[asterisk-users] Pickup cmd
Hi all,
I have a GXP2000 with BLF configured. I follow the configuration
guide to enable the pickup cmd as follow and include it under
corresponding content.
[BLF_group_pickup]
exten = _**1XX,1,Pickup(${EXTEN:2})
exten = _**1XX,n,Hangup
The I press the single key to pickup the call to extension 100 when
there is a call to it. From CLI, I can see it issue **100 to asterisk
but failed to pickup the call.
-- Executing [EMAIL PROTECTED]:1] Pickup(SIP/102-08373480, 100)
in new stack
[Dec 7 16:47:42] NOTICE[31079]: app_directed_pickup.c:159
pickup_exec: No target channel found for 100.
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/102-08373480, ) in new
stack
Anyone can tell me if I make something wrong for the pickup cmd?
asterisk version: 1.4.15
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Re: [asterisk-users] Any idea how making Asterisk transparent?
Maybe a Patton Smartnode or similar would do the trick : ISDN SmartNode PBX | Asterisk I would be very curious to hear opinions from others on this. Regards ___ --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 AGI script
Nicholas Blasgen wrote: I've got a very nice PHP AGI script but I want to be able to do some database cleanup when the user hangs up the phone. I wish everyone would hang up when they were suposed to, but some people don't. So what does Asterisk send to an AGI file when the line has been disconnected? If I remember reading somewhere correctly, I don't need to use DeadAGI. Instead I'm able to use normal AGI but I just need to catch a SIGTERM or something like that and process it. Nicholas, I solved that using the following extension: exten = h,1,DeadAGI(log_exit.php) If you catch any signals you're 'cheating' asterisk. Using the 'h' extension and DeadAGI should be fine. -Andreas ___ --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 AGI script
On Friday 07 December 2007 04:42:34 Andreas Brodmann wrote: Nicholas Blasgen wrote: I've got a very nice PHP AGI script but I want to be able to do some database cleanup when the user hangs up the phone. I wish everyone would hang up when they were suposed to, but some people don't. So what does Asterisk send to an AGI file when the line has been disconnected? If I remember reading somewhere correctly, I don't need to use DeadAGI. Instead I'm able to use normal AGI but I just need to catch a SIGTERM or something like that and process it. Nicholas, I solved that using the following extension: exten = h,1,DeadAGI(log_exit.php) If you catch any signals you're 'cheating' asterisk. Using the 'h' extension and DeadAGI should be fine. I don't see how that's cheating. In fact, if you need to cleanup connections, catching the signal is EXACTLY what you ought to do. Just bear in mind that as soon as you get a signal, the current version of AGI will stop interacting with your script. This has been changed for the next release cycle, so AGI will transparently switch over to what is now DeadAGI behavior (after sending a HUP) and continue to interact with the script until the script dies (which is what most people wanted anyway). -- Tilghman ___ --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 performance
by 3rd call do you mean over the internet? if the answer is yes, then I wouldn't be surprised. another thing what codec are you using? On 12/6/07, jorain [EMAIL PROTECTED] wrote: Hi all, We are using - a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size bus 2MB cache) as the asterisk server - dell 400sc(Intel P4) as a SER server - digium isdn card, TE120P at Asterisk server - Bandwidth: 2Mbps/512kbps All SIP Phones are registered to SER server, and SER will route all outgoing calls to Asterisk server. My problem is the sound quality goes down if more than 3 concurent calls to PSTN. Logically i think our system and bandwidth are more than enough to handle 3 concurent calls, but as the 4th person use it, the sound become jerky and a bit delay. So how can we improve the sound quality? Thanks Regards, jorain ___ --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] Any idea how making Asterisk transparent?
artifex, if you want call recording transparently, check out orecX.com they have a commercial and an open source SIP call recording package... no zap recording but if you are forwarding to sip exensions, you should be golden! saw them at VON 2007 boston... they have a recorded calls database lookup and web interface too... very interesting... daveC Artifex Maximus wrote: Hello! I am using Asterisk as transparent voice recorder for calls (isdn - asterisk - pbx). Voice recording (therefore voice forwarding) is working great but seems that Asterisk does not route/bridge/forward D-Channel messages which means PBX cannot get time synchronization answer from provider and tarification impulse too. With direct connection PBX works great and use both synchronization and give impulse value so there must be problem on Asterisk side. Machine is using lastest versions of Asterisk 1.2 branch (at time of writing: zaptel 1.2.22, libpri 1.2.6, asterisk 1.2.24) on Fedora Core 4. I tried with facilityenable=yes as well without success. I do not exactly know what facilityenable for. Is Asterisk capable forwarding D-Channel and making Asterisk box totally transparent? If yes which version? If branch 1.2 is capable how should I setup it right? Thanks. bye, a ___ --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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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] Cisco power injector with GXP2000 phones
Hi, Am Donnerstag, den 06.12.2007, 11:30 -0500 schrieb Jon Pounder: Quoting Ricardo Carvalho [EMAIL PROTECTED]: I only see one explanation to my problem... GXP2000 phones only implement PoE mode A of the IEEE 802.3af protocol, and the power injector does only PoE mode B of the IEEE 802.3af protocol. The switch does mode A. The problem is that I can't prove this! can't find documentation with this kind of detail. If someone does, please tell. what is the difference between the modes - is it just the pins used ? Mode A means inline power. The 48 V is transferred over the wire-pairs 1/2 and 3/6 which are also used for data transfer. Before the power is switched on, endpoint and power supply must negotiate this process. Mode B means midspan power. The 48 V is transferred over the wire-pairs 4/5 and 7/8, which are unused and 10/100 networks. Mode B is not possible on GigE (afaik). Regards, Karsten ___ --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] Sidetone with Snom 370
Hi all, I'm not getting any sidetone on my Snom 370. I searched the web and the snom wiki, but I don't see any place to enable/adjust it. Callers say I sound great on the other end, but I don't hear myself so it is a little off-putting. Any suggestions would be appreciated. On a related note, some times (maybe 1 out of 10 calls) I get the side tone, but its delayed by a second or so, so its like an echo. Is that something to do with asterisk? I'm using SIP trunking on a fast connection, and only 1 call at a time right now. Regards, Zaheer ___ --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] Any idea how making Asterisk transparent?
