Re: [asterisk-users] Newbie Queue: greetings when first joiningqueue
> I would think you'll need to do a Playback() of this message before the > caller enters the queue, as I'm not aware of such an option provided by > app_queue. > > Exten=>100,1,Answer() > Exten=>100,n,Playback(greetings-earthling) > Exten=>100,n,Queue(xyzqueue) > Exten=>100,n,Hangup Thanks Mark for your suggestion. The issue with this is the first caller will always have to listen to this greeting regardless. Is there anyway to check if there is anyone in the queue before this greeting is played? In other words, if the queue is empty, then just "Queue". If queue is not empty, then "Playback + Queue". Is this possible? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Queue: greetings when first joining queue
I would think you'll need to do a Playback() of this message before the caller enters the queue, as I'm not aware of such an option provided by app_queue. Exten=>100,1,Answer() Exten=>100,n,Playback(greetings-earthling) Exten=>100,n,Queue(xyzqueue) Exten=>100,n,Hangup -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee, John (Sydney) Sent: March 19, 2008 2:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Newbie Queue: greetings when first joining queue I was trying to find out how I could put in a greeting when a caller ***first*** joins the queue. I searched high and low but could only find (in queues.conf): . "announce", which is announcement to the agent . "announce-frequency" which is announcement of queue position . "periodic-announcement-frequency", "periodic-announce" which may seem applicable to my case but it does not do it the first time. It only announce at the end of the first interval. I just want a greeting like "Thank you for calling XYZ. Your call is important to us and we will answer your call shortly". This only needs to be announced once and it is done at the start. Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting config from SPA-941 or 942 phones
James Lamanna wrote: >Hi, >Is it possible to get the XML config off of a Linksys SPA-941 or 942 phone? >I've tried http://[ip address]/admin/spacfg.xml however that file >doesn't appear to exist. > > > Yes it is. It requires a 2 step process. 1. Configure the Provisioning Parameter "Report Rule" to point to your tftp server and define a file. For example: Report Rule tftp://192.168.3.110/spa941.xml Create the file using the touch command #touch /tftpboot/spa941.xml chmod 777 /tftpboot/spa941.xml Also make sure the SPA Parameter Auth Resync-Reboot is set to NO. 2. Now generate a SIP Notify Message from your server with the 'report' header. I generaly use sipsak for this (http://sipsak.org/) First I generate a text file called report.txt # cat report.txt NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 From: sip:$srchost$:$port$ To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 NOTIFY Max-Forwards: 70 Event: report User-Agent: sipsak Content-Length: 0 ...and then I execute the command: sipsak -s sip:[EMAIL PROTECTED] -G -f ./report.txt (were 1002 is the username configured on the SPA and 192.168.3.12 is it's IP Address) This will cause the SPA to do a tftp put with all the XML config into your spa941.xml file. Andres http://www.neuroredes.com >Thanks. > >___ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie Queue: greetings when first joining queue
I was trying to find out how I could put in a greeting when a caller ***first*** joins the queue. I searched high and low but could only find (in queues.conf): . "announce", which is announcement to the agent . "announce-frequency" which is announcement of queue position . "periodic-announcement-frequency", "periodic-announce" which may seem applicable to my case but it does not do it the first time. It only announce at the end of the first interval. I just want a greeting like "Thank you for calling XYZ. Your call is important to us and we will answer your call shortly". This only needs to be announced once and it is done at the start. Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Screening feature using asterisk
Have you looked at the privacymanager function in Asterisk? PaulH On Wed, 2008-03-19 at 10:31 +0530, Janu Mukherjee wrote: > Hi, > > I have our software with SIP running on it.I configured asterisk > server as proxy. How do I implement the call screening > features(incoming and outgoing) using asterisk server.Please suggest > me how to proceed on this. > > Thanks & Regards, > Jahnavi. > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Deadair in queues.
Hello,
Asterisk Server A makes an outbound call, and upon connect:
exten
=>1,n,RetryDial(/var/lib/asterisk/sounds/connecting,0,3,SIP/${connectto},,tT
)
(${connectto} most of the time happens to be [EMAIL PROTECTED] or 54321 {IP
masqueraded ofcourse})
..transfers it to * Server B (i.e 66.xx.xx.66) via SIP.
(Background info, Server B registers on Server A as 1000, and Server A
registers on Server B as 1000. Both of them are on direct IPs, and not
behind a hardware firewall. Server A has no iptables, and Server B has udp
ports 1 to 2 open, and tcp/udp 5060)
Server B has two queues, where there are agents logged in waiting to take
this call. Depending on what extension the calls comes to, i.e 12345 or
54321, it goes to separate queues. One queue has agents, who are on direct
IPs, not behind a firewall, all open ports, no XP firewall and using
eyebeam. The other queue has agents who are on NAT.
BOTH these queue agents complain of deadair. The call comes in, but the
agents say they don't hear anybody on the other side.
Server B looks like this (extensions.conf):
[test]
exten => 12345,1,Set(CALLERID(num)=${CALLERID(num)})
exten => 12345,n,Set(CALLERID(name)="PayMaker")
exten => 12345,n,Set(QUEUE_PRIO=5)
exten => 12345,n,Goto(collections,100,1)
[collections]
exten=> 100,1,Answer
exten=> 100,n,Verbose(CID: ${CALLERID(num)})
exten=> 100,n,Ringing
exten=> 100,n,Wait(2)
exten=> 100,n,Queue(collections|Tt|0)
exten=> 100,n,Voicemail(100,u)
exten=> 100,n,Hangup
And [collections] in queues.conf looks like this:
[collections]
autofill = yes
musiconhold=default
strategy=rrmemory
timeout=5
retry=1
eventwhencalled=yes
wrapuptime=0
ringinuse=no
joinempty=strict
leavewhenempty=yes
maxlen = 0
memberdelay=1
announce-frequency = 60
announce-holdtime = no
;member => Agent/:1
member => Agent/10
member => Agent/11
member => Agent/12
member => Agent/13
-
Sometimes, instead of transferring via SIP to Server B, we transfer to a DID
(an external queue/callcenter). Even they complain of deadair.
Server B recently got iptables just this Monday. Before that Server B had no
iptables at all.
I'm really desperate in getting this dead air issue resolved. Because I've
been asking for some time now.
Best regards and much thanks,
Mark.
___
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[asterisk-users] Getting config from SPA-941 or 942 phones
Hi, Is it possible to get the XML config off of a Linksys SPA-941 or 942 phone? I've tried http://[ip address]/admin/spacfg.xml however that file doesn't appear to exist. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardphone SIP phone costs
> when you look at the iPhone with all its amazing features for less than $500.00 it just doesn't make sense. Am I the only one that thinks this? Remember that the service providers such as AT&T, Cingular, Sprint, Verizon and so forth, subsidize the cost of the phones because they make it up over the course of the contract. Hence the reason that some phones that have an initial cost when sold with a 1 year contract may be free initially with a 2 year contract. Even some VoIP phones and ATA's are done this way but only through service providers. Take the subsidies away and that iPhone is pretty pricey. John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Screening feature using asterisk
Hi, I have our software with SIP running on it.I configured asterisk server as proxy. How do I implement the call screening features(incoming and outgoing) using asterisk server.Please suggest me how to proceed on this. Thanks & Regards, Jahnavi. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardphone SIP phone costs
--Original Message Text--- From: Anciso, Roy Date: Tue, 18 Mar 2008 23:03:52 -0400 Hardphone SIP phone costs Im trying to understand something that just doesnt seem to compute. How can companies like Cisco justify selling their hard phones for as much as they do? I know there is a matter of recouping R&D costs but when you look at the iPhone with all its amazing features for less than $500.00 it just doesnt make sense. Am I the only one that thinks this? Yep, Cisco phones cost a lot. Too much in my opinion. Do they work. Yes, they work well. But as long as I can get Polycom and Aastra phones that work as well or better why pay the Cisco premium? I once heard a rumour that the Cisco phones were actually made by Polycom for Cisco under contract. Not sure if that's true or not. Michael -- Michael Graves mgravesmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardphone SIP phone costs
I'm trying to understand something that just doesn't seem to compute. How can companies like Cisco justify selling their hard phones for as much as they do? I know there is a matter of recouping R&D costs but when you look at the iPhone with all its amazing features for less than $500.00 it just doesn't make sense. Am I the only one that thinks this? Roy Anciso Director of Technology Manistee Intermediate School District 772 East Parkdale Avenue Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-398-3036 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] capacity
I did basically the same thing via T1 on the Definity. It took a bit of tinkering on the Definity to get the coverage path right. For your use, I would go for a RAID 5, dual power supply box with quite a bit of storage. RAM and CPU should not be an issue with anything new. I would go with a T1/E1 card with more than one port just for future possible growth or options. Echo cancellation is probably not needed but if in the budget, it can never hurt (never say never, seldomly or rarely I guess is more appropriate). I would probably go with an HP DL380. The dialplan should be very simple. It should actually be pretty fun project. Thanks, Steve Totaro On Tue, Mar 18, 2008 at 3:15 PM, Eve-Ellen Cole <[EMAIL PROTECTED]> wrote: > > > > I have an Avaya Definity G3R. Calls to students will be routed through the > G3R, to the Asterisk system so the caller can leave a message. I'm not sure > how many channels I'll really need, but I expect no more than 23 > simultaneous calls. In fact, maybe no more than 10 simultaneously. > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro > Sent: Tuesday, March 18, 2008 3:05 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] capacity > > On Tue, Mar 18, 2008 at 1:55 PM, Eve-Ellen Cole <[EMAIL PROTECTED]> > wrote: > > > > > > > > > > Hi, > > > > > > > > I am planning to deploy an Asterisk system to supply 4-6,000 students > with > > voicemail capabilities. The system will be set up with non-DIDs, route > > incoming calls to voicemail, then send an email notification. Anyone > with > > some ideas on how I should go about spec'ing the server this use? > > > > > > > > - Eve Ellen > > Strictly VM? How are the calls going to arrive? How many > simultaneous accesses, both leaving messages and retrieving (highest > peak). > > I believe Vonage uses Asterisk for their VM (not sure where I heard that). > > Thanks, > Steve Totaro > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Queue: Simple Queue Problem
