Re: [asterisk-users] Queues and penalties

I'm fairly sure the patch to App Queue that was added to Asterisk 13+ should do the job... It causes agent priorities to "float up" over time so that new agents are included without excluding old agents. I can't find it right now but there can't be that many app_queue patches to ast 13 in the

Re: [asterisk-users] RTP Timestamp rewind

Mark, You have cropped the image you inserted above and removed a very important part of the line you highlighted. I think is says ",Mark" after the time value - You can even see the un-cropped comma in your picture. RTP timestamps can be reset mid-stream if needed - It is part of the spec, and

Re: [asterisk-users] Asterisk sip_autodestruct messages - extensions locked

Based on the line number of that error in chan_sip.c, it looks like you're running Asterisk 1.8 or earlier. AFAIK, The issue you are seeing was fixed years ago, but not THAT many years ago! If I'm right, you should upgrade to fix that issue. Cheers, Steve On Fri, 30 Jun 2017 at 13:39 Stefan

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

I am also getting this, three or four times in the last month after years of no problems. I agree that Gmail is the likely common factor, but I would love to have access to these bounce messages to know whether it is actually an overly-paranoid list server! Steve On Mon, 12 Jun 2017 at 09:09

Re: [asterisk-users] Using queue priorities to add agents

member => SIP/104.2,4,Debbie > member => SIP/105.2,4,Luci > member => SIP/106.2,5,Sheila > member => SIP/107.2,6,Mike > > So every 20 seconds it jumps up to the next Penalty and every few minutes > it resets the penalty back down to 1 and starts again. > > > On Thu, Ma

Re: [asterisk-users] Using queue priorities to add agents

t queue to timeout after 60 secs. Then send to the overflow > queue with all agents/members as same priority. > > > > > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steve Davies > *Sent:* Thur

[asterisk-users] Using queue priorities to add agents

Hi, I have a scenario that I am failing to implement using the Queue app, but which I had thought would be commonplace... 1) (this bit works fine) I want a queue caller to have access to the basic set of agents initially, with an overflow to additional agents if they are busy - This is done

Re: [asterisk-users] ** in extensions.conf

On Wed, 26 Apr 2017 at 20:29 Jerry Geis wrote: > I just tried this in my extensions.conf > > exten => **,1,Noop(Testing) > exten => **,n,Playback(demo-congrats) > > Did a reload... and the above does not happen. > I created as 12 instead of the ** and that works fine. > >

Re: [asterisk-users] Cisco IP 8841 asterisk integration

If you need to know what the provisioning XML should look like for a 3PCC build of a Cisco 78xx or 88xx phone, then let it boot without provisioning, and then log in to its web interface. Select admin mode and log-in if necessary. Then edit the URL in the browser from: http://ip-address/admin/ to

Re: [asterisk-users] Cisco IP 8841 asterisk integration

able from the Cisco website? > > Actually it came with sip88xx firmware. > > Regards . > > > On Fri, 2 Dec 2016, 10:38 p.m. Steve Davies, <davies...@gmail.com> wrote: > > Hi, > > You have to buy the 3PCC version for this to work. Once you have this, > the

Re: [asterisk-users] Cisco IP 8841 asterisk integration

Hi, You have to buy the 3PCC version for this to work. Once you have this, they work very much like the Cisco SPA handsets. I also ended up with a non-3PCC handset and it is useless, and as far as I can tell they cannot be re-flashed. Cheers, Steve On Fri, 2 Dec 2016 at 16:16 Gopalakrishnan

Re: [asterisk-users] pedantic=yes in sip.conf

Alan, A little more context would be useful. Where are you putting the '#' and why? ( If all else fails, print it out and mail it to them ;-) ) %23 is the correct encoding for a hash '#' symbol in many SIP contexts, and should be decoded by a properly functioning far-end. Regards, Steve On

Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

Hi, In my experience, all Yealink phones work just fine with Asterisk, we have hundreds (perhaps even low-thousands) out there with customers on Asterisk 1.2, 1.6.2, 1.8 and 11. If you are accurately representing the SIP trace on the phone and the SIP trace on Asterisk, then I would strongly