On Friday 07 December 2007 04:39:37 Artifex Maximus wrote: On Dec 7, 2007 11:07 AM, Philipp Kempgen [EMAIL PROTECTED] wrote: Artifex Maximus wrote: I am using Asterisk as transparent voice recorder for calls (isdn - asterisk - pbx). Voice recording (therefore voice forwarding) is working great but seems that Asterisk does not route/bridge/forward D-Channel messages which means PBX cannot get time synchronization answer from provider and tarification impulse too. With direct connection PBX works great and use both synchronization and give impulse value so there must be problem on Asterisk side. Is Asterisk capable forwarding D-Channel and making Asterisk box totally transparent? No. We need recording calls with using nice functions like time synchro and tarification. The real issue is that Asterisk needs to be able to understand all of those messages and transmit them across a channel bridge in the form of a control frame. This isn't as big of a deal with the two messages that you've suggested as it is with vendor proprietary extensions. If we don't know what the messages say, we can't correctly interpret them in a protocol agnostic way. But those two messages could certainly be done; they just haven't been (yet). -- Tilghman ___ --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] Probems receiving 200ok message
No. My asterisk server had two NIC, one for public internet and another to LAN for phones. The problem is when I receive SIP 200 from public internet. Thanks. Fred Em Qui, 2007-12-06 às 21:53 -0500, C F escreveu: is this machine or the phone behind nat? On 12/6/07, Frederico Madeira [EMAIL PROTECTED] wrote: Hi guys, Using tcpdump I could see the messages sip 200 arriving on my server, but enabling sip debug on asterisk console I only saw Invite and 180 message. What can be the source of this problem ? Thanks. Fred ___ --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] Any idea how making Asterisk transparent?
On Dec 7, 2007 11:07 AM, Philipp Kempgen [EMAIL PROTECTED] wrote: Artifex Maximus wrote: Is Asterisk capable forwarding D-Channel and making Asterisk box totally transparent? No. Thanks Philipp. Bad news. We need recording calls with using nice functions like time synchro and tarification. Where should I look for solution? Is there any software only solution? On hardware side might EyeSDN device (or similar) do the trick? bye, a ___ --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 AGI script
Nicholas Blasgen wrote: I've got a very nice PHP AGI script but I want to be able to do some database cleanup when the user hangs up the phone. I wish everyone would hang up when they were suposed to, but some people don't. So what does Asterisk send to an AGI file when the line has been disconnected? SIGHUP If I remember reading somewhere correctly, I don't need to use DeadAGI. Instead I'm able to use normal AGI but I just need to catch a SIGTERM or something like that and process it. You need to catch it or the script will terminate. Not sure if this is possible at all in PHP. I think there's an inofficial POSIX module which can do it. Perl can do it for sure. Regards, Philipp ___ --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] Open Asterisk Exchange Project
Is there anyone interested in developing an open source Asterisk / MS Exchange solution? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Attachment encrypted? click here http://www.highpoweredhelp.com/tutorials/wincrypt/ . ___ --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 performance
Hi all, We are using - a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size bus 2MB cache) as the asterisk server - dell 400sc(Intel P4) as a SER server - digium isdn card, TE120P at Asterisk server - Bandwidth: 2Mbps/512kbps All SIP Phones are registered to SER server, and SER will route all outgoing calls to Asterisk server. My problem is the sound quality goes down if more than 3 concurent calls to PSTN. Logically i think our system and bandwidth are more than enough to handle 3 concurent calls, but as the 4th person use it, the sound become jerky and a bit delay. So how can we improve the sound quality? Thanks Regards, jorain ___ --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] Perspective on Asterisk
I’ve put out a project to bid that involves telephony and SMS. These technologies are not in my comfort zone so I’m trying to understand the landscape a little better. Several of the bidders plan to use Asterisk for the backend. From what I understand so far, that would apply to the telephony piece. I pasted the requirements at the end of this email. I’d appreciate any insight as to where Asterisk is applicable. Also, I’d appreciate any thoughts on implementing on Amazon EC2. While not the purpose of this post, if anyone is interested in bidding, please provide contact info. Thanks, Steve The purpose is to give users the ability to set off a series of communications as follows: 1. Actions that can set-off a notification 1.1.Web form 1.2.Incoming SMS message 1.3.Incoming telephone call with keyed-in and/or IVR data 1.4.Incoming email 2. Content of the outgoing notification 2.1.Name of account owner (from database) 2.2.Call back number (from incoming message) 2.3.Voice message or digital content of the communication 2.4.Time and date 3. Means of delivery of notification – one or more of: 3.1.Email 3.1.1. If source is a voice message, convert the message to a wav file. 3.2.Telephone automated message 3.3.SMS 3.3.1. If the source is a voice message, provide a means to listen to the message (method to be determined) 4. Message routing 4.1.Incoming message will contain a user id. 4.2.Each ID can have one or more contact records associated with it. 4.2.1. If more than one, the database will have contact priority options and messages will be routed to contact records as per the priorities selected. 4.2.2. Next priority will be activated with each subsequent request within a limited period (ie within two hours) 4.2.3. Next priority will be activated if the system is unable to successfully send a message. 4.2.4. One priority option will be to contact all records simultaneously. 4.3.Each contact record will have one or more contact destinations. 4.3.1. The destinations can be email, SMS or phone 4.3.1.1.Whenever a contact record is chosen, all contact destinations for that record will be used simultaneously. 5. Log 5.1.Every action will create a log entry. 5.2.Incoming requests will create a record. 5.3.Each destination (ie phone number, SMS) attempt will create a record. 5.4.Failures will create a record. 5.5.Portions of the log details will be sent to the message source after specified periods (ie 5 min, 1 hr, 2 hrs.) GET FREE 5GB EMAIL - Check out spam free email with many cool features! Visit http://www.inbox.com/email to find out more! ___ --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] 7960 Won't Register Yet Multiple Attempts?
