[asterisk-users] Asterisk Billing
Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi List. I'm in need of something that will allow me to analyze cdr details either via .csv or mysql that will give me call durations as well as call costs. This is so that we can see in what areas/staff are costing what per month/week on outbound phone calls. Can anyone recommend a system? I've looked at Asterisk CDR and while this works perfect it doesn't allow for actual call costs. I'm also looking at Astbill but not so sure if it will suit this application as that seems more for a provider - end user and Astbill wants to control the workings/creating of users/peers or am I mistaken? Thanks, Richard . Best Regards Richard Soderblom Network Configurations Cell: E-Mail: [EMAIL PROTECTED] Number of Attachments: 0 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Querying channel variables via the Manager API
DumpChan (it's there in 1.2 as well) would be great, if it were a manager command where you can choose the channel to dump and not a diaplan function that outputs the current channel config to the CLI. l. In data Wed, 18 Apr 2007 02:30:09 +0200, Philipp von Klitzing [EMAIL PROTECTED] ha scritto: Hi! I was thinking that there must be a way to tell Asterisk give me a complete dump of all the available channel information including variables In Asterisk 1.4: show application DumpChan Cheers, Philipp -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR datasets
Well, the larger the better :) l. In data Wed, 18 Apr 2007 04:15:28 +0200, Melcon Moraes [EMAIL PROTECTED] ha scritto: How large is large for you? []'s MM -Original Message- From: Lenz [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Tue, 17 Apr 2007 23:46:28 +0200 Delivered: Tue, 17 Apr 2007 18:45:47 Subject:[asterisk-users] CDR datasets Hello list, I have been working lately on a small CDR parsing utility, and would like to do some performance testing on it. I am looking for some - possibly large - real-life Asterisk CDR datasets to run some performance monitoring. Anybody's got some CDRs that can be shared? Thanks in advance, l. -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Default lenguage
Manolet Gmail wrote: Hi to all! i have installed asterisk 1.4.2 and asterisknow from the digium svn repository, when i was installing i select using menuselect utility the spanish voice lenguage pack. everythink is ok but i dont know how or where to tell asterisk to use the spanish as the default lenguage... i check on /var/lib/asterisk/sounds and i have the es directory with all the voices in spanish thanks in advanced! Put a line language=es in the general section of your sip.conf and iax.conf. If you installed the samples than this is also shown in the sample configuration files. In the dialplan (extensions.conf) you can add exten = s,n,Set(CHANNEL(language)=es) to the default context so that incoming calls hit the spanish version of the voicemail prompts. If you want the caller to switch his/her language by dialing 1 or 2 you can add something like this: exten = 1,1,Set(CHANNEL(language)=en) exten = 1,n,Goto(s,start) exten = 2,1,Set(CHANNEL(language)=es) exten = 2,n,Goto(s,start) exten = s,1,BackGround(xw_change_lang); Explain how to switch language exten = s,n(start),BackGround(welcome); Welcome and instruction exten = s,n,WaitExten(5) exten = s,n,BackGround(change_lang) ; Explain how to switch language exten = s,n,Goto(start) Theo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connection between Asterisk - Cisco 2851
callmanager can also be running in ios firmware in router (callmanager express), with near all funcionality as server version... Adam KOSA wrote: Antonopoulos Angelos wrote: Thanks for your help..But i dont know yet whether is CCM embeded on cisco 2851 or it is an extra element? Practically speaking, CCM is a standalone pc with software on it. Or maybe two, which are called publisher (master) and subscriber (slave). It's not embedded on the router. They are usually hp rack mountable servers, but you may install the CCM software on any hardware, at your own risk. hope this helps. best regards adam ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfercapability DIGITAL
yes and it is still set to speech I've even tried to port the old patch here http://bugs.digium.com/view.php?id=6251 to the system with no luck robb Melcon Moraes wrote: Have you tried: exten = s,n,SetTransferCapability(DIGITAL) ? []'s MM -Original Message- From: robert boardman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Tue, 17 Apr 2007 23:17:13 +0100 Delivered: Tue, 17 Apr 2007 19:15:09 Subject:[asterisk-users] Transfercapability DIGITAL Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/cgi-bin/imail.cgi?+_u=levelz_l=1,1176848736.557345.22480.arrino.hst.terra.com.br,4235,Des15,Des15 --Original Message Ends-- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] sending an SMS via Asterisk?
Yuan LIU wrote:
My dialplan looks like this:
exten = 0900900941,1,Goto(smsmotx,${CALLERIDNUM},1)
exten = 0794998990,1,Goto(smsmotx,${CALLERIDNUM},1)
How do callers get into these extensions?
They're specified on the smsq command, e.g.:
smsq --concurrent=3 --mo --motx-channel='mISDN/2/0900900941'
--motx-callerid=0434439000
079nnn 'testing 1234567890'
I'm a bit confused about your procedures. On one hand, if you use
smsq, you don't need to use SMS application
Oh. Then the confusion is clearly on my part.
I got most of the config from
http://www.voip-info.org/wiki-Asterisk+cmd+Sms
If I don't need the SMS application, can you tell me what I need to do,
or where I need to look?
Thanks for helping with this - it doesn't seem like sending SMS over
Asterisk is much used.
/Per Jessen, Zürich
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[asterisk-users] SIP failover between Sip Providers
Hi all, lets say I've registered at several Sip-Providers. Provider A offers best rates but is often too busy to get a line. Sip Provider B is stable (but more expensive). The asterisk box has a high call volume therefore problems at provider A will get obvious after a few calls stalled. In this case astersik shall switch temporarily to provider B but shall test periodically for selected calls if provider A is available again. I think it can be done by using the dialplan and the database to store the statistical information but maybe there is an easier way that integrates better with asterisk!? regards, -- Knud A. Müller ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP failover between Sip Providers
On Wed, 18 Apr 2007 09:04:22 +0200, Knud Müller wrote: I think it can be done by using the dialplan and the database to store the statistical information but maybe there is an easier way that integrates better with asterisk!? i dont think you'd even need a database with statistics. just have all calls sent to provider A with an automatic failover to provider B if the call can't be completed through A. you'd need to go look at the DIALSTATUS variable for that. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [OT] Nokia E60 firmware update break SIP
On 2007-04-17 00:53:56 -0700, Dinesh Nair [EMAIL PROTECTED] said: On Mon, 16 Apr 2007 20:14:40 -0700, Martin Joseph wrote: The phone no longer registers with asterisk, although it displays the little icon as though it has, and it doesn't even seem to try to pass calls to asterisk... So, I would avoid 3.06330904 20-11-06 RM-49 i've got an E61 running the same firmware revision and it works fine and dandy with asterisk 1.2.17. one thing you may want to do is to delete all your SIP profiles in the phone and reconfigure it from scratch. upgrading firmware from 2.x to 3.x broke something which wasnt forward compatible. we had similar issues, but deleting all profiles and reconfiguring from scratch fixed it. Yes, thank you Dinesh, that's exactly right. I deleted the SIP profiles and recreated them (tedious to be sure), and it seems to be working again. Also, there are clearly improvements with regards to how it can switch internet phone profiles automatically now... Thanks for the help! Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Hello Dan, What version of Asterisk are you using? I've had recording working with SVN before 1.4, the 1.4 betas and currently 1.4.1. *** Update *** Recordings are tied to a moderator joining the conference at this time. I may need to change that based on feedback/requests to do so. *** Update *** Please include a note in the documentation for that (and maybe even note that in the web page for configuring conferences) !! It is really needed. Also please update the web page of each (past) conference with the link from where the recording could be downloaded I've never user the sql option for the user/participant. It was contributed by another user of the suite. Depending on the technology the caller used to call into the conference you should have their Caller-id number and possibly their Caller-id name. What additional Information would you like to see? Also - this is probably again a problem of the missing documentation, but let me clarify my problem in detail: If I create a conference, there is a button email participants. If I click that button, nothing happens (). How does the whole email procedure works? How does the web-meetme gather the email addresses of the participants? There is no way how to configure participants to the conference. My understanding was, that participants are informed about conference start/end/extended by this procedure. But since there is no way how the application could find their email addresses, I just do not know how it should work. From the sources I see that it uses SQL database users - but since I use AD, my users database is empty Contributions welcome. There is a new How-To up on SF that covers the installation on a step by step basis. I've tried to comment the configuration files to make it clear how each setting works. Some features have been contributed to the project, and I am sorry to say that beyond making sure they integrate cleanly, I have not taken enough time to document their setup and use. I guess I should ask for supporting documentation before merging the changes/features. I agree, because without any documentation is the feature de-facto unusable. I am happy to contribute to the project but at this stage it is (due to the bugs mentioned above) for me unfortunately still quite far from being promising. Lot of work has been done, but there are still some important pieces to be done. Thank you for the feedback. I am surprised almost daily how many people have found it useful. I did not really expect it to be as popular as it has become, and I am more than happy to try and address any problems. Glad to hear that :-) Ondrej Thanks, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this e-mail and in any attachments is confidential and is designated solely for the attention of the intended recipient(s). If you are not an intended recipient, you must not use, disclose, copy, distribute or retain this e-mail or any part thereof. If you have received this e-mail in error, please notify the sender by return e-mail and delete all copies of this e-mail from your computer system(s). Please direct any additional queries to: [EMAIL PROTECTED] Thank You. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING
Hi Moises,
the Asterisk SVN-branch-1.4-r60989 make a change in the
ast_channel_alloc function:
This is a big improvement over the current CDR fixes. It may still
need refinement, but this won't have as many folks bothered.
here the patch for chan_unicall.c ;p
--
Humberto Figuera - Using Linux 2.6.20
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603
--- chan_unicall.c.orig 2007-04-18 03:32:17.0 -0400
+++ chan_unicall.c 2007-04-18 03:32:26.0 -0400
@@ -2485,7 +2485,7 @@
}
while (x 3);
-if ( ( tmp = ast_channel_alloc(0, state, 0, 0, chan_name) ) == NULL)
+if ( ( tmp = ast_channel_alloc(0, state, 0, 0, i-accountcode, i-exten, i-context, i-amaflags, chan_name) ) == NULL)
{
ast_log(LOG_WARNING, Unable to allocate channel structure\n);
return NULL;
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Re: [asterisk-users] peers are using wrong contexts
Tnaks for your answer. Sorry, if I'm missing something obvious here. Yes, it's asterisk 1.4. I've configured a user entry in users.conf. One of the lines is context = numberplan-custom-1. I suppose that should make that user use the dialplan context [numberplan-custom-1]. I have [numberplan-custom-1] configured in extensions.conf. However the user uses [default]. users.conf ... [951XX] callwaiting = yes cid_number = 951XX context = numberplan-custom-1 email = fullname = New User group = hasagent = no hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 951XX secret = 00 threewaycalling = yes vmsecret = 1234 zapchan = registeriax = no registersip = yes extensions.conf . [default] exten = _X.,1,NoOp(This is default) [numberplan-custom-1] exten = _X.,1,NoOp(This is numberplan-custom-1) Output of sip show peer 951XX CLI sip show peer 951XX * Name : 951XX Secret : Set MD5Secret: Not set Context : numberplan-custom-1 Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: 1 Pickupgroup : 1 Mailbox : 951XX VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Dynamic : Yes Callerid : New User 951XX MaxCallBR: 384 kbps Expire : 26 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : X.X.X.X Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 951XX SIP Options : (none) Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (none) Auto-Framing: No Status : Unmonitored Useragent: Sipura/SPA3000-2.0.11(GWg) Reg. Contact : sip:[EMAIL PROTECTED]:5060 Is this Asterisk 1.4.x? from samples/extensions.conf... ; ; User context is where entries from users.conf are registered. The ; default value is 'default' ; ;userscontext=default ; Is this any help? regards, Drew dima wrote: Hello, everyone. Today I've installed an asterisk svn trunk (r61667). The problem I'm having is no matter what context I set in the config file for that peer, default is always being used. The output of sip show peers shows the context correctly, but when I try to make a call, using that peer, I can only dial the numbers set in the default context. Please, could anyone help me resolve this. Thanks in advance. This is a part of users.conf [951XX] callwaiting = yes cid_number = 951XX context = numberplan-custom-1 email = fullname = New User group = hasagent = no hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 951XX secret = 00 threewaycalling = yes vmsecret = 1234 zapchan = registeriax = no registersip = yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING
Hi Moises,
the Asterisk SVN-branch-1.4-r60989 make a change in the
ast_channel_alloc function:
This is a big improvement over the current CDR fixes. It may still
need refinement, but this won't have as many folks bothered.
