: Date: 60 here is an example: From: 7807560785 To: s Date: 2012-12-11 00:46:04 Status: ANSWERED From: 5 To: 4331235 Date: 60 Status: 4 What version of Asterisk? Where are you seeing the data you display above? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 This is on my Asterisk 1.8.15.1 Here are other examples: From: 14034566015 To: s Date: tr Status: 8 From: 877 To: 218 Date: 25 Status: 65 From: 322 To: 77586476 Date: 60 Status: 45 From: 5 To: 218 Date: 25 Status: 67 -- Joseph -- _ -- 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] date - outgoing call
On 12/10/12 20:45, Steve Edwards wrote:
On Mon, 10 Dec 2012, Joseph wrote:
When a call comes in asterisk records the date correctly but when I cake a
call out I get only something like:
Date: 60
here is an example:
From: 7807560785
To: s
Date: 2012-12-11 00:46:04
Status: ANSWERED
From: 5
To: 4331235
Date: 60
Status: 4
What version of Asterisk?
Where are you seeing the data you display above?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
I'm using this command to pull the records of the cvs:
grep -P '\d{4}-\d{2}-\d{2}\s\d{2}:\d{2}:\d{2}' /var/log/asterisk/cdr-csv/*.csv | cut -d, -f2,3,10,15 | awk -F, {'print
From:\t$1\nTo:\t$2\nDate:\t$3\nStatus:\t$4\n\n'} |tail -60
the actual last records in Master.csv is:
,5,218,internal,Cerra 5,IAX2/192.168.141.1:4569-4374,SIP/11-0180,Dial,SIP/11SIP/321SIP/218,25,m(penguin)w,2012-12-11
02:58:53,2012-12-11 02:58:59,2012-12-11 03:00:00,67,61,ANSWERED,DOCUMENTATION,1355194733.524,
but with the above command it display:
From: 5
To: 218
Date: 25
Status: 67
--
Joseph
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