Re: [asterisk-users] hangup call gw FXO

Looking at the pastebin, the Vega device sends a CANCEL with reason: Reason: Q.850 ;cause=16. Cause 16 is normal clearing and suggests that the original caller has disconnected. I would take a look at the Vega's logs Regards, Steve On Thu, 5 Mar 2015 at 11:41 ricky gutierrez

Re: [asterisk-users] Connected Line presentation in 1.8.x upwards

On 29 July 2013 16:55, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: From:Steve Davies davies...@gmail.com To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date:07/29/2013 10:53 AM Subject:[asterisk-users

[asterisk-users] Connected Line presentation in 1.8.x upwards

Hi, I've searched the asterisk.org and voip-info wiki sites, but not found an answer that seems to match. Hopefully this is a simple question. COLP is working very well on our system - Unfortunately it is working a bit TOO well in some circumstances. We have some untrusted trunks. On these

Re: [asterisk-users] ignore 183 session progress in parallel call scenarios

I am sure I submitted the following alternative behaviour to the bug-tracker in the past, but cannot find any reference to it. Here is the patch I use to IMHO improve this behaviour. In case it is not officially uploaded, I will state here that this code is disclaimed and unencumbered as if

Re: [asterisk-users] IPcortex: ERROR[23444] chan_zap.c: You cannot use cause 17 number when in state 6

Hi Xavier, The issue you are seeing is an old Asterisk/Bristuff bug that was fixed years ago. Basically ISDN is unable to understand a call going from RING state to BUSY state, so Asterisk converts the BUSY into a HANGUP/Normal Clearing, and warns that this is happening. Sadly, in that old

Re: [asterisk-users] FW: IPcortex: ERROR[23444] chan_zap.c: You cannot use cause 17 number when in state 6

Xavier, DoNotDisturb generates a Busy indication. Insert that into my earlier response, and you have an explanation of why the call tries to go from RING to BUSY, and confirms my theory. No you cannot replace the Zaptel card driver on its own (and the problem was bigger than that anyway), as

Re: [asterisk-users] Fundemental changes to CDR within single asterisk family

On 4 April 2013 09:05, Ishfaq Malik i...@pack-net.co.uk wrote: On Tue, 2013-03-26 at 07:26 -0500, Matthew Jordan wrote: On 03/26/2013 05:22 AM, Ishfaq Malik wrote: Hi In asterisk 1.8.7.0, an inbound call that was transferred to another peer would have 2 cdr entries. In

Re: [asterisk-users] Congestion() forcing PRI channels to be not available

On 19 December 2012 21:54, Christopher Harrington ch...@acsdi.com wrote: You probably already know this, but 1.4x is very old (released in 2006) and is officially end-of-life. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions You might get more help or better behavior by updating

Re: [asterisk-users] SIP Question - Early audio one-way or 2-way?

On 1 September 2012 09:08, Olle E. Johansson o...@edvina.net wrote: 31 aug 2012 kl. 13:13 skrev Steve Davies davies...@gmail.com: On 31 August 2012 07:49, Olle E. Johansson o...@edvina.net wrote: 24 aug 2012 kl. 16:18 skrev Steve Davies davies...@gmail.com: Hi SIP Gurus, I've tried

[asterisk-users] SIP Question - Early audio one-way or 2-way?

Hi SIP Gurus, I've tried to find the relevant RFCs, but am struggling. I can find the odd opinion online, but was wondering if anyone could give a definitive answer. If a SIP call is initiated (INVITE) and receives either a 180 with SDP, or a 183 with SDP, then the remote party will start to

Re: [asterisk-users] SIP Question - Early audio one-way or 2-way?

On 24 August 2012 15:34, Faisal Hanif fai...@vopium.com wrote: Steve Davies davies...@gmail.com wrote: Hi SIP Gurus, I've tried to find the relevant RFCs, but am struggling. I can find the odd opinion online, but was wondering if anyone could give a definitive answer. If a SIP call is initiated

Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?

