similar to: No audio after a reinvite changing codec ----> with SIP DEBUG!!

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

We are having an issue with the SPA962/932 combo where the phone and the sidecar will reboot unexpectedly ? could be onhook, could be on a call, doesn?t seem to matter. I read that certain early firmware revisions could cause this so I?m running what was a week ago the newest available (5.1.18). A call to Linksys support suggested that I ensure that the phones are using a recent firmware version

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 directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite

Greetings all. First off, thank you for your time on this. I have spent literally 12 hours searching every forum and article I can find, and I'm going cross-eyed, so I need to bother everyone with this. I am running * 1.2.37, and I am trying to get the hints working, so I can turn one of my SPA962's LED's red when someone parks a call. I have used Button #3 on my SPA962 to

Greetings list, We recently installed some Linksys SPA962 + SPA932 sidecars into a client's offices. The BLF functionality works fine, but call pickup using the BLF is something of an issue. Following the advice on voip-info.org, I configured part of their dialplan as follows: exten => _**2XX,1,Pickup(SIP/${EXTEN:2}) exten => _**2XX,n,Dial(SIP/${EXTEN:2},15,tw) exten =>

Hi, I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft Regards ASLAY

I'm no programming wiz, so I need some hand-holding on this one... I have a couple of programs used to update my phone. First problem was that they showed the wrong IP address for my PC. This I fixed by editing the /etc/hosts file with the right IP. In order for the update to work, you must enter the IP address of your phone. Every time I do this, they come back with "invalid IP

Dear all I wrote a routine. At the end of each cycle of the loop I would like to save the result (plot) in a postcriptfile. Of course if I just use dev.print in the following way: dev.print(device=postcript, 'c:/Rfigures/plot_1.ps") I overwrite my results with the second cycle of the loop. I suppose there is a way to define the file name so that several plots are

Using control_work instead of config_work as the 3rd argument to container_of results in an invalid portdev pointer. Indeed, the work structure is initialized as below: INIT_WORK(&portdev->config_work, &config_work_handler); It leads to a crash when portdev->vdev is dereferenced later. This bug is triggered when the guest uses a virtio-console without multiport feature and

Using control_work instead of config_work as the 3rd argument to container_of results in an invalid portdev pointer. Indeed, the work structure is initialized as below: INIT_WORK(&portdev->config_work, &config_work_handler); It leads to a crash when portdev->vdev is dereferenced later. This bug is triggered when the guest uses a virtio-console without multiport feature and

Hi list, is there a way to achieve in asterisk (version 1.4.x) the behavior described below? * a caller place a call to an extension, and I want the caller hears the extension ringing for some seconds, and then hears the music on hold (or a courtesy message) _in the same call;_ * the called extension must continue to ring until answered. With the m(...) option in the Dial

Hi, I've been asked sometimes to tailor call history features embeded in SIP hardphones. For example, a cutomer wanted internal call to be taken out. Another wanted calls to sorted according specific criteria. 1. Have you identified a phone offering the possibility to display as Call History, an XML list produced on a distant web server ? With this feature, you would simply have to tell the

Hi, I've been searching a solution for this with no luck for the last 5 days, so I thought I'd finally need help. We have a ADS primary domain, samba 3.0..4-6.3E on RHEL as a domain member. The parent domain say MEDIA.COM has a child domain CHILD.MEDIA.COM. People from the parent domain can access tha shares with no problem , but I can't find a way to make users from the child

I am looking for a simple way to send occasional faxes via the FXO port on my SPA3102 -- without having to connect a fax modem to an ATA. In an ideal world, this would be some sort of "softfax" that runs on my Linux desktop and talks (via Asterisk) to the SPA3102 with T.38. This is one of those things that I thought would be relatively straightforward, but a couple of hours of Googling

https://bugzilla.mindrot.org/show_bug.cgi?id=3193 Bug ID: 3193 Summary: Add separate section in sshd_config man page on Access Control Product: Portable OpenSSH Version: 8.3p1 Hardware: Other OS: Linux Status: NEW Severity: enhancement Priority: P5 Component:

Hi all, There are a lot of posts around the web about my question; unfortunately I have not been able to get any of the solutions to work. I'm using Asterisk 1.6.2.8 under CentOS 5.5. I'm trying to get call pickup working for the secretaries that monitor their bosses' phones. The BLF and the speed dial works great on the Linksys phones. Call pickup is the problem. My features.conf

Hola, LLevo buscando desde hace tiempo como hacer el Goodness of fit test en R. Es decir, me explico, intento hacer una cosa parecida que se hace en Minitab, por ejemplo, yo tengo un conjunto de datos, y lo que quiero es sabes que tipo de distibución es, en minitab se hace un histograma para ver si se ajusta bien o no a la campana de Gauss, luego vemos si aproximar la distribución de la muestra

Hi, I'm trying to use samba with ldap because I've several server with samba and it would be impossible to add the users to every server. I'm using samba from debian unstable andfollowing the smb-ldap-3-howto but I'm having this error: Sep 12 12:26:08 tele nmbd[24587]: [2002/09/12 12:26:08, 1] nmbd/nmbd_processlogon.c:process_logon_packet(100) Sep 12 12:26:08 tele nmbd[24587]:

Dear all In order to clearly mark values wich are larger than a treshold value, I would like to color the surface below the line given by plot (yy~xx). To color is only the surface between abline (treshold) and yy if they are larger than the specific limit. I guess I can use the function polygon, but I can not find any valuable solution. I'm grateful to you for an advice or an example.