similar to: SIP Lines

Hi, I am trying to parse a date (that is read in as a factor) and add it to a dataframe. The length of the parsed date is shorter than the length of unparsed date and I therefore cannot add it to the dataframe: > x [1] 20030807 20030807 20030807 20030808 20030809 20030809 20030809 20030809 [9] 20030808 20030808 20030809 20030808 20030808 20030819 20030819 20030821 . .

When I place a call using Iconnecthere as my sip provider, I hear no ringback when making a call. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm

Ok I have tried to post to this list server but have just gotten the automated reply saying the moderator has to approve it to the list first which was my mistake for sending from the wrong email account. So if the moderator finally approves my questions and you see the same post again "Sorry". My situation is this: I havn't installed Asterisk yet but am curious the general way

Does anyone know if this unit works with Asterisk? Thx. B.

Does anyone have any insignt on this? Any client programs that could be used? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030925/0e9e9920/attachment.htm

Hi all, I installed the estara softphone and had no problem registering it with asterisk. I could make calls to other hardware SIP phones (Cisco 7960) from the softphone, but I couldn't call the softphone from the Cisco 7960s. The asterisk console gave me an error message saying "unable to create channel" to my softphone. What could be the problem? I searched the archive with no

Hello, I resolved my echo issue using grandstream/estara etc etc sip phones and wcfxo interfaces from digium. I swapped out my via kt400 based msi kt4vl motherboard for an asus p4pe? i845? based motherboard and the echo has completly gone away along with aggressive suppressor option in the makefile. I hope this helps others. Brian J. Schrock Anistone Technologies, LLC 6926 Avery Rd. Dublin, OH

Hello people. I would like to know if someone knows about any applet to include in a web page to start calls. What I am looking for is something that doesn't allow users to change numbers, or any other option, so I can include it in my web page and force them to call to me and no one else. I have tried JIAXClient, but it allows people to call anywhere, and what I want is just a configurable

Hi, Companies that offer telephone service to the public are obliged to offer tapping to all kind of authorities. Does anyone know how to tap in Asterisk? I.e. record (or copy) a conversation based upon their telephone number? Thanks Dan _________________________________________________________________ Stay in touch with absent friends - get MSN Messenger http://www.msn.co.uk/messenger

Can anyone recommend a client / phone that runs on Windows XP, with either a sound card or some other hardware? Ideally free, but does not have to be. Thanks...

Does anybody know why I can NOT hear the MusicOnHold - using SJphone on another PC in our network (normal playback is not a problem) . See the * output and the line configured in extension.conf below (also mp3player does not function) Any suggestions? *Asterisk output:* *CLI> -- Executing WaitMusicOnHold("SIP/jeroen-bf54", "30") in new stack --

On onsdag 29 augusti 2018 kl. 10:00:39 EEST Sandro Bonazzola wrote: > 2018-08-28 13:52 GMT+02:00 Dag Nygren <dag at newtech.fi>: > > > We have a desperate need for TPM support and: > > > > 1. Tried the "standard" distro install. linvirt supports > > TPM passthrough but kvm-qemu barfs: > > "unsupported configuration: The QEMU executable

Does anyone have a copy of spc.exe they could send me? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061117/ac0b6a44/attachment.htm

I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error

Does anyone know what version of Asterisk Switchvox uses, and if it is modified in any way? FWIW, I am dealing with a provider that claims compatibility with Switchvox but not Asterisk for their SIP trunking service.

On onsdag 29 augusti 2018 kl. 15:37:47 EEST Alvin Starr wrote: > On 08/29/2018 07:38 AM, Dag Nygren wrote: > > > On onsdag 29 augusti 2018 kl. 10:00:39 EEST Sandro Bonazzola wrote: > >> 2018-08-28 13:52 GMT+02:00 Dag Nygren <dag at newtech.fi>: > >> > >>> We have a desperate need for TPM support and: > >>> > >>> 1. Tried the

I used to be able to pass dial strings to IAX2 providers with # characters at the end of the string. This is how we end dial strings for international calls. So, I would like to be able to selectivity chop off any # characters at the end of string, only if they exist. Basically as follows (chopping off the leading '9' with ${EXTEN:1} syntax: EXTEN from Phone EXTEN for Dial String

Asterisk 1.2.24 seems to crash repeatedly under any substantial call load (and sometimes without a substantial call load - just one SIP leg is enough to do it) when using the G.729 pre-compiled binaries from: http://asterisk.hosting.lv/ As per: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing Time to crash is variable, but seems to require at least an hour of production performance

On onsdag 29 augusti 2018 kl. 15:37:47 EEST Alvin Starr wrote: > You could try using Xen. > A quick search implies that Xen from 4.3 onward will virtualize TPM. > I am not sure if the libvirt drivers for xen will support the feature > but some work around may be possible. Nice attitude and helpfulness in this list! Just had a look and it doesn't seem to be that an intrusive

A long time ago (Asterisk 0.x, 1.0.x) my experience is that there were alot of interoperability issues, a common troubleshooting issue was to make sure all endpoints where using the latest version of Asterisk. I have not seen these issues in a while. However I've been working with a customer of mine and this ITSP called IP Communications (IPComms.net) well turns out we have had constant