> So you either need to go a Goto(context,4000,1) or to drop it to the queue > with Queue(console) etc. I have chosen to use Goto(context,4000,1) from a programmer's perspective although queue(console) works just as good. Thanks guys. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Have you tried disabling highpriority=yes in asterisk.conf? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH4G9SDQNt8rg0Kp4RAjIoAKCQEP/e8pR27gbz9p1ilGw8AvWA+wCgs7qX mIrPzDRPWsGt9goKwljsT0Q= =W2og -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with lumenvox
Hello all, how are you? I would like to know from someone uses or has used the engines of LumenVox for integration with the asterisk for voice recognition. Best Regards Josué ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Avaya 4610 handset
i was reading posts on wiki and noticed lots of posts about Avaya 4610 handset having issue with MWI, Anyone has any more updates? Is this still the case? Any good tutorial for configuring these phones and Asterisk? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2008-005: HTTP Manager ID is predictable
Asterisk Project Security Advisory - AST-2008-005 ++ | Product| Asterisk| |--+-| | Summary| HTTP Manager ID is predictable | |--+-| | Nature of Advisory | An attacker could hijack a manager session | |--+-| |Susceptibility| All users using the HTTP manager port | |--+-| | Severity | Minor | |--+-| |Exploits Known| No | |--+-| | Reported On | February 25, 2008 | |--+-| | Reported By | Dino A. Dai Zovi < ddz AT theta44 DOT org > | |--+-| | Posted On | March 18, 2008 | |--+-| | Last Updated On| March 18, 2008 | |--+-| | Advisory Contact | Tilghman Lesher < tlesher AT digium DOT com > | |--+-| | CVE Name | CVE-2008-1390 | ++ ++ | Description | Due to the way that manager IDs are calculated, this | | | 32-bit integer is likely to have a much larger than | | | average number of 1s, which greatly reduces the number | | | of guesses an attacker would have to make to | | | successfully predict the manager ID, which is used | | | across multiple HTTP queries to hold manager state. | | | | | | "The issue is the generation of session ids in the | | | AsteriskGUI HTTP server. | | | | | | When using Glibc, the implementation and state of rand() | | | and random() is | | | | | | shared. Asterisk uses random() to issue MD5 digest | | | authentication | | | | | | challenges and rand() bitwise-ORed with a malloc'd | | | pointer to generate | | | | | | AsteriskGUI session identifiers. An attacker can | | | synchronize with | | | | | | random() by retrieving 32 successive challenges and | | | predict all subsequent | | | | | | output of calls to random() and rand(). Because a| | | pointer returned by | | | | | | malloc has at best 21 bits of entropy, the attacker will | | | on average only | | | | | | need to guess 1448 session identifiers in order to steal | | | an established | | | | | | session. | | | | | | "The crux of the problem is that under Glibc, the| | |
[asterisk-users] AST-2008-004: Format String Vulnerability in Logger and Manager
Asterisk Project Security Advisory - AST-2008-004 ++ | Product | Asterisk | |+---| | Summary | Format String Vulnerability in Logger and Manager | |+---| | Nature of Advisory | Denial of Service | |+---| | Susceptibility | Remote Unauthenticated Sessions | |+---| | Severity | Moderate | |+---| | Exploits Known | No| |+---| |Reported On | March 13, 2008| |+---| |Reported By | Steve Davies (bugs.digium.com user stevedavies) | || | || Brandon Kruse (bugs.digium.com user bkruse) | |+---| | Posted On | March 18, 2008| |+---| | Last Updated On | March 18, 2008| |+---| | Advisory Contact | Joshua Colp <[EMAIL PROTECTED]>| |+---| | CVE Name | CVE-2008-1333 | ++ ++ | Description | Logging messages displayed using the ast_verbose logging | | | API call are not displayed as a character string, they | | | are displayed as a format string.| | | | | | Output as a result of the Manager command "command" is | | | not appended to the resulting response message as a | | | character string, it is appended as a format string. | | | | | | It is possible in both instances for an attacker to | | | provide a formatted string as a value for input which| | | can cause a crash. | ++ ++ | Resolution | Input given to both the ast_verbose logging API call and | || astman_append function is now interpreted as a character | || string and not as a format string.| ++ ++ | Affected Versions| || | Product | Release | | || Series | | |+-+-| |Asterisk Open Source| 1.0.x | Unaffected | |+-+-| |Asterisk Open Source| 1.2.x | Unaffected | |+-+-| |Asterisk Open Source| 1.4.x | Unaffected | |+-+-| |Asterisk Open Source| 1.6.x | All versions prior to | || | 1.6.0-beta6 | |+-+-| | Asterisk Business Edition | A.x.x | Unaffected | |+-+-| | Asterisk Business Edition | B.x.x | Unaffected | |---
Re: [asterisk-users] Newbie Queue: Simple Queue Problem
On Tue, 2008-03-18 at 18:20 +1100, Lee, John (Sydney) wrote: > I am trying to build a simple queue for the receptionist phone. > In other words, there is only 1 agent and that is the receptionist > phone. > > I just defined a few lines in queues.conf > [console] > strategy = ringall > member => SIP/4000 ;4000 is the console extension > > In extensions.conf, it is: > exten => 4000,1,Answer() > exten => 4000,n,Queue(console) > exten => 4000,n,HangUp() > > I pressed DND on 4000 and then call from another SIP phone (say 4001). > As expected, I saw 1 caller in the queue by "queue show" and that is > great. > exten => 4001,1,SetMusicOnHold() > exten => 4001,n,Dial(SIP/4001,20) > exten => 4001,n,VoiceMail,4001 > exten => 4001,n,Playback(vm-goodbye) > exten => 4001,n,Wait(2) > exten => 4001,n,HangUp() > > However, when I call from an outside line to another extension which I > then forward to 4000, I cannot get into the queue. > exten => 98786983,1,Answer() > exten => 98786983,n,Dial(SIP/4000,20) > exten => 98786983,n,HangUp() > > Any thoughts? The outside line coding should be exten => 98786983,1,Answer() exten => 98786983,n,Queue(console) exten => 98786983,n,HangUp() later, PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2008-003: Unauthenticated calls allowed from SIP channel driver
Asterisk Project Security Advisory - AST-2008-003 ++ | Product | Asterisk | |+---| | Summary | Unauthenticated calls allowed from SIP channel| || driver| |+---| | Nature of Advisory | Authentication Bypass | |+---| | Susceptibility | Remote Unauthenticated Sessions | |+---| | Severity | Major | |+---| | Exploits Known | No| |+---| |Reported On | March 12, 2008| |+---| |Reported By | Jason Parker <[EMAIL PROTECTED]> | |+---| | Posted On | March 18, 2008| |+---| | Last Updated On | March 18, 2008| |+---| | Advisory Contact | Jason Parker <[EMAIL PROTECTED]> | |+---| | CVE Name | CVE-2008-1332 | ++ ++ | Description | Unauthenticated calls can be made via the SIP channel| | | driver using an invalid From header. This acts similarly | | | to the SIP configuration option 'allowguest=yes', in | | | that calls with a specially crafted From header would be | | | sent to the PBX in the context specified in the general | | | section of sip.conf. | ++ ++ | Resolution | A fix has been added which checks for the option | || 'allowguest' to be enabled before determining that| || authentication is not required. | || | || As a workaround, modify the context in the general| || section of sip.conf to point to a non-trusted location| || (example: a non-existent context, or a context that does | || nothing but hang up the call).| ++ ++ | Affected Versions| || | Product| Release | | | | Series | | |--+-+---| | Asterisk Open Source | 1.0.x | All versions | |--+-+---| | Asterisk Open Source | 1.2.x | All versions prior to 1.2.27 | |--+-+---| | Asterisk Open Source | 1.4.x | All versions prior to | | | | 1.4.18.1 and 1.4.19-rc3 | |--+-+---| | Asterisk Business Edition | A.x.x | All versions | |--+-+---| | Asterisk Business Edition | B.x.x | All versions prior to B.2.5.1 | |--+-+---| | Asterisk Business Edition | C.x.x | All versions prior to C.1.6.2 | |--+-+---| | Aster
Re: [asterisk-users] Turn off MusicOnHold for individual User
> I might of got my wires crossed here, but I'm looking for a way to disable > musiconhold for individual users. Good question Adrian. I never thought about that but I googled a bit and here seems to be the answer: http://lists.digium.com/pipermail/asterisk-users/2007-August/193721.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call screening feature
The 'PrivacyManager' application in Asterisk would probably be a good bet. PaulH On Tue, 2008-03-18 at 11:54 +0530, Janu Mukherjee wrote: > Hi, > > I have our software with SIP running on it.I configured asterisk > server as proxy. How do I implement the call screening > features(incoming and outgoing) using asterisk server.Please suggest > me how to proceed on this. > > Thanks & Regards, > Jahnavi. > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2008-002: Two buffer overflows in RTP Codec Payload Handling
Asterisk Project Security Advisory - AST-2008-002 ++ | Product | Asterisk | |+---| | Summary | Two buffer overflows in RTP Codec Payload | || Handling | |+---| | Nature of Advisory | Exploitable Buffer Overflow | |+---| | Susceptibility | Remote Unauthenticated Sessions | |+---| | Severity | Critical | |+---| | Exploits Known | No| |+---| |Reported On | March 11, 2008| |+---| |Reported By | Mu Security Research Team | |+---| | Posted On | March 18, 2008| |+---| | Last Updated On | March 18, 2008| |+---| | Advisory Contact | Joshua Colp <[EMAIL PROTECTED]>| |+---| | CVE Name | CVE-2008-1289 | ++ ++ | Description | Two buffer overflows exist in the RTP payload handling | | | code of Asterisk. Both overflows can be caused by an | | | INVITE or any other SIP packet with SDP. The request may | | | need to be authenticated depending on configuration of | | | the Asterisk installation. | | | | | | The first overflow is caused by sending a payload number | | | that surpasses the programmed maximum payload number of | | | 256. This causes an invalid memory write outside of the | | | buffer. While this does not allow the attacker to write | | | arbitrary data it does allow the attacker to write a 0 | | | to other memory locations. | | | | | | The second overflow is caused by sending more than 32| | | RTP payloads. This causes a buffer on the stack to | | | overflow allowing the attacker to write values between 0 | | | and 256 (the maximum payload number) to memory locations | | | after the buffer.| ++ ++ | Resolution | Two fixes have been added to check the provided data to | || ensure it does not exceed static buffer sizes.| || | || When removing internal information regarding an RTP | || payload the given payload number will now be checked to | || make sure it does not exceed the maximum acceptable | || payload number. | || | || When reading RTP payloads from SDP a maximum limit of 32 | || in total will be enforced. Any further RTP payloads will | || be discarded. | ++ ++ | Affected Versions| || | Product | Release | | |
[asterisk-users] AEL2 Hint & Parking
I've been reading most of the day and can't seem to find a clear
definition of the syntax for parking lot hints in AEL2. I have tried
all of the following and they either do not light up the line button on
my Snom 300 or give syntax errors:
hint(park/701) 701 => {
ParkedCall(701);
}
hint(park:701) 701 => {
ParkedCall(701);
}
hint(park/[EMAIL PROTECTED]) 701 => {
ParkedCall(701);
}
hint(park:[EMAIL PROTECTED]) 701 => {
ParkedCall(701);
}
I have this in my context as well:
includes {
parkedcalls;
}
I do not see any indication on the CLI that Asterisk is attempting to
notify my sip phone of the status change and I have verbose and debug at 20.