Port 5060 should be udp as well. Sent from my iPhone On Dec 7, 2007, at 12:14 AM, [EMAIL PROTECTED] wrote: Hi List, I've got a 7960 that's behind NAT (nat_enabled: 1 and nat_received_processing: 1) and for whatever reason doesn't seem to register, or at least hold a registration. If both the phone and the router (netgear) are rebooted, the phone will register, take a few incoming/outgoing calls no problems, then a few hours later, it drops the registration and never re-registers. If the phone itself is rebooted, I see a mess of registration attempts via SIP channels: 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER Is there something that I'm missing. Short of replacing the customers router (which I have admin access to) is there anything else I should try? Any sort of packet filtering is disabling, nat is enabled in the SIP config, and port forwarding was also setup to forward 5060-5070 TCP and 1+ UDP to the phone to no avail. Note that if the phone is plugged directly into the customer's modem (thus removing the router out of the picture) the phone works perfectly. Thanks - Any input is appreciated -Robert Norton ___ --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 AGI script
On Thu, 6 Dec 2007, Nicholas Blasgen wrote: I've got a very nice PHP AGI script but I want to be able to do some database cleanup when the user hangs up the phone. I wish everyone would hang up when they were suposed to, but some people don't. So what does Asterisk send to an AGI file when the line has been disconnected? If I remember reading somewhere correctly, I don't need to use DeadAGI. Instead I'm able to use normal AGI but I just need to catch a SIGTERM or something like that and process it. When the user hangs up, your AGI will receive a HUP. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] dtmf detection not working on sip trunks using asterisk-1.4.15
Hi all, I am using an asterisk-1.4.13 connected to our carrier via SIP trunk. I use rfc2833 as dtmf detection method. After upgrading to asterisk-1.4.15 our system would not detect dtmf from a caller from PSTN anymore. When investigating the SIP traffic at call initiation I realized that in the SDP message asterisk is no longer offering the telephone-event/8000 capability. So the carrier does not send the rfc2833 messages anymore. Does anyone know about this or has seen an open bug case for it (I haven't found any myself)? Thanks for help and feedback. Kind Regards, Andreas Brodmann ___ --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] AMQP Support for Asterisk?
Are there any plans to implement AMQP directly in Asterisk or is it best to use a third party bridge like Mule? https://jira.amqp.org/confluence/display/AMQP/Advanced+Message+Queuing+Protocol -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --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] Happy Birthday Asterisk
Philip Prindeville wrote: [...] There were earlier experimental versions of IP, but v4 got it right. and v6 will get it even more right. ;) -- Jason Parker Digium ___ --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] SIP-Realtime and sip reload
Am Donnerstag, den 06.12.2007, 21:06 +0100 schrieb Torbjörn Abrahamsson: Our current approach is to use the #exec directive, and call a script which creates static friends by reading information from the DB. We still use the remote ITSP peers with realtime, as they do not need the OPTIONS. This way when we call a reload the users registration is still there, and we have the flexibility of using a DB as the user database. Could you explain that a litte bit to me? I just tried to find something about #exec, but not very successfully. Is there any documentation? Do you reload asterisk and generate the sip.conf by reading the users from a database with a script? And omit the usage of realtime for these users? Could you perhaps post/send your configuration/script? thanks Henrik ___ --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] Happy Birthday Asterisk
Philip Prindeville wrote: So I'd venture to say that by August, the Internet will really be *30* years old. As Al Gore was born in 1948, I can see that the Internet could be as old as 30, but not much more. 35 years ago would put him at 25 years old. And inventing the whole Internet at 25 is pretty ambicious, even for Al! :) ___ --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] Strange ISDN-problem with incoming calls out of the same city
Hi, Does this number (you are dialing) has been ported from a different Telco? When you dial from the other city and you get service not available you may be dialing from a different Telco that either has no route aggreement for the dialed network, or the number portability database (of Out of city Operator) is not up to date. before we switched from the old pbx to the asterisk server, these people had no problems calling our client. With some more debugging we saw what happens with these specific calls. For some reason local calls and calls from a few other cities cause trouble, because asterisk doesn't get the whole number that has been dialed. If e.g. someone from the same town dials 123456, asterisk only gets 12345 or 1234. This extension doesn't exist in the dialplan and so the call fails. And this is not a single failure, it happens every time. The telco has checked the lines and they are okay, so it might be the ISDN card (EICON) or the driver. I have made a trace log from one of these failed calls and will forward it to EICON. Meanwhile we catch all these calls with the i extension. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --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] Limit participants in Meetme...
On Friday 07 December 2007 12:04:04 Carlos Chavez wrote:
Is there an easy way to limit the number of participants on a Meetme
room? Lets say we only want 10 people to be able to enter a particular
meetme conference, how can I prevent number 11 from entering this
conference? We will not have a pin to enter.
Use group counting:
Set(GROUP()=foo)
GotoIf($[${GROUP_COUNT(foo)} 10]?hangup)
Meetme(1234)
--
Tilghman
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[asterisk-users] Limit participants in Meetme...
Is there an easy way to limit the number of participants on a Meetme room? Lets say we only want 10 people to be able to enter a particular meetme conference, how can I prevent number 11 from entering this conference? We will not have a pin to enter. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ --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-users] Show calls in progress
In article [EMAIL PROTECTED], Steve Johnson [EMAIL PROTECTED] wrote: Is there an Asterisk CLI command to show a list of calls in progress (for all channels: Zap/SIP/IAX2 etc). Restart when convenient waits until the system is idle, but is there an obvious way of seeing what's going on at the moment? show channels Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] CDR Function in Hangup Channel
On Thu, 2007-12-06 at 12:54 -0800, Douglas Garstang wrote:
Ok, this is a little crazy...
billsec and duration are 0, but disposition is ANSWERED.
Huh?
h = {
NoOp(*** LEG B HANGUP ${CDR(duration)} ${CDR(billsec)}
${CDR(disposition)});
AddCallLeg(${LEGB_SOURCE},${LEGB_DEST},1,2,${HANGUPCAUSE});
};
Douglas--
Check out cdr.conf--
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may
be
; retrieved inside of of this extension.
