similar to: [Asterisk-Dev] q931 dial errors

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

Hi Guys, We have a problem with Caller ID not being displayed. I want to test everything to see where the problem is with the incoming Caller ID and why it's not displaying. I am tracking this down to "Presentation prohibited of network provided number" even though the Caller doesn't use *67 and even though they haven't asked their provider to block their CLID for outbound.

Hi Guys, Can q931.c be re-compiled using gcc or something else without the need to re-do the whole libpri? Some changes were made to q931.c and I want those to be reflected in .a .o .so .lo files as I think those are the files read by Asterisk vs the .c