Any ideas?
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[asterisk-users] (Critical Updates) Asterisk 1.2.27, 1.4.18.1, 1.4.19-rc3, 1.6.0-beta6 Released
The Asterisk.org development team has released four new versions of Asterisk to address critical security vulnerabilities. AST-2008-002 details two buffer overflows that were discovered in RTP codec payload type handling. * http://downloads.digium.com/pub/security/AST-2008-002.pdf * All users of SIP in Asterisk 1.4 and 1.6 are affected. AST-2008-003 details a vulnerability which allows an attacker to bypass SIP authentication and to make a call into the context specified in the general section of sip.conf. * http://downloads.digium.com/pub/security/AST-2008-003.pdf * All users of SIP in Asterisk 1.0, 1.2, 1.4, or 1.6 are affected. AST-2008-004 details some format string vulnerabilities that were found in the code handling the Asterisk logger and the Asterisk manager interface. * http://downloads.digium.com/pub/security/AST-2008-004.pdf * All users of Asterisk 1.6 are affected. Asterisk 1.2.27 and 1.4.18.1 are releases that only contain changes to fix these security vulnerabilities. In addition to fixes for these security issues, 1.4.19-rc3 and 1.6.0-beta6 contain a number of other bug fixes over the previous release candidates and beta releases for the upcoming 1.4.19 and 1.6.0 releases. We encourage all affected users of these security vulnerabilities to upgrade their installations as time permits. Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip Line Status/Pickup
Does anyone know of a way to make a Snom 300 phone monitor the parking lot extensions and allow one-button pickup with the programmable buttons? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On 3/18/08, Benny Amorsen <[EMAIL PROTECTED]> wrote: > "Steve Totaro" <[EMAIL PROTECTED]> writes: > > > I will probably continue this train of thought (1.2.X is more > > production ready) until these threads stop popping up on the list. > > > I think you're being too kind to 1.2.x. It has numerous problems, most > especially with locking in chan_sip. 1.4.x is a HUGE improvement. Who uses chan_sip? Long live IAX! :) But seriously, several of my clients use SIP exclusively, passing tens of thousand of calls a day on Asterisk 1.2.X with no issues. I have noticed that the load is slightly lower for SIP-only in 1.4, but I have not noticed any stability issues revolving around SIP on 1.2.X. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
"Steve Totaro" <[EMAIL PROTECTED]> writes: > I will probably continue this train of thought (1.2.X is more > production ready) until these threads stop popping up on the list. I think you're being too kind to 1.2.x. It has numerous problems, most especially with locking in chan_sip. 1.4.x is a HUGE improvement. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Queue: Simple Queue Problem
Robert Lister <[EMAIL PROTECTED]> writes: > So you either need to go a Goto(context,4000,1) or to drop it to the queue > with Queue(console) etc. There's also Dial(Local/[EMAIL PROTECTED]). Goto is almost always a better idea though. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] capacity
I have an Avaya Definity G3R. Calls to students will be routed through the G3R, to the Asterisk system so the caller can leave a message. I'm not sure how many channels I'll really need, but I expect no more than 23 simultaneous calls. In fact, maybe no more than 10 simultaneously. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, March 18, 2008 3:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] capacity On Tue, Mar 18, 2008 at 1:55 PM, Eve-Ellen Cole <[EMAIL PROTECTED]> wrote: > > > > > Hi, > > > > I am planning to deploy an Asterisk system to supply 4-6,000 students with > voicemail capabilities. The system will be set up with non-DIDs, route > incoming calls to voicemail, then send an email notification. Anyone with > some ideas on how I should go about spec'ing the server this use? > > > > - Eve Ellen Strictly VM? How are the calls going to arrive? How many simultaneous accesses, both leaving messages and retrieving (highest peak). I believe Vonage uses Asterisk for their VM (not sure where I heard that). Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call forward on Telco line
On Tue, Mar 18, 2008 at 1:37 PM, Tim Litwiller <[EMAIL PROTECTED]> wrote: > Is there any way we can make use of the call forwarding feature on our > Telco phone line. I've seen this question asked on this list before but > looking in the archive i don't see that it has been answered. > > If someone has this working or knows how please let me know. > > Thanks. You could assign an exten for call forwarding and then use dial to get that feature like *68 or whatever the code is and then use senddtmf. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] capacity
On Tue, Mar 18, 2008 at 1:55 PM, Eve-Ellen Cole <[EMAIL PROTECTED]> wrote: > > > > > Hi, > > > > I am planning to deploy an Asterisk system to supply 4-6,000 students with > voicemail capabilities. The system will be set up with non-DIDs, route > incoming calls to voicemail, then send an email notification. Anyone with > some ideas on how I should go about spec'ing the server this use? > > > > - Eve Ellen Strictly VM? How are the calls going to arrive? How many simultaneous accesses, both leaving messages and retrieving (highest peak). I believe Vonage uses Asterisk for their VM (not sure where I heard that). Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] capacity
Hi, I am planning to deploy an Asterisk system to supply 4-6,000 students with voicemail capabilities. The system will be set up with non-DIDs, route incoming calls to voicemail, then send an email notification. Anyone with some ideas on how I should go about spec'ing the server this use? - Eve Ellen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How is uniqueid computed
Thanks Mindaugas. Regards, Sanjay. - Original Message - From: "Mindaugas Kezys" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, March 18, 2008 10:26:37 PM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] How is uniqueid computed Hello, Uniqueid = (call initiation time in unix time format) . (call count since asterisk restart / 2 ) If call is transfered or it is leg2 then: Uniqueid = (call initiation time in unix time format) . (call count since asterisk restart / 2 + 1) This is from observations, i can be mistaken. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing Solution for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, March 18, 2008 6:12 PM To: asterisk-users Subject: [asterisk-users] How is uniqueid computed Can anyone let me know how the uniqueid for a call is computed in asterisk? Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call forward on Telco line
Is there any way we can make use of the call forwarding feature on our Telco phone line. I've seen this question asked on this list before but looking in the archive i don't see that it has been answered. If someone has this working or knows how please let me know. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk.conf uniquename or sysname for uniqueid field in CDR
On 3/18/08, Vieri <[EMAIL PROTECTED]> wrote: > > --- Vieri <[EMAIL PROTECTED]> wrote: > > > I set uniquename = MYHOST in asterisk.conf (under > > [options]) so that my uniqueid data shows up as > > MYHOST.time.seq. > > > > First of all, I would like to know if uniquename (or > > sysname?) will still be valid across future * > > versions > > (mainly 1.6). > > > > Secondly, is there a way to specify uniquename as an > > asterisk option at the command line? (asterisk -h > > doesn't show me anything regarding this feature) > > > > Finally, how can I set uniquename to a system value > > (say, dynamically set to whatever `hostname` > > yields)? > > Something like > > uniquename = `hostname` > > so that I don't have to statically set it on each > > asterisk server? > > > I just realized that uniquename is only available > after applying the BRISTUFF patches. > So let me rephrase my question: > will Asterisk ever include the "uniquename" feature in > its base code? If not, why? > (I would prefer not to apply BRIstuff since I don't > have Junghanns hardware). Look into doc/asterisk-conf.txt - probably you can use "systemname". Asterisk config files also support #exec directive, so you can create your regular asterisk.conf without sysname and create shell script: #!/bin/bash cat asterisk.conf.template echo "sysname=`hostname`." Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call signalling on BT FeatureLine Compact(Sangoma A200)
Thanks, sorry for not being thougher, Yes I swapped the cables and the fault moved to Channel 2 and Channel 3, I did this to test the Sangoma FXS modules and they all work fine with the "working fine" lines. So I believe the card,modules and cables to be good. PaulG. On Tue, Mar 18, 2008 at 4:55 PM, Ade Vickers <[EMAIL PROTECTED]> wrote: > Paul Goodyear wrote: > > > I have had a BT phone plugged into these lines for about 3 week > > prior to testing on asterisk, and all the lines are fine. Even > > the first line, it rings and answers ok. > > Apologies if this seems dumb, but have you done the "swap the cables around" > test? i.e. swap the cables plugged into BT1 & BT2 to make sure the fault > stays on BT1? > > If it does - then it's probably something on BT's end; if it moves, you've > eliminated BT from the equation... > > From what's been posted so far, I'd anticipate a cable fault (either between > Asterisk & the BT socket, or on the other side of the BT socket...) > > Cheers, > Ade. > > No virus found in this outgoing message. > Checked by AVG. > Version: 7.5.519 / Virus Database: 269.21.7/1331 - Release Date: 16/03/2008 > 10:34 > > > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy problem causing playback () to fail
Atis Lezdins wrote: > On 3/18/08, Pete Kay <[EMAIL PROTECTED]> wrote: > ztdummy is required by meetme application. If you have no intention to > use it, you might very well remove. > > I've seen this problem once, however recompiling everything and > restarting helped me. I would suggest you just doing "make clean" on > zaptel and asterisk, then compile first zaptel, then asterisk. > There is something with Playback() where ztdummy helps, I have had issues with skipping audio or gsm audio files that won't playback until ztdummy is loaded. I have seen this in 1.4 and in the 1.6 betas. I also think this guy needs to recompile zaptel and look for errors in compiling. -Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
I believe most of them will be in 1.4.19-rc3 (and in SVN), but I applied patches to 1.4.19-rc2 from: Patches from 11712 and 12098. Plus another one I reported as 12162. Norman Franke Answering Service for Directors, Inc. www.myasd.com On Mar 18, 2008, at 12:11 PM, [EMAIL PROTECTED] wrote: On Tue, 2008-03-18 at 11:05 -0400, Norman Franke wrote: I've also applied a few SIP-related patches from various bug reports and things are much, much more stable. Mind sharing which patches you have applied? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy problem causing playback () to fail
On Tue, Mar 18, 2008 at 11:28:55PM +0800, Pete Kay wrote: > Hi, I am having problem with my Asterisk installation and find out it > has to do with ztdummy. > > if the ztdummy module is loaded, the asterisk playback() command > will not play files. DTMF is still properly received. If the ztdummy > module is unloaded, sound playback works again. > > Here is my version > zaptel-1.4.9.2 > linux-source-2.6.18 > asterisk-1.4.18 > > > Can anyone tell me how to fix it? Or should I just have ztdummy > removed forever and the system will work? > > I saw from manual that ztdummy is required. What Linux distribution is it? What is the output of 'zttest -v -c 6' ? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Off-topic note: On Tue, Mar 18, 2008 at 05:45:04PM +0200, Atis Lezdins wrote: > If you're not using safe_asterisk script to start it, you should > execute also "ulimit -c unlimited" before launching asterisk.. Without -g (at least on Linux) Asterisk will refuse to generate core dumps. With -g it will generate core files but will also set the ulimit to "unlimited". With safe_asterisk you have -g enabled by default, and hence ulimit -c unlimited on by default. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using dedicated eth2 card for SIP trunk line to ISP provider - how to setup ?