On 25 April 2012 18:05, Kevin P. Fleming kpflem...@digium.com wrote: On 04/25/2012 11:54 AM, Steve Davies wrote: A further question... It appears that for SIP endpoints, this facility only updates RPID and PAI headers? I have found that there appear to be 4 different SIP CID-update mechanisms

[asterisk-users] CONNECTEDLINE() updated during SIP events?

Hi, I have read the excellent information here: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information and believe I have an understanding of what is offered. I have a couple of questions: - Is it possible to update COLP/COLR when a SIP redirect occurs, or when a SIP

Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?

On 25 April 2012 16:55, Richard Mudgett rmudg...@digium.com wrote: [snip] - Is it possible to have the COLP/COLR information updated when a SIP attended transfer is completed? If so how? Transfers generate connected line update events automatically.  The connected line interception macros

Re: [asterisk-users] Asterisk ACL

On 2 April 2012 14:06, Mark Farmer mark.far...@gagenetworks.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: 02 April 2012 13:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive

On 6 February 2012 10:45, Jonas Kellens jonas.kell...@telenet.be wrote: ** Hello, is there anyone that can give me some more information on these deadlocks ?! How can these deadlocks occur and what is good practise to avoid these problems ?? Jonas. The only way to avoid deadlocks is

Re: [asterisk-users] Asterisk CLI unresponsive

On 3 February 2012 12:12, Jonas Kellens jonas.kell...@telenet.be wrote: On 02/03/2012 01:05 PM, Mikhail Lischuk wrote: Jonas Kellens писал 03.02.2012 12:09: using asterisk 1.6.2.22 What is wrong with Asterisk when the CLI becomes unresponsive ?! Greetings. I am using the same version, 

[asterisk-users] SIP and NAT best practices since recent changes?

Hi, Since the recent update to the NAT configuration options and defaults in chan_sip.so, I am interested in any SIP/NAT best practices advice. What I've always done in the past is: Global: nat=no SIP handsets that are local: nat=no SIP handsets that are remote: nat=yes ITSP SIP trunks: nat=yes

Re: [asterisk-users] SIP and NAT best practices since recent changes?

On 11 January 2012 15:43, Kevin P. Fleming kpflem...@digium.com wrote: On 01/11/2012 05:29 AM, Steve Davies wrote: Hi, Since the recent update to the NAT configuration options and defaults in chan_sip.so, I am interested in any SIP/NAT best practices advice. What I've always done

Re: [asterisk-users] Is Asterisk 1.4 compatible with 1.8.7 ?

On 28 December 2011 03:02, Joseph syscon...@gmail.com wrote: No, it makes no difference, on the other end is asterisk 1.4.39 and 1.8.8 is still giving me:  Executing [4@internal:1] Dial(SIP/11-0003, IAX2/home_server:@192.168.141.1/4,30,rw) in new stack    -- Called

Re: [asterisk-users] Asterisk 1.4.x segfaulting daily

On 14 December 2011 12:56, Paulo Santos paulo.r.san...@sapo.pt wrote: Hello list, An Asterisk installation that was doing fine suddenly stared segfaulting a couple of times per day. I enabled all the logging and debugging to try to find a pattern but there was too much information to see

Re: [asterisk-users] one way voice with IVR

On 17 October 2011 11:01, gincantalupo gincantal...@fgasoftware.com wrote: Hi, found where the problem is.I tried with a Grandstream phone and it works!!! The problem is my (crappy) Snom phone. I'm investigating the probhope to find the cause asap. FYI: snom firmware 7.3.30 is

Re: [asterisk-users] Reinvite dialplan application [Was: OT - SIP - Toggle to autoanswer after ringing]

On 5 October 2011 10:21, Nasir Iqbal na...@ictinnovations.com wrote: You can do this by an AMI based transfer (Redirect) to Local channel, and then in local channel's dialplan you need to add your desired custom sip header followed by a dial command. Nasir Iqbal ICT Innovations

[asterisk-users] Possibly odd sip.conf security requirements. Possible?