here the patch for chan_unicall.c ;p
--- chan_unicall.c.orig 2007-04-18 03:32:17.0 -0400
+++ chan_unicall.c 2007-04-18 03:32:26.0 -0400
@@ -2485,7 +2485,7 @@
}
while (x 3);
-if ( ( tmp = ast_channel_alloc(0, state, 0, 0, chan_name) ) == NULL)
+if ( ( tmp = ast_channel_alloc(0, state, 0, 0, i-accountcode,
i-exten, i-context, i-amaflags, chan_name) ) == NULL)
{
ast_log(LOG_WARNING, Unable to allocate channel structure\n);
return NULL;
--
Humberto Figuera - Using Linux 2.6.20
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603
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Re: [asterisk-users] Asterisk 1.2.16 - No Caller ID
Hi, have you tried different values of callerid? Maybe setting *useincomingcalleridonzaptransfer* to yes can help you. Giorgio Incantalupo OCOSA ListAcc wrote: Hello, When I upgraded a while back the caller ID stop working I have tried everything and searched the lists no answer. Please help!! I have two pots lines coming into the Asterisk Box caller ID is set in the zapta.conf Here is what our zapata.conf looks like [channels] busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=10.5 txgain=5.5 group=1 pickupgroup=1-4 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel=1 channel=2 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
Hello I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921 I'd like to have some user feedback about how those phones perform, and whether their LCD screen displays both the caller ID name and number (The GrandStream BT-100 only displays numbers, which isn't very helpful). Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
While I can't speak for the Linksys SPA-921, I /can/ comment on the Grandstream GXP-2000. We're running half a dozen of these at the moment, primarily for testing. I can confirm that the LCD display /does/ display both caller name and number - assuming of course that both are presented. We've had the very occasional problem with the phone locking up, but nothing overly serious. I'm fairly happy with voice quality (using both aLaw and GSM) and the BLF indicators work quite nicely. Gilles Ganault wrote: Hello I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921 I'd like to have some user feedback about how those phones perform, and whether their LCD screen displays both the caller ID name and number (The GrandStream BT-100 only displays numbers, which isn't very helpful). Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bad case of buzzing
Hi, are you using PoE or power supplies? As power supllies usually are not grounded it could be that it's comming from the power source. You could try using a grounded PoE switch or probably a power backup to test if this is the case. Cheers Tim On 3/30/07, Louis-David Mitterrand [EMAIL PROTECTED] wrote: Hello, We are at wit's end on this. One (and only one) of our five asterisk installation is giving us real headaches. Buzzing and/or choppy sound interfere with conversations. I recorded some conversations with monitor() and no problem whatsoever appear in the recording, while the local user was hearing the buzz and half my words. This is a 1.2.16 installation with mISDN but mostly using SIP to our central PRI-equipped asterisk. Phones are Polycom 430, 601, Cisco 7960, 7912 all to the latest firmware. We tried everything: changing the switch, network cards, auditing every network drop with fluke, re-certifying our wan, swapping some phones to no effect. Has anyone gone through that ordeal? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- snom technology AG Tim Koehler Partner Manager ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP failover between Sip Providers
Dinesh Nair wrote: On Wed, 18 Apr 2007 09:04:22 +0200, Knud Müller wrote: I think it can be done by using the dialplan and the database to store the statistical information but maybe there is an easier way that integrates better with asterisk!? i dont think you'd even need a database with statistics. just have all calls sent to provider A with an automatic failover to provider B if the call can't be completed through A. you'd need to go look at the DIALSTATUS variable for that. The disadvantage of that solution is, that I'll always try to make a connection with a provider for that I know by experience it wouldn't work. In the failover case the time between starting to dial and the first ring gets longer. If I know that Provider A fails 60% of Calls then I don't need to start with a but can start with b directly. -- Knud A. Müller ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
I have experience with both. GXP is a great phone for its low price and it has all the features of the IP phones. It doesn't have any considerable issues with it. On the other hand Linksys 921 is superior in voice quality, look, and TFTP support but limited in features, like limited line appearances, no PoE or inline ethernet, and many other software features. In my opinion, where you have busy environment and high usage of phones, use GXP-2000 phones. For a few executive desks where there is not much call volume, use Linksys. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
Gilles Ganault wrote: Hello I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921 I'd like to have some user feedback about how those phones perform, and whether their LCD screen displays both the caller ID name and number (The GrandStream BT-100 only displays numbers, which isn't very helpful). I both have the BT-100 and the SPA-921. The BT-100 displays caller name or number. Since the display is made of 7 segment characters, text is hard to display. I noticed that some characters are displayed where other are just ignored. This is not very handy. If the callerid is specified with callernum like callerid=name number, then the number is only displayed (no name). The SPA-921. Has a dot-matrix display so everything is displayed very nicely. Some points to consider for the BT-100: It does not have a phone book. It just remembers callers and called numbers. The speaker quality for handsfree calling is just unacceptable (no echo cancellation) The mute button works, but there is no feedback at all (led for instance). So you have to ask the other party can you hear me? when using the mute Some points to consider for the SPA-921: Very complex web interface (yes you have the freedom to tweak everything, I prefer a simpler interface) I had problem with this phone were the called party could not hear me. I had to fix the codec in the phone to ulaw to get it working. (In asterisk I have an disallow=all, allow=alaw, allow=ulaw setup but this does not seem to work for this phone). The display has no backlight (no problem in an office environment, but the BT-100 looks better in that respect) The users of both phones are very satisfied (good sound quality). Both give a clear indication for message waiting for instance. I plan to buy more SPA-921 because of the before mentioned reasons. Theo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
spa-922/942 has backlighted display, inline power (PoE), internal switch, audio gain/attenuation can be tunned, works great in bussines environment (voice vlan negotiation through cdp from ci$co switch), solid design, robust chassis lack of features like programable buttons for pickup or busy lamp field ... PJ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
On Wed, 18 Apr 2007, Rob Hillis wrote: While I can't speak for the Linksys SPA-921, I /can/ comment on the Grandstream GXP-2000. We're running half a dozen of these at the moment, primarily for testing. I can confirm that the LCD display /does/ display both caller name and number - assuming of course that both are presented. We've had the very occasional problem with the phone locking up, but nothing overly serious. I'm fairly happy with voice quality (using both aLaw and GSM) and the BLF indicators work quite nicely. I've deployed a fair number of GXP2000's over the past few months and generally found them to perform very well. However, search back in the archives and you'll find a lot of negavtive comments. Make sure they're flashed to the latest versions. Sound quality has been good for me, setup via their web interface is also easy - they can provision via TFTP but if you only have a small number to provision it's just as easy to use the web interface. There is a 3rd party PERL program which I'm now using to help me provision them a little quicker than going via the web interface. (it drive sthe web interface directly for you!) The display does show caller name number - if the sending system sends it. The handset is heavy enough and the buttons easy to use. It is cheap cheerful but after flashing some of the early ones I've had with the latest firmware, I've not really had a problem. You put them on the desk (or wall!), give a few minutes training to the puntes using them (how to transfer calles, etc) and generally that's it. I've jsut worked out how to put a custom logo on the phone display too. I'd avoid the BT100/200 in anything other than a demo situation though, or where the client really is strapped for cash... (although I've only ever used the 100's and they're OK, but caller ID numbers only. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] sending an SMS via Asterisk?
Per Jessen wrote:
Yuan LIU wrote:
My dialplan looks like this:
exten = 0900900941,1,Goto(smsmotx,${CALLERIDNUM},1)
exten = 0794998990,1,Goto(smsmotx,${CALLERIDNUM},1)
How do callers get into these extensions?
They're specified on the smsq command, e.g.:
smsq --concurrent=3 --mo --motx-channel='mISDN/2/0900900941'
--motx-callerid=0434439000 079nnn 'testing 1234567890'
I'm a bit confused about your procedures. On one hand, if you use
smsq, you don't need to use SMS application
Oh. Then the confusion is clearly on my part.
OK, part of the confusion is now clearing up. But I'm not getting much
further. When I try to send an SMS, I see the call going through, but
no SMS is ever sent.
I'm wondering if this is a matter of which protocols the Swisscom SMSCs
support? I understand that the SMS app uses ETSI ES 201 912 protocol
1, and I've also patched app_sms.c to try out protocol 2 (longer fixnet
messages). Still no joy.
/Per Jessen, Zürich
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Re: [asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
On Apr 18, 2007, at 6:50 AM, Rob Hillis wrote: We've had the very occasional problem with the phone locking up, but nothing overly serious. Are you using DHCP on the GXPs that are locking up? I have one and it would lock up almost every night requiring the power to be pulled in the morning. Knowing my DHCP server can sometimes be a PITA and not renew leases properly, I on a hunch changed my GXP to a static IP address and so far it has yet to lock up again. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bad case of buzzing
On Wed, Apr 18, 2007 at 01:04:31PM +0200, Tim Koehler wrote: Hi, are you using PoE or power supplies? As power supllies usually are not grounded it could be that it's comming from the power source. We are using PoE You could try using a grounded PoE switch or probably a power backup to test if this is the case. The problem was solved by changing the server and installing a fresh OS image on it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reminder: HITBSecConf2007 - Malaysia: Call for Papers closing in 2 weeks
Greetings from sunny Malaysia! This is a reminder that the Call for Papers for the upcoming HITBSecConf2007 - Malaysia is closing on the 1st of May. HITBSecConf2007 - Malaysia is set to take place from the 3rd till the 6th of September in Kuala Lumpur. Our event last year attracted over 600 attendees from all corners of the globe and this year we are expecting this number to grow to well over 800. In addition, the event will feature 4 keynote speakers, 40 researchers, 7 tracks of hands-on technical trainings, a dual-track security conference, capture the flag competition, a lock picking village, zone-h/hitb hacking challenge, bzflag competition and one MASSIVE post conference party!!! If you only attend ONE event this year; make sure its HITBSecConf2007 - Malaysia; Asia's largest network security conference! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon-Vonage Lawsuit
Salvatore, most, if not all VoIP providers support LNP. We do. On 4/17/07, Salvatore Giudice [EMAIL PROTECTED] wrote: Can anyone recommend a VoIP provider who supports LNP? I need to move to a new provider for inbound calling and I want to keep my current numbers. My current provider is a gaggle of retards. Any recommendation? I need a service that is reliable. TIA, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phones working with 1.2.17, not with 1.4.2
Hello, I've got various phones (mostly SPA-922) behind NAT registered to Asterisk. I've set nat=yes and canreinvite=no, and everything seemed to work great with 1.2.17. After upgrading to 1.4.2 using users.conf and macro-stdexten my spa-922 can't call other extensions. -- Executing [EMAIL PROTECTED]:1] Macro(SIP/22-b72006f0, stdexten|23| SIP/23) in new stack -- Executing [EMAIL PROTECTED]:1] Dial(SIP/22-b72006f0, SIP/23) in new stack -- Called 23 [Apr 18 12:29:16] NOTICE[3831]: chan_sip.c:2757 auto_congest: Auto-congesting SIP/23-081db528 -- SIP/23-081db528 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/22-b72006f0' status is 'CONGESTION' Debugging SIP messages seems that the called exten is not replying to invites, but it registers correctly. Other phones (Siemens C450 IP) seem to be able to call other extensions: -- Executing [EMAIL PROTECTED]:1] Macro(SIP/27-b72020e0, stdexten|22| SIP/22) in new stack -- Executing [EMAIL PROTECTED]:1] Dial(SIP/27-b72020e0, SIP/22) in new stack -- Called 22 -- SIP/22-081de4f8 is ringing Phones configuration is unaltered. What could it be? thanks Luca ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] sending an SMS via Asterisk?