Hi, I'm about to test VOIP connection (from my ISP provider) directly through dedicated network card instead of going through ADSL gateway with analog phone port - SPA 3000 - Asterisk. I need to have eth2 set on dhcp (to retrieve IP automatically) and then work with it under Asterisk as dedicated VOIP trunk. Anyone with more insight how to setup such situation ? Any more info anywhere ? Thanks in advance, regards, Bulek. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call signalling on BT FeatureLine Compact(Sangoma A200)
Paul Goodyear wrote: > I have had a BT phone plugged into these lines for about 3 week > prior to testing on asterisk, and all the lines are fine. Even > the first line, it rings and answers ok. Apologies if this seems dumb, but have you done the "swap the cables around" test? i.e. swap the cables plugged into BT1 & BT2 to make sure the fault stays on BT1? If it does - then it's probably something on BT's end; if it moves, you've eliminated BT from the equation... >From what's been posted so far, I'd anticipate a cable fault (either between Asterisk & the BT socket, or on the other side of the BT socket...) Cheers, Ade. No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.21.7/1331 - Release Date: 16/03/2008 10:34 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy problem causing playback () to fail
Hi, I found another weird problem. If I don't have ztdummy, then when I connect using a xlite in another computer within the same lan, I get errored: 00:40:08 Registering user '[EMAIL PROTECTED]' 00:40:08 Failed registration for '[EMAIL PROTECTED]' with cause 'service or option not implemented' 00:50:50 Registering user '[EMAIL PROTECTED]' If I turned ztdummy on, I can connect. Any idea why? Pete On Tue, Mar 18, 2008 at 11:53 PM, Atis Lezdins <[EMAIL PROTECTED]> wrote: > On 3/18/08, Pete Kay <[EMAIL PROTECTED]> wrote: > > > > Hi, I am having problem with my Asterisk installation and find out it > > has to do with ztdummy. > > > > if the ztdummy module is loaded, the asterisk playback() command > > will not play files. DTMF is still properly received. If the ztdummy > > > > module is unloaded, sound playback works again. > > > > Here is my version > > zaptel-1.4.9.2 > > linux-source-2.6.18 > > asterisk-1.4.18 > > > > > > Can anyone tell me how to fix it? Or should I just have ztdummy removed > > forever and the system will work? > > > > > > I saw from manual that ztdummy is required. > > ztdummy is required by meetme application. If you have no intention to > use it, you might very well remove. > > I've seen this problem once, however recompiling everything and > restarting helped me. I would suggest you just doing "make clean" on > zaptel and asterisk, then compile first zaptel, then asterisk. > > Regards, > Atis > > -- > Atis Lezdins, > VoIP Project Manager / Developer, > [EMAIL PROTECTED] > Skype: atis.lezdins > Cell Phone: +371 28806004 > Cell Phone: +1 800 7300689 > Work phone: +1 800 7502835 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How is uniqueid computed
Hello, Uniqueid = (call initiation time in unix time format) . (call count since asterisk restart / 2 ) If call is transfered or it is leg2 then: Uniqueid = (call initiation time in unix time format) . (call count since asterisk restart / 2 + 1) This is from observations, i can be mistaken. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing Solution for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, March 18, 2008 6:12 PM To: asterisk-users Subject: [asterisk-users] How is uniqueid computed Can anyone let me know how the uniqueid for a call is computed in asterisk? Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [SOLVED] GXP2000 and asterisk 1.0.9
Switching the dtmf mode to RFC2833 solved my problem, thanks a lot Sam Good work everyone -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Lutgring, Sam Inviato: giovedì 14 febbraio 2008 13.55 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] R: GXP2000 and asterisk 1.0.9 Try switching your DTMF mode on both asterisk and the phone to RFC2833. I have not seen the issue that you are describing, but I had some very strange issues like call hang-ups when I was using INFO. After switching the issues were gone and I have had no further troubles. Hope this helps you. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis Sent: Thursday, February 14, 2008 3:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] R: GXP2000 and asterisk 1.0.9 Thanks Henry, anyway the phone is always registered when i get the busy tone * Name : 502 Secret : MD5Secret: Context : local Language : it FromUser : FromDomain : Callgroup: 1 (2) Pickupgroup : 1 (2) Mailbox : LastMsgsSent : -1 Dynamic : Yes Expire : 703 seconds Expiry : 900 Insecure : No Nat : No ACL : No CanReinvite : No PromiscRedir : No DTMFmode : info LastMsg : 0 ToHost : Addr->IP : 192.168.13.171 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Username : 502 Codecs : 0x8010f (g723|gsm|ulaw|alaw|g729|h263) Codec Order : (alaw|ulaw|gsm|g729|g723) Status : OK (22 ms) Useragent: Grandstream GXP2000 1.1.5.15 Full Contact : sip:[EMAIL PROTECTED]:5060;transport=udp;user=phone Any idea? Thanks again to all -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Henry Devito Inviato: mercoledì 13 febbraio 2008 22.01 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9 Is your phone actually registered to the switch. go to the CLI and do a 'sip show peers' see if extension 502 is registered. Making an outbound call does not prove that the phone is registered. - Original Message - From: "C F" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, February 13, 2008 2:09 PM Subject: Re: [asterisk-users] GXP2000 and asterisk 1.0.9 > Just check DND if it's on on the phone or not. > What is the CLI output when you try making a phone call? > Why don't you try it with a later version of astrisk and a Phone? > > On Feb 13, 2008 10:58 AM, Giordano Grandis <[EMAIL PROTECTED]> wrote: >> >> >> Hi all gusy, >> i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a >> few >> go in "busy" state, if you call it get the busy tone but the phone can >> male >> any type of call. >> This is my sip.conf >> >> [502] >> language = it >> username = 502 >> secret = >> host = dynamic >> type = friend >> context = local >> canreinvite = yes >> dtmfmode = info >> callgroup = 1 >> pickupgroup = 1 >> callerid = 502 <502> >> >> Under Grandstream's support suggest, I set "Use randmom port" to yes and >> "Nat traversal (STUN)" to "No, but send keep alive" but without success. >> This is the firmware version: Program-- 1.1.5.15Bootloader-- 1.1.5.6 >> >> Anyone can help me ? >> >> Thanks in advance >> >> Giordano >> >> >> No virus found in this outgoing message. >> Checked by AVG Free Edition. >> Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: >> 12/02/2008 >> 15.20 >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 15.20 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1277 - Release Date: 13/02/2008 20.00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/lis
Re: [asterisk-users] Turn off MusicOnHold for individual User
Anyone have an idea on this? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Marsh Sent: 17 March 2008 17:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Turn off MusicOnHold for individual User Hi All, I might of got my wires crossed here, but I'm looking for a way to disable musiconhold for individual users. I had thought that putting the sip.conf entry to: [690] type=friend context=from-sip secret=* qualify=yes host=dynamic canreinvite=no nat=yes mailbox=2090 callerid=2090 musiconhold=silent and then putting an entry in musiconhold.conf like: [silent] mode=files directory=/var/lib/asterisk/mohmp3-empty ;(no files in this dir) I thought this would do it.. but testing shows it still uses the "default" class. I know I could use SetMusicOnHold and in extensions.conf, but that would require a special dialplan prefix or something, so hoped the sip.conf would work. Obviously I've missed something, but what? Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On 3/18/08, Ben Willcox <[EMAIL PROTECTED]> wrote: > > A million calls sounds good, but 2 weeks, not so good. It's a bit > disappointing to me that crashing /ever/ is acceptable, I had always had > the understanding that asterisk was supposed to be rock-solid. I suppose > it's some consolation that its not just me that has problems! > > Thanks for all the input. I think short term I will restart asterisk > daily, then the action plan is to revert back to Debian Etch, and then > install asterisk 1.4.18 from source, and hopefully this will improve > things. Keep in mind that my tests go from 0 to 400 calls in about 1 minute then they keep that volume for several hours, and I kept running them for two weeks, and about 6 hours into the last test is when it crashed. I should mention that 1.2.26.2 is what I still use on all of my production servers and they will go for months without a crash. As for rebooting nightly or weekly, that is something we do on a lot of our high-volume servers just to be safe. When pushing Asterisk to high concurrent call volumes it is a good idea to give it a fresh start every day if you can. If Asterisk is being used as a standard office PBX it should be able to run for months with no crashes. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On Tue, Mar 18, 2008 at 8:05 AM, Gordon Henderson <[EMAIL PROTECTED]> wrote: > On Tue, 18 Mar 2008, Steve Totaro wrote: > > > Why not try a different OS such as CentOS for now? That would be my next > step. > > I wouldn't suggest chasing distros is the way to solve issues, especially > if you're happy with the hardware. > > Personally, I'd go back to Debian, but stick to stable (Etch) and then > compile and install a custom kernel tailored exactly to your hardware, > then compile and install your own asterisk from source. > > But only because that's what I do, and it works for me ... > > Gordon Well personally, I would go to 1.2.x unless there was some feature in 1.4 that is absolutely needed but the OP said that was not a long term option. I have deployed ONE 1.4 system and that is because I had to, no work arounds due to hardware (unless zaptel 1.4 plays nice with Asterisk 1.2). I will probably continue this train of thought (1.2.X is more production ready) until these threads stop popping up on the list. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)
