Re: [asterisk-users] Choose IAX or SIP trunking?
On Tue, 2009-10-06 at 22:11 -0700, Kirill 'Big K' Katsnelson wrote: On 091001 0406, Mindaugas Kezys wrote: We had many problems with IAX2, changing to SIP solved them all. Let me paste link to wise-words which clearly illustrates our experience: http://wiki.kolmisoft.com/index.php/Why_we_do_not_suggest_to_use_IAX2 Thanks for the wise-words. From the responses I got, SIP seems a much better option. -kkm Obviously, it is an outdated document that should be revised. It stated: Now sip supports more than asterisk sip supports. SIP RFC requires tcp support for example, yet its not in trunk yet. This feature is already in the 1.6-branch, production. You just need to enable it in the config. I used it in connecting to Microsoft-stuff, quite some months ago. So the url seems no more than FUD. It should be updated or removed. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to answer to an incoming call with alsa (or OSS).
Hi, Dial(ALSA/hw:0,0); - false syntax Dial(ALSA/hw:0,0); - WARNING[1292]: channel.c:3914 ast_request: No channel type registered for 'ALSA' WARNING[1292]: app_dial.c:1528 dial_exec_full: Unable to create channel of type 'ALSA' (cause 66 - Channel not implemented) Do you have another way to answer a call with ALSA (or maybe OSS) Thank you Fabien -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Philipp Kempgen Envoyé : mardi 6 octobre 2009 17:55 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] How to answer to an incoming call with alsa. Fabien Comte schrieb: I try to use asterisk as softphone with alsa. I search how to answer to an incoming sip call (from wan). Does anyone did it (extensions.conf exemple) ? Maybe something like Dial(ALSA/hw:0,0); (untested) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MPG123 Dying
- Trevor Peirce tpei...@digitalcon.ca wrote: | --[ UxBoD ]-- wrote: | Please how do I stop the following ??? | | Asterisk ended with exit status 127 | Asterisk died with code 127. | Automatically restarting Asterisk. | mpg123: no process killed | | | You figure out why asterisk is crashing. :) | | This has nothing to do with mpg123, which is just an innocent | bystander. | Thanks for all the replies ... Will go over the logs this evening with a fine tooth comb. It is a basic install, nothing fancy, though I wonder if it is down to the hardware I am using; Intel Atom Mini-ITX. Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is anyone doing real time updates to where asterisk registers?
Hi Could you possibly have your providers set up in your sip.conf then handle your outgoing calls via an AGI where you check the number of channels in use (core show channels) and then route the call through one of the providers depending on how many channels are in use? Ish Eric Chamberlain wrote: Hello, We need Asterisk to register with a variable and changing number (hundreds) of VoIP providers, is there a way to do this in a database and without reloading the entire sip config? Where Asterisk needs to register is determined by downstream users, so we need to do it real time and with minimal impact on the server. If Asterisk can't do this, is anyone using anything else to handle the registrations for Asterisk? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Networking Concept
B.Masoud @ SH wrote: China too wide, but regardless! How is asterisk take care such situation? -regardless- I think I pointed it out already, please do some research, simply googling for asterisk + reinvite leads you here: http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drop calls when using Flash Operator Panel
hin lee wrote: Doug, I have tested both ends and got the same results. I was able to using FOP to drag to Conferences, just not the Parking Lot. Another strange thing I found is this: I'm not familiar with how Freepbx handles things, and I've personally never used FOP for parking or on hold, but I think there are instructions on how to set this up within the FOP documentation. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Integration with RDP Property Management Software?