Hi, Is the following possible in some way? I want to have several SIP providers able to send me calls, each provider may send calls into many possible DDIs. Each provider has a cluster of servers, but is unable to authenticate with me, so the following would be a sort of pseudo-code sip.conf

Re: [asterisk-users] 1.6.2.20 ${DIALSTATUS} disagrees with CDR(answered)

On 14 August 2011 08:36, Eric Wieling ewiel...@nyigc.com wrote: I am having a problem with ${DIALSTATUS} and )=CDR(disposition) disagreeing.   Below is a dialplan snippet and the resulting CLI output.  This is running in an 'h' extension.        Noop(DIALSTATUS=${DIALSTATUS})        

Re: [asterisk-users] Seg Faults with 1.6.2.19

On 18 July 2011 12:03, Lee Archer lee.arc...@thebigword.com wrote: Hi, is anyone else having problems with the reload command crashing Asterisk 1.6.2.19?  I’ve run a few tests and 1.6.2.18.2 doesn’t have this problem but 1.6.2.19 after a few reloads is just dumping and restarting. Thanks

Re: [asterisk-users] FAX with SIP

On 18 July 2011 12:20, Eduardo Carpes car...@bsd.com.br wrote: Hello guys I need some help to do works FAX using SIP, anybody know the secret to this? Have asterisk 1.6. Thanks!! -- Enviado do meu celular Eduardo Carpes E-mail: car...@bsd.com.br www.freebsd.org The magic sauce that you

Re: [asterisk-users] Seg Faults with 1.6.2.19

On 18 July 2011 13:00, Lee Archer lee.arc...@thebigword.com wrote: Hi Steve, I think it's related to my ODBC connection.  I started with a random hang where it looked ODBC related which led me to try a few things.   Reloading the config a few times is causing core dumps which 1.6.2.18.2 just

Re: [asterisk-users] Seg Faults with 1.6.2.19

On 18 July 2011 14:05, Lee Archer lee.arc...@thebigword.com wrote: Seems to be an already reported problem but since no more fixes for 1.6 it's back to 1.6.2.18.2 for me. https://issues.asterisk.org/jira/browse/ASTERISK-18103 Regards Lee If it is a regression introduced in 1.6.2.19, then

Re: [asterisk-users] OT: Google Plus

On Saturday, 9 July 2011, Gordon Henderson gordon+aster...@drogon.net wrote: On Sat, 9 Jul 2011, Steve Davies wrote: On 9 July 2011 12:34, randulo rand...@randulo.com wrote: Go ahead and lambast me for this post, it isn't specific to Asterisk, but: G+ has only been open at all for a week

Re: [asterisk-users] OT: Google Plus

On 9 July 2011 12:34, randulo rand...@randulo.com wrote: Go ahead and lambast me for this post, it isn't specific to Asterisk, but: G+ has only been open at all for a week and I already am chatting with over 200 people who are into VoIP, Asterisk and all the rest of the stuff we here care

[asterisk-users] iLBC re-licence

Does anybody know if the updated licence on iLBC makes it safe to include in Asterisk when used in a commercial environment again? https://sites.google.com/site/webrtc/ilbc-freeware It seems to require that the Google iLBC licence document is on the box, but that otherwise it is free-to use by

Re: [asterisk-users] iLBC re-licence

On 22 June 2011 17:14, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 06/22/2011 03:32 PM, Steve Davies wrote: Does anybody know if the updated licence on iLBC makes it safe to include in Asterisk when used in a commercial environment again?   https://sites.google.com/site/webrtc

Re: [asterisk-users] Aastra phone # key in dialplan

On 22 June 2011 17:09, marvin horst fivehor...@gmail.com wrote: I want to use extension numbers that begin with the # key in my dialplan, but I can't get my Aastra phone (6731i) to transmit the # key to asterisk. It works fine for the * key. I've tried numerous Local Dial Plan patterns in the

Re: [asterisk-users] Asterisk 1.8 broken MWI

On 9 June 2011 15:49, satish patel satish...@hotmail.com wrote: Date: Wed, 8 Jun 2011 18:15:14 +0100 From: davies...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI On 8 June 2011 17:20, satish patel satish...@hotmail.com wrote:

Re: [asterisk-users] Asterisk 1.8 broken MWI

On 8 June 2011 17:20, satish patel satish...@hotmail.com wrote: Interesting thing is when i reload sip.conf  i got MWI lamp working on polycom 501 But its not working when anyone leave voicemail. Do you know its some timeout or polling setting in sip.conf ? Still my question is my my

[asterisk-users] Refactor of CDR - Comments please.

