similar to: Dial() command h and H options for SIP channel

Displaying 20 results from an estimated 8000 matches similar to: "Dial() command h and H options for SIP channel"

2007 Sep 25
4
Anyone else having problems with the list
I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian
2007 Jun 06
4
Slow list
Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998
2008 Jan 10
3
OT - Is handover included in DECT GAP ?
Hi, Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support roaming and handover and are these functions transparent for handset (then, these functions are implemented in DECT base stations) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080110/4254f602/attachment.htm
2005 Jul 20
5
Grandstream GXP2000 resetting all the time
All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly
2008 Sep 12
2
Setup speed dials on Cisco 7921
I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Aug 15
2
No translator path exists for channel type MGCP & Comfort noise support incomplete
ONLY ON MONDAY! Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this?
2005 Jul 25
4
Fritz PCI card in ptp mode with chan_misdn
Hello ! I would like to get working a Fritz PCI card using chan_misdn operating in ptp mode. Afer compiling mISDN into the kernel and building chan_misdn Asterisk stops loading with : [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables
2005 Sep 02
6
Looking for better "Follow Me"
Hi everybody :) I am a new member here and hope that someone gives me a hint for my problem: Let's say I am at work and my SIP phone (KPhone in my case) is connected to my private Asterisk. I want to call my wife at home so her SIP phone rings. She does not pick up the phone (maybe she is somewhere in the house and has to run to the phone) so after 15 seconds her cell phone should ring.
2006 Mar 03
9
Program Buttons on Cisco 79xx Phones
Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware...thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060303/e5e63834/attachment.htm
2005 Sep 21
7
add 0 (zero) to incoming callerID - how?
I have an asterisk box and SIP / IAX2 phones. To call out, users have to add 0 (zero) before a real telephone number. That means, that if they want to call someone that has a number 123456, they have to call 0-123456. Simple, right? This has a serious drawback though - when someone calls us from the number 123456, we see the callerID 123456, and we're unable to use the callback/redial
2007 Oct 23
5
Asterisk under VMWare
Anyone had any experience with an Asterisk server as a VMWare virtual machine?
2006 Jun 28
9
Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
Hello, Here is a breakdown of the issue I am experiencing. I have three remote employees, in various states, who have Polycom 501 phones. They are unable to receive incoming calls after a few minutes of the phones being plugged in. They work immediately after being plugged in, but they lose the ability shortly thereafter. They can always make outbound calls, but only to real phone numbers, not
2006 Feb 25
2
sipgate.de question
Hi, Anyone here using sipgate.de ? It worked for months, but for a couple of days now I'm unable to register with them. My account is ok, because I can login to the website. Asterisk keeps showing me: Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n) I looked at the sip debug stuff, and all I
2006 Feb 12
2
IP phone with many speed dial buttons
Hello, I'm looking for IP phones with at least 10 or so speed dial buttons. Can you recommend something which works with Asterisk and does not cost fortune? An option can be analog phone combined with ATA adapter. So hints for good analog phones (EU) are welcomed as well. Thank you, -- David -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Aug 16
2
SIP "agent" phone w/ headset
I have a call center where we're looking at converting it from a traditional PBX w/ digital phone "agent" sets (keyless phones) that have headsets to a SIP based environment. I am having trouble finding anything on the market that resembles this in the VoIP world. For reference, we're currently using Inter-Tel Agent Sets, which are basically a digital phone with out any keypad,
2007 Feb 28
1
extensions.conf & sccp.conf howto call external number
Hello @List, i'm using a Cisco 7970 / 7914 phone with Asterisk 1.4 & sccp part of my sccp.conf type = 7914 (Cisco 7970 with 7914 phone extension) autologin = 117 description = Test speeddial = 10,Test (10),10@mycompany my speeddial line is for internal call howto call a external number eg. +xx467584933 ?? i want to press a button and call directly out thank you greetings, Daniel
2007 Jul 19
1
Idefisk softphone - official 2.0 release - Zoiper
Hello guys, The so expected 2.0 release of Idefisk 2.0 softphone is a fact. Idefisk and Zoiper became one - Zoiper 2.06. Here are some of the features: SIP and IAX, TCP, TLS support, Multi-language support, Automatic provisioning (XML), URL handling, Outlook Integration, Native conferencing, API, Changeable number of lines.... You could read the complete Press Release here:
2007 Dec 23
1
Asterisk 1.2.26 badly broken?
After upgrading from 1.2.25 to 1.2.26 i noticed that IAX -> IAX calls always result in Asterisk just exiting without any message. Asterisk also seems to die when using a TDM400 with 2 FXO modules, placing 2 outgoing calls on both lines as Zap/g2 and then trying to make a 3rd call. Went back to 1.2.25 for the moment
2006 Jun 26
3
This is getting really annoying - re: POSTFIX
What on earth is going on with the list?!?! Some of my messages never make it... then days later I get something like this back: Final-Recipient: rfc822; asterisk-users@lists.digium.com Action: failed Status: 5.0.0 Diagnostic-Code: X-Postfix; mail forwarding loop for asterisk-users@lists.digium.com
2006 Oct 19
1
bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now
Bristuff has been updated; http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz -- Vidar