;endbeforehexten=no
Try setting endbeforehexten=yes and see if you get what you need
murf
- Original Message
From: Douglas Garstang [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, December 6, 2007 12:04:29 PM
Subject: CDR Function in Hangup Channel
So... I'm trying to access CDR(duration) and CDR(billsec) inside h...
I keep getting 0. Can I access the CDR function inside a hangup
extensions?
Asterisk 1.4.13
Thanks, Doug.
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[asterisk-users] Using XML for configuration management, single-source-of-truth, etc.
I'm starting work on some provisioning tools to simplify plugging in and configuring hard SIP handsets and conference bridges (maybe eventually MPEG-4 PoE video cameras that speak SIP as well). Issue is that I'd like to glean as much information out of the configuration files... but don't want to write a whole new parser to do it (especially not one that understands templates and macros). For instance, from the voicemail.conf, extensions.conf, and sip.conf files, I should be able to generate 90% of the configuration state needed for provisioning an out-of-the-box Sipura SPA941... if only those files were in some more parsable format, like XML. How much effort would it be to add an application that traverses the configuration state and writes it out as an XML flat file? Or perhaps at some point in the future, Asterisk's configuration files could be represented as XML natively (did someone in the back row just show gconf???). I'm a relative newbie, so if I'm missing something obvious or there's been a religious war on the subject in the past, apologies... -Philip ___ --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] Print CALLERID in CLI during pri debug
What don't you tell us what you are ultimately trying to do. You want the callerid next to the connect message in debug output... why? What will that help you to accomplish? On Dec 7, 2007 4:42 PM, Arpit Mehta [EMAIL PROTECTED] wrote: Ok so the call reference is the 'cr' field (q931.c) and how do I retrieve the caller id from this call reference ? On Dec 7, 2007 4:29 AM, Richard Revels [EMAIL PROTECTED] wrote: When the call sets up the 'call reference' is assigned. It will be unique for the duration of the call and other messages, like Connect, will reference it. At the same time, the setup will have indicated the caller ID info. Sent from my iPhone On Dec 6, 2007, at 10:28 PM, Arpit Mehta [EMAIL PROTECTED] wrote: Or in other words is there a way to map which message is from which CallerID ? On Dec 6, 2007 6:40 PM, Arpit Mehta [EMAIL PROTECTED] wrote: Hi all, I was wondering if it is possible to print the callerid value in the CLI when doing 'pri debug span 1' For example Call Ref: len= 2 (reference 2707/0xA93) (Terminator) Message type: CONNECT (7) [18 03 a9 83 97] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 23 ] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) I would like to print '1234567890 Message type: CONNECT (7) ... ... ' where 1234567890 is the callerid Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --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 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --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] Problem with the ring timeout in dial command for local extensions
Hi all, I don't know if this is the right list to ask, since I'm using Trixbox version 1.0.0.28, that has asterisk 1.2.17. I'm trying to configure the ring timeout value for my local extensions (when dialing from one to another), and the dial command simply ignores my values... I have one extension 0017 in my box, so I used the command Dial(SIP/0017|100|rTtWw) to dial to it. The call gets completed without a problem, but it only rings for 30 seconds, when it should ring for a 100 seconds. I'm pretty sure this is my mistake here, but I didn't find a solution. I also tried changing the value directly in trixbox web interface that says Number of seconds to ring phones before sending callers to voicemail and nothing happens. I know that trixbox does weird things to my configuration files, but I edited extenions.conf, since it does not get messed up by trixbox. If I use the dial command to dial out with my termination provider (runs on IAX2) the timeout option works just fine. All help is very welcome, Thiago Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/ ___ --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] Function vs. Application?
On Fri, 2007-12-07 at 21:04 +0100, Vincent wrote: Out of curiosity, what's the difference between a function and an application? In a nutshell, an application is something that performs an action on a channel (such as playing a sound prompt, gathering DTMF input, putting the call into a call queue, etc.). A function, on the other hand, is used to get or set values, and doesn't directly manipulate the channel. These values *might* have something to do with the channel (such as is the case with the CDR function), but don't necessarily have to (such as is the case with the CUT and LEN functions). Hopefully I've explained it in such a way that it's clearer to you know. If not, let me know and I'll try to be more clear. --- Jared Smith Community Relations Manager Digium, Inc. ___ --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] Happy Birthday Asterisk
On Friday 07 December 2007 09:56:56 Bill Andersen wrote: Philip Prindeville wrote: So I'd venture to say that by August, the Internet will really be *30* years old. As Al Gore was born in 1948, I can see that the Internet could be as old as 30, but not much more. 35 years ago would put him at 25 years old. And inventing the whole Internet at 25 is pretty ambicious, even for Al! In actuality, most people produce all of the great inventions of their life by the time they hit 30. Einstein, for one, produced his great theory of relativity at the ripe old age of 26. Mark Spencer came up with Asterisk at age 22. So this idea that 25 is too young to produce a great achievement is baloney. BTW, Al Gore was credited with introducing the legislation that permitted commercial organizations onto the network that would become known as the Internet. So in a way, he did create the Internet, by changing the circumstances you would have to have in order to access this decentralized computer network. If you doubt the importance of having commercial organizations on the network, consider where the Internet would be, if Amazon, eBay, and Linux Support Services (d/b/a Digium) had never been allowed onto the network. -- Tilghman ___ --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] astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR
