similar to: So many Asterisk Patches - Which do I choose and use?

Pulled Stable. ( CVS-v1-0-01/12/05-11:43:32) This does compile cleanly. Now I just have to weed out the features from CVS-HEAD that I was using and patch them into STABLE. -brian On Wed, 12 Jan 2005 at 11:37 Paul Rodan (asterisk@glitch.cc) wrote: > Problems with CVS-HEAD, but what about CVS-STABLE? > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com

I've just installed asterisk on a Debian Linux (apt-get it) And i have placed two sip phones in sip.conf and i'm testing parking with them I have phone1-SIP/1000 and phone2-SIP/1007 The following happens if i park from calling party and everything is OK 1. Pickup Phone2 and call to Phone1 2. Talk 3. Phone2 dials #700 and parks the call (it is placed in 701) 4. Phone2 is hangup 5. Pickup

Hi, Reading from www.voip-info.org, i can see that "Junghann's chan_capi is now part of bristuff <http://www.voip-info.org/wiki/view/Bristuff> , as of version 0.3.0-pre". What does that really mean ? Shall I understand I can share a Junghanns QuadBRI board between 2 CAPI-enabled software (like a 0.3.0-pre bristuffed Asterisk for instance) ? If positive, what are the pros and

Hi! Asterisk in Debian/Lenny claims to be bristuffed, not? At least the the Debian patch tracking system shows the bristuff-patches: [1] http://bit.ly/bRRHe7 We have a QuadBRI-Card and recently needed support from Junghanns.net but they refused telling us there is no bristuff installed because of the show version output: *CLI> show version Asterisk 1.4.21.2~dfsg-3+lenny1 built by pbuilder @

Hi, I am currently building a 1.4.4 Asterisk box for a client and they are interested in GSM functionality. Does anyone have any experience with a GSM card, preferably Quad Span (4 GSM modules or higher) for use in the UK. I have seen the Junghanns* version but I am not keen on the limitation of having to use a BriStuffed version of Asterisk. Do Digium make one ? as I am unable to find

Hi, I'm getting unusable DTMF detection with DISA on incoming ZAP channel (bristuffed *) on quadbri from GSM gateway. DTMF detection works ok in normal ISDN incoming line. How can I check what's going on ? What settings to check ? Anyone with more experience on such scenarios ? Thanks in advance, regards, Rob.

Hi, I'm getting unreliable dtmf recognition (it works fine for 4-5 digits, errors (duplicates) on more), when transferred inband from gsm gateway to NT port of quadbri under bristuffed Asterisk..... Since Asterisk is claimed to have good dtmf recognizer, I suspect there are some settings to workarouned... I've tried dtmf relax, but didn't help, so I suspect gain settings.... Is

Hello, I have a problem with the Bristuffed version of Asterisk. I have tried Bristuff-0.3.0-pre-1m,n,o,p (Asterisk 1.2.6 to 1.2.7.1) but they all have the same problem it seems: The setup: A machine with a single hfc-s PCI BRI adapter running Gentoo 2.6.15. Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1) installed and working perfectly. Grandstream gxp-2000 as a SIP phone, and a normal mobile

Hi, I'm using spandsp-0.0.3 [http://www.soft-switch.org/downloads/snapshots/spandsp/ spandsp-20061116.tar.gz] on a bristuffed asterisk (1.2.13) [http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0- PRE-1v.tar.gz] libtiff is at version 3.6.0 Running on: Linux router2 2.6.17-2-686 #1 SMP Wed Sep 13 16:34:10 UTC 2006 i686 GNU/Linux Debian testing distro. I've tried many

Hi, I'm trying to setup a small BRI ISDN <-> voip gateway. The ISDN card is based on Cologne chipset, so I try set it up with zaphfc. The versions i'm running: kernel-2.4.27 Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e zaptel modules 1.0.7 zaphfc is from bristuff-0.2.0-RC8e When I'm doing the insmod on zaptel, zaphfc, zaprtc: Zapata Telephony Interface Registered on major 196 PCI:

Hello, I recently bought a Junghanns Octobri Card. I have some problems with this card to make outbound calls but I can receive calls. I have 3 lines to PSTN and 3 lines to my existing PBX FRANCE TELECOM <-- OctoBRI --> Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h <-- OctoBRI --> PABX e-Generis <----> ISDN Phones |

I'm using bristuffed Asterisk with ISDN/ZAPHFC I have VERY ugly (outgoing) sound through ISDN/HFC if echocancel=yes in zapata.conf, but without echocancel I have bad (incoming) echo Through PSTN/FXO sound is ok with or without echocancel. I tried other echo cancellers (in zconfig.h) two times: ECHO_CAN_KB1 (this was default) ECHO_CAN_MARK2 ECHO_CAN_MG2 after any change I compiled (make

Are IVR and "Auto Attendant" interchangeable terms? They both do the "Press 1 for" thing. Sales is asking me how to word it and I've always used both terms interchangeably.

Anybody have a guide to the Cisco 79xx phones? One that I can give the 7 or 8 ppl in my office so that they can stop asking me questions. I was going to type up a basic guide but then decided I don't want to reinvent the wheel, one of you may already have one. I tried to use Cisco's guide but it's for their own protocol, a lot of options are different or rearranged. I need a basic

I have a serious problem that repeats very often, after 30 - 50 calls and I can only solve it by stopped and restarting * :-( After a while, * seems to loose track of something. When an ISDN call from PBX needs to go to the Telco, I get 'Ring requested on unconfigured channel 255/255 span 2' It's always channel 255/255 ??, the 'span' number is random Setup: * between Local

Using FireFly, all other codecs but G711 Ulaw is selected. But whenever I place a call, I get: Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Umm, wtf? I thought Inband was ONLY supported on G.711 u-law. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I need advanced Asterisk training in the SouthEast area of the U.S; I don't need to know how to install linux and Asterisk and compile the modules and load them and such. I don't need to know what extensions.conf does or sip.conf does; What I do need is a better understanding of what every single little option in sip.conf or iax.conf does, and I need to learn a lot more about all the

Is there a way to monitor a call mid-stream? I did look on the Wiki and found that AstGUI can do it, but it's a bit of an overkill. What I want is for a customer service rep, sitting in front of a Cisco 7960, to be able to hit a button (either on their phone, or maybe a specific webpage) that will start recording the call from that point on. I'm thinking the services button on the

I have a problem with CDR recording in Asterisk 1.2.x. This is the situation: An Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1 with florz) machine with a single HFC-S ISDN BRI card. I log the call records to both the Master.csv and MySQL. The problem is that when an incoming call from the ISDN line is logged to the CDR, the "src" and the "clid" field show up as something like

We're having a horrid problem with our asterisk setup. Sometimes calls just go dead - we can't hear what the other end is saying. (I think they can't hear us either). The call doesn't hang up until one of the callers gets bored. Internaly we use Thomson ST2030 SIP phones. Externaly we have 3 ISDN BRI lines (6 channels total), connected to an Eicon Diver server card (4BRI).