similar to: dialling out

I am not having any luck placing out going calls I dial the number 08 82420173 ( our outside line ) But all I get is engaged signal and log this. Oct 14 08:40:14 DEBUG[16401]: File pbx_wilcalu.c, Line 65 (autodial): Entered Wil-Calu fd=20 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 548

When dialling in and dialling my extension, when answered I get " Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[20499]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3) -- Hungup on vpb/1-3 complete --

When setting include => daytime|9:00-21:00|mo-fri|*|* How does this determine what is different between 9 AM and 9 PM And after hours ??? I want different hours on Saturday and Sunday And a different welcome message after hours Any help appreciated Regards Mick

Has anyone gotten the voicetronix boards to work with Asterisk, what would it take? Or does anyone know where I can get 4 ports or more fxs PCI cards that do work with asterisk? Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

<FONT face="Default Sans Serif, Verdana, Arial, Helvetica, sans-serif" size=2><div>I am using some of the sample recordings included with asterisk for my conferencing application.&nbsp; They seem to be rather choppy at times, with the overall quality not being quite where I'd like it.&nbsp; I think I saw in a previous mailing list on here that people suggested we

Hi all How can I make * ring one phone then if no answer Go to a different extension ?? Any help always appreciated Regards Mick

If you ring into * and leave voicemail It does not reset the line Any ideas would be appreciated Regards Mick

Has anyone got a mobile wireless phone working with * yet ???? Is it possible to use the Cisco 7920 with skinny ???? Regards Mick West

Hi, I have a Cisco 7920 that I'm trying to get working with my * box. When the phone boots it requests XMLDefault.cnf.xml and SEP<MACADDRESSHERE>.cnf. I assume I set the line number, etc in the latter of the two. However I cannot find any reference to how this file is structured. Anyone know? I assume this is why I'm getting the errors below: Oct 17 19:47:24

Thanks for the reply Trey, should of said I'm using chan_sip. Regards Mick On 16 Feb 2016 18:03, "Trey Hilyard" <kctrey at gmail.com> wrote: > Are you using res_pjsip or chan_sip? > > For PJSIP, it's as easy as passing the parameters to the Dial. For example: > Dial(PJSIP/${ARG1}\;phone-context=mydomain.com at pjsippeer,60) > > I am pretty sure it was

Hi, I'm trying to boot 64bit version of syslinux.efi through network. But booting is freezes with following message: Getting cached packets My IP is X.X.X.X the last file that was requested through tftp is syslinux.efi. I'm using precompiled binary from official syslinux-6.02.tar.gz. Here is the dump in ASCII - http://brom.in/dumps/syslinux-20131220.txt Any help is appreciated.

hi, Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call. ie, the operator answers the call, and presses hash key to transfer, and then enters the extension number, some times, it timeouts too quickly before the operator enters the whole extension number (may be bcos the operator is slow). I tried the following, but it doesn't seems to be helping

Hey all, I'm having problems reliably dialling out my FXO card. About 30% of the time I'll get a "your call cannot be completed as dialed". I'm thinking it might be the dialling speed, but I can't find any configs that change that setting. Any suggestions for troubleshooting? Thanks, Brad Waite

Hello, I'm wondering if it is possible to ring X number of extensions simultaneously, and each answered call can be handled with some code. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up. I hope this makes sense.. If not please say.. Many thanks! Andy -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello everybody, one of our customers which wants a soft transfer between his old pbx to asterisk and sip. the setup is as follows: telco <---pri---> asterisk <---pri---> legacy pbx everything is fine exept that when dialling from the legacy pbx it takes about 3 seconds before the asterisk start to dial. where does this delay come from? has it to do with

Hello, On Sat, Dec 21, 2013 at 6:07 PM, Geert Stappers <stappers at stappers.nl>wrote: > Op 2013-12-20 om 12:05 schreef Roman: > > Hi, > > > > I'm trying to boot 64bit version of syslinux.efi through network. But > > booting is freezes with following message: > > Getting cached packets > > My IP is X.X.X.X > > That is a very strange IPv4

Hi all. I am seeing incoming calls from digital lines (mobiles e.g.) dialling my main number + 3-digit extension just fine ("Accepting voice call from '11234567' to '250' on channel 0/1, span 1"). The problem however is with calls from analog lines: "Accepting voice call from '13331846' to '25' on channel 0/1, span 1" * just sees 2 digits, not

Hello, I'm experimenting with Overlap Dialling in asterisk 1.6. I've enabled this in sip.conf and on the SNOM 300 phone. My problem is that asterisk dials out as soon as it matches an extension without waiting to see if the user is going to type in more digits. Is there a way to set a timeout per channel or globally? I'd like Asterisk to wait for a few seconds once its found a match

hi there was wondering if anybody knows this.. have successfully installed shady dial and the agent is now logging in successfully i've enabled postgres debugging and i found out that no request had been made by the shady dialer to query the database for the numbers... also i am able to login to the queue simply by entering the agent id... it doesnt ask for the password...it simply plays the

Hi :) I have a little 'slow dialling' problem. When I dial, e.g. 200# for the Asterisk 'echo test' demo application from my PBX extension 1010, I see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 then exactly 3 seconds elapses, and