What you are looking for is probably a HOBIC (Hotel Billing Information Center) interface for asterisk, there was a similar thread in this list five years ago, I came across this some time ago and I did not find a working implementation. The links are dead, but you may found it of interest: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg39090.html PS: Please use your mail client New/Compose button when creating a new thread, do no reply an existing different one. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] system cmd + fax line
Hi I have a site that has asterisk install with a tdm410 one port is connected to a pstn that is used as a backup outbound line when/if the internet/voip is unavailable. Currently my dial plan for this line is to ignore it, I just basically do a s,1,noop s,n,wait (60) s,n,hangup what I would like to do is if I call from a certain number be given an option to run some commands or for the beginning just run a simple command via system something like s/041100,1,goto(specialcontext,s,1) s,1,noop s,n,wait(60) s,n,hangup [specialcontext] s,1,system(do a linux command) is that about right. Alex signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
Thanks, What if I want to group a TDM2400 into 3 groups, r0/0 to r0/7 , r1/8 to r1/15 , r2/16 to r2/23 How to do that? Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Monday, October 05, 2009 10:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm outgoing B.Masoud @ SH schrieb: I have defined the card g0 to have 24 channels, but every time I try to call, if all ports are off the call always go to the first port, how can I balance the calls over all ports??? http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels#DialingaGroup Dial(Dahdi/r0/...) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
On Wed, Oct 07, 2009 at 01:24:01PM +0300, B.Masoud @ SH wrote: Thanks, What if I want to group a TDM2400 into 3 groups, r0/0 to r0/7 , r1/8 to r1/15 , r2/16 to r2/23 How to do that? DAHDI channel numbers are 1-based and not 0-based. HEnce you should probably use 1-8, 9-16 and 17-24 group = 0 ... channel = 1-8 ... group = 1 ... channel = 9-16 ... group = 2 ... channel = 17-24 Alternatively, for Asterisk = 1.6.1: [channells] ; Global settings go here. echocancel = yes ; The name is arbitrary and ignored. Just don't use a reserved one [firsts] group = 0 dahdichan = 1-8 [seconds] group = 1 dahdichan = 9-16 [thirds] group = 2 dahdichan = 17-24 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.6 suddently restarts ...
2009/10/7 Leif Madsen leif.mad...@asteriskdocs.org Please test some of the newly released RCs, such as 1.6.1.7-rc2 or 1.6.2.0-rc3 (depending on the branch you're using). Leif. Hi Leif, We've just followed your advice and installed 1.6.1.7-rc2. Before this, we had lines (3 times in 4 hours of production) like : [Oct 7 14:42:43] VERBOSE[5599] app_dial.c: -- SIP/7246-0a471ac0 answered SIP/7200-0a2e6c58 [Oct 7 14:42:43] VERBOSE[4360] chan_sip.c: == Extension Changed 7246[subs] new state InUse for Notify User 7200 [Oct 7 14:42:46] VERBOSE[4371] res_musiconhold.c: -- Stopped music on hold on SIP/patton-0a2be250 [Oct 7 14:44:31] VERBOSE[5635] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log Each time, the last visible line before restart was res_musiconhold.c: -- Stopped music on hold. I looked at 1.6.1.7-rc2's changelog and saw : 2009-10-06 Leif Madsen lmad...@digium.com * Release Asterisk 1.6.1.7-rc2 2009-10-06 01:36 + [r222112-222186] Kevin P. Fleming kpflem...@digium.com * apps/app_queue.c, channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c, /, channels/chan_sip.c, funcs/func_dialgroup.c, include/asterisk/astobj2.h, res/res_phoneprov.c, channels/chan_console.c, res/res_musiconhold.c: Merged revisions 222176 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ As this rc2 version was published just after this latest bugfix, shall I suppose this bug is considered serious and the best suspect at hand for the issues we're facing here and did you mention this upgrade as general practice ? Regards Olivier wrote: Hi, In dev-list, some people reported Asterisk 1.6.2-rc2 would suddenly restart. Here, a platform running 1.6.1.6 is also suddenly restarting (once or twice a day with moderate load (40 users)). I don't have much details to report here at the moment. Has someone met something similar ? Thoughts ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] system cmd + fax line