Hi, Since raising this ticket about broken CDR data: https://issues.asterisk.org/jira/browse/ASTERISK-17826 I have been researching how CDR records work in various circumstances. CEL will do most things that people want, but that does not change that CDR records are likely to persist into

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

On 1 June 2011 15:10, randall rand...@songshu.org wrote: On 06/01/2011 03:55 PM, randall wrote: On 06/01/2011 03:41 PM, Tzafrir Cohen wrote: On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote: Hi all, After running fine for a few months now asterisk seems to hang frequently , still

Re: [asterisk-users] AstManProxy

On 20 May 2011 16:16, Ishfaq Malik i...@pack-net.co.uk wrote: On Fri, 2011-05-20 at 10:58 -0400, Leif Madsen wrote: On 11-05-20 09:37 AM, Ishfaq Malik wrote: Do many people use this? Is it reliable and safe? It may still work, but that code is quite old, and I'm not even sure it's

Re: [asterisk-users] AstManProxy

On 24 May 2011 10:43, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 05/24/2011 11:02 AM, Steve Davies wrote: [snip] I use astmanproxy with Asterisk 1.6.2.18 - It works fine. The most recent version is on Github, and is not that old. In fact that reminds me that I really must upload

Re: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering

On 6 May 2011 16:30, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Friday, May 06, 2011 11:23 AM To: Asterisk Users Mailing List -

[asterisk-users] IAX2 codec selection and video

Hi, Can anyone let me know how I can enable video (h.263) on SIP, but if a video call is passed over IAX, it will remove the video and pass the audio only. What I tried was: SIP - videosupport=yes - disallow=all - allow=alaw - allow=h263 IAX - disallow=all - allow=alaw

Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)

On 15 April 2011 13:02, Vlasis Hatzistavrou vh...@kinetix.gr wrote: Hello, On an Asterisk 1.4.33.1 in a simple scenario: [test] exten = _X.,1,Dial(SIP/12345@peer01,,,) exten = i,1,Hangup(${HANGUPCAUSE}) exten = t,1,Hangup(${HANGUPCAUSE}) exten = h,1,Hangup(${HANGUPCAUSE}) I have

Re: [asterisk-users] No ringback even though progressinband=yes is set

: 505.327.7300 . -Original Message- From: Steve Davies [mailto:davies...@gmail.com] Sent: Thursday, April 07, 2011 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No ringback even though progressinband=yes is set On 7 April 2011 17:02

Re: [asterisk-users] No ringback even though progressinband=yes is set

On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote: Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set

[asterisk-users] SIP channel able to add codecs once up and running?

From my observations, if a video capable device starts the call in non-video mode, it is never able to add video to the channel? Is this correct, or am I missing something? It looks as if the codec 'jointcapability' is calculated at the start of the call, and can never be added to (with

Re: [asterisk-users] Registration from '000000 x 1000

On 2 April 2011 09:46, Jonas Kellens jonas.kell...@telenet.be wrote: Hello list, I often see the following in my message log : [Apr  2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00 sip:00@MY-IP' failed for '184.106.109.168' - No matching peer found [Apr  2 08:15:01]

[asterisk-users] DAHDI, IAX2 and SIP considerations for Early-Media / Alerting

Hi, Short version: Is it possible or even legal to convert an IAX2 PROGRESS/EARLY-MEDIA indication into a DAHDI/q.931 ALERTING signal when your ISDN provider does not pass early media on receipt of an PROGRESS(8) indication? Long version: I have an Asterisk 1.6.2.18-rc1 based system with a

Re: [asterisk-users] One Way Audio

On 10 March 2011 11:17, Ishfaq Malik i...@pack-net.co.uk wrote: Just fixed our problem with directmedia=no but this only applies if your extensions are behind a nat Ish There are several reasons why directmedia=no might be the correct configuration. 1) NAT - probably the most common

Re: [asterisk-users] IAX channel name incorrect - Found in 1.2 still happens in 1.6

*Bump* No takers? Perhaps no-one else thinks this is a bug? Regards, Steve On 7 February 2011 16:45, Steve Davies davies...@gmail.com wrote: Hi, The following IAX config (slightly edited) causes an issue for me in version 1.6.2.16.1, where my CDR data is unreliable. [user1] type=friend

Re: [asterisk-users] Aastra phones cannot transfer calls?