Per Jessen wrote: Per Jessen wrote: OK, part of the confusion is now clearing up. But I'm not getting much further. When I try to send an SMS, I see the call going through, but no SMS is ever sent. This is a bit of what I see in the debug output: (this is sending a longer message, protocol 2): P[ 2] -- caps:Speech pi:0 keypad: sending_complete:0 P[ 2] -- None -- mISDN/3-u54 answered Local/[EMAIL PROTECTED],2 Channel Local/[EMAIL PROTECTED],1 was answered. Launching SMS(062210|t) on Local/[EMAIL PROTECTED],1 P[ 2] * IND: Got Fixup State:CONNECTED L3id:50012 == Spawn extension (Internal, 062210, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' P[ 2] I IND :FACILITY oad:0434439000 dad:062210 pid:19 state:CONNECTED P[ 2] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:0062210 P[ 2] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 2] -- caps:Speech pi:0 keypad: sending_complete:0 P[ 2] -- AOCD currency: currency:FR. amount:10 multiplier:1 typeOfChargingInfo:-1220842403 P[ 2] I IND :INFORMATION oad:0434439000 dad:062210 pid:19 state:CONNECTED P[ 2] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:0062210 P[ 2] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 2] -- caps:Speech pi:0 keypad: sending_complete:0 P[ 2] -- None -- SMS[-1] RX 93 00 6D -- SMS[0] TX 10 98 96 00 10 01 00 00 11 06 00 00 00 00 00 00 00 12 03 00 02 00 04 13 65 00 53 65 63 75 72 69 74 79 20 72 65 73 65 61 72 63 68 65 72 73 20 68 61 76 65 20 74 72 61 63 65 64 20 73 70 61 6D 2D 73 65 6E 64 69 6E 67 20 62 6F 74 6E 65 74 20 63 6C 69 65 6E 74 73 20 62 61 63 6B 20 74 6F 20 6E 65 74 77 6F 72 6B 73 20 72 75 6E 20 62 79 20 74 68 65 20 55 53 20 6D 69 6C 69 74 61 72 79 2E 17 01 00 01 18 0A 00 30 34 33 34 34 33 39 30 30 30 1B 01 00 01 1C 03 00 00 00 00 E8 P[ 2] I IND :DISCONNECT oad:0434439000 dad:062210 pid:19 state:CONNECTED P[ 2] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:0062210 P[ 2] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 2] -- caps:Speech pi:8 keypad: sending_complete:0 P[ 2] -- org:1 nt:0, inbandavail:1 state:10 P[ 2] -- queue_hangup In all the other examples I've come across on the 'net, there are multil lines beginning SMS[x] RX/TX .. /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] QueueMetrics 1.3.4 released today
Hello list, QueueMetrics 1.3.4 has been released today. Among other features, it provides realtime cluster monitoring through the manager API and, by popular demand, user defined time intervals in the daily call breakdown. You can find the latest version at http://queuemetrics.com and support at http://forum.queuemetrics.com For those who don't know it, QueueMetrics is an industrial-grade commercial solution available free of charge to small CCs / SOHOs / individual hackers. Comments and ideas are welcome! l. -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk unable to create files, too many files open
hello, im having trouble with asterisk with medium load, it seems im running out of files, here is a chunk of the logs with grep \(file\|pipe\): Apr 18 15:40:46 WARNING[11644] res_agi.c: Unable to create toast pipe: Too many open files Apr 18 15:40:46 WARNING[11574] channel.c: Channel allocation failed: Can't create alert pipe! Apr 18 15:40:46 WARNING[11574] channel.c: Channel allocation failed: Can't create alert pipe! Apr 18 15:40:46 ERROR[11643] cdr_csv.c: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files Apr 18 15:40:46 ERROR[11643] cdr_custom.c: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Too many open files Apr 18 15:40:46 ERROR[11645] cdr_csv.c: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files i tried to increase the number in /proc/sys/fs/file-max , which was: 203511 and file-nr was 21120 203511 so i did : echo 400176 /proc/sys/fs/file-max but it didn't help, what could possibly make this happen, and does asterisk need that huge number of files ? this machine takes less than ~40 calls at peaks! this is asterisk 1.2.17 running on Debian etch 2.6.18-4-amd64 #1 SMP on xeon cpu, i got the same behaviour with 1.2.16 too! -- uwe maysara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP failover between Sip Providers
Am Mittwoch, den 18.04.2007, 13:18 +0200 schrieb Knud Müller:
Dinesh Nair wrote:
On Wed, 18 Apr 2007 09:04:22 +0200, Knud Müller wrote:
I
think it can be done by using the dialplan and the database to store the
statistical information but maybe there is an easier way that integrates
better with asterisk!?
i dont think you'd even need a database with statistics. just have all
calls sent to provider A with an automatic failover to provider B if the
call can't be completed through A. you'd need to go look at the DIALSTATUS
variable for that.
The disadvantage of that solution is, that I'll always try to make a
connection with a provider for that I know by experience it wouldn't
work. In the failover case the time between starting to dial and the
first ring gets longer. If I know that Provider A fails 60% of Calls
then I don't need to start with a but can start with b directly.
Hi Knud,
I think what you want is a combination of both.
If indeed DIALSTATUS reveals that provider A is having his five
minutes (again), the first call that notices this could set a database
flag, say, DB(a-is-crappy) to the current time value.
All calls could, before trying provider A, retrieve this value - if the
last crap moment was less than 300 seconds ago, just skip A and go for
B immediately. This way, no more than one call per 300 seconds should be
delayed - except of course, when those 300 timed out and two outgoing
calls start before any of those returns the bad DIALSTATUS. Anyway, they
will block the provider A again if he continues having moments, but
will allow using A as long as that works fine.
You should take extra care to distinguish DIALSTATUS cases; a call that
could not be terminated because the number was invalid should better not
block provider A if this can be distinguished.
Anselm('s 2 cent)
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Re: [asterisk-users] Transfercapability DIGITAL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi robb, Have you just seen the bearer capability in asterisk or is the call nat working? I've seen that a digital call shows up as speech. You are using Zap? Or are you using mISDN? Cause there you have to set an extra parameter in the dial statement. chris... robert boardman schrieb: yes and it is still set to speech I've even tried to port the old patch here http://bugs.digium.com/view.php?id=6251 to the system with no luck robb Melcon Moraes wrote: Have you tried: exten = s,n,SetTransferCapability(DIGITAL) ? []'s MM -Original Message- From: robert boardman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Tue, 17 Apr 2007 23:17:13 +0100 Delivered: Tue, 17 Apr 2007 19:15:09 Subject:[asterisk-users] Transfercapability DIGITAL Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/cgi-bin/imail.cgi?+_u=levelz_l=1,1176848736.557345.22480.arrino.hst.terra.com.br,4235,Des15,Des15 --Original Message Ends-- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - -- Dipl.-Ing. Kurt Krenn - IT-Beratung Franz-Josef-Strasse 33/4/43, 5020 Salzburg Tel: +43 662 879512 Fax: +43 662 875960 IP-Tel: +43 780 kkrenn (557366) Email: [EMAIL PROTECTED] sip: [EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGJhXnR0exH8dhr/YRAoFkAJ0UEmz8y+XqLYqDhBTTDl7VbdEkjACfabkX X5mowtdnhs9qiX26oPxJJbA= =aBxY -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] peers are using wrong contexts
Try adding userscontext = numberplan-custom-1 to the [general] section of extensions.conf to see if that helps regards, Drew dima wrote: Tnaks for your answer. Sorry, if I'm missing something obvious here. Yes, it's asterisk 1.4. I've configured a user entry in users.conf. One of the lines is context = numberplan-custom-1. I suppose that should make that user use the dialplan context [numberplan-custom-1]. I have [numberplan-custom-1] configured in extensions.conf. However the user uses [default]. users.conf ... [951XX] callwaiting = yes cid_number = 951XX context = numberplan-custom-1 email = fullname = New User group = hasagent = no hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 951XX secret = 00 threewaycalling = yes vmsecret = 1234 zapchan = registeriax = no registersip = yes extensions.conf . [default] exten = _X.,1,NoOp(This is default) [numberplan-custom-1] exten = _X.,1,NoOp(This is numberplan-custom-1) Output of sip show peer 951XX CLI sip show peer 951XX * Name : 951XX Secret : Set MD5Secret: Not set Context : numberplan-custom-1 Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: 1 Pickupgroup : 1 Mailbox : 951XX VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Dynamic : Yes Callerid : New User 951XX MaxCallBR: 384 kbps Expire : 26 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : X.X.X.X Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 951XX SIP Options : (none) Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (none) Auto-Framing: No Status : Unmonitored Useragent: Sipura/SPA3000-2.0.11(GWg) Reg. Contact : sip:[EMAIL PROTECTED]:5060 Is this Asterisk 1.4.x? from samples/extensions.conf... ; ; User context is where entries from users.conf are registered. The ; default value is 'default' ; ;userscontext=default ; Is this any help? regards, Drew dima wrote: Hello, everyone. Today I've installed an asterisk svn trunk (r61667). The problem I'm having is no matter what context I set in the config file for that peer, default is always being used. The output of sip show peers shows the context correctly, but when I try to make a call, using that peer, I can only dial the numbers set in the default context. Please, could anyone help me resolve this. Thanks in advance. This is a part of users.conf [951XX] callwaiting = yes cid_number = 951XX context = numberplan-custom-1 email = fullname = New User group = hasagent = no hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 951XX secret = 00 threewaycalling = yes vmsecret = 1234 zapchan = registeriax = no registersip = yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
Gilles Ganault wrote: I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921 I'd like to have some user feedback about how those phones perform, and whether their LCD screen displays both the caller ID name and number (The GrandStream BT-100 only displays numbers, which isn't very helpful). I've just bought a SPA-921 and a SPA-941. I've been testing the 921 for a while already, and I'm quite happy with it. It would have been nice if it had also had SMS capability, but it's not critical. /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue App - Free agent and waiting calls
Asterisk 1.4 I have strategy= leastrecent and autofill = yes I have 2 agents, one is answering a call and the other is free and have some calls waiting in the queue. Only when the first agent hangup the second agent receive the first call in the queue. It happends some times. This behavior still happend in 1.4.1 version. Thanks a lot. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
Theo Band wrote: Some points to consider for the SPA-921: Very complex web interface (yes you have the freedom to tweak everything, I prefer a simpler interface) But the SPA-921 can also be remote provisioned/configured over TFTP, which is just perfect. IMHO. The display has no backlight (no problem in an office environment, Yes, good point. That would also be nice to have. /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk unable to create files, too many files open
Hi, On Wed, 2007-04-18 at 15:56 +0300, Maysara A. Abdulhaq wrote: hello, i tried to increase the number in /proc/sys/fs/file-max , which was: 203511 and file-nr was 21120 203511 so i did : echo 400176 /proc/sys/fs/file-max but it didn't help, what could possibly make this happen, and does asterisk need that huge number of files ? this machine takes less than ~40 calls at peaks! that value is a system value, not a process one. You should increase asterisk process file limit with ulimin -n before starting *. Eg ulimit -n 8192 will increase max files from the default 1024 to 8192. Greetings, matteo. -- Matteo Brancaleoni RD Director Tel :+39.02.70633354 Voip :sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk unable to create files, too many files open
Hi Maysara, I have your same problem. are you using mISDN? If yes update your driver. Giorgio Incantalupo Maysara A. Abdulhaq wrote: hello, im having trouble with asterisk with medium load, it seems im running out of files, here is a chunk of the logs with grep \(file\|pipe\): Apr 18 15:40:46 WARNING[11644] res_agi.c: Unable to create toast pipe: Too many open files Apr 18 15:40:46 WARNING[11574] channel.c: Channel allocation failed: Can't create alert pipe! Apr 18 15:40:46 WARNING[11574] channel.c: Channel allocation failed: Can't create alert pipe! Apr 18 15:40:46 ERROR[11643] cdr_csv.c: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files Apr 18 15:40:46 ERROR[11643] cdr_custom.c: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Too many open files Apr 18 15:40:46 ERROR[11645] cdr_csv.c: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files i tried to increase the number in /proc/sys/fs/file-max , which was: 203511 and file-nr was 21120 203511 so i did : echo 400176 /proc/sys/fs/file-max but it didn't help, what could possibly make this happen, and does asterisk need that huge number of files ? this machine takes less than ~40 calls at peaks! this is asterisk 1.2.17 running on Debian etch 2.6.18-4-amd64 #1 SMP on xeon cpu, i got the same behaviour with 1.2.16 too! -- uwe maysara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Querying channel variables via the Manager API
On 17 Apr 2007, at 22:32, Lenz wrote: Hello list, we are developing a new application that uses the Manager API in order to find a set of channels where variables are set in a predefined way. To do this, we currently send a Status command to obtain all available channels and then query them all, one by one, fot the status of a certain dialplan variable. As you can imagine, this gets rapidly pretty tedious as the number of active channels on a server grows and requires a lot of round-trips to and from the Asterisk server. I was wondering if there are more efficient ways to get: 1. a variable as set on all channels 2. the complete list of channel variables for one channel, using standard manager response block and not reverting to an execute CLI command show channel Local/[EMAIL PROTECTED] Anybody has ideas/hints on how to make all this a bit less cumbersome? You could query via SNMP. it has the astChanVariables for each active channel as a DisplayString I can't promise that this is less cumbersome, but the overhead might be smaller. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 sluggish keys: found the problem!
