similar to: Setting gain levels with mISDN

The value of 14800 is correct. I had issues with my TDM400p with 2x FXO's installed and using the Xlite client. I could not get the echo stable at the initial call. Changing to a hard phone made everything work correctly. I still had problems with the off location I called, but mostly worked great. To see the TX gain I created an extension with the Milliwatt command attached to it and called

Hello *, I have a strange problem with the MAPI proxy AstManProxy: sometimes it happens that I send a request and I receive a response to ANOTHER request that it got in the frame time between my request and my response. Did anyone else notice this behaviour? How can this be solved? I've been reading the source code, but I didn't find a solution. Thanks in advance, -- Dr. Andrea

Hello, We're in the process of setting up an Asterisk server, and are having echo problems. We have a Digium TE110P, and have tried the MG and MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and training times, and with both trunk and 1.2 branch versions of Zaptel, Libpre, and Asterisk. In all cases, callers from the PSTN hear their own voice echoed back after 1.5-2 seconds;

Hello everybody, I've got a 56k usb modem, lsusb says: Bus 002 Device 002: ID 0572:130 Conexant Systems (Rockwell), Inc. I'd like to let it work with Asterisk. I think that I should use chan_modem and/or chan_modem_bestdata, but I found little or no documentation. Can anybody please post some instructions? Thanks in advance, -- Dr. Andrea Spadaccini Multimedia Technologies Institute

Hello everybody, I'd like to have more detailed records for calls related to queues. For instance, if A enters in queue X, waits for Y secs and then talks to peer Z for T seconds, I'd like to have two entries in my CDR: - src: A, dst: X, duration: Y, state: ANSWERED - src: A, dst: Z, duration: T, state: ANSWERED This independently from how many peers the Queue app calls without success

Hello *, do queues allow me to set an announce like the A() option of the Dial() cmd? The announce that I've found is a message that is heard by the caller. I'd like to send a message to the member of the queue that picks up the call. Thanks in advance, -- Dott. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l.

Hello everybody, I'm trying to understand how can I set the MoH class for parked calls. I set the "incoming" class for calls, and it works correctly. When I park the call, the music on hold is ok, but when I close the communication on the parking side, the parked call gets the default music on hold class. Can someone explain me what's going on? Thanks in advance, -- Dott.

Hi all, i downloaded and installed mISDN with 2.6.8 kernel, but when i try mISDN-init scan (or config) i get this error: [!!] FATAL: bc not in path, please install. Anyone can help me. Tnx Giordano -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.0/639 - Release Date: 18/01/2007 18.47 -------------- next part

Hello all, I know this has been covered on list but can not find the answer I need, lots of references to no authority found, but none with an answer. I have two * servers, one behind firewall with nat the other on a dmz with nat. Both servers register with each other successfully. home is today's CVS-HEAD away is Asterisk 1.0.7 on away: Registered to '165.xxx.xxx.xxx', who sees

I'm having a bit of an issue with one of the bargain x100p clones, and I'm not sure what the right approach is. My symptom started as way loud offset delayed echo from voip hardphones -> PSTN through the clone card. I played with and then learned everything I could about echo cancelling, and have managed to bring the echo down to acceptible levels, where it's not too intrusive. In

I have been trying to adjust the gain as per this document without any success: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html I have a PSTN and VoIP (SIP) connection via *. I disabled all echo cancel/training in zapata.conf and set tx/rxgain to 0. I then changed my extensions.conf so that when I call the VoIP line from the PSTN line, it plays the Milliwatt

Hey Guys! We had echo issue before so we replaced old PRI card with Sangoma A102D HWEC. Now my question is i set rx/txgain level 0.0 default do i need to touch this value or default is best. I have read on google and people say it should around 14844 on ztmonitor for rx/tx level same. I just use milliwatt and test my default 0.0 rx/tx level and it come around 4600. Do you think i need to make

Hey guys, I'm having some trouble with gain settings using a Wildcard X100P and Zaptel 1.4 Beta 2 on Asterisk 1.4 beta 3. My transmit volume between an IAX client (idefisk) and a POTS device through my 1.4 box is great, but my receive volume is terrible. I can hardly hear a word they say. I've cranked my gains up to 100, inserted the wcfxo module with boost=1, but haven't had

Hi, Until now I've been using my Digium B410P cards with misdn 1.0.x. I would like to upgrade my systems and am now wondering which is the "best" route to take: - use the latest release of misdn v1 - upgrade to the latest "stable" kernel and use the built-in misdn v2 - use misdn v2 as a seperate package (disable misdn in the kernel) - use dahdi's support for misdn

Hi, I try to compile mISDN 1_1_9_2 (kernel: 2.6.32.13). I had the following compiling error: make[1]: Entering directory `/usr/src/linux-2.6.32.13' CC [M] /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.o /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.c: In function ?mISDN_register_sysfs_obj?:

Hello I fixed my problem. I changed user and group in /etc/mISDN.conf: <devnode user="asterisk" group="asterisk" mode="644">mISDN</devnode> now it works again! Thanx 4 help! ;) Ingrid -- "Bonnie & Clyde der Postmaster-Szene!" approved by Postfix-God http://wetterstation-pliening.info http://dokuwiki.nausch.org --------------

Hi i've installed this software: ******************** SOFTWARE mISDN-1_1_7 mISDNuser-1_1_7 Asterisk-1.4.15 ******************** SOFTWARE misdn is correctly loaded by misdn-inist start Here there is the misdn.conf (copied from an existing and working installation with Asterisk 1.2.x and one BN8S0) ******************** MISDN.CONF [general] misdn_init=/etc/misdn-init.conf debug=0 bridging=no

Hi All I have noticed that when calling over mISDN channels the call itself is made over the channels listed in misdn.conf eg: 1,2,3,4 as misdn/1, misdn/2, misdn/3, misdn/4 etc. However, when looking through the CDR's the call gets logged as misdn/0, misdn/1, misdn/2, misdn/3, misdn/4, misdn/5 etc. I also noted that busy, failed, congested calls link mainly to misdn/0. The only logic I

Hi, I'm installing Asterisk on Ubuntu 6.10 When I first compiled the zaptel package I used: make clean make make install So far so good, but the following command failed: make b410p I did some digging on google and found a guide on how to install it manually, but the result was the same. I got these files ftp.digium.com/pub/telephony/zaptel/b410p/misdn-b410p.tar.gz

Hi All, We've got msidn configured: Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1 (Euro ISDN) -> childcnt: 2 -------- mISDN_close: fid(3) isize(131072) inbuf(0x8fd5060) irp(0x8fd5060) iend(0x8fd5060) and running on Asterisk 1.4.21.2: pbx*CLI> misdn show stacks BEGIN STACK_LIST: * Port 1 Type TE Prot. PMP L2Link UP L1Link:UP Blocked:0 Debug:0