I would do something like this: s/041100,1,answer() s,1,noop s/041100,2,background(message) s/041100,3,wait(10,m) s/041100,4,playback(vm-goodbye) s/041100,5,hangup 1/041100,1,goto(specialcontext,s,1)s,n,wait(60) s,n,hangup This would play message to let you know you are connected on the special number, then run the command when you pressed 1. If you don't press 1 in 10 seconds, the system says goodbye and hangs up. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Samad Sent: Wednesday, October 07, 2009 5:22 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] system cmd + fax line Hi I have a site that has asterisk install with a tdm410 one port is connected to a pstn that is used as a backup outbound line when/if the internet/voip is unavailable. Currently my dial plan for this line is to ignore it, I just basically do a s,1,noop s,n,wait (60) s,n,hangup what I would like to do is if I call from a certain number be given an option to run some commands or for the beginning just run a simple command via system something like s/041100,1,goto(specialcontext,s,1) s,1,noop s,n,wait(60) s,n,hangup [specialcontext] s,1,system(do a linux command) is that about right. Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfers from Queue Calls
A number of our clients has such issues. What we suggest for escalation is to do a blind transfer to a second-level queue, so that the logging is correct and even if second-line support cannot handle the call immediately, you get the functionality and the logging. Just my two euro cents, l. 2009/10/6 Darrin Henshaw darrin.aster...@gmail.com Hello, I thought to post this here before my manager starts his own coding project to give us a workaround. My situation I'm running into is as follows: 1. A call comes into our Asterisk system, it's trunked from one office to another via IAX. 2. Call enters a queue and is picked up by one of the agents. 3. That agent has to transfer the call, could be for a number of reasons the client wanted someone in particular, or maybe an escalation(we are a helpdesk). My problem is that the second part of the conversation after the transfer is not logged in the queue_log. Now this is by design from what I've found out, but we want the second part of the conversation to be recorded in the queue_log as well, for stats reporting for reviews of employee performance. Is anyone aware of a relatively easy way of implementing this? Whether it's by a patch or something else? Basically something similar to audiohook_inherit, which we use to allow mixmonitor to continue recording the call after it's been transferred. I've looked around, but haven't found anything. Thanks. Cheers, Darrin Henshaw -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] put some IVR into a queue after the call queuing
any interest in it? I'm evauating to add this feature but before to do that i'd like to know if there is some other approach that can avoid some developement. Regards On Wed, Sep 30, 2009 at 12:48 PM, nik600 nik...@gmail.com wrote: Dear all is it possible to handle a queue using a programmed IVR? As i understood, is possible to: - do some IVR in the dialplan BEFORE to queue the call - put a timeout to exit from the call and then do some IVR in the dialplan - intercept a single dialtone to exit the queue and performe some IVR in the dialplan (context setting in the queue) I've tested these things but in each case if i re-queue the call thi queue_log file reports the wrong total queued time. I'm wondering if is possible to bluild a script like that: 1) queue the call 2) after x seconds prompt message A 3) after y seconds prompt message B 4) after z seconds prompt message C 5) after t seconds prompt message Z with DTMF options 1,2,3 if option is 1 = remain in queue if option is 2 = ask the user to be recalled if option is 3 = transfer to In each moment (1,2,3,4,5) if a member queue gets available the call is routed to him. I belive that the only thing to do that is to do something like: 1) Queue A ... timeount 2) Queue B ... timeout 3) Queue C ...Timeout 4) Queue D ...periodic-announce - context set to xxx [xxx] 1,1,Queue D 2,1,Goto (.IVR to be recalled) 3,1,Goto ( transfer) And then manually match information between unique ID and queue_log to consider info on queue A,B,C,D, as a single queue. Or is there some magic sauce to specify an IVR script that is executed when a call is in a queue? Thanks -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need provider recommendations for the UK