Josué, MFC/R2 signaling use pair of frequencies, not letters or numbers. For older packages of spandsp and libmfcr2 the letter E represent the last of this pair of frequencies. Your telco was asking for F, because for the telco F is the last signal of the 15 signals used for MFC/R2. In newer packages of libmfcr2 and spandsp F is also the last frequency. The fact that you see an E for old libraries does not means the signal is incorrect, just the letter used to represent that signal does not match with what the telco says, but in the end, both F of the telco and E of the old spandsp represent the same signal. So, you should not have problems. Can you send me a trace of your working installation? Thanks. On Dec 7, 2007 10:20 AM, Josué Conti [EMAIL PROTECTED] wrote: Hi Steve and Hi Moises, how are you? Greetings! :) would like to thank the you for always helping and to all this community. Steve already helped me some times in 2005, heheheh! I remade all the installation and now I used the following packages: asterisk-1.2.21.1, libpri-1.2.5, zaptel-1.2.19, libsupertone-0.0.2, libunicall-0.0.3, spandsp-0.0.2(spandsp-0.0.2pre26), libmfcr2-0.0.3 and wanpipe-3.2.1. The compilation occurred normally and is functioning normally same with the signalling E that it continues appearing in the messages of exchange of signalling MFC with loglevel=255. This will be able to cause some problem? Best Regards. Josué 2007/12/6, Moises Silva [EMAIL PROTECTED]: Josue, This version of spandsp should work. http://www.soft-switch.org/downloads/spandsp/old/spandsp-0.0.2pre26.tar.gz The odd thing is that astunicall for 1.2 was packaged by a guy who had running these versions in México, probably he made a mistake and I never tested myself that the libraries worked well together blindly trusting the contributor of 1.2 working versions. I will change spandsp from the package. Regards, On Dec 6, 2007 6:17 PM, Steve Underwood [EMAIL PROTECTED] wrote: Hi Josué, Those E/F mismatch issues are due to using incompatible versions of spandsp and unicall. MFC/R2 defines 15 tone signals. These are called 1 to 15 in the R2 documentation. I wanted a single character code for these, so I used 0-9 for the digits, and A-E for the other 5 codes. This confused people, who complained they say I should have signal 15 here, and the log is saying E'. I changed the internal codes for the signals to be 0-9 and B-F, so the B-F codes match their hexadecimal equivalent. Now people seem to find the logs clearer. However, this change occured in spandsp AND unicall. If you don't use versions of these two things which match, you get the result you are seeing. Regards, Steve Josué Conti wrote: Hi All, as good? I am trying to make a call for the Unicall channels and after the exchange of signalling and sending of the digits asterisk locks up with the sending of the signalling E and the TELCO says that asterisk would have to send signalling F, as to make for asterisk to send signalling F? The TELCO says that the signalling E is suppresor insertion of ECHO in the destination. F is end of the digits. They could help me? Best Regards Josué -- Executing Dial(SIP/1196082068-082a6b78, Unicall/g1/01197831234|90|tT) in new stack Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(1) Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Make call Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32769 Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0001 - [1/ 1/Idle /Idle ] -- Called g1/01197831234 Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Dialing Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - [1/ 40/Seize /Idle ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 on [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 off [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 on [2/ 40/Group I /DNIS ]
Re: [asterisk-users] astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR
Hi Steve and Hi Moises, how are you? Greetings! :) would like to thank the you for always helping and to all this community. Steve already helped me some times in 2005, heheheh! I remade all the installation and now I used the following packages: asterisk-1.2.21.1, libpri-1.2.5, zaptel-1.2.19, libsupertone-0.0.2, libunicall-0.0.3, spandsp-0.0.2(spandsp-0.0.2pre26), libmfcr2-0.0.3 and wanpipe-3.2.1. The compilation occurred normally and is functioning normally same with the signalling E that it continues appearing in the messages of exchange of signalling MFC with loglevel=255. This will be able to cause some problem? Best Regards. Josué 2007/12/6, Moises Silva [EMAIL PROTECTED]: Josue, This version of spandsp should work. http://www.soft-switch.org/downloads/spandsp/old/spandsp-0.0.2pre26.tar.gz The odd thing is that astunicall for 1.2 was packaged by a guy who had running these versions in México, probably he made a mistake and I never tested myself that the libraries worked well together blindly trusting the contributor of 1.2 working versions. I will change spandsp from the package. Regards, On Dec 6, 2007 6:17 PM, Steve Underwood [EMAIL PROTECTED] wrote: Hi Josué, Those E/F mismatch issues are due to using incompatible versions of spandsp and unicall. MFC/R2 defines 15 tone signals. These are called 1 to 15 in the R2 documentation. I wanted a single character code for these, so I used 0-9 for the digits, and A-E for the other 5 codes. This confused people, who complained they say I should have signal 15 here, and the log is saying E'. I changed the internal codes for the signals to be 0-9 and B-F, so the B-F codes match their hexadecimal equivalent. Now people seem to find the logs clearer. However, this change occured in spandsp AND unicall. If you don't use versions of these two things which match, you get the result you are seeing. Regards, Steve Josué Conti wrote: Hi All, as good? I am trying to make a call for the Unicall channels and after the exchange of signalling and sending of the digits asterisk locks up with the sending of the signalling E and the TELCO says that asterisk would have to send signalling F, as to make for asterisk to send signalling F? The TELCO says that the signalling E is suppresor insertion of ECHO in the destination. F is end of the digits. They could help me? Best Regards Josué -- Executing Dial(SIP/1196082068-082a6b78, Unicall/g1/01197831234|90|tT) in new stack Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(1) Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Make call Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32769 Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0001 - [1/ 1/Idle /Idle ] -- Called g1/01197831234 Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Dialing Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - [1/ 40/Seize /Idle ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 on [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 off [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 on [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 off - [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 off [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 on [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 off - [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 off [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]:
Re: [asterisk-users] SIP-Realtime and sip reload
Torbjörn Abrahamsson [EMAIL PROTECTED] writes: Our current approach is to use the #exec directive, and call a script which creates static friends by reading information from the DB. Brilliant idea! That'll definitely be the replacement for our current realtime system. Thanks! /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
[asterisk-users] Any idea how making Asterisk transparent?