Kenneth Padgett wrote:
I have learned the hard way that using old configs with new firmware is
asking for trouble. It is much better to keep your custom configurations
in a MAC specific overrides file and replace the sip.cfg and phone1.cfg
files completely.
This doesn't guarantee that you won't have problems, but it's a lot
easier to troubleshoot an overrides file with a dozen items in it than
to sift through big, customized sip.cfg files.
Where can I find documentation on how to setup an override file using
the phone's MAC? I see a (MAC)-phone.cfg file the phone uploads has
something about overrides in it, but it looks like settings that the
phone re-reads...
Any help appreciated! Thanks.
There is a Polycom white paper, part number 3725-17461-001/A, that is
available from the Polycom website. The white paper title is
Configuration File Management on SoundPoint IP Phones. It outlines how
configuration files are specified and loaded by the phone.
The document is useful as a starting point, with these caveats:
It specifically warns *against* using the {MACADDR}-phone.cfg file for
configuring custom settings on the phone, because this is the name of
the configuration override file generated by the phone when the user
changes a setting such as the preferred ring type.
The instructions recommend creating a totally custom file specific to
the phone and then calling the file in the {MACADDR}.cfg file. This
didn't work for us, whether we used a name like local-settings.cfg or
custom1.cfg or a name like {MACADDR}-custom.cfg. The only thing that
did work was putting the configuration changes in the
{MACADDR}-phone.cfg and making the file {MACADDR}-phone.cfg read-only.
(This is on the SIP 2.1.0 firmware.)
It's possible that we were specifying the files incorrectly in the
{MACADDR}.cfg file; they are read from left to right, and that's how
they were entered. Nevertheless, the only thing that has worked is to
put the custom configurations in an overrides file.
Some people may need users to be able to configure overrides, but that's
not the case for us.
-Stephen-
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Re: [asterisk-users] peers are using wrong contexts
Try adding userscontext = numberplan-custom-1 to the [general] section of extensions.conf Done that. No change happened. Extesions are still executed in default context. One strange thing I've noticed is that in lines like SIP/80.1.61.21-092c23b0 before I used to see a number of extension that was calling. Now its my IP address, not a number. CLI -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/80.1.61.21-092c23b0, This is default) in new stack == Auto fallthrough, channel 'SIP/80.1.61.21-092c23b0' status is 'UNKNOWN' Another thing I didn't mention is that I used GUI for the initial configuration. dima wrote: Tnaks for your answer. Sorry, if I'm missing something obvious here. Yes, it's asterisk 1.4. I've configured a user entry in users.conf. One of the lines is context = numberplan-custom-1. I suppose that should make that user use the dialplan context [numberplan-custom-1]. I have [numberplan-custom-1] configured in extensions.conf. However the user uses [default]. users.conf ... [951XX] callwaiting = yes cid_number = 951XX context = numberplan-custom-1 email = fullname = New User group = hasagent = no hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 951XX secret = 00 threewaycalling = yes vmsecret = 1234 zapchan = registeriax = no registersip = yes extensions.conf . [default] exten = _X.,1,NoOp(This is default) [numberplan-custom-1] exten = _X.,1,NoOp(This is numberplan-custom-1) Output of sip show peer 951XX CLI sip show peer 951XX * Name : 951XX Secret : Set MD5Secret: Not set Context : numberplan-custom-1 Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: 1 Pickupgroup : 1 Mailbox : 951XX VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Dynamic : Yes Callerid : New User 951XX MaxCallBR: 384 kbps Expire : 26 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : X.X.X.X Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 951XX SIP Options : (none) Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (none) Auto-Framing: No Status : Unmonitored Useragent: Sipura/SPA3000-2.0.11(GWg) Reg. Contact : sip:[EMAIL PROTECTED]:5060 Is this Asterisk 1.4.x? from samples/extensions.conf... ; ; User context is where entries from users.conf are registered. The ; default value is 'default' ; ;userscontext=default ; Is this any help? regards, Drew dima wrote: Hello, everyone. Today I've installed an asterisk svn trunk (r61667). The problem I'm having is no matter what context I set in the config file for that peer, default is always being used. The output of sip show peers shows the context correctly, but when I try to make a call, using that peer, I can only dial the numbers set in the default context. Please, could anyone help me resolve this. Thanks in advance. This is a part of users.conf [951XX] callwaiting = yes cid_number = 951XX context = numberplan-custom-1 email = fullname = New User group = hasagent = no hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 951XX secret = 00 threewaycalling = yes vmsecret = 1234 zapchan = registeriax = no registersip = yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and
RE: [asterisk-users] Queue App - Free agent and waiting calls
Try ringall or roundrobbin. You only have two agents. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Wednesday, April 18, 2007 9:21 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queue App - Free agent and waiting calls Asterisk 1.4 I have strategy= leastrecent and autofill = yes I have 2 agents, one is answering a call and the other is free and have some calls waiting in the queue. Only when the first agent hangup the second agent receive the first call in the queue. It happends some times. This behavior still happend in 1.4.1 version. Thanks a lot. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Billing
the cdr analyzer should work for most of what you need. The call costs will be the hard part. If you know how much each type of call should cost (based on destination number, location, etc), then you could do the math on your own. But if you don't, then you'll have to wait for your provider to give you call detail back with the costs calculated. Rob Richard Soderblom wrote: Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi List. I'm in need of something that will allow me to analyze cdr details either via .csv or mysql that will give me call durations as well as call costs. This is so that we can see in what areas/staff are costing what per month/week on outbound phone calls. Can anyone recommend a system? I've looked at Asterisk CDR and while this works perfect it doesn't allow for actual call costs. I'm also looking at Astbill but not so sure if it will suit this application as that seems more for a provider - end user and Astbill wants to control the workings/creating of users/peers or am I mistaken? Thanks, Richard . Best Regards Richard Soderblom Network Configurations Cell: E-Mail: [EMAIL PROTECTED] Number of Attachments: 0 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk svn and zaptel
Hi all!! I have downloaded the asterisk from svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk-trunk (is the asterisk 1.4 subversion). I also downloaded the patch for cellphone and make it work fine. Then I bought the tdm11b board to have phone connection in my computer. I installed the hardware for zapte and the libpri modules in my Mandriva 2007 and the lights of the pci card switch on. I can see zaptel working by lsmod. Now I go to my asterisk recompile it but I realize there is no chan_zap.so! When I recompile it, i check the make menuselect and the channel zapata is not appearing there. Does any body know any patch for that? Or how to sort out this problem Thanks a lot iban _ ¿Estás pensando en cambiar de coche? Todas los modelos de serie y extras en MSN Motor. http://motor.msn.es/researchcentre/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP failover between Sip Providers
Anselm Martin Hoffmeister wrote:
Am Mittwoch, den 18.04.2007, 13:18 +0200 schrieb Knud Müller:
Dinesh Nair wrote:
On Wed, 18 Apr 2007 09:04:22 +0200, Knud Müller wrote:
I
think it can be done by using the dialplan and the database to store the
statistical information but maybe there is an easier way that integrates
better with asterisk!?
i dont think you'd even need a database with statistics. just have all
calls sent to provider A with an automatic failover to provider B if the
call can't be completed through A. you'd need to go look at the DIALSTATUS
variable for that.
The disadvantage of that solution is, that I'll always try to make a
connection with a provider for that I know by experience it wouldn't
work. In the failover case the time between starting to dial and the
first ring gets longer. If I know that Provider A fails 60% of Calls
then I don't need to start with a but can start with b directly.
Hi Knud,
I think what you want is a combination of both.
If indeed DIALSTATUS reveals that provider A is having his five
minutes (again), the first call that notices this could set a database
flag, say, DB(a-is-crappy) to the current time value.
All calls could, before trying provider A, retrieve this value - if the
last crap moment was less than 300 seconds ago, just skip A and go for
B immediately. This way, no more than one call per 300 seconds should be
delayed - except of course, when those 300 timed out and two outgoing
calls start before any of those returns the bad DIALSTATUS. Anyway, they
will block the provider A again if he continues having moments, but
will allow using A as long as that works fine.
You should take extra care to distinguish DIALSTATUS cases; a call that
could not be terminated because the number was invalid should better not
block provider A if this can be distinguished.
Anselm('s 2 cent)
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Yeah,
I think that would be a solution. I'm a little confused. It seems like
I'm the first one with such a demand? I'd expected that there is
something out of the box as asterisk has for nearly every problem
something someone already solved
When I made it I'll post it...