> Why you got > featureline on one line and not the other 2 is odd to me, but that's BT > saledroids for you Sorry, this must be me then, I was told that FeatureLine was on the first line, but I do need to dial 9 for the other two lines, so I would presume that FeatureLine Compact is on all 3 lines. > But before you go any further, I'd suggest going to Argos and getting 1 or > 2 standard £1.99 analogue phones and plugging them in and test the lines > with the phones first I have had a BT phone plugged into these lines for about 3 week prior to testing on asterisk, and all the lines are fine. Even the first line, it rings and answers ok. If the first line is setup the same as the other lines, and one isn't working (the first line) but the others are, would this mean there is a fault on that line? Thanks, PaulG. On Tue, Mar 18, 2008 at 2:50 PM, Gordon Henderson <[EMAIL PROTECTED]> wrote: > > On Tue, 18 Mar 2008, Paul Goodyear wrote: > > > Hi, > > > > I have a TrixBox install with a Sangoma A200 and 4 FXO ports, there > > are 3 BT lines connected directly to these ports. > > > > One of the lines has BT FeatureLine Compact and this is the line I am > > having problems with, the other 2 lines are working perfectly, > > detecting CID, answering incoming calls and placing external calls via > > SIP devices. > > > > I am receiving a error log entry: > > > > chan_zap.c: Ring/Off-hook in strange state 6 on channel 1 > > > > Incoming calls are detected by asterisk, however answering the SIP > > devices does not answer the call, and placing a call via line one does > > nothing, just silence. > > > > I contacted BT about it (I know, what was I expecting!) they informed > > me that I must use the number 9 to access a external number! I have > > asked them to pass it to the technical department to see if they have > > any input. > > > > Is there someother signalling I should be using to detect the incoming > > calls on a BT FeatureLine? I have tried Groudstart but asterisk fails > > to load chan_zap due to: > > > > Mar 18 14:01:26 ERROR[28951] chan_zap.c: Signalling requested on > > channel 1 is FXS Groundstart but line is in FXS Kewlstart signalling > > Mar 18 14:01:26 ERROR[28951] chan_zap.c: Unable to register channel '1' > > > > Any help, or ideas on what to try? > > Mark it down to experience. > > BT nearly always try to sell featureline on business lines these days. > "Would sir like a 3 of 5 year feature line contract"? > > When what you really wanted was just 3 lines in a hunt-group on a single > number (possibly, I don't know exactly what you want) > > As for signalling, it's no different on the feature line to any other BT > POTS line, you just need to prefix outgoing calls with '9'. Why you got > featureline on one line and not the other 2 is odd to me, but that's BT > saledroids for you > > So for the dialling issues, I'd suggest a trixbox list/forum to start > with, but if that fails, then you'll need to post your configs here - > zapata.conf, zaptel.conf, etc. to start with, then the dialplan to carry > on with... > > But before you go any further, I'd suggest going to Argos and getting 1 or > 2 standard £1.99 analogue phones and plugging them in and test the lines > with the phones first... Just in-case.. Stranger things have been know to > happen (Then again, this is BT and I'm now no-longer surprised when > thing's aren't quite to plan...) > > Good luck, > > Gordon > (Also in the UK, facing similar fristrations with BT at times too!) > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy problem causing playback () to fail
Atis Lezdins wrote: > ztdummy is required by meetme application. If you have no intention to > use it, you might very well remove. > And music on hold, if you don't have a timing source. -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How is uniqueid computed
Can anyone let me know how the uniqueid for a call is computed in asterisk? Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy problem causing playback () to fail
On 3/18/08, Pete Kay <[EMAIL PROTECTED]> wrote: > > Hi, I am having problem with my Asterisk installation and find out it > has to do with ztdummy. > > if the ztdummy module is loaded, the asterisk playback() command > will not play files. DTMF is still properly received. If the ztdummy > > module is unloaded, sound playback works again. > > Here is my version > zaptel-1.4.9.2 > linux-source-2.6.18 > asterisk-1.4.18 > > > Can anyone tell me how to fix it? Or should I just have ztdummy removed > forever and the system will work? > > > I saw from manual that ztdummy is required. ztdummy is required by meetme application. If you have no intention to use it, you might very well remove. I've seen this problem once, however recompiling everything and restarting helped me. I would suggest you just doing "make clean" on zaptel and asterisk, then compile first zaptel, then asterisk. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On Tue, 2008-03-18 at 11:05 -0400, Norman Franke wrote: > I've also applied a few SIP-related patches from various bug reports > and things are much, much more stable. Mind sharing which patches you have applied? Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
I would suggest taking latest 1.4 branch from SVN (or 1.4.19-rc3 when it's out). There has been few deadlocks fixed since rc2. Recompile asterisk with DEBUG_THREADS enabled (in "make menuselect"), If you're not using safe_asterisk script to start it, you should execute also "ulimit -c unlimited" before launching asterisk.. When your asterisk is deadlocked, open CLI and execute "core show locks". Copy that output, and submit to bugs.digium.com - it will tell developers where exactly is problem. Then, do "killall -11 asterisk". It will dump asterisk to core file, and that might provide helpful information later. If your have been requested backtraces, look in /tmp (or in directory you launched asterisk from) for core file. Open that core file with "gdb /usr/sbin/asterisk core." and take a dump of "thread apply all bt full" (make sure you set "set pagination off" in gdb before this) Regards, Atis On 3/18/08, Norman Franke <[EMAIL PROTECTED]> wrote: > > Check around on bugs.digium.com. You'll find a number of issues reported > that sound similar. I'm hoping that 1.4.19 will fix a lot of stuff, since > the release candidates seem much more stable to me. I couldn't keep Asterisk > up for more than a few days before on 1.4.18. I've also applied a few > SIP-related patches from various bug reports and things are much, much more > stable. > > 1.4.17, which you mentioned, is also very buggy. 1.4.18 fixed many issues. > > Norman Franke > Answering Service for Directors, Inc. > www.myasd.com > > On Mar 18, 2008, at 7:40 AM, > [EMAIL PROTECTED] wrote: > > > We have been experiencing some ongoing reliability problems with > > Asterisk for quite some time, and I am trying to find out if anyone else > > has experienced the same problems. > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Hi All, Thanks for all the replies. Here are my responses to the responses: On Tue, 2008-03-18 at 06:13 -0400, Al Baker wrote: > Curious, you mention "a number of problems" that have "gone on for months" > Question: Have you reported ANY or ALL of them to DIGIUM and if so > what has been their response on each of these problems ? We have been working very closely with the reseller that supplied us with the system, and although we have made progress over this time and they have given us a lot of technical support, I now feel that it will be quicker to progress the current issues independently. I don't know if the issues were escalated as far as Digium though. Tzafrir Cohen wrote: > The symptoms you mention suggest some sort of deadlock. Please enable > debug and the full log. Maybe this will provide some hints. But please > check that the full log is rotated in /etc/logrotate.d/asterisk . > > Can you reproduce this situation? e.g.: by extensive usage of the > manager interface? If so, it might help for testing. I will enable full debug logging. I suspect that we could reproduce the original problem with the manager interface by stress testing it with multiple connections, but I'm not sure if this is the same problem that we are currently experiencing. I also want to avoid causing problems on our production system at the moment, as it is rather 'delicate' as far as the users are concerned at the moment. Steve Totaro wrote: > Why not try a different OS such as CentOS for now? That would be my > next step. I have considered this, to at least to establish whether it is a Debian specific problem, either with the asterisk packages themselves, or some other configuration or package issue. I am umming and ahhing between this and Gordon's suggestion below: Gordon Henderson wrote: > Personally, I'd go back to Debian, but stick to stable (Etch) and > then > compile and install a custom kernel tailored exactly to your > hardware, > then compile and install your own asterisk from source. I'm thinking that this may be the way I should go, then I will have the freedom to install any version of asterisk that I need, whilst also keeping my favourite distro. Doug Lytle wrote: > Two things, > > 1.) On your queue setup, avoid using AgenCallbackLogin, it's known > to > cause deadlocked channels. > 2.) Restart the Asterisk service once a week. I do this via a CRON > job > at 3am on Sundays. We're actually not using Agents on our queues, just SIP channels, so hopefully this is not the problem. We simulate 'agents' logging in and out by pausing and unpausing queue members. I am now going to add a cron job to restart asterisk daily, in the hope that until the problem is resolved properly, at least it will help relieve some of the pain by making it stable for a full 24hrs at a time. Matt Florell wrote: > I would suggest upgrading to at least 1.4.18. I was able to run it for > about 2 weeks and almost one million calls before I could get it to > crash, and the 1.4.19RC2 seems to fix even more of the locking issues > as well. I know a lot of these problems still existed under 1.4.17. A million calls sounds good, but 2 weeks, not so good. It's a bit disappointing to me that crashing /ever/ is acceptable, I had always had the understanding that asterisk was supposed to be rock-solid. I suppose it's some consolation that its not just me that has problems! Thanks for all the input. I think short term I will restart asterisk daily, then the action plan is to revert back to Debian Etch, and then install asterisk 1.4.18 from source, and hopefully this will improve things. Thanks, Ben ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call signalling on BT FeatureLine Compact (SangomaA200)
Maybe http://www.voipuser.org/forum_topic_1791.html ? > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Paul Goodyear > Sent: 18 March 2008 14:07 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Call signalling on BT FeatureLine > Compact (SangomaA200) > > Hi, > > I have a TrixBox install with a Sangoma A200 and 4 FXO ports, > there are 3 BT lines connected directly to these ports. > > One of the lines has BT FeatureLine Compact and this is the > line I am having problems with, the other 2 lines are working > perfectly, detecting CID, answering incoming calls and > placing external calls via SIP devices. > > I am receiving a error log entry: > > chan_zap.c: Ring/Off-hook in strange state 6 on channel 1 > > Incoming calls are detected by asterisk, however answering > the SIP devices does not answer the call, and placing a call > via line one does nothing, just silence. > > I contacted BT about it (I know, what was I expecting!) they > informed me that I must use the number 9 to access a external > number! I have asked them to pass it to the technical > department to see if they have any input. > > Is there someother signalling I should be using to detect the > incoming calls on a BT FeatureLine? I have tried Groudstart > but asterisk fails to load chan_zap due to: > > Mar 18 14:01:26 ERROR[28951] chan_zap.c: Signalling requested > on channel 1 is FXS Groundstart but line is in FXS Kewlstart > signalling Mar 18 14:01:26 ERROR[28951] chan_zap.c: Unable to > register channel '1' > > Any help, or ideas on what to try? > > Thanks > > PaulG. > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztdummy problem causing playback () to fail