Hello! I am using Asterisk as transparent voice recorder for calls (isdn - asterisk - pbx). Voice recording (therefore voice forwarding) is working great but seems that Asterisk does not route/bridge/forward D-Channel messages which means PBX cannot get time synchronization answer from provider and tarification impulse too. With direct connection PBX works great and use both synchronization and give impulse value so there must be problem on Asterisk side. Machine is using lastest versions of Asterisk 1.2 branch (at time of writing: zaptel 1.2.22, libpri 1.2.6, asterisk 1.2.24) on Fedora Core 4. I tried with facilityenable=yes as well without success. I do not exactly know what facilityenable for. Is Asterisk capable forwarding D-Channel and making Asterisk box totally transparent? If yes which version? If branch 1.2 is capable how should I setup it right? Thanks. bye, a ___ --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-users] Show calls in progress
Is there an Asterisk CLI command to show a list of calls in progress (for all channels: Zap/SIP/IAX2 etc). Restart when convenient waits until the system is idle, but is there an obvious way of seeing what's going on at the moment? Thanks, Steve ___ --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] Using XML for configuration management, single-source-of-truth, etc.
On Friday 07 December 2007 20:12:12 Philip Prindeville wrote: Darryl Dunkin wrote: You can store most of the configurations in a database which may be more accessable to you. Perl can also parse these configurations quickly enough if you know how to use the input record seperator ($/) properly. The only thing Asterisk will not store which you would probably need is the actual MAC address of the phones themselves. This may be done easily enough as comments in the users sip.conf section. That's sort of my point: that you have to reinvent it, and it's easy to get wrong. XML wouldn't make it any less wrong. There's a difference between parsing it syntactically (which XML fixes) and parsing it semantically (which XML does not). In fact, I find the configuration files, as they are now are much EASIER to parse than XML. With XML, you need to load up a whole state engine to ensure the config is properly formatted. At the simplest level, the config file as-is is simply a set of key/value pairs, which syntactically is very easy to parse. Part of the allure of the current format is also that it is human readable, which assists in manual editing. I'm not sure what part of the universe you have be from to make XML human readable (or more importantly, human-editable), but I am quite sure it is not from this planet. -- Tilghman ___ --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 performance
Your 512k outbound bandwidth will tend to be the defining factor in call quality here. Does your connection only gets used for voip? Or is it shared with other uses? Can you use more compressed codecs? G729 will quadruple you call capacity. What sort of QoS and traffic shaping do you use? Note that these are separate matters, and you need both. Michael --Original Message Text--- From: jorain Date: Thu, 6 Dec 2007 17:47:18 +0800 Hi all, We are using - a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size bus 2MB cache) as the asterisk server - dell 400sc(Intel P4) as a SER server - digium isdn card, TE120P at Asterisk server - Bandwidth: 2Mbps/512kbps All SIP Phones are registered to SER server, and SER will route all outgoing calls to Asterisk server. My problem is the sound quality goes down if more than 3 concurent calls to PSTN. Logically i think our system and bandwidth are more than enough to handle 3 concurent calls, but as the 4th person use it, the sound become jerky and a bit delay. So how can we improve the sound quality? Thanks Regards, jorain -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --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] Using XML for configuration management, single-source-of-truth, etc.
That's sort of my point: that you have to reinvent it, and it's easy to get wrong. Darryl Dunkin wrote: You can store most of the configurations in a database which may be more accessable to you. Perl can also parse these configurations quickly enough if you know how to use the input record seperator ($/) properly. The only thing Asterisk will not store which you would probably need is the actual MAC address of the phones themselves. This may be done easily enough as comments in the users sip.conf section. ___ --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] Polycom 601 stops ringing
On 12/7/2007 at 2:33 PM, Doug [EMAIL PROTECTED] wrote: At 10:58 12/7/2007, Joe Acquisto wrote: I have an odd issue, where a polycom 601 stops ringing, or more properly, maybe, stops *being* rung, when a call comes in. Other phones/extensions, continue to work fine, they being run at the same time. My dial plan works fine (?) seems it will ring properly, right after a reboot. It works fine for outgoing calls at all times. Hints? Is it behind a firewall? joe a. My entire network is behind a firewall, but there is only a switch between asterisk and the phones. joe a. ___ --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] Playback file and detect a key press
I would like to do the following: Play back a file, and during the playback be able to detect a DTMF tone that may be pressed. I do not want to interrupt the playing of the file, but when the file finishes I would like to be able to tell if a key was pressed and which key it was. Anyway to do this? In AGI: o Wait for Digit waits for a digit to be pressed, and I don't see how to play a file at the same time. o Stream File can detect a digit, but then the file playback is interrupted. In a call plan: o Playback plays a file but does not detect pressed digits. o Background plays a file, but stops the playback when a key is pressed. Is there anyway to do what I want to do? Thanks! -- Bob Smither [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] Using XML for configuration management, single-source-of-truth, etc.
Tilghman Lesher wrote: On Friday 07 December 2007 20:12:12 Philip Prindeville wrote: Darryl Dunkin wrote: You can store most of the configurations in a database which may be more accessable to you. Perl can also parse these configurations quickly enough if you know how to use the input record seperator ($/) properly. The only thing Asterisk will not store which you would probably need is the actual MAC address of the phones themselves. This may be done easily enough as comments in the users sip.conf section. That's sort of my point: that you have to reinvent it, and it's easy to get wrong. XML wouldn't make it any less wrong. There's a difference between parsing it syntactically (which XML fixes) and parsing it semantically (which XML does not). In fact, I find the configuration files, as they are now are much EASIER to parse than XML. With XML, you need to load up a whole state engine to ensure the config is properly formatted. At the simplest level, the config file as-is is simply a set of key/value pairs, which syntactically is very easy to parse. Part of the allure of the current format is also that it is human readable, which assists in manual editing. I'm not sure what part of the universe you have be from to make XML human readable (or more importantly, human-editable), but I am quite sure it is not from this planet. Well, after hand-coding HTML and SGML for 15+ years, XML isn't all that much of a stretch. More to the point though, there are some excellent schema-driven configuration managers for XML, so you wouldn't have to edit the files by hand. -Philip ___ --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] Function vs. Application?