Hi, I realise this is probably the wrong list for such a question, but I need a pointer to somewhere I can get some feedback on experience of (business class) voip providers for the UK? Situation is that we are currently with Gradwell and use them for an inbound/outbound single line for a business and their quality has gone from excellent to abysmal in the last few weeks. I'm sure they will work it out, but right now I just need a reliable provider that I can port a number to. I'm not especially price sensitive, reliability is the main requirement. IAX preferred, but not fussy. Possibly multiple incoming numbers in future, single incoming at the moment - in general we rarely have more than 1 line in use, but occasionally hit 2-3 simultaneous calls Note, it's going to be important that we can port our number across from Gradwell... Grateful if anyone can offer some really solid recommendations, or point me towards a more appropriate forum to request the same? Thanks Ed W ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Issues
I have a block of DID's that I ported to Vitelity about 7 days ago. The problem is if a POTS caller dials into the system, his dtmf is not heard at READ() or Background() while a prompt is played. After the prompt is finished, then dtmf is heard. I've been working with their support, but it still not resolved. SIP callers are not effected. Yesterday, I purchased a DID from Flowroute. The setting are the same as Vitelity. And amazingly, this DID works perfectly. This to me would indicate the problem whatever it is, is Vitelity or its upstream provider. - Am I right here?? Also, I have other DID blocks that have been with Vitelity for a while that do not have this issues. I'm told the upline carrier is the same. So I'm reaching out to you guys to give me ideas of what could be causing the problem and how I can resolve Thanks, Bart attachment: bhfisher.vcf___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.6 suddently restarts ...
As a follow up and unfortunately, I must say now that upgrading to 1.6.1.7-rc2 didn't help. We downgraded to a 1.6.1.0 with which we never met the problem we're facing now. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] system cmd + fax line
Use a GoToIf based on the callerid number. - Original Message - From: Alex Samad a...@samad.com.au To: asterisk-users@lists.digium.com Sent: Wednesday, October 07, 2009 12:21 Subject: [asterisk-users] system cmd + fax line Hi I have a site that has asterisk install with a tdm410 one port is connected to a pstn that is used as a backup outbound line when/if the internet/voip is unavailable. Currently my dial plan for this line is to ignore it, I just basically do a s,1,noop s,n,wait (60) s,n,hangup what I would like to do is if I call from a certain number be given an option to run some commands or for the beginning just run a simple command via system something like s/041100,1,goto(specialcontext,s,1) s,1,noop s,n,wait(60) s,n,hangup [specialcontext] s,1,system(do a linux command) is that about right. Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MPG123 Dying
- Original Message - From: Trevor Peirce tpei...@digitalcon.ca To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 06, 2009 23:14 Subject: Re: [asterisk-users] MPG123 Dying --[ UxBoD ]-- wrote: Please how do I stop the following ??? Asterisk ended with exit status 127 Asterisk died with code 127. Automatically restarting Asterisk. mpg123: no process killed You figure out why asterisk is crashing. :) This has nothing to do with mpg123, which is just an innocent bystander. I had an issue with mpg123 a few days ago where all of a sudden Asterisk was using 100% of the CPU. It happened over and over and I decided to just remove it. Any particular reason why you need to use mpg123 ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MPG123 Dying
If you use Iphones to call in, the regular MOH application renders sporatically. MPG123 allows you to increase/decrease the volume. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender Sent: Wednesday, October 07, 2009 12:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MPG123 Dying - Original Message - From: Trevor Peirce tpei...@digitalcon.ca To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 06, 2009 23:14 Subject: Re: [asterisk-users] MPG123 Dying --[ UxBoD ]-- wrote: Please how do I stop the following ??? Asterisk ended with exit status 127 Asterisk died with code 127. Automatically restarting Asterisk. mpg123: no process killed You figure out why asterisk is crashing. :) This has nothing to do with mpg123, which is just an innocent bystander. I had an issue with mpg123 a few days ago where all of a sudden Asterisk was using 100% of the CPU. It happened over and over and I decided to just remove it. Any particular reason why you need to use mpg123 ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Issues