Hi, I am having problem with my Asterisk installation and find out it has to do with ztdummy. if the ztdummy module is loaded, the asterisk playback() command will not play files. DTMF is still properly received. If the ztdummy module is unloaded, sound playback works again. Here is my version zaptel-1.4.9.2 linux-source-2.6.18 asterisk-1.4.18 Can anyone tell me how to fix it? Or should I just have ztdummy removed forever and the system will work? I saw from manual that ztdummy is required. Thanks, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Check around on bugs.digium.com. You'll find a number of issues reported that sound similar. I'm hoping that 1.4.19 will fix a lot of stuff, since the release candidates seem much more stable to me. I couldn't keep Asterisk up for more than a few days before on 1.4.18. I've also applied a few SIP-related patches from various bug reports and things are much, much more stable. 1.4.17, which you mentioned, is also very buggy. 1.4.18 fixed many issues. Norman Franke Answering Service for Directors, Inc. www.myasd.com On Mar 18, 2008, at 7:40 AM, [EMAIL PROTECTED] wrote: We have been experiencing some ongoing reliability problems with Asterisk for quite some time, and I am trying to find out if anyone else has experienced the same problems. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)
On Tue, Mar 18, 2008 at 02:06:44PM +, Paul Goodyear wrote: > Is there someother signalling I should be using to detect the incoming > calls on a BT FeatureLine? I have tried Groudstart but asterisk fails > to load chan_zap due to: > > Mar 18 14:01:26 ERROR[28951] chan_zap.c: Signalling requested on > channel 1 is FXS Groundstart but line is in FXS Kewlstart signalling > Mar 18 14:01:26 ERROR[28951] chan_zap.c: Unable to register channel '1' You should configure zaptel.conf the same way (or in 1.6: just configure zaptel.conf , and use signalling=auto in zapata.conf) . But then again: really use groundstart? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)
On Tue, 18 Mar 2008, Paul Goodyear wrote: Hi, I have a TrixBox install with a Sangoma A200 and 4 FXO ports, there are 3 BT lines connected directly to these ports. One of the lines has BT FeatureLine Compact and this is the line I am having problems with, the other 2 lines are working perfectly, detecting CID, answering incoming calls and placing external calls via SIP devices. I am receiving a error log entry: chan_zap.c: Ring/Off-hook in strange state 6 on channel 1 Incoming calls are detected by asterisk, however answering the SIP devices does not answer the call, and placing a call via line one does nothing, just silence. I contacted BT about it (I know, what was I expecting!) they informed me that I must use the number 9 to access a external number! I have asked them to pass it to the technical department to see if they have any input. Is there someother signalling I should be using to detect the incoming calls on a BT FeatureLine? I have tried Groudstart but asterisk fails to load chan_zap due to: Mar 18 14:01:26 ERROR[28951] chan_zap.c: Signalling requested on channel 1 is FXS Groundstart but line is in FXS Kewlstart signalling Mar 18 14:01:26 ERROR[28951] chan_zap.c: Unable to register channel '1' Any help, or ideas on what to try? Mark it down to experience. BT nearly always try to sell featureline on business lines these days. "Would sir like a 3 of 5 year feature line contract"? When what you really wanted was just 3 lines in a hunt-group on a single number (possibly, I don't know exactly what you want) As for signalling, it's no different on the feature line to any other BT POTS line, you just need to prefix outgoing calls with '9'. Why you got featureline on one line and not the other 2 is odd to me, but that's BT saledroids for you So for the dialling issues, I'd suggest a trixbox list/forum to start with, but if that fails, then you'll need to post your configs here - zapata.conf, zaptel.conf, etc. to start with, then the dialplan to carry on with... But before you go any further, I'd suggest going to Argos and getting 1 or 2 standard £1.99 analogue phones and plugging them in and test the lines with the phones first... Just in-case.. Stranger things have been know to happen (Then again, this is BT and I'm now no-longer surprised when thing's aren't quite to plan...) Good luck, Gordon (Also in the UK, facing similar fristrations with BT at times too!)___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)
Hi, I have a TrixBox install with a Sangoma A200 and 4 FXO ports, there are 3 BT lines connected directly to these ports. One of the lines has BT FeatureLine Compact and this is the line I am having problems with, the other 2 lines are working perfectly, detecting CID, answering incoming calls and placing external calls via SIP devices. I am receiving a error log entry: chan_zap.c: Ring/Off-hook in strange state 6 on channel 1 Incoming calls are detected by asterisk, however answering the SIP devices does not answer the call, and placing a call via line one does nothing, just silence. I contacted BT about it (I know, what was I expecting!) they informed me that I must use the number 9 to access a external number! I have asked them to pass it to the technical department to see if they have any input. Is there someother signalling I should be using to detect the incoming calls on a BT FeatureLine? I have tried Groudstart but asterisk fails to load chan_zap due to: Mar 18 14:01:26 ERROR[28951] chan_zap.c: Signalling requested on channel 1 is FXS Groundstart but line is in FXS Kewlstart signalling Mar 18 14:01:26 ERROR[28951] chan_zap.c: Unable to register channel '1' Any help, or ideas on what to try? Thanks PaulG. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma FXO/FXS config
Excellent, thanks for that Tzafrir. PaulG. On Tue, Mar 18, 2008 at 1:18 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > On Tue, Mar 18, 2008 at 12:02:26PM +, Paul Goodyear wrote: > > Hi all, > > > > I bought a Sangoma A200 card from an online supplier and explained > > exactly what I wanted, > > > > 3 incoming phone lines to PBX and a life line (some where to connect a > > standard BT phone to the PBX incase the power goes, making the BT > > phone ring). > > > > I was told to order > > > > 1 x FXS module (2 FXS ports) > > 2 x FXO modules (4 FXO ports) > > > > However being a complete noob, I have connected the 3 lines to the PBX > > and have all but 1 line working (BT Featureline problems), but after a > > month or playing, I realised I have 4 FXS ports and 2 FXO ports. > > > > --- > > [EMAIL PROTECTED] ~]# ztcfg -vvv > > > > Zaptel Version: 1.4.9.2 > > Echo Canceller: MG2 > > Configuration > > == > > > > > > Channel map: > > > > Channel 01: FXS Kewlstart (Default) (Slaves: 01) > > Channel 02: FXS Kewlstart (Default) (Slaves: 02) > > Channel 03: FXS Kewlstart (Default) (Slaves: 03) > > Channel 04: FXS Kewlstart (Default) (Slaves: 04) > > Channel 07: FXO Kewlstart (Default) (Slaves: 07) > > Channel 08: FXO Kewlstart (Default) (Slaves: 08) > > > > 6 channels to configure. > > --- > > > > Does this mean they sent me 2 FXS instead of FXO's or is this the a > > FXS is a FXO and FXO is a FXS thing? Calls are detected and answered > > perfectly via Channels 2 and 3! > > FXO signalling is used for FXS channels and vice versa, so all's well. > > -- >Tzafrir Cohen > icq#16849755 jabber:[EMAIL PROTECTED] > +972-50-7952406 mailto:[EMAIL PROTECTED] > http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] php web chat + asterisk -> callcenter
I would recommend you Asterisk for Voice and Video and XMPP for Chat. Asterisk in parallel with Jabberd2 (XMPP server) may feet your requirements, and if you use a XMPP MSN Transport Gateway you can do even more. On Mon, Mar 17, 2008 at 5:50 PM, Carlos Carvalhar < [EMAIL PROTECTED]> wrote: > Hello, > > > > How can I make a live chat (mainly text, but with voice/video chat if > possible) interacting with asterisk? > > Can asterisk control simultaneously the queue between people calling by > phone and people by web chat? > > > > At my work, there is a call center using asterisk to control the queue of > the clients (by phone) already. This part is ok. > > But now I need to make a chat room at the website and someone of the call > center will need to answer that client. > > > > So my doubt is how to implement a solution that identifies an operator who > is free and put him to talk by chat and then make him busy to phone calls. > > After the web chat is finished, the operator turns automatically free > again. > > > > I'm planning to use php to set an asterisk variable telling the agent is > free or busy. > > Can you tell me the asterisk apis involved with busy agents? > > Eg.: how do I set one agent as busy? I can set it by php, don't I? > > > > Is there any software like this one, Centriphone Millennium, for free? > > http://www.vocalcom.com/asterisk.html > > > > Is there any free solution? > > > > Where can I find information about how to settle asterisk variables (to > get and to set) with php programming? > > > > I need to make a php page that settles a property of asterisk in runtime. > > Is it possible? How do I do it? > > > > Thanks in advance, > > Carlos > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call screening feature
Your solution is Asterisk Manager Interface http://www.voip-info.org/wiki-Asterisk+manager+API On Tue, Mar 18, 2008 at 6:24 AM, Janu Mukherjee <[EMAIL PROTECTED]> wrote: > Hi, > > I have our software with SIP running on it.I configured asterisk server as > proxy. How do I implement the call screening features(incoming and outgoing) > using asterisk server.Please suggest me how to proceed on this. > > Thanks & Regards, > Jahnavi. > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with incoming calls on Broadvoice after upgrade to 1.4.18
Hi Raj, Sorry for the delay. The NIC in my server running Asterisk died so I wasn't able to verify until just now. After commenting out the secret= line, calls go through. I'll contact their support, but I'm sure they'll be as useless as ever. This may be the last straw for them. Thanks again Raj On Sun, Mar 16, 2008 at 6:44 PM, Raj Jain <[EMAIL PROTECTED]> wrote: > Based on the trace alone, it seems like a problem on their end. You > may want to try shutting off INVITE authentication (by commenting out > secret= line in your sip.conf) to see if the call goes through. > > > > > > On Sun, Mar 16, 2008 at 6:27 PM, Jon Miron <[EMAIL PROTECTED]> wrote: > > Hi Raj, > > > > Thanks for your response. > > > > I'm a little confused though. Does this look as if it's a problem > > with Broadvoice itself, and not my configuration? Any time I've > > called them with problems where it's clearly not my fault (ie nothing > > on my end has changed), they're never very helpful. > > > > > > > > On Sun, Mar 16, 2008 at 4:45 PM, Raj Jain <[EMAIL PROTECTED]> wrote: > > > Looking at the trace, the entity sending you the INVITE is not > > > resubmitting INVITE with credentials after the initial INVITE was > > > challenged with a 401 response by Asterisk. The trace shows two > > > independent calls and both have the same problem. > > > > > > -- > > > Raj Jain > > > > > > mailto:rj2807 at gmail dot com > > > sip:rjain at iptel dot org > > > > > > > > > > > > > > > On Sun, Mar 16, 2008 at 4:10 PM, Jon Miron <[EMAIL PROTECTED]> wrote: > > > > Hi all, > > > > > > > > I just upgraded to Asterisk 1.4.18 a few days ago and I don't use > > > > Broadvoice TOO often, however I have a Vermont number with them and > so > > > > my mother in law calls it to talk to my wife once in a while, so > > > > that's why it took me so long to notice it wasn't working. Anyway, > > > > when she calls she gets a busy signal (as I've tested when calling > it > > > > from my cell). > > > > > > > > When I enable debugging I get the following: > > > > > > > > SIP Debugging Enabled for IP: 147.135.0.128 > > > > net-xero*CLI> > > > > <--- SIP read from UDP://147.135.0.128:5060 ---> > > > > INVITE sip:@:5060 SIP/2.0 > > > > Call-ID: [EMAIL PROTECTED] > > > > CSeq: 1 INVITE > > > > From: "Toronto ON" #>@147.135.0.128;user=phone>;tag=prtu > > > > To: ""> > > > > Via: SIP/2.0/UDP 147.135.0.128:5060 > > > > Contact: @147.135.0.128:5060> > > > > Supported: 100rel > > > > Content-Length: 309 > > > > Content-Type: application/sdp > > > > > > > > v=0 > > > > o=2475098871 10 10 IN IP4 147.135.2.247 > > > > s=- > > > > c=IN IP4 147.135.2.250 > > > > t=0 0 > > > > m=audio 28274 RTP/AVP 0 8 18 96 97 101 > > > > a=rtpmap:0 PCMU/8000 > > > > a=rtpmap:8 PCMA/8000 > > > > a=rtpmap:18 G729/8000 > > > > a=fmtp:18 annexb=no > > > > a=rtpmap:96 iLBC/8000 > > > > a=fmtp:96 mode=30 > > > > a=rtpmap:97 t38/8000 > > > > a=rtpmap:101 telephone-event/8000 > > > > > > > > <-> > > > > --- (10 headers 14 lines) --- > > > > == Using SIP RTP CoS mark 5 > > > > Sending to 147.135.0.128 : 5060 (no NAT) > > > > Using INVITE request as basis request - [EMAIL PROTECTED] > > > > No user '' in SIP users list > > > > Found peer 'sip.broadvoice.com' for '' from > 147.135.0.128:5060 > > > > net-xero*CLI> > > > > <--- Reliably Transmitting (no NAT) to 147.135.0.128:5060 ---> > > > > SIP/2.0 401 Unauthorized > > > > Via: SIP/2.0/UDP 147.135.0.128:5060;received=147.135.0.128 > > > > From: "Toronto ON" #>@147.135.0.128;user=phone>;tag=prtu > > > > To: "">;tag=as77a74c13 > > > > Call-ID: [EMAIL PROTECTED] > > > > CSeq: 1 INVITE > > > > User-Agent: Asterisk PBX SVN-trunk-r106946 > > > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > > > > Supported: replaces, timer > > > > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", > nonce="06b61489" > > > > Content-Length: 0 > > > > > > > > > > > > <> > > > > Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in > > > > 32000 ms (Method: INVITE) > > > > net-xero*CLI> > > > > <--- SIP read from UDP://147.135.0.128:5060 ---> > > > > ACK sip:@:5060 SIP/2.0 > > > > Call-ID: [EMAIL PROTECTED] > > > > CSeq: 1 ACK > > > > From: "Toronto ON" #>@147.135.0.128;user=phone>;tag=prtu > > > > To: "">;tag=as77a74c13 > > > > Via: SIP/2.0/UDP 147.135.0.128:5060 > > > > Content-Length:0 > > > > > > > > > > > > <-> > > > > --- (7 headers 0 lines) --- > > > > [Mar 16 15:52:39] NOTICE[27524]: chan_sip.c:9020 sip_reregister: > -- > > > > Re-registration for @sip.broadvoice.com > > > > REGISTER 12 headers, 0 lines > > > > Reli
Re: [asterisk-users] Sangoma FXO/FXS config
On Tue, Mar 18, 2008 at 12:02:26PM +, Paul Goodyear wrote: > Hi all, > > I bought a Sangoma A200 card from an online supplier and explained > exactly what I wanted, > > 3 incoming phone lines to PBX and a life line (some where to connect a > standard BT phone to the PBX incase the power goes, making the BT > phone ring). > > I was told to order > > 1 x FXS module (2 FXS ports) > 2 x FXO modules (4 FXO ports) > > However being a complete noob, I have connected the 3 lines to the PBX > and have all but 1 line working (BT Featureline problems), but after a > month or playing, I realised I have 4 FXS ports and 2 FXO ports. > > --- > [EMAIL PROTECTED] ~]# ztcfg -vvv > > Zaptel Version: 1.4.9.2 > Echo Canceller: MG2 > Configuration > == > > > Channel map: > > Channel 01: FXS Kewlstart (Default) (Slaves: 01) > Channel 02: FXS Kewlstart (Default) (Slaves: 02) > Channel 03: FXS Kewlstart (Default) (Slaves: 03) > Channel 04: FXS Kewlstart (Default) (Slaves: 04) > Channel 07: FXO Kewlstart (Default) (Slaves: 07) > Channel 08: FXO Kewlstart (Default) (Slaves: 08) > > 6 channels to configure. > --- > > Does this mean they sent me 2 FXS instead of FXO's or is this the a > FXS is a FXO and FXO is a FXS thing? Calls are detected and answered > perfectly via Channels 2 and 3! FXO signalling is used for FXS channels and vice versa, so all's well. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
I would suggest upgrading to at least 1.4.18. I was able to run it for about 2 weeks and almost one million calls before I could get it to crash, and the 1.4.19RC2 seems to fix even more of the locking issues as well. I know a lot of these problems still existed under 1.4.17. MATT--- On 3/18/08, Patrick <[EMAIL PROTECTED]> wrote: > > On Tue, 2008-03-18 at 07:04 -0400, Al Baker wrote: > > Could you clarify what you mean by a "Dead Locked Channel" ? > > That is not a term I am familiar with used in context to "channels", > > databases yes, channels ??? > > > A channel got locked but never unlocked causing all sorts of funky > behavior. It's a bug. The developers have fixed a ton of these deadlocks > in 1.4 so it's usually a good plan to try the latest and greatest > version to see if the problem goes away. > > I'm not very familiar with queue setups but Doug Lytle's advice sounds > like a plan. And try 1.4.19-rc2 to see if the deadlock problem persists. > If it does then please file a bug so it can be looked at. > > Regards, > > Patrick > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On Tue, 18 Mar 2008, Steve Totaro wrote: > Why not try a different OS such as CentOS for now? That would be my next > step. I wouldn't suggest chasing distros is the way to solve issues, especially if you're happy with the hardware. Personally, I'd go back to Debian, but stick to stable (Etch) and then compile and install a custom kernel tailored exactly to your hardware, then compile and install your own asterisk from source. But only because that's what I do, and it works for me ... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma FXO/FXS config
Hi all, I bought a Sangoma A200 card from an online supplier and explained exactly what I wanted, 3 incoming phone lines to PBX and a life line (some where to connect a standard BT phone to the PBX incase the power goes, making the BT phone ring). I was told to order 1 x FXS module (2 FXS ports) 2 x FXO modules (4 FXO ports) However being a complete noob, I have connected the 3 lines to the PBX and have all but 1 line working (BT Featureline problems), but after a month or playing, I realised I have 4 FXS ports and 2 FXO ports. --- [EMAIL PROTECTED] ~]# ztcfg -vvv Zaptel Version: 1.4.9.2 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) Channel 07: FXO Kewlstart (Default) (Slaves: 07) Channel 08: FXO Kewlstart (Default) (Slaves: 08) 6 channels to configure. --- Does this mean they sent me 2 FXS instead of FXO's or is this the a FXS is a FXO and FXO is a FXS thing? Calls are detected and answered perfectly via Channels 2 and 3! Thanks, PaulG. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On Tue, 2008-03-18 at 07:04 -0400, Al Baker wrote: > Could you clarify what you mean by a "Dead Locked Channel" ? > That is not a term I am familiar with used in context to "channels", > databases yes, channels ??? A channel got locked but never unlocked causing all sorts of funky behavior. It's a bug. The developers have fixed a ton of these deadlocks in 1.4 so it's usually a good plan to try the latest and greatest version to see if the problem goes away. I'm not very familiar with queue setups but Doug Lytle's advice sounds like a plan. And try 1.4.19-rc2 to see if the deadlock problem persists. If it does then please file a bug so it can be looked at. Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On Tue, Mar 18, 2008 at 5:40 AM, Ben Willcox <[EMAIL PROTECTED]> wrote: > Hello All, > > We have been experiencing some ongoing reliability problems with > Asterisk for quite some time, and I am trying to find out if anyone else > has experienced the same problems. > > We are running asterisk 1.4.17~dfsg-2+b1 on Debian Lenny, with a Digium > PRI card, and have approximately 120 sip peers, mostly Snom 360s, with a > few Grandstream GXP2000 and a handful of Handytone 486 units. > > The symptoms, when they occur, are as follows: > > -The inability to receive incoming calls to our ISDN PRI (callers get a > busy tone), this starts off becoming intermittent but becomes permanent. > > -Asterisk cli commands work once, but then no longer return any data > until disconnecting and reconnecting to the cli, i.e. sip show peers, > show channels etc. > > -Internal SIP calls stop working > > -Calls remain stuck in queues, the queue members do not ring, and show > as Busy when issuing a 'queue show' command. > > > We've actually had these sort of problems for many months now, which > originally started when we were running Asterisk 1.2 on Gentoo. We have > done a large amount of fault finding and testing, which has involved a > replacement ISDN card, reinstall on complete different server hardware, > and changing to Asterisk 1.4 on Debian Lenny. > > I believe there may be two separate issues here - we did track down one > problem to our cacti and nagios monitoring scripts, which were > connecting and disconnecting to the manager interface several times per > minute, which eventually caused asterisk to give the above symptoms, > although in addition to the above, asterisk would consume 100% cpu on > the box, and eventually need a hard-reboot of the server. I posted about > this to the list a few weeks ago, and it was confirmed that this could > cause such a problem. After stopping these services the problems were > much reduced. > > However, we have now completely disabled the manager interface > (enabled=no in manager.conf), and yesterday the problem occurred again - > a restart of asterisk got everything going again. > So really I'm at a loss as to where to go from here. A colleague of mine > also has the same problem at his site running Asterisk 1.4 on Debian > Lenny, he has never used the manager interface, and has completely > different server hardware and ISDN card, so I wonder if it's a Debian > specific problem? > > One option is to try reverting back to Asterisk 1.2, but that isn't > really a long-term solution. We also had major problems with 1.2 with > our Snom 360 phones, as with any Snom firmware > 6.2.2 there was a > serious problem whereby on hangup the channels were not cleared down, > meaning we had many outgoing ISDN calls held open for many hours until > we realised the problem. This problem does not occur in Asterisk 1.4, > although we have many log messages such as: > > chan_sip.c: Remote host can't match request BYE to call > > so I don't know if this is anything to worry about? > > Any help would be gratefully received! > > Thanks, > Ben I have seen this when banging on the AMI but you eliminated that. Why not try a different OS such as CentOS for now? That would be my next step. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Al Baker wrote: > Could you clarify what you mean by a "Dead Locked Channel" ? > That is not a term I am familiar with used in context to "channels", > databases yes, channels ??? > Non functional, but showing up within the console and not being released. core show channels, sip show channels, etc. Channels within Asterisk link technology types. IAX,SIP,ZAP, Whatever. I may have it incorrect; if so, someone will correct me. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Queue: Simple Queue Problem