Knud
--
Knud A. Müller
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RE: [asterisk-users] Feedback on Linksys SPA-921 and GrandStreamGXP-2000
Feedback on the GXP2000 - we have around 10 of them: 1) Great if the firmware's recent (but not too recent - see GS info over at http://www.voip-info.org/wiki/view/GXP-2000) 2) Good caller ID 3) Speakerphone OK 4) Good features - Asterisk friendly and they support paging/announcements 5) BLF works fairly well but has the occasional hiccup 6) Power plug/sockets are a loose fit and moving a phone will often 'glitch' it so it reboots - this is the biggest PITA we have found - go with PoE where possible 7) LCD backlight LEDs (white) fade within a month or so if they are left on permanently, which can make the display hard to read in some conditions. Aiming to take a look at how easy these are to replace. 8) We have 4 phones connected back to base via 512K ADSL and NAT/STUN works well, plus the phones do not tend to disconnect randomly and fail to re-register (like our test Safecom phones) Overall, the GXP-2000 seems to be good for the money. It's our phone of choice for the spec/price. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk svn and zaptel
On Wed, 2007-04-18 at 14:58 +, Iban Lopetegi Zinkunegi wrote: I go to my asterisk recompile it but I realize there is no chan_zap.so! When I recompile it, i check the make menuselect and the channel zapata is not appearing there. Does any body know any patch for that? Or how to sort out this problem You need to rerun the configure script for asterisk *after* you have the zaptel drivers installed. If configure doesn't detect the zaptel drivers (i.e. they weren't installed when it was run), then it won't build the chan_zap module. I ran into this too. What you have to do is make distclean, then rerun configure and recompile asterisk, now that you have the zaptel drivers installed. --Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk svn and zaptel
do you have also compiled latest svn-trunk zaptel? Iban Lopetegi Zinkunegi wrote: Hi all!! I have downloaded the asterisk from svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk-trunk (is the asterisk 1.4 subversion). I also downloaded the patch for cellphone and make it work fine. Then I bought the tdm11b board to have phone connection in my computer. I installed the hardware for zapte and the libpri modules in my Mandriva 2007 and the lights of the pci card switch on. I can see zaptel working by lsmod. Now I go to my asterisk recompile it but I realize there is no chan_zap.so! When I recompile it, i check the make menuselect and the channel zapata is not appearing there. Does any body know any patch for that? Or how to sort out this problem Thanks a lot iban _ ¿Estás pensando en cambiar de coche? Todas los modelos de serie y extras en MSN Motor. http://motor.msn.es/researchcentre/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue App - Free agent and waiting calls
Hi, sometimes I have only two agents, but most of time I have four or five. On 4/18/07, Steve Totaro [EMAIL PROTECTED] wrote: Try ringall or roundrobbin. You only have two agents. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *equis software *Sent:* Wednesday, April 18, 2007 9:21 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Queue App - Free agent and waiting calls Asterisk 1.4 I have strategy= leastrecent and autofill = yes I have 2 agents, one is answering a call and the other is free and have some calls waiting in the queue. Only when the first agent hangup the second agent receive the first call in the queue. It happends some times. This behavior still happend in 1.4.1 version. Thanks a lot. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DISABLE 9?
I preffer not dialing 9 and have set up my server like this. One thing that does puzzle me is would it be possible to dial +441232345634 I come accross this problem as all my cell phone contacts are preffixed + I then sync these contacts with my laptop and sometimes cut / past the number into a softphone. Aother time I come accross this problem is if I use callback to be mobile phone, then I send DTMF to dial a number in my phone's memory the + makes it fail... - Original Message - From: Remco Post [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, April 15, 2007 10:15 AM Subject: Re: [asterisk-users] DISABLE 9? JNA wrote: Is there a way to make it so you do not have to dial 9 by default to dial a outside number? I would like it if we could just dial the number any pointers? the asterisk dialplan matches most specific entries first. So you could have one set for one or two ditgit internal numbers, one set for 7 digit local numbers, one set for 10 digit national numbers and one set for n digit international numbers all starting with an international prefix. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP failover between Sip Providers
On Wed, 18 Apr 2007, Knud Müller said something to this effect: Hi all, lets say I've registered at several Sip-Providers. Provider A offers best rates but is often too busy to get a line. Sip Provider B is stable (but more expensive). The asterisk box has a high call volume therefore problems at provider A will get obvious after a few calls stalled. In this case astersik shall switch temporarily to provider B but shall test periodically for selected calls if provider A is available again. I think it can be done by using the dialplan and the database to store the statistical information but maybe there is an easier way that integrates better with asterisk!? Best way to do this in my opinion is to deputise this logic to a SIP proxy and have Asterisk trunk all of its calls through that. -- Alex Balashov [EMAIL PROTECTED]___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk svn and zaptel
1)i downloaded the zaptel drivers from svn checkout http://svn.digium.com/svn/zaptel/trunk. 2) I did make distclean, ./configure while my zaptel is already running. However now i check in make menuselect and still can not see the zaptel module. Any other idea? Thanks iban From: Pavel Jezek [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk svn and zaptel Date: Wed, 18 Apr 2007 17:12:13 +0200 do you have also compiled latest svn-trunk zaptel? Iban Lopetegi Zinkunegi wrote: Hi all!! I have downloaded the asterisk from svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk-trunk (is the asterisk 1.4 subversion). I also downloaded the patch for cellphone and make it work fine. Then I bought the tdm11b board to have phone connection in my computer. I installed the hardware for zapte and the libpri modules in my Mandriva 2007 and the lights of the pci card switch on. I can see zaptel working by lsmod. Now I go to my asterisk recompile it but I realize there is no chan_zap.so! When I recompile it, i check the make menuselect and the channel zapata is not appearing there. Does any body know any patch for that? Or how to sort out this problem Thanks a lot iban _ ¿Estás pensando en cambiar de coche? Todas los modelos de serie y extras en MSN Motor. http://motor.msn.es/researchcentre/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Descarga gratis la Barra de Herramientas de MSN http://www.msn.es/usuario/busqueda/barra?XAPID=2031DI=1055SU=http%3A//www.hotmail.comHL=LINKTAG1OPENINGTEXT_MSNBH ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DISABLE 9?
Wireless wrote:
I preffer not dialing 9 and have set up my server like this. One thing that
does puzzle me is would it be possible to dial +441232345634 I come accross
this problem as all my cell phone contacts are preffixed + I then sync these
contacts with my laptop and sometimes cut / past the number into a
softphone. Aother time I come accross this problem is if I use callback to
be mobile phone, then I send DTMF to dial a number in my phone's memory the
+ makes it fail...
Assuming you need to dial 00 instead of + then:
exten = _+X.,1,Goto(00${EXTEN:1},1)
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RE: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Ondrej wrote: What version of Asterisk are you using? I've had recording working with SVN before 1.4, the 1.4 betas and currently 1.4.1. *** Update *** Recordings are tied to a moderator joining the conference at this time. I may need to change that based on feedback/requests to do so. *** Update *** Please include a note in the documentation for that (and maybe even note that in the web page for configuring conferences) !! It is really needed. Also please update the web page of each (past) conference with the link from where the recording could be downloaded The links to download a recording are already on the past conference page IF the conference was recorded. I will try to make time to update the README and installation How-To on SF. I also plan to add mouse-over help text to the UI, but I do not know when I will get to it (real work takes priority) I've never user the sql option for the user/participant. It was contributed by another user of the suite. Depending on the technology the caller used to call into the conference you should have their Caller-id number and possibly their Caller-id name. What additional Information would you like to see? Also - this is probably again a problem of the missing documentation, but let me clarify my problem in detail: If I create a conference, there is a button email participants. If I click that button, nothing happens (). How does the whole email procedure works? How does the web-meetme gather the email addresses of the participants? There is no way how to configure participants to the conference. My understanding was, that participants are informed about conference start/end/extended by this procedure. But since there is no way how the application could find their email addresses, I just do not know how it should work. OK, I get it now. This is a side effect of offering too much flexibility. I use and prefer the client-side mailer, and my users simply get an new message draft in their email client that they can add the participants to. If you use the server-side mailer, then there is currently no way to add participants to the notice other than to email the details to yourself and forward them. I'd happily integrate an AD address book function, but it is Not a feature I or my users would use, so I cannot dedicate too much time to writing it myself. From the sources I see that it uses SQL database users - but since I use AD, my users database is empty Contributions welcome. There is a new How-To up on SF that covers the installation on a step by step basis. I've tried to comment the configuration files to make it clear how each setting works. Some features have been contributed to the project, and I am sorry to say that beyond making sure they integrate cleanly, I have not taken enough time to document their setup and use. I guess I should ask for supporting documentation before merging the changes/features. I agree, because without any documentation is the feature de-facto unusable. I am happy to contribute to the project but at this stage it is (due to the bugs mentioned above) for me unfortunately still quite far from being promising. Lot of work has been done, but here are still some important pieces to be done. I'm sorry to hear that. I know it has some rough edges, but many people are using it. Some feature combinations work better/are better documented than others. If you are interested in following the development progress, I recommend monitoring the forums for the project on SF. I also hope I am not sounding if I do not care about the changes Or suggestions you are making. I agree will most if not all of them, but I need to focus on the problems that impact my users first and if anytime is left I can work on features that will not be used by them, but that others will enjoy. Thank you for the feedback. I am surprised almost daily how many people have found it useful. I did not really expect it to be as popular as it has become, and I am more than happy to try and address any problems. Glad to hear that :-) Ondrej Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming SIP call
Hello all, I'm having a quite simple configuration like: SIP provider = asterisk SIP = lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch=z9hG4bK67c2df66;rport From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch=z9hG4bK253c1a3d;rport From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net;tag=00-31057-001dc208-591e1ca81 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df66 Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net;tag=00-32700-001dc209-6fc2b3303 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' sip.conf [general] context=incoming realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all allow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register = 09:[EMAIL PROTECTED] registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=6 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXX username=09XXX dtmfmode=inband qualify=6 fromdomain=freephonie.net [freephonie_inbound] type=peer context=incoming host=freephonie.net qualify=6 allow=all deny=0.0.0.0/0.0.0.0 permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net etension.conf ... [incoming] exten = s,1,Ringing exten = s,2,Noop(I receive a sip call); exten = s,n,Goto(home,1000,1) exten = s,n,Congestion ; ... !DSPAM:462643f450705772331342! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping
Kevin P. Fleming wrote: Eric ManxPower Wieling wrote: I'll be sending Digium support the info they requested later today. I hope it helps. We have a developer working on extending Zaptel to support pre-echo audio capture right now, so that we can work on debugging these issues with real data instead of just conjectures :-) Stay tuned, a patch should be available for testing in the very near future. Any updates on this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Also - this is probably again a problem of the missing documentation, but let me clarify my problem in detail: If I create a conference, there is a button email participants. If I click that button, nothing happens (). How does the whole email procedure works? How does the web-meetme gather the email addresses of the participants? There is no way how to configure participants to the conference. I have seen this with my setup, I am using the client mode for emails, when using firefox. Strange enough IE works. Most of our users are on IE so I have not researched the why. -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Also - this is probably again a problem of the missing documentation, but let me clarify my problem in detail: If I create a conference, there is a button email participants. If I click that button, nothing happens (). How does the whole email procedure works? How does the web-meetme gather the email addresses of the participants? There is no way how to configure participants to the conference. I have seen this with my setup, I am using the client mode for emails, when using firefox. Strange enough IE works. Most of our users are on IE so I have not researched the why. -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk svn and zaptel