- Original Message - From: Barton Fisher bpvoip...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, October 07, 2009 18:25 Subject: [asterisk-users] DTMF Issues I have a block of DID's that I ported to Vitelity about 7 days ago. The problem is if a POTS caller dials into the system, his dtmf is not heard at READ() or Background() while a prompt is played. After the prompt is finished, then dtmf is heard. I've been working with their support, but it still not resolved. SIP callers are not effected. Yesterday, I purchased a DID from Flowroute. The setting are the same as Vitelity. And amazingly, this DID works perfectly. This to me would indicate the problem whatever it is, is Vitelity or its upstream provider. - Am I right here?? Also, I have other DID blocks that have been with Vitelity for a while that do not have this issues. I'm told the upline carrier is the same. So I'm reaching out to you guys to give me ideas of what could be causing the problem and how I can resolve Thanks, Bart Enable dtmf under full in logger.conf do tail -f /var/log/asterisk/full, call in and enter some DTMF. If you see that it is delayed chances are it is your carrier. Also what are you using for DTMF (rfc2833, Inband etc.) ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need provider recommendations for the UK
- Original Message - From: Ed W li...@wildgooses.com To: Asterisk Users Mailing List asterisk-users@lists.digium.com Sent: Wednesday, October 07, 2009 17:22 Subject: [asterisk-users] Need provider recommendations for the UK Hi, I realise this is probably the wrong list for such a question, but I need a pointer to somewhere I can get some feedback on experience of (business class) voip providers for the UK? Situation is that we are currently with Gradwell and use them for an inbound/outbound single line for a business and their quality has gone from excellent to abysmal in the last few weeks. I'm sure they will work it out, but right now I just need a reliable provider that I can port a number to. I'm not especially price sensitive, reliability is the main requirement. IAX preferred, but not fussy. Possibly multiple incoming numbers in future, single incoming at the moment - in general we rarely have more than 1 line in use, but occasionally hit 2-3 simultaneous calls Note, it's going to be important that we can port our number across from Gradwell... Grateful if anyone can offer some really solid recommendations, or point me towards a more appropriate forum to request the same? Thanks Ed W Try asking on the biz list. There are plenty of people there that would help you out there. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk,click2talk, webphone
Hello We have launched another Gadget Option (Call Extension Button) to use with Asterisk: For instance: extensions.conf: [callButtonRoute] exten = 444,1,Dial(SIP/18874...@10.0.0.1,90,t) exten = 444,2,Hangup sip.conf: [callButton] type=peer username=callButton secret=1qa2ws3ed context=callButtonRoute host=dynamic nat=yes disallow=all allow=alaw allow=gsm Doddle Call Extension Button Gadget: iframe src= http://www.doddling.com/doddle/embed/doddleEmbedded.jsp?sipserver=10.0.0.1username=callButtonpassword= 1qa2ws3edcallto=444label1=My%20Asterisklabel2=Hangup%20Nowauto=yesmode=button width=243 height=160 frameborder=0 scrolling=no /iframe You can check all URL parameters options at: http://www.doddplephone.com Regards Sergio On Fri, Oct 2, 2009 at 7:38 PM, Doddle WebPhone doddleph...@gmail.comwrote: Hi,This can be useful for Asterisk / TI integrators: How to create a Free click2call application using Asterisk: We can build click2talk / webphone application empowering webpages with VoIP Telephony using online DoddlePhone and Asterisk Invoke doddle webphone (http://www.doddlephone.com) as follows: sipserver=Asterisk_SERVERusername=USERpassword=PASSWORDcallto=PHONE_NUMBER_TO_CALL Just create sip account on Asterisk, define its route and trigger click2call as above Notice that we can set a fixed route / context to the click2talk sip peer. check out http://www.doddlephone.com for details Regards Sergio ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VPS Server
Looking for Genuine VPS Server for 250 ports on Rent. Anyone can help ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can dial long distance but not local?
AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI (single span). I'm sure I just have something goofed up in the dialplans? I have a bunch of Polycom 331 IP phones connecting to the server. I can dial the other extensions in the system fine and I can dial long distance outgoing but cannot seem to get it to dial local (7 digit) calls. I see this in the CLI: -- Executing [...@macro-dialout-trunk:19] Dial(SIP/801-09b6e498, DAHDI/g1/5551212|300|) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/5551212 -- Channel 0/1, span 1 got hangup, cause 28 -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [...@macro-dialout-trunk:20] Goto(SIP/801-09b6e498, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-chanunav...@macro-dialout-trunk:1] GotoIf(SIP/801-09b6e498, 1?noreport) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) -- Executing [s-chanunav...@macro-dialout-trunk:3] NoOp(SIP/801-09b6e498, TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 2 - failing through to other trunks) in new stack Also when I check the PRI DEBUG I see an Error 28 which indicates an invalid number format. But I'm just sending 5551212, which should be o.k. I'm a newbie at this...any suggestions welcomed. -Ben- image001.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can dial long distance but not local?
Ben Schorr wrote: AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI (single span). I’m sure I just have something goofed up in the dialplans? I have a bunch of Polycom 331 IP phones connecting to the server. I can dial the other extensions in the system fine and I can dial long distance outgoing but cannot seem to get it to dial local (7 digit) calls. My guess is that your provider requires the full 10 digits even for local calls. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can dial long distance but not local?
Perhaps send it as 10 digits or 1+? Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Wednesday, October 07, 2009 3:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Can dial long distance but not local? AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI (single span). I'm sure I just have something goofed up in the dialplans? I have a bunch of Polycom 331 IP phones connecting to the server. I can dial the other extensions in the system fine and I can dial long distance outgoing but cannot seem to get it to dial local (7 digit) calls. I see this in the CLI: -- Executing [...@macro-dialout-trunk:19] Dial(SIP/801-09b6e498, DAHDI/g1/5551212|300|) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/5551212 -- Channel 0/1, span 1 got hangup, cause 28 -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [...@macro-dialout-trunk:20] Goto(SIP/801-09b6e498, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-chanunav...@macro-dialout-trunk:1] GotoIf(SIP/801-09b6e498, 1?noreport) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) -- Executing [s-chanunav...@macro-dialout-trunk:3] NoOp(SIP/801-09b6e498, TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 2Cool - failing through to other trunks) in new stack Also when I check the PRI DEBUG I see an Error 28 which indicates an invalid number format. But I'm just sending 5551212, which should be o.k. I'm a newbie at this.any suggestions welcomed. -Ben- image001.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can dial long distance but not local?
Thanks Cary, If I send as 10 digits (dialing the local area code) I get an error from the phone co that I've dialed an incorrect number. Unfortunately I live in Hawaii where the 808 area code covers the whole state but is treated differently on neighbor-islands. So if I dial 8085551212 I get information for the neighbor islands rather than for our local island. I think I tried the 1+ trick with the same results, but I'll give it another try just in case I overlooked it. Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com http://www.rolandschorr.com/ b...@rolandschorr.com mailto:b...@rolandschorr.com Twitter: http://www.twitter.com/bschorr From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Wednesday, October 07, 2009 11:04 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Can dial long distance but not local? Perhaps send it as 10 digits or 1+? Cary Fitch From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Wednesday, October 07, 2009 3:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Can dial long distance but not local? AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI (single span). I'm sure I just have something goofed up in the dialplans? I have a bunch of Polycom 331 IP phones connecting to the server. I can dial the other extensions in the system fine and I can dial long distance outgoing but cannot seem to get it to dial local (7 digit) calls. I see this in the CLI: -- Executing [...@macro-dialout-trunk:19] Dial(SIP/801-09b6e498, DAHDI/g1/5551212|300|) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/5551212 -- Channel 0/1, span 1 got hangup, cause 28 -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [...@macro-dialout-trunk:20] Goto(SIP/801-09b6e498, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-chanunav...@macro-dialout-trunk:1] GotoIf(SIP/801-09b6e498, 1?noreport) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) -- Executing [s-chanunav...@macro-dialout-trunk:3] NoOp(SIP/801-09b6e498, TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 2 - failing through to other trunks) in new stack Also when I check the PRI DEBUG I see an Error 28 which indicates an invalid number format. But I'm just sending 5551212, which should be o.k. I'm a newbie at this...any suggestions welcomed. -Ben- image001.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can dial long distance but not local?
No, unfortunately not, our local area code is split among multiple islands. You have to dial it to reach a neighbor island. If you dial it with a local number it tries to find that number on a neighbor island (then usually fails). I'm trying to find out from them if they want us to dial something else first though - like a 1 or a 9. I tried pridialplan=local as well as pridialplan=unknown. No improvement either way. Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com Twitter: http://www.twitter.com/bschorr -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Wednesday, October 07, 2009 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can dial long distance but not local? Ben Schorr wrote: AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI (single span). I'm sure I just have something goofed up in the dialplans? I have a bunch of Polycom 331 IP phones connecting to the server. I can dial the other extensions in the system fine and I can dial long distance outgoing but cannot seem to get it to dial local (7 digit) calls. My guess is that your provider requires the full 10 digits even for local calls. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.6 suddently restarts ...
Please follow up on the issue tracker at http://issues.asterisk.org Thanks! Leif. Olivier wrote: As a follow up and unfortunately, I must say now that upgrading to 1.6.1.7-rc2 didn't help. We downgraded to a 1.6.1.0 with which we never met the problem we're facing now. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk debug message --- stopped sounds ???
I have seen this message stopped sounds while I am watching asterisk debug: -- Called 9/0532828384 -- Call accepted by 192.168.10.220 (format ulaw) -- Format for call is ulaw -- IAX2/9-69 stopped sounds -- IAX2/9-69 answered SIP/xxx.xxx.xxx.xxx-b7d009a0 What does it mean?? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Drop Call on ICMP Port Unreachable?
One of our users recently had a powerfail while connected to our meetme gateway. (Asterisk 1.4.17 on debian 4.0) Through the course of it, asterisk never hung up. His system came back up, and started sending ICMP port unreachables, but the stream went on, flooding him with silence media stream packets (there was nobody else in the conference). Is asterisk aware of ICMP unreachables? Is there a tunable I can set to make it be? I found a thread here that discusses it briefly: http://lists.digium.com/pipermail/asterisk-users/2005-March/086626.html However, there's no real resolution there. If it's not aware of it, how difficult would it be to add? -Dan Mahoney -- Dan Mahoney Techie, Sysadmin, WebGeek Gushi on efnet/undernet IRC ICQ: 13735144 AIM: LarpGM Site: http://www.gushi.org --- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choose IAX or SIP trunking?
On 091007 0001, Hans Witvliet wrote: Obviously, it is an outdated document that should be revised. It stated: Now sip supports more than asterisk sip supports. SIP RFC requires tcp support for example, yet its not in trunk yet. This feature is already in the 1.6-branch, production. You just need to enable it in the config. I used it in connecting to Microsoft-stuff, quite some months ago. I would agree, in assessing how bad SIP is the article is outdated. My goal is, however, different. We are already using SIP, and I wanted to know how bad is IAX2. I am afraid the information in the article matches what others suggested: IAX2 support requires an overhaul to become viable in a high volume production environment. It is suitable for home or a very small office with no trained personnel, though, where traffic is low. If you disagree, please tell me where do you think I am mistaken. s/bad/good/g for a more optimistic point of view. -kkm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