On Tue, Mar 18, 2008 at 06:20:02PM +1100, Lee, John (Sydney) wrote: > I am trying to build a simple queue for the receptionist phone. > In other words, there is only 1 agent and that is the receptionist > phone. > > However, when I call from an outside line to another extension which I > then forward to 4000, I cannot get into the queue. > exten => 98786983,1,Answer() > exten => 98786983,n,Dial(SIP/4000,20) > exten => 98786983,n,HangUp() SIP devices defined in sip.conf do not magically become extensions in extensions.conf by virtue of them being there. i.e, a dialplan (extensions.conf) entry of "4000" bears no relation to the SIP device [4000]". You just happen to have called them the same thing. Therefore, your: exten => 98786983,n,Dial(SIP/4000,20) Is routing to the SIP device 4000 rather than the queue 'console'. So you either need to go a Goto(context,4000,1) or to drop it to the queue with Queue(console) etc. R. -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Could you clarify what you mean by a "Dead Locked Channel" ? That is not a term I am familiar with used in context to "channels", databases yes, channels ??? Thx Doug Lytle wrote: > Ben Willcox wrote: > >> Hello All, >> >> One option is to try reverting back to Asterisk 1.2, but that isn't >> really a long-term solution. We also had major problems with 1.2 with >> >> > > Two things, > > 1.) On your queue setup, avoid using AgenCallbackLogin, it's known to > cause deadlocked channels. > 2.) Restart the Asterisk service once a week. I do this via a CRON job > at 3am on Sundays. > > Doug > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Ben Willcox wrote: > Hello All, > > One option is to try reverting back to Asterisk 1.2, but that isn't > really a long-term solution. We also had major problems with 1.2 with > Two things, 1.) On your queue setup, avoid using AgenCallbackLogin, it's known to cause deadlocked channels. 2.) Restart the Asterisk service once a week. I do this via a CRON job at 3am on Sundays. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48)
On 17/03/2008, Faraz Khan <[EMAIL PROTECTED]> wrote: > Good Idea and done. It is now available here: > > http://www.voip-info.org/wiki/view/LDAP The correct LDAP Schema is included: /asterisk-1.6.0-beta4/contrib/scripts/asterisk.ldap-schema and /asterisk-1.6.0-beta4/contrib/scripts/asterisk.ldif Good work though. I'm just uploading some fixes to it at: http://bugs.digium.com/view.php?id=12177 Gavin. -- http://www.suretecsystems.com/services/openldap/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Curious, you mention "a number of problems" that have "gone on for months" Question: Have you reported ANY or ALL of them to DIGIUM and if so what has been their response on each of these problems ? Ben Willcox wrote: > Hello All, > > We have been experiencing some ongoing reliability problems with > Asterisk for quite some time, and I am trying to find out if anyone else > has experienced the same problems. > > We are running asterisk 1.4.17~dfsg-2+b1 on Debian Lenny, with a Digium > PRI card, and have approximately 120 sip peers, mostly Snom 360s, with a > few Grandstream GXP2000 and a handful of Handytone 486 units. > > The symptoms, when they occur, are as follows: > > -The inability to receive incoming calls to our ISDN PRI (callers get a > busy tone), this starts off becoming intermittent but becomes permanent. > > -Asterisk cli commands work once, but then no longer return any data > until disconnecting and reconnecting to the cli, i.e. sip show peers, > show channels etc. > > -Internal SIP calls stop working > > -Calls remain stuck in queues, the queue members do not ring, and show > as Busy when issuing a 'queue show' command. > > > We've actually had these sort of problems for many months now, which > originally started when we were running Asterisk 1.2 on Gentoo. We have > done a large amount of fault finding and testing, which has involved a > replacement ISDN card, reinstall on complete different server hardware, > and changing to Asterisk 1.4 on Debian Lenny. > > I believe there may be two separate issues here - we did track down one > problem to our cacti and nagios monitoring scripts, which were > connecting and disconnecting to the manager interface several times per > minute, which eventually caused asterisk to give the above symptoms, > although in addition to the above, asterisk would consume 100% cpu on > the box, and eventually need a hard-reboot of the server. I posted about > this to the list a few weeks ago, and it was confirmed that this could > cause such a problem. After stopping these services the problems were > much reduced. > > However, we have now completely disabled the manager interface > (enabled=no in manager.conf), and yesterday the problem occurred again - > a restart of asterisk got everything going again. > So really I'm at a loss as to where to go from here. A colleague of mine > also has the same problem at his site running Asterisk 1.4 on Debian > Lenny, he has never used the manager interface, and has completely > different server hardware and ISDN card, so I wonder if it's a Debian > specific problem? > > One option is to try reverting back to Asterisk 1.2, but that isn't > really a long-term solution. We also had major problems with 1.2 with > our Snom 360 phones, as with any Snom firmware > 6.2.2 there was a > serious problem whereby on hangup the channels were not cleared down, > meaning we had many outgoing ISDN calls held open for many hours until > we realised the problem. This problem does not occur in Asterisk 1.4, > although we have many log messages such as: > > chan_sip.c: Remote host can't match request BYE to call > > so I don't know if this is anything to worry about? > > Any help would be gratefully received! > > Thanks, > Ben > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Queue: Simple Queue Problem
Lee, John (Sydney) wrote: > However, when I call from an outside line to another extension which I > then forward to 4000, I cannot get into the queue. > exten => 98786983,1,Answer() > exten => 98786983,n,Dial(SIP/4000,20) > > My guess would be that extension 4000 matches somewhere else within your dial plan and that it's hitting before your context with the queue). Seeing the console output would be of help here. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 reliability problems
Hello All, We have been experiencing some ongoing reliability problems with Asterisk for quite some time, and I am trying to find out if anyone else has experienced the same problems. We are running asterisk 1.4.17~dfsg-2+b1 on Debian Lenny, with a Digium PRI card, and have approximately 120 sip peers, mostly Snom 360s, with a few Grandstream GXP2000 and a handful of Handytone 486 units. The symptoms, when they occur, are as follows: -The inability to receive incoming calls to our ISDN PRI (callers get a busy tone), this starts off becoming intermittent but becomes permanent. -Asterisk cli commands work once, but then no longer return any data until disconnecting and reconnecting to the cli, i.e. sip show peers, show channels etc. -Internal SIP calls stop working -Calls remain stuck in queues, the queue members do not ring, and show as Busy when issuing a 'queue show' command. We've actually had these sort of problems for many months now, which originally started when we were running Asterisk 1.2 on Gentoo. We have done a large amount of fault finding and testing, which has involved a replacement ISDN card, reinstall on complete different server hardware, and changing to Asterisk 1.4 on Debian Lenny. I believe there may be two separate issues here - we did track down one problem to our cacti and nagios monitoring scripts, which were connecting and disconnecting to the manager interface several times per minute, which eventually caused asterisk to give the above symptoms, although in addition to the above, asterisk would consume 100% cpu on the box, and eventually need a hard-reboot of the server. I posted about this to the list a few weeks ago, and it was confirmed that this could cause such a problem. After stopping these services the problems were much reduced. However, we have now completely disabled the manager interface (enabled=no in manager.conf), and yesterday the problem occurred again - a restart of asterisk got everything going again. So really I'm at a loss as to where to go from here. A colleague of mine also has the same problem at his site running Asterisk 1.4 on Debian Lenny, he has never used the manager interface, and has completely different server hardware and ISDN card, so I wonder if it's a Debian specific problem? One option is to try reverting back to Asterisk 1.2, but that isn't really a long-term solution. We also had major problems with 1.2 with our Snom 360 phones, as with any Snom firmware > 6.2.2 there was a serious problem whereby on hangup the channels were not cleared down, meaning we had many outgoing ISDN calls held open for many hours until we realised the problem. This problem does not occur in Asterisk 1.4, although we have many log messages such as: chan_sip.c: Remote host can't match request BYE to call so I don't know if this is anything to worry about? Any help would be gratefully received! Thanks, Ben ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk.conf uniquename or sysname for uniqueid field in CDR
--- Vieri <[EMAIL PROTECTED]> wrote: > I set uniquename = MYHOST in asterisk.conf (under > [options]) so that my uniqueid data shows up as > MYHOST.time.seq. > > First of all, I would like to know if uniquename (or > sysname?) will still be valid across future * > versions > (mainly 1.6). > > Secondly, is there a way to specify uniquename as an > asterisk option at the command line? (asterisk -h > doesn't show me anything regarding this feature) > > Finally, how can I set uniquename to a system value > (say, dynamically set to whatever `hostname` > yields)? > Something like > uniquename = `hostname` > so that I don't have to statically set it on each > asterisk server? I just realized that uniquename is only available after applying the BRISTUFF patches. So let me rephrase my question: will Asterisk ever include the "uniquename" feature in its base code? If not, why? (I would prefer not to apply BRIstuff since I don't have Junghanns hardware). Thanks. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialstatus and cancelled calls
--- Matt Riddell <[EMAIL PROTECTED]> wrote: > http://bugs.digium.com/view.php?id=12230 Thanks Matt. However, "I may be wrong" but this isn't exactly what I'm looking for. I would like Asterisk to "transparently" set my CDR(disposition) field to reflect if a call has simply timed out (NO ANSWER) or if the caller hung up prior to ANSWER (thus CANCEL). I think that it's all in the cdr.h, cdr.c and app_dial.c files. cdr.h has: #define AST_CDR_NULL0 #define AST_CDR_FAILED (1 << 0) #define AST_CDR_BUSY(1 << 1) #define AST_CDR_NOANSWER(1 << 2) #define AST_CDR_ANSWERED(1 << 3) So I guess we would need an AST_CDR_CANCEL. cdr.c has: void ast_cdr_noanswer(struct ast_cdr *cdr) Here too I would add something like void ast_cdr_cancel(struct ast_cdr *cdr) then would add a condition to: char *ast_cdr_disp2str(int disposition) such as case AST_CDR_CANCEL: return "CANCEL"; in app_dial.c static struct ast_channel *wait_for_answer would call ast_cdr_cancel(in->cdr); whenever it subsequently calls strcpy(status, "CANCEL"); Now the problem is: can I define AST_CDR_CANCEL in cdr.h? And how? The source code I'm referring to is 1.2 but I think it's similar to 1.4/1.6. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie Queue: Simple Queue Problem
I am trying to build a simple queue for the receptionist phone. In other words, there is only 1 agent and that is the receptionist phone. I just defined a few lines in queues.conf [console] strategy = ringall member => SIP/4000 ;4000 is the console extension In extensions.conf, it is: exten => 4000,1,Answer() exten => 4000,n,Queue(console) exten => 4000,n,HangUp() I pressed DND on 4000 and then call from another SIP phone (say 4001). As expected, I saw 1 caller in the queue by "queue show" and that is great. exten => 4001,1,SetMusicOnHold() exten => 4001,n,Dial(SIP/4001,20) exten => 4001,n,VoiceMail,4001 exten => 4001,n,Playback(vm-goodbye) exten => 4001,n,Wait(2) exten => 4001,n,HangUp() However, when I call from an outside line to another extension which I then forward to 4000, I cannot get into the queue. exten => 98786983,1,Answer() exten => 98786983,n,Dial(SIP/4000,20) exten => 98786983,n,HangUp() Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