On Friday 07 December 2007 14:19:49 Jared Smith wrote: On Fri, 2007-12-07 at 21:04 +0100, Vincent wrote: Out of curiosity, what's the difference between a function and an application? In a nutshell, an application is something that performs an action on a channel (such as playing a sound prompt, gathering DTMF input, putting the call into a call queue, etc.). A function, on the other hand, is used to get or set values, and doesn't directly manipulate the channel. These values *might* have something to do with the channel (such as is the case with the CDR function), but don't necessarily have to (such as is the case with the CUT and LEN functions). Hopefully I've explained it in such a way that it's clearer to you know. If not, let me know and I'll try to be more clear. You could also think of it as the difference between a procedure and a function. A procedure does something and returns nothing. A function may or may not be doing something, but its primary function is to return a value. Unlike other languages, in Asterisk, the return value of a function may not be directly ignored (i.e. you HAVE to get it, even if you do nothing with it). Of course, setting a dialplan function completely ruins this nice dichotomy. ;-) -- Tilghman ___ --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] Strange ISDN-problem with incoming calls out of the same city
On Fri, 7 Dec 2007, Stefan Guenther wrote: Hi, Does this number (you are dialing) has been ported from a different Telco? When you dial from the other city and you get service not available you may be dialing from a different Telco that either has no route aggreement for the dialed network, or the number portability database (of Out of city Operator) is not up to date. before we switched from the old pbx to the asterisk server, these people had no problems calling our client. With some more debugging we saw what happens with these specific calls. For some reason local calls and calls from a few other cities cause trouble, because asterisk doesn't get the whole number that has been dialed. If e.g. someone from the same town dials 123456, asterisk only gets 12345 or 1234. This extension doesn't exist in the dialplan and so the call fails. And this is not a single failure, it happens every time. The telco has checked the lines and they are okay, so it might be the ISDN card (EICON) or the driver. I have made a trace log from one of these failed calls and will forward it to EICON. Meanwhile we catch all these calls with the i extension. How does your dialplan look like? If you have e.g. exten = _.,1, in the context for capi incoming calls, then asterisk (chan-capi) accept these calls even if not all numbers are dialed (transmitted) yet. Anyway, you talk about external calls, but you set ntmode=yes which does not make sense. Also, you should set isdnmode= to whatever isdn mode you have on your line. Please have a look at the example capi.conf of chan-capi package. Some of your general settings are possible in the interface sections only. Armin ___ --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] Happy Birthday Asterisk
Bill Andersen wrote: Philip Prindeville wrote: So I'd venture to say that by August, the Internet will really be *30* years old. As Al Gore was born in 1948, I can see that the Internet could be as old as 30, but not much more. 35 years ago would put him at 25 years old. And inventing the whole Internet at 25 is pretty ambicious, even for Al! :) I wrote my first RFC at 22, and I was never Vice President... ;-) -Philip ___ --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] Any idea how making Asterisk transparent?
Artifex Maximus wrote: I am using Asterisk as transparent voice recorder for calls (isdn - asterisk - pbx). Voice recording (therefore voice forwarding) is working great but seems that Asterisk does not route/bridge/forward D-Channel messages which means PBX cannot get time synchronization answer from provider and tarification impulse too. With direct connection PBX works great and use both synchronization and give impulse value so there must be problem on Asterisk side. Machine is using lastest versions of Asterisk 1.2 branch (at time of writing: zaptel 1.2.22, libpri 1.2.6, asterisk 1.2.24) on Fedora Core 4. I tried with facilityenable=yes as well without success. I do not exactly know what facilityenable for. Is Asterisk capable forwarding D-Channel and making Asterisk box totally transparent? No. Regards, Philipp ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 601 stops ringing
I have an odd issue, where a polycom 601 stops ringing, or more properly, maybe, stops *being* rung, when a call comes in. Other phones/extensions, continue to work fine, they being run at the same time. My dial plan works fine (?) seems it will ring properly, right after a reboot. It works fine for outgoing calls at all times. Hints? joe a. ___ --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 performance
2007/12/7, C F [EMAIL PROTECTED]: by 3rd call do you mean over the internet? if the answer is yes, then I wouldn't be surprised. Oh my god! If it is over internet and you get crap quality.. you have to be surprised.. It is depends by Latency (Traffic congestion, Network congestion) and Packet loss - jorain, What do you mean for quality problem ? Different quality problems are generated by different parameter braking ? echo? low volume ? Cheers 2007/12/7, C F [EMAIL PROTECTED]: by 3rd call do you mean over the internet? if the answer is yes, then I wouldn't be surprised. another thing what codec are you using? On 12/6/07, jorain [EMAIL PROTECTED] wrote: Hi all, We are using - a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size bus 2MB cache) as the asterisk server - dell 400sc(Intel P4) as a SER server - digium isdn card, TE120P at Asterisk server - Bandwidth: 2Mbps/512kbps All SIP Phones are registered to SER server, and SER will route all outgoing calls to Asterisk server. My problem is the sound quality goes down if more than 3 concurent calls to PSTN. Logically i think our system and bandwidth are more than enough to handle 3 concurent calls, but as the 4th person use it, the sound become jerky and a bit delay. So how can we improve the sound quality? Thanks Regards, jorain ___ --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 -- Giovanni Miano ___ --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] Print CALLERID in CLI during pri debug
Well my project is an experimental project at my university. I need to collect experiment results which could tag every isdn message to the callerid, so it is clear which message belongs to which callerid (as multiple calls could be going on at one time). Thanks Arpit On Dec 7, 2007 5:34 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: What don't you tell us what you are ultimately trying to do. You want the callerid next to the connect message in debug output... why? What will that help you to accomplish? On Dec 7, 2007 4:42 PM, Arpit Mehta [EMAIL PROTECTED] wrote: Ok so the call reference is the 'cr' field (q931.c) and how do I retrieve the caller id from this call reference ? On Dec 7, 2007 4:29 AM, Richard Revels [EMAIL PROTECTED] wrote: When the call sets up the 'call reference' is assigned. It will be unique for the duration of the call and other messages, like Connect, will reference it. At the same time, the setup will have indicated the caller ID info. Sent from my iPhone On Dec 6, 2007, at 10:28 PM, Arpit Mehta [EMAIL PROTECTED] wrote: Or in other words is there a way to map which message is from which CallerID ? On Dec 6, 2007 6:40 PM, Arpit Mehta [EMAIL PROTECTED] wrote: Hi all, I was wondering if it is possible to print the callerid value in the CLI when doing 'pri debug span 1' For example Call Ref: len= 2 (reference 2707/0xA93) (Terminator) Message type: CONNECT (7) [18 03 a9 83 97] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 23 ] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) I would like to print '1234567890 Message type: CONNECT (7) ... ... ' where 1234567890 is the callerid Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --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 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --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 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --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] Function vs. Application?