Sorry about that!!! IS WORKING!! you were right, i had to make distclean!! I was confused because i could not see zaptel channel in make menuselect, but i can not even see sip channel. I just followed normally with make and make install and is working fine for me!! Thank you Iban From: Greg Woods [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk svn and zaptel Date: Wed, 18 Apr 2007 09:12:00 -0600 On Wed, 2007-04-18 at 14:58 +, Iban Lopetegi Zinkunegi wrote: I go to my asterisk recompile it but I realize there is no chan_zap.so! When I recompile it, i check the make menuselect and the channel zapata is not appearing there. Does any body know any patch for that? Or how to sort out this problem You need to rerun the configure script for asterisk *after* you have the zaptel drivers installed. If configure doesn't detect the zaptel drivers (i.e. they weren't installed when it was run), then it won't build the chan_zap module. I ran into this too. What you have to do is make distclean, then rerun configure and recompile asterisk, now that you have the zaptel drivers installed. --Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ ¿Estás pensando en cambiar de coche? Todas los modelos de serie y extras en MSN Motor. http://motor.msn.es/researchcentre/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] openvz resources
I didn't do anything special, I just used the command to split the resources into four equal nodes, I think its called vzsplit. The only possible extra step I remember was I had to play around with the tty variable and how its used in safe_asterisk but I don't remember what I actually did or the problem was, I haven't had a need to modify the box in almost two years apart from the odd security patch and yum upgrade. --- Shidan On 4/16/07, Voip Asterisk [EMAIL PROTECTED] wrote: Awesome, any chance you can share your resource specs? Thanks Miles Asterisk works great with openvz. Ive run 4 VE's with combined average around 32 simultaneous calls at any time and you wouldn't know the difference. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.16 - No Caller ID
Giorgio, That does not work it just shows up as useincomingcalleridonzaptransfer I set the following: callerid=useincomingcalleridonzaptransfer. Are you referring to something else like useincomingcalleridonzaptransfer=yes Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp Giorgio Incantalupo wrote: Hi, have you tried different values of callerid? Maybe setting *useincomingcalleridonzaptransfer* to yes can help you. Giorgio Incantalupo OCOSA ListAcc wrote: Hello, When I upgraded a while back the caller ID stop working I have tried everything and searched the lists no answer. Please help!! I have two pots lines coming into the Asterisk Box caller ID is set in the zapta.conf Here is what our zapata.conf looks like [channels] busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=10.5 txgain=5.5 group=1 pickupgroup=1-4 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel=1 channel=2 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] internal sounds of asterisk / freePBX
CallWeaver is the new name for OpenPBX -Original Message- From: Carlos Jerónimo [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] internal sounds of asterisk / freePBX i use xlite and kphone in a diferent pc's. i can phone well. the problem is internal asterisk sounds. I think i not use Call Weaver, what is call weaver, i search at google but i'm was confused. i hope more help's. thanks 2007/4/17, Andrew Joakimsen [EMAIL PROTECTED]: If that's what your phone is setup. Are you even using a SIP phone? What does the PEER context contain? Also, while Asterisk 1.2 and CALL WEAVER are basically the same (besides that fact that CALL WEAVER is trying to fully support faxing and Asterisk/Digium refuse to support correctly faxing) they do not share sound files. So if you are indeed using CALL WEAVER and their sounds you shouldn't be asking about that here. On 4/17/07, Carlos Jerónimo [EMAIL PROTECTED] wrote: HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks 2007/4/17, EWV2 [EMAIL PROTECTED]: It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Asterisk COLP (COnnected Line Presentation)
Hi, I would just like to know if any work was ever done on COLP or its related cousins? The last evidence of it seems to be about 2 years old when K.Flemming and Olle both showed some mild interest. I am not sure how well that code would apply to today's Asterisk. (I realise that this is sort of a duplicate posting, sorry about that.) Thanks for any feedback. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping
Eric ManxPower Wieling wrote: Any updates on this? The code is done and initially tested; it is being reviewed internally and should be available on Friday or Monday. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Hello Dan, The links to download a recording are already on the past conference page IF the conference was recorded. Aha, I see, intelligent. I will give it a try. OK, I get it now. This is a side effect of offering too much flexibility. I use and prefer the client-side mailer, and my users simply get an new message draft in their email client that they can add the participants to. If you use the server-side mailer, then there is currently no way to add participants to the notice other than to email the details to yourself and forward them. I'd happily integrate an AD address book function, but it is Not a feature I or my users would use, so I cannot dedicate too much time to writing it myself. Ok, I understand that now as well - you click that button and thunderbird should popup with the mail composer open, right? Does not happen to me - most likely problem w/ my firefox settings. Now it all make a sense, sorry for being too pessimistic! One thing that does not work for sure - I had some problems to terminate the running conference from within the web page - I just clicked the button and nothing happened. Anyway - thanks a lot for the explanation - I will give it a try! Ondrej The information contained in this e-mail and in any attachments is confidential and is designated solely for the attention of the intended recipient(s). If you are not an intended recipient, you must not use, disclose, copy, distribute or retain this e-mail or any part thereof. If you have received this e-mail in error, please notify the sender by return e-mail and delete all copies of this e-mail from your computer system(s). Please direct any additional queries to: [EMAIL PROTECTED] Thank You. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial out from AGI and then connect it to another dialled out call
Hi there,
I'm converting a dialplan callback type application to fastagi as I'm
hitting the buffers with respects to getting useful results from CDRs.
It works by a spool call file triggering a Local extension, that extension
then does the first dial to a client. I dial to a local context from the
spool file as I need proper return codes as in ${DIALSTATUS} which are not
available from the spoolfile (even using the failed priority trick). They
then get some IVR prompts followed by being connected to another dialled
number.
Dialplan wise I do this with two contexts... the first being the one that
the local dial in the spoolfile calls, it does various bits of set up, sets
a few variables so they'll be inherited and does a :
exten = s,19,Dial(Zap/g1/${extnum},,G(anewextdialbridgev2^s^1))
in anewextdialbridge priority 1 (for the caller leg) I have a Goto which
just calls congestion and then hangup - ie. it waits around for the end of
the call. At priority 2 which the dial with G option will put the callee
into I start my more usual IVR type prompts before doing my final dial (no G
option this time) to the number we're connecting the user to.
What's a nice way of replicating the first Dial in AGI? At the moment I
suspect I'll have to do a bodge with it dropping back to dialplan and then
calling my AGI again? I'd rather not if at all possible.
Any advice appreciated...
--
Tony
_
Solve the Conspiracy and win fantastic prizes.
http://www.theconspiracygame.co.uk/
___
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Re: [asterisk-users] internal sounds of asterisk / freePBX
Did you have any E1/T1 cards in your server? On 4/18/07, shadowym [EMAIL PROTECTED] wrote: CallWeaver is the new name for OpenPBX -Original Message- From: Carlos Jerónimo [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] internal sounds of asterisk / freePBX i use xlite and kphone in a diferent pc's. i can phone well. the problem is internal asterisk sounds. I think i not use Call Weaver, what is call weaver, i search at google but i'm was confused. i hope more help's. thanks 2007/4/17, Andrew Joakimsen [EMAIL PROTECTED]: If that's what your phone is setup. Are you even using a SIP phone? What does the PEER context contain? Also, while Asterisk 1.2 and CALL WEAVER are basically the same (besides that fact that CALL WEAVER is trying to fully support faxing and Asterisk/Digium refuse to support correctly faxing) they do not share sound files. So if you are indeed using CALL WEAVER and their sounds you shouldn't be asking about that here. On 4/17/07, Carlos Jerónimo [EMAIL PROTECTED] wrote: HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks 2007/4/17, EWV2 [EMAIL PROTECTED]: It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com --
Re: [asterisk-users] Asterisk 1.2.16 - No Caller ID
I don't know where he got the bizarre useincomingcalleridonzaptransfer option, but it does not exist as you can see below: [EMAIL PROTECTED] ~]# grep useincomingcalleridonzaptransfer /home/software/asterisk/asterisk-1.2.17/configs/zapata.conf.sample [EMAIL PROTECTED] ~]# Maybe the option is specific to BRIstuff patches to Zaptel. You want the following before your FXO ports in /etc/asterisk/zapata.conf: usecallerid=yes callerid=asreceived You will also want to watch the console when a call comes in to see if there are any Caller*ID errors. OCOSA ListAcct wrote: Giorgio, That does not work it just shows up as useincomingcalleridonzaptransfer I set the following: callerid=useincomingcalleridonzaptransfer. Are you referring to something else like useincomingcalleridonzaptransfer=yes Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp Giorgio Incantalupo wrote: Hi, have you tried different values of callerid? Maybe setting *useincomingcalleridonzaptransfer* to yes can help you. Giorgio Incantalupo OCOSA ListAcc wrote: Hello, When I upgraded a while back the caller ID stop working I have tried everything and searched the lists no answer. Please help!! I have two pots lines coming into the Asterisk Box caller ID is set in the zapta.conf Here is what our zapata.conf looks like [channels] busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=10.5 txgain=5.5 group=1 pickupgroup=1-4 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel=1 channel=2 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.16 - No Caller ID
Eric ManxPower Wieling wrote: I don't know where he got the bizarre useincomingcalleridonzaptransfer option, but it does not exist as you can see below: *snipped just a note, not sure if it is still in 1.4 tree, but it used to be in CVS-TRUNK as an option for chan_zap ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
Gilles Ganault wrote: Hello I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921 I don't know where you live, but I've seen significant price-differences on the SPA-921 across Europe. Very pricey in the UK, less so in Germany, but absolutely rock-bottom in Switzerland at SFr124. /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] internal sounds of asterisk / freePBX
no i don't have any card. 2007/4/18, Leonardo Kamache (Gmail) [EMAIL PROTECTED]: Did you have any E1/T1 cards in your server? On 4/18/07, shadowym [EMAIL PROTECTED] wrote: CallWeaver is the new name for OpenPBX -Original Message- From: Carlos Jerónimo [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] internal sounds of asterisk / freePBX i use xlite and kphone in a diferent pc's. i can phone well. the problem is internal asterisk sounds. I think i not use Call Weaver, what is call weaver, i search at google but i'm was confused. i hope more help's. thanks 2007/4/17, Andrew Joakimsen [EMAIL PROTECTED]: If that's what your phone is setup. Are you even using a SIP phone? What does the PEER context contain? Also, while Asterisk 1.2 and CALL WEAVER are basically the same (besides that fact that CALL WEAVER is trying to fully support faxing and Asterisk/Digium refuse to support correctly faxing) they do not share sound files. So if you are indeed using CALL WEAVER and their sounds you shouldn't be asking about that here. On 4/17/07, Carlos Jerónimo [EMAIL PROTECTED] wrote: HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks 2007/4/17, EWV2 [EMAIL PROTECTED]: It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update
Re: [asterisk-users] Asterisk 1.2.16 - No Caller ID
Richard Lyman wrote: Eric ManxPower Wieling wrote: I don't know where he got the bizarre useincomingcalleridonzaptransfer option, but it does not exist as you can see below: *snipped just a note, not sure if it is still in 1.4 tree, but it used to be in CVS-TRUNK as an option for chan_zap He is, of course, running 1.2.6. If the option exists in 1.2.x then it is not documented. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.16 - No Caller ID
Eric ManxPower Wieling wrote:
Richard Lyman wrote:
Eric ManxPower Wieling wrote:
I don't know where he got the bizarre
useincomingcalleridonzaptransfer option, but it does not exist as
you can see below:
*snipped
just a note, not sure if it is still in 1.4 tree, but it used to be
in CVS-TRUNK as an option for chan_zap
He is, of course, running 1.2.6. If the option exists in 1.2.x then
it is not documented.
current 1.2 tree: } else if (!strcasecmp(v-name,
useincomingcalleridonzaptransfer)) {
current 1.4 tree: } else if (!strcasecmp(v-name,
useincomingcalleridonzaptransfer)) {
current svn-trunk: } else if (!strcasecmp(v-name,
useincomingcalleridonzaptransfer)) {
so yep, still there, still undocumented.