On 16 February 2011 00:22, Ernie Dunbar maill...@lightspeed.ca wrote: At 12:12 PM 2/15/2011, you wrote: I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk v1.6.2.1. They can call each other's extensions (and make and receive calls otherwise), but they cannot transfer calls,

Re: [asterisk-users] Barge in.

On 16 February 2011 10:13, Peter den Hartog peterdenhar...@gmail.com wrote: I'm running Asterisk 1.6 and was wondering if anybody have a workig barge in solution running. I was thinking of using chanspy, but i would like that the original call would be dropped, and the new call would be the

[asterisk-users] IAX channel name incorrect - Found in 1.2 still happens in 1.6

Hi, The following IAX config (slightly edited) causes an issue for me in version 1.6.2.16.1, where my CDR data is unreliable. [user1] type=friend auth=md5 accountcode=user1 notransfer=yes context=context1 host=10.0.0.250 username=user1 secret=secret1 disallow=all allow=alaw [user2] type=friend

Re: [asterisk-users] queue_log in MySQL database

On 13 January 2011 16:28, Jonas Kellens jonas.kell...@telenet.be wrote: I actually found this : http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL But a second question : how can I know how long a caller stayed inside the queue untill it was answered by a member ?? The

Re: [asterisk-users] No MOH with parked call

On 24 December 2010 15:44, Steve Davies davies...@gmail.com wrote: On 23 December 2010 18:01, Steve Davies davies...@gmail.com wrote: Hi Again, I thought I had this sorted, but it appears that in a clean environment I did not in fact fix it. There appears to be a bit of a contradiction. 1

Re: [asterisk-users] One way crappy audio in iax call - Asterisk 1.6.2.15

On 24 December 2010 14:40, Administrator TOOTAI ad...@tootai.net wrote: Hi, We had 2 asterisk 1.4 connected together in iax, all was fine. One of them was upgraded (server and Asterisk) in 1.6.2.15, the other end is in 1.4.38 When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But

Re: [asterisk-users] No MOH with parked call

On 23 December 2010 18:01, Steve Davies davies...@gmail.com wrote: Hi Again, I thought I had this sorted, but it appears that in a clean environment I did not in fact fix it. There appears to be a bit of a contradiction. 1) In 1.6.2.x, musiconhold requires DAHDI (which we have) 2) In 1.6.2

Re: [asterisk-users] No MOH with parked call

On 7 December 2010 17:47, Steve Davies davies...@gmail.com wrote: On 7 December 2010 15:00, Steve Davies davies...@gmail.com wrote: On 7 December 2010 14:17, Lee Archer lee.arc...@thebigword.com wrote: Hi, try unloading res_timing_dahdi.so then trying again. Lee -Original Message

Re: [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.

On 21 December 2010 22:06, Tilghman Lesher tilgh...@meg.abyt.es wrote: On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server

Re: [asterisk-users] Asterisk hangs up call after 20s

On 22 December 2010 12:44, Gilles codecompl...@free.fr wrote: Hello        I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT

[asterisk-users] UDP buffer overflows?

Hi, On one of our asterisk systems that is quite busy, we are seeing the following from 'netstat -s': Udp: 17725210 packets received 36547 packets to unknown port received. 44017 packet receive errors 17101174 packets sent RcvbufErrors: 44017 --- this When

Re: [asterisk-users] UDP buffer overflows?