On Fri, 07 Dec 2007 15:19:49 -0500, Jared Smith [EMAIL PROTECTED] wrote: Hopefully I've explained it in such a way that it's clearer to you know. If not, let me know and I'll try to be more clear. Nope, good enough for me :-) 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] Happy Birthday Asterisk
Tilghman Lesher wrote: On Friday 07 December 2007 09:56:56 Bill Andersen wrote: Philip Prindeville wrote: So I'd venture to say that by August, the Internet will really be *30* years old. As Al Gore was born in 1948, I can see that the Internet could be as old as 30, but not much more. 35 years ago would put him at 25 years old. And inventing the whole Internet at 25 is pretty ambicious, even for Al! In actuality, most people produce all of the great inventions of their life by the time they hit 30. Einstein, for one, produced his great theory of relativity at the ripe old age of 26. Mark Spencer came up with Asterisk at age 22. So this idea that 25 is too young to produce a great achievement is baloney. BTW, Al Gore was credited with introducing the legislation that permitted commercial organizations onto the network that would become known as the Internet. So in a way, he did create the Internet, by changing the circumstances you would have to have in order to access this decentralized computer network. If you doubt the importance of having commercial organizations on the network, consider where the Internet would be, if Amazon, eBay, and Linux Support Services (d/b/a Digium) had never been allowed onto the network. Let's give credit where it's due: a lot of people in Washington were being lobbied by Bill Shrader, Vint Cerf, and Dave Van Bellengem (sp?) to be honest. All people like Senator Gore did was carry their water. The argument being put forward was that various groups (like SRI, Rand, Mitre, etc) would get onto ARPAnet because they had been awarded a DARPA or DISA or DMA contract... as would other vendors (like Boeing or Wellfleet or Raytheon...). Since SRI, Rand, Mitre, etc. would have a constant stream of contracts in progress, their ARPAnet access never went away. Others, like the latter group, would get their access yanked when their contracts ended (or got suspended while DoD budgets languished in Congress). Their argument was that such collaboration with other companies and universities would continue after contracts were completed, and that the Internet was a powerful collaboration tool (duh!)... ergo a permanent Internet was needed, even if the users had to pay for it themselves (rather than it being a perk of getting a DARPA contract). Even small companies (like FTP Software, who I was working for at the time), could benefit from being able to ship new binaries to government agencies or other partners on government contract (like HP, who we were writing a DOS TCP/IP stack for with a Sockets API... sound familiar?). In some ways, these were dark days: the future was very uncertain. On the other hand, we didn't have spam. ;-) -Philip ___ --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] Function vs. Application?
Hello Out of curiosity, what's the difference between a function and an application? asterisk*CLI core show functions asterisk*CLI core show applications 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] Polycom 601 stops ringing
At 10:58 12/7/2007, Joe Acquisto wrote: I have an odd issue, where a polycom 601 stops ringing, or more properly, maybe, stops *being* rung, when a call comes in. Other phones/extensions, continue to work fine, they being run at the same time. My dial plan works fine (?) seems it will ring properly, right after a reboot. It works fine for outgoing calls at all times. Hints? Is it behind a firewall? joe a. ___ --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] Print CALLERID in CLI during pri debug
Ok so the call reference is the 'cr' field (q931.c) and how do I retrieve the caller id from this call reference ? On Dec 7, 2007 4:29 AM, Richard Revels [EMAIL PROTECTED] wrote: When the call sets up the 'call reference' is assigned. It will be unique for the duration of the call and other messages, like Connect, will reference it. At the same time, the setup will have indicated the caller ID info. Sent from my iPhone On Dec 6, 2007, at 10:28 PM, Arpit Mehta [EMAIL PROTECTED] wrote: Or in other words is there a way to map which message is from which CallerID ? On Dec 6, 2007 6:40 PM, Arpit Mehta [EMAIL PROTECTED] wrote: Hi all, I was wondering if it is possible to print the callerid value in the CLI when doing 'pri debug span 1' For example Call Ref: len= 2 (reference 2707/0xA93) (Terminator) Message type: CONNECT (7) [18 03 a9 83 97] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 23 ] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) I would like to print '1234567890 Message type: CONNECT (7) ... ... ' where 1234567890 is the callerid Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --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 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --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] Using XML for configuration management, single-source-of-truth, etc.
You can store most of the configurations in a database which may be more accessable to you. Perl can also parse these configurations quickly enough if you know how to use the input record seperator ($/) properly. The only thing Asterisk will not store which you would probably need is the actual MAC address of the phones themselves. This may be done easily enough as comments in the users sip.conf section. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philip Prindeville Sent: Friday, December 07, 2007 13:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Using XML for configuration management,single-source-of-truth, etc. I'm starting work on some provisioning tools to simplify plugging in and configuring hard SIP handsets and conference bridges (maybe eventually MPEG-4 PoE video cameras that speak SIP as well). Issue is that I'd like to glean as much information out of the configuration files... but don't want to write a whole new parser to do it (especially not one that understands templates and macros). For instance, from the voicemail.conf, extensions.conf, and sip.conf files, I should be able to generate 90% of the configuration state needed for provisioning an out-of-the-box Sipura SPA941... if only those files were in some more parsable format, like XML. How much effort would it be to add an application that traverses the configuration state and writes it out as an XML flat file? Or perhaps at some point in the future, Asterisk's configuration files could be represented as XML natively (did someone in the back row just show gconf???). I'm a relative newbie, so if I'm missing something obvious or there's been a religious war on the subject in the past, apologies... -Philip ___ --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