___
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RE: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Ondrej wrote: Ok, I understand that now as well - you click that button and thunderbird should popup with the mail composer open, right? Yes. Does not happen to me - most likely problem w/ my firefox settings. Browser security settings most likely Now it all make a sense, sorry for being too pessimistic! No worries. One thing that does not work for sure - I had some problems to terminate the running conference from within the web page - I just clicked the button and nothing happened. This is likely a manager.conf security issue, but it could be a problem in the php code. I just tested branches/3.0 and trunk against 1.4.1 and it worked as expected. If you set core verbose to 10 and click on 'End Now' the console should display a message like this: app_meetme.c:941 meetme_cmd: Cmdline: 41251|k|1 At this point this would be a topic better suited for the support forums on SF. Anyway - thanks a lot for the explanation - I will give it a try! I just committed a simple set of mouse-over text popups to provide details about the options/settings in 'Add Conference' that might not be obvious to everyone. Since I know what the fields are for, I may have over/under thought which ones need more explanation, and the text I used to explain the fields may be poor. If you'd care to check svn branches/3.0, I'd love to know what needs more work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
You could buy one of those X100P clones for ~$20 shipped and use that for timing (and also an added FXO port), or a bare TDM400P with no modules for ~$100 and have the option of adding modules for future upgrades. Thanks, Steve Bryan M. Johns wrote: Install zaptel and only enable the ztdummy module. As long as you are not running in a VM, this will supply you the timing that you are looking for. Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Arun Kumar [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Thomas Kenyon [EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 4:54:47 AM (GMT-0500) America/New_York Subject: Re: [asterisk-users] No of Calls how do I check that whether trunking is working or not ? No I don't any timing soure (like zaptel card) b'coz these are test server. what else I can use for timing. thanks On 4/17/07, *Thomas Kenyon* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Arun Kumar wrote: I've tried this but stil some problem Like if I use this link that you gave me it shows for 10 call 136.08KBps in one direction, but, when I place call using my phone for 10 calls it comes 210KBps in one direction. Ar eyou sure trunking is working? Do both asterisk servers have a timing source? ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
I've installed zaptel on FreeBSD and when I try to load ztdummy module I get this error kldload: can't load ztdummy.ko No such file or directory. and when I do ztcfg:- Notice: Configuration file is /usr/local/etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' Keyword: [loadzone], Value: [us] Keyword: [defaultzone], Value: [us] 1 error(s) detected thanks On 4/17/07, Bryan M. Johns [EMAIL PROTECTED] wrote: Install zaptel and only enable the ztdummy module. As long as you are not running in a VM, this will supply you the timing that you are looking for. Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Arun Kumar [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Thomas Kenyon [EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 4:54:47 AM (GMT-0500) America/New_York Subject: Re: [asterisk-users] No of Calls how do I check that whether trunking is working or not ? No I don't any timing soure (like zaptel card) b'coz these are test server. what else I can use for timing. thanks On 4/17/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Arun Kumar wrote: I've tried this but stil some problem Like if I use this link that you gave me it shows for 10 call 136.08KBps in one direction, but, when I place call using my phone for 10 calls it comes 210KBps in one direction. Ar eyou sure trunking is working? Do both asterisk servers have a timing source? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.16 - No Caller ID
Eric, I have watched the CLI before and it said nothing although I did change the position of the callerid = asreceived to right below and nothing it still shows up on the phones asterisk and in voice mail sent via e-mail unknown caller: Here is an output from a while back but it stopped so I do not know what is up. -- Starting simple switch on 'Zap/1-1' Apr 6 16:54:39 ERROR[4726]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-92) Apr 6 16:54:39 WARNING[4726]: chan_zap.c:6255 ss_thread: CallerID feed failed: Success Apr 6 16:54:39 WARNING[4726]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/1-1' Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp Eric ManxPower Wieling wrote: I don't know where he got the bizarre useincomingcalleridonzaptransfer option, but it does not exist as you can see below: [EMAIL PROTECTED] ~]# grep useincomingcalleridonzaptransfer /home/software/asterisk/asterisk-1.2.17/configs/zapata.conf.sample [EMAIL PROTECTED] ~]# Maybe the option is specific to BRIstuff patches to Zaptel. You want the following before your FXO ports in /etc/asterisk/zapata.conf: usecallerid=yes callerid=asreceived You will also want to watch the console when a call comes in to see if there are any Caller*ID errors. OCOSA ListAcct wrote: Giorgio, That does not work it just shows up as useincomingcalleridonzaptransfer I set the following: callerid=useincomingcalleridonzaptransfer. Are you referring to something else like useincomingcalleridonzaptransfer=yes Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp Giorgio Incantalupo wrote: Hi, have you tried different values of callerid? Maybe setting *useincomingcalleridonzaptransfer* to yes can help you. Giorgio Incantalupo OCOSA ListAcc wrote: Hello, When I upgraded a while back the caller ID stop working I have tried everything and searched the lists no answer. Please help!! I have two pots lines coming into the Asterisk Box caller ID is set in the zapta.conf Here is what our zapata.conf looks like [channels] busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=10.5 txgain=5.5 group=1 pickupgroup=1-4 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel=1 channel=2 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] gxp2000 expansion module blf leds not working
Today a 56-button expansion module for the GXP2000 came in. When I program the buttons+leds on the expansion module for BLF, then speed-dial works fine: when I press the button the programmed ext number is called properly. However the LEDs are always off: neither green nor red They are not broken, because on reboot the LEDs flash red! On the GXP2000 itself, this function works fine, with LEDs being green when the ext is free, or red whenever it is busy. Does anybody know this problem? Or can anyone confirm that the LEDs on the GXP2000 expansion module should be working properly? Thanks, Z. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] peers are using wrong contexts
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of dima Sent: Tuesday, April 17, 2007 10:39 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] peers are using wrong contexts Hello, everyone. Today I've installed an asterisk svn trunk (r61667). The problem I'm having is no matter what context I set in the config file for that peer, default is always being used. The output of sip show peers shows the context correctly, but when I try to make a call, using that peer, I can only dial the numbers set in the default context. Please, could anyone help me resolve this. Thanks in advance. This is a part of users.conf [951XX] callwaiting = yes cid_number = 951XX context = numberplan-custom-1 Do you have the context numberplan-custom-1 in your extensions.conf file? I think if you don't have it in extensions.conf then it goes back to using default. email = fullname = New User group = hasagent = no hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 951XX secret = 00 threewaycalling = yes vmsecret = 1234 zapchan = registeriax = no registersip = yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.16 - No Caller ID
OCOSA ListAcc wrote: Eric, I have watched the CLI before and it said nothing although I did change the position of the callerid = asreceived to right below and nothing it still shows up on the phones asterisk and in voice mail sent via e-mail unknown caller: Here is an output from a while back but it stopped so I do not know what is up. -- Starting simple switch on 'Zap/1-1' Apr 6 16:54:39 ERROR[4726]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-92) Apr 6 16:54:39 WARNING[4726]: chan_zap.c:6255 ss_thread: CallerID feed failed: Success Apr 6 16:54:39 WARNING[4726]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/1-1' These errors usually indicate that your rxgain for the FXO ports is either too high or too low. Change the rxgain in /etc/asterisk/zapata.conf in increments of 2 either up or down until, but you generally don't want it to be less than -10 or greater than 10. reload chan_zap.so should apply the gain changes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
On 4/18/07, Gilles Ganault [EMAIL PROTECTED] wrote: Hello I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921 I'd like to have some user feedback about how those phones perform, and whether their LCD screen displays both the caller ID name and number (The GrandStream BT-100 only displays numbers, which isn't very helpful). My main complaint about both phones is there is no way to reject a call once the phone starts to ring. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
On the GXP-2000 press the Mute/DEL button while the phone is ringing, and it will return 486 (Busy). This works to bounce new incoming calls while already in a call as well (call waiting). - Anthony Kepler Andrew Joakimsen wrote: My main complaint about both phones is there is no way to reject a call once the phone starts to ring. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Segmentation Fault
Hello..I own a server running Slackware 10.2 with kernel 6.1.13 and I tried unsuccessfully to install recently Asterisk 1.4.0. I install all packages but when I execute the command asterisk -vc in order to start asterisk, I get a message Segmentation Fault and the debugging stops suddenly. Does anyone can help me?Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: OT (a little): IPV6 Ramifications Article
Hi guys, I know it's a little off topic but..Wondering if you can help. My wife has been asked to find a writer to produce a story on The dramatic ramifications of IPV6 on commercial businesses and how it will change the product designs for ordinary household/commercial use in a 5-10 year time frame So her company hired someone who should have been able to deliver the goods (ex magazine editor - maybe a little too 'ex') He has come back with the story angle that is boring (and just plain wrong) that says; - IPV6 is a big cost to companies like the Y2k bug was. - That it will stop spam (hmmm Cringley you have a lot to answer for) - That Asia is leading the way but we can ignore it as the USA have many many IPV4 addresses to use for the future. So now my wife has egg on her face and her boss thinks that IPV6 is of no interest to anyone in their customers companies, apart from the CIO who needs to implement it, when I'm telling her that there are dramatic applications; eg. - That Ford needs to consider how your car having an IP addresses changes the way they should be building cars (oh and the streetlights have one as well). - That Sharp needs to consider what your TV having an IP address means (and your set top box and your front door bell as well) - That Verizon needs to consider what every mobile phone having an IP address means (and your desk phone and your office phone) - That Chase needs to consider what IPV6 means to your wallet, the ATM and the POS cash registers. Can anyone help with some url's for some really good articles on 'super networking' and related applications that dramatically change the products that companies should be manufacturing today? (also the less technical the better) Or if you are a writer who has published something on this exact topic that has been run at a national print level..want a gig? Regards, Dean Collins ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.16 - No Caller ID
so to fix the no caller id thing will need to adjust the rx gain and tx gain? Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp Eric ManxPower Wieling wrote: OCOSA ListAcc wrote: Eric, I have watched the CLI before and it said nothing although I did change the position of the callerid = asreceived to right below and nothing it still shows up on the phones asterisk and in voice mail sent via e-mail unknown caller: Here is an output from a while back but it stopped so I do not know what is up. -- Starting simple switch on 'Zap/1-1' Apr 6 16:54:39 ERROR[4726]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-92) Apr 6 16:54:39 WARNING[4726]: chan_zap.c:6255 ss_thread: CallerID feed failed: Success Apr 6 16:54:39 WARNING[4726]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/1-1' These errors usually indicate that your rxgain for the FXO ports is either too high or too low. Change the rxgain in /etc/asterisk/zapata.conf in increments of 2 either up or down until, but you generally don't want it to be less than -10 or greater than 10. reload chan_zap.so should apply the gain changes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audio playback problems with FC6 and Zaptel 1.2.16