On 10 December 2010 16:45, Steve Davies davies...@gmail.com wrote: Hi, On one of our asterisk systems that is quite busy, we are seeing the following from 'netstat -s': Udp:    17725210 packets received    36547 packets to unknown port received.    44017 packet receive errors    17101174

Re: [asterisk-users] UDP buffer overflows?

On 10 December 2010 17:21, Shaun Ruffell sruff...@digium.com wrote: On 12/10/2010 11:02 AM, Steve Davies wrote: On 10 December 2010 16:45, Steve Davies davies...@gmail.com wrote: Hi, On one of our asterisk systems that is quite busy, we are seeing the following from 'netstat -s': Udp

Re: [asterisk-users] UDP buffer overflows?

On 10 December 2010 17:33, Steve Davies davies...@gmail.com wrote: On 10 December 2010 17:21, Shaun Ruffell sruff...@digium.com wrote: On 12/10/2010 11:02 AM, Steve Davies wrote: On 10 December 2010 16:45, Steve Davies davies...@gmail.com wrote: Hi, On one of our asterisk systems

[asterisk-users] No MOH with parked call

Hi, Has anybody else noticed that MOH does not play on parked calls in 1.6.2.14? Or is it just my setup? MOH seems to work in every other respect (Call Held or in-Queue), but when a call is parked, the logs show MOH being started, but the parked party hears nothing. The verbose logs show the

Re: [asterisk-users] No MOH with parked call

On 7 December 2010 14:17, Lee Archer lee.arc...@thebigword.com wrote: Hi, try unloading res_timing_dahdi.so then trying again. Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies Sent: 07

Re: [asterisk-users] No MOH with parked call

On 7 December 2010 15:00, Steve Davies davies...@gmail.com wrote: On 7 December 2010 14:17, Lee Archer lee.arc...@thebigword.com wrote: Hi, try unloading res_timing_dahdi.so then trying again. Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk

Re: [asterisk-users] Avoided deadlock Error(solved)

On 25 November 2010 13:02, bayardo.sanc...@gmail.com wrote: The proble is dialplan I configure fine -- Sent from my BlackBerry® VoIP, Windows/Linux Administration and Network Management US Numbers: 561-886-0664 Nicaragua Mobile: +505.8488.6876 -Original Message- From: Stefan

[asterisk-users] Contradiction in GROUP() function

I am confused. In Asterisk 1.2 and 1.4, in the code there is an error: Setting a group requires an argument (group name) But the syntax is shown as: Syntax: GROUP([category]) The [category] square brackets indicate to me an optional parameter, which contradicts the error. Verison 1.6 is

Re: [asterisk-users] Contradiction in GROUP() function

On 24 November 2010 10:12, Steve Davies davies...@gmail.com wrote: I am confused. In Asterisk 1.2 and 1.4, in the code there is an error: Setting a group requires an argument (group name) But the syntax is shown as: Syntax: GROUP([category]) The [category] square brackets indicate to me

Re: [asterisk-users] Asterisk parking question

On 18 November 2010 17:43, Mike l...@net-wall.com wrote: I tried thator I think I did something similar, but that may or may not apply (depending on my understanding of parking lots) Here is my relevant contexts.  The SIP phones are registered under this context: [some_context]

Re: [asterisk-users] Licensing of Default MOH

On 22 October 2010 14:24, Miguel Molina mmol...@millenium.com.co wrote: I think the OP is asking for the old MoH sound (fpm-world-mix, etc) that came with asterisk. I wonder why the change from the fpm sounds to the opsound ones, it was a licensing issue? I think the original 'fpm' files were

[asterisk-users] SIP authentication - Thoughts please

Hi, We have a scenario where we need multiple discrete SIP trunks (peers) from/to a single endpoint. Because the authentication system starts by matching IP address, it only ever matches on one of the SIP peer entries, and ignores the others. This is documented behaviour and as such is correct. I

Re: [asterisk-users] SIP authentication - Thoughts please

On 7 October 2010 10:10, Stefan Schmidt s...@sil.at wrote: Am 07.10.10 10:52, schrieb Steve Davies: Hi, snipped Hello, i just want to say something about point 4 which comes to my mind about security. 4) I am not sure whether it is worth dropping through and testing auth against other

Re: [asterisk-users] High volume BLF - Suggestions?

On 13 September 2010 19:12, Cassius Smith cass...@cassius.org wrote: Steve I have 64 channels being monitored with an SPA962 with two SPA932 sidecars. It works perfectly with Asterisk 1.6.2.9; my users are very happy with this. Latest firmware is a must. HTH Cassius Smith Any chance you

Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.

On 11 September 2010 20:36, Antonio Berrios anto...@sheffieldcitytaxis.com wrote: On 09/09/10 17:59, Steve Davies wrote: On 9 September 2010 17:52, Antonio Berrios anto...@sheffieldcitytaxis.com  wrote: Steve Davies wrote: Hi, I am using 1.6.2.11, and I need to be able to include the name

[asterisk-users] High volume BLF - Suggestions?

Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow

Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.

On 13 September 2010 11:07, Antonio Berrios anto...@sheffieldcitytaxis.com wrote: Gotcha. Yeah, I'm looking at implementing that (searching call recordings by agent that took the call) here but since our asterisk call recording is a separate server to the ones dealing with queues I'll be

Re: [asterisk-users] High volume BLF - Suggestions?

On 13 September 2010 11:43, Olivier oza_4...@yahoo.fr wrote: 2010/9/13 Steve Davies davies...@gmail.com [snip] Our test involves about 10 BLF-NOTIFY messages per second to each handset with a 5-second pause every 5 seconds. This will either crash or render unusable all of the following

Re: [asterisk-users] High volume BLF - Suggestions?

On 13 September 2010 12:16, Stefan Schmidt s...@sil.at wrote: Hello, Am 13.09.10 11:56, schrieb Steve Davies: Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up

Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.

On 13 September 2010 16:58, Carlos Chavez cur...@telecomabmex.com wrote: On Mon, 2010-09-13 at 11:22 +0100, Steve Davies wrote: On 13 September 2010 11:07, Antonio Berrios anto...@sheffieldcitytaxis.com wrote: Gotcha. Yeah, I'm looking at implementing that (searching call recordings

Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.

On 9 September 2010 17:52, Antonio Berrios anto...@sheffieldcitytaxis.com wrote: Steve Davies wrote: Hi, I am using 1.6.2.11, and I need to be able to include the name of the channel that answered a call in the call-recording filename. At a guess we need to use the Queue(name,,macro

[asterisk-users] Queue/Dial Recording - Capture answering channel name.

Hi, I am using 1.6.2.11, and I need to be able to include the name of the channel that answered a call in the call-recording filename. At a guess we need to use the Queue(name,,macro) or Dial(chan1chan2,,M(macro)) and use the macro to update the call recording filename. But, the macro runs

Re: [asterisk-users] How to debug this specific issue?

On 25 August 2010 08:22, Matt Riddell li...@venturevoip.com wrote: On 25/08/10 7:20 PM, Tilghman Lesher wrote: I really thought that the canary should have sounded if Asterisk got in a loop - or maybe that only happens with high priority? The canary only runs in high priority mode, and it's

Re: [asterisk-users] How to debug this specific issue?

On 24 August 2010 08:07, Stefan Schmidt s...@sil.at wrote: Steve Davies schrieb: On 23 August 2010 18:32, Stefan Schmidt s...@sil.at wrote: hello, have you allready tried strace ? you could just easily start asterisk with this command: strace asterisk - Yes, I tried

Re: [asterisk-users] How to debug this specific issue?

On 24 August 2010 14:34, Steve Davies davies...@gmail.com wrote: On 24 August 2010 08:07, Stefan Schmidt s...@sil.at wrote: Steve Davies schrieb: On 23 August 2010 18:32, Stefan Schmidt s...@sil.at wrote: hello, have you allready tried strace ? you could just easily start asterisk

[asterisk-users] How to debug this specific issue?

Hi, I am happy with the usual GDB backtrace methods and so forth, but have an issue that I cannot work out how to trace on 1.6.2.10. If I use either the Bridge() app, or the manager Action: Bridge() in a certain scenario (Basically to bridge 2 SIP channels, like an attended transfer, resulting

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