I'm chasing down some issues at a call center. Today I received a complaint that audio file playback ceased after they upgraded the system from FC4 to FC6, Asterisk 1.2.14 to 1.2.17. Zaptel is at 1.2.16. The system in question takes inbound calls via IAX2 and has a TE410P with a couple of channel banks connected to it for analog extensions. I ultimately found that the problem goes away if I load ztdummy alone or prior to wct4xxp. I realize ztdummy should not be used when there's real hardware available, but it appears to solve/mask the problem at least for troubleshooting. No errors or clues in the logs, dmesg, etc. I even tried transcoding the gsm audio files in ulaw with no luck. As an aside, I noticed that zttranscode loads itself when Asterisk is started. I haven't found anything in Mantis, Google, etc. Before I file a bug report, I wanted to see if anyone else has seen this weirdness. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
On Wed, 18 Apr 2007, Anthony Kepler wrote: On the GXP-2000 press the Mute/DEL button while the phone is ringing, and it will return 486 (Busy). This works to bounce new incoming calls while already in a call as well (call waiting). And push it when the phone isn't ringing and it set Do Not Disturb mode... Gordon - Anthony Kepler Andrew Joakimsen wrote: My main complaint about both phones is there is no way to reject a call once the phone starts to ring. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.16 - No Caller ID
Eric, Thanks when I took the rx and tx to 0.0 on both the caller id showed up I guess I will play with. My main reasoning for adjusting the rx and tx was to get rid of the echo...What other tips do you suggest or anyone out there? Thank you! Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp Eric ManxPower Wieling wrote: OCOSA ListAcc wrote: Eric, I have watched the CLI before and it said nothing although I did change the position of the callerid = asreceived to right below and nothing it still shows up on the phones asterisk and in voice mail sent via e-mail unknown caller: Here is an output from a while back but it stopped so I do not know what is up. -- Starting simple switch on 'Zap/1-1' Apr 6 16:54:39 ERROR[4726]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-92) Apr 6 16:54:39 WARNING[4726]: chan_zap.c:6255 ss_thread: CallerID feed failed: Success Apr 6 16:54:39 WARNING[4726]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/1-1' These errors usually indicate that your rxgain for the FXO ports is either too high or too low. Change the rxgain in /etc/asterisk/zapata.conf in increments of 2 either up or down until, but you generally don't want it to be less than -10 or greater than 10. reload chan_zap.so should apply the gain changes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.2 + Cisco 7960G not registering
Hi all, I've recently upgraded to Asterisk 1.4.2, coming from 1.2.14. Using my existing Cisco 7960G handset(s). I've tried multiple installs of asterisk 1.4.2 with multiple handsets and SIP will not authorize my phone. I'll include some verbose log messages below to show a VALID registration and one where I'm having difficulty registering the phone. Thanks to anyone who can lend a helping hand with this matter or offer any insight on how to further debug. I've gone as far as packet capture and cannot understand why using the same configs will not allow registration of these handsets. - sf -- Working excerpt: REGISTER sip:10.2.7.2 SIP/2.0 Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK1029a1dd From: sip:[EMAIL PROTECTED];tag=0017e0134094000d045a4ac2-6d217bab To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 104 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;+sip.instance=urn:uuid:----0017e0134094;+u.sip!model.ccm.cisco.com=7 Content-Length: 0 Expires: 120 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK1029a1dd;received=10.2.7.254 From: sip:[EMAIL PROTECTED];tag=0017e0134094000d045a4ac2-6d217bab To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK1029a1dd;received=10.2.7.254 From: sip:[EMAIL PROTECTED];tag=0017e0134094000d045a4ac2-6d217bab To: sip:[EMAIL PROTECTED];tag=as010f0581 Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=3f28f962 Content-Length: 0 REGISTER sip:10.2.7.2 SIP/2.0 Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK2596e8d7 From: sip:[EMAIL PROTECTED];tag=0017e0134094000e1fcda391-7d3d3796 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 104 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;+sip.instance=urn:uuid:----0017e0134094;+u.sip!model.ccm.cisco.com=7 Content-Length: 0 Expires: 120 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK2596e8d7;received=10.2.7.254 From: sip:[EMAIL PROTECTED];tag=0017e0134094000e1fcda391-7d3d3796 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK2596e8d7;received=10.2.7.254 From: sip:[EMAIL PROTECTED];tag=0017e0134094000e1fcda391-7d3d3796 To: sip:[EMAIL PROTECTED];tag=as0a5554ff Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5f5d830d Content-Length: 0 REGISTER sip:10.2.7.2 SIP/2.0 Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK5153628e From: sip:[EMAIL PROTECTED];tag=0017e0134094000d045a4ac2-6d217bab To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 105 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;+sip.instance=urn:uuid:----0017e0134094;+u.sip!model.ccm.cisco.com=7 Authorization: Digest username=6096,realm=asterisk,uri=sip:10.2.7.2,response=6bec57e7aaedd046469fab89b39c024a,nonce=3f28f962,algorithm=MD5 Content-Length: 0 Expires: 120 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK5153628e;received=10.2.7.254 From: sip:[EMAIL PROTECTED];tag=0017e0134094000d045a4ac2-6d217bab To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 105 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK5153628e;received=10.2.7.254 From: sip:[EMAIL PROTECTED];tag=0017e0134094000d045a4ac2-6d217bab
[asterisk-users] MeetMe Error
Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [EMAIL PROTECTED]:1] MeetMe(SIP/600-09111e58, 700|MI) in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]: app_meetme.c:755 build_conf: Unable to open pseudo channel - trying device WARNING[20055]: app_meetme.c:758 build_conf: Unable to open pseudo device SIP/600-09111e58 Playing 'conf-invalid' (language 'es') Spawn extension (numberplan-custom-1, 700, 1) exited non-zero on 'SIP/600-09111e58' i dont have any zap interface. how to solve this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Error
Manolet Gmail wrote: Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [EMAIL PROTECTED]:1] MeetMe(SIP/600-09111e58, 700|MI) in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]: app_meetme.c:755 build_conf: Unable to open pseudo channel - trying device WARNING[20055]: app_meetme.c:758 build_conf: Unable to open pseudo device SIP/600-09111e58 Playing 'conf-invalid' (language 'es') Spawn extension (numberplan-custom-1, 700, 1) exited non-zero on 'SIP/600-09111e58' i dont have any zap interface. how to solve this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Compile and install ztdummy from zaptel package, I think that will fix your issue ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IM
hi, i donwload XLITE and see there is a fuction to send Instant Messages. when i try to use it i get this error: Error: Method Not Allowed. there is anyway to enable IM on asterisk 1.4.2? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Error
2007/4/18, Rodrigo Gonzalez [EMAIL PROTECTED]: Manolet Gmail wrote: Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [EMAIL PROTECTED]:1] MeetMe(SIP/600-09111e58, 700|MI) in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]: app_meetme.c:755 build_conf: Unable to open pseudo channel - trying device WARNING[20055]: app_meetme.c:758 build_conf: Unable to open pseudo device SIP/600-09111e58 Playing 'conf-invalid' (language 'es') Spawn extension (numberplan-custom-1, 700, 1) exited non-zero on 'SIP/600-09111e58' i dont have any zap interface. how to solve this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Compile and install ztdummy from zaptel package, I think that will fix your issue 1) dont cares if i dont have any zap device?, 2) how to check if i have ztdummy installed? thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitor application inestability and high load
Hi, I'm having high load, choppy sound and slow responsives with an asterisk server (version 1.2.12.1) that make a peak of 90 channels (around 60 phones calling at max, isn't necessary to reach this peak to get the problem). All the traffic is SIP, with recording for every call. The server has: Intel(R) Xeon(TM) CPU 3.20GHz (with HyperThreading disabled for inestability) 4G RAM 2 DD SCSI 150GB in RAID I via hardware. The problems are detected in the high count of asterisk processes and sh wrappers to soxmix which could be as old as 1 hour in the server without a reason to stay idle, but for some unknow reason this sh don't die fast. This is when the dialplan calls Monitor obviously. I already tried to switch to MixMonitor but yesterday users reported that in some calls the recording isn't complete. Which is similar to a bug that is mentioned in mantis but for versions prior to 1.2.7. The asterisk logs don't show any particular message in verbose level 3. Apart from the recording, I have a high use of Manager and the mysql is used for some bussines logic but I think that nothing to high load, indeed mysql never is the most important part in processor, memmory and disk access statistics. In http://linuxuanl.org/eald/random/ps.txt there is an example (no very espectacular but is more or less what happens) of the status of the computer with problems. I see that there are many sh without their soxmix or rm; Usually this is done faster indeed I changed soxmix to a script that only copy the files in an attempt to low the load of the server. Any knows a solution to this problem? or has an explanation for it? Thanks, Edgar Luna ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] OMG Verizon is terrible
Had an appointment for these schmoes to come out and install another line. Was supposed to be 8-12. Its now 6PM and not even call. Missed 3 sales calls waiting on these jerks. No wonder customers were jumping ship to Vonage. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Error
Hi Manolet, You have to install zaptel in order to make MeetMe application to work. MeetMe needs a kind of timer device that is provided by zaptel package. Eventhough you don't have a zaptel card you need to install its package. Search for MeetMe application in http://www.voip-info.org/ and you will find documentation about how to do that. Good Luck. Ronaldo Manolet Gmail wrote: Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [EMAIL PROTECTED]:1] MeetMe(SIP/600-09111e58, 700|MI) in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]: app_meetme.c:755 build_conf: Unable to open pseudo channel - trying device WARNING[20055]: app_meetme.c:758 build_conf: Unable to open pseudo device SIP/600-09111e58 Playing 'conf-invalid' (language 'es') Spawn extension (numberplan-custom-1, 700, 1) exited non-zero on 'SIP/600-09111e58' i dont have any zap interface. how to solve this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] sending an SMS via Asterisk?
From: Per Jessen [EMAIL PROTECTED] Date: Wed, 18 Apr 2007 14:48:45 +0200 Per Jessen wrote: Per Jessen wrote: OK, part of the confusion is now clearing up. But I'm not getting much further. When I try to send an SMS, I see the call going through, but no SMS is ever sent. This is a bit of what I see in the debug output: (this is sending a longer message, protocol 2): P[ 2] -- caps:Speech pi:0 keypad: sending_complete:0 P[ 2] -- None -- mISDN/3-u54 answered Local/[EMAIL PROTECTED],2 Channel Local/[EMAIL PROTECTED],1 was answered. Launching SMS(062210|t) on Local/[EMAIL PROTECTED],1 P[ 2] * IND: Got Fixup State:CONNECTED L3id:50012 == Spawn extension (Internal, 062210, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' P[ 2] I IND :FACILITY oad:0434439000 dad:062210 pid:19 state:CONNECTED P[ 2] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:0062210 P[ 2] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 2] -- caps:Speech pi:0 keypad: sending_complete:0 P[ 2] -- AOCD currency: currency:FR. amount:10 multiplier:1 typeOfChargingInfo:-1220842403 P[ 2] I IND :INFORMATION oad:0434439000 dad:062210 pid:19 state:CONNECTED P[ 2] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:0062210 P[ 2] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 2] -- caps:Speech pi:0 keypad: sending_complete:0 P[ 2] -- None -- SMS[-1] RX 93 00 6D -- SMS[0] TX 10 98 96 00 10 01 00 00 11 06 00 00 00 00 00 00 00 12 03 00 02 00 04 13 65 00 53 65 63 75 72 69 74 79 20 72 65 73 65 61 72 63 68 65 72 73 20 68 61 76 65 20 74 72 61 63 65 64 20 73 70 61 6D 2D 73 65 6E 64 69 6E 67 20 62 6F 74 6E 65 74 20 63 6C 69 65 6E 74 73 20 62 61 63 6B 20 74 6F 20 6E 65 74 77 6F 72 6B 73 20 72 75 6E 20 62 79 20 74 68 65 20 55 53 20 6D 69 6C 69 74 61 72 79 2E 17 01 00 01 18 0A 00 30 34 33 34 34 33 39 30 30 30 1B 01 00 01 1C 03 00 00 00 00 E8 P[ 2] I IND :DISCONNECT oad:0434439000 dad:062210 pid:19 state:CONNECTED P[ 2] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:0062210 P[ 2] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 2] -- caps:Speech pi:8 keypad: sending_complete:0 P[ 2] -- org:1 nt:0, inbandavail:1 state:10 P[ 2] -- queue_hangup In all the other examples I've come across on the 'net, there are multil lines beginning SMS[x] RX/TX .. The operator seems to hang up on you. Good thing is, the operator is at least responding to your call and sending you that initial answer. This may sound bizarre but try the s option and operate in mttx mode. I vaguely remember seeing a comment about one operator does some role reversal. (May not be due to protocol 2.) If you have an extra channel to spare with (seems you do), can also try to set up a context to receive SMS so you know all your commands/dial plan are working before testing against operator. (I always test via SIP channel to simplify my debugging. You can do so, too.) Yuan Liu /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Timestamp in recorded calls filename
Hi, I need to add the timestamp to the recorded call filename, I use this
variable ${TIMESTAMP} in the Monitor() function, but when I look for this
call, the TIMESTAMP is missing in the filename.
I try to export this as a environment variable but nothing changes.
Any help is welcome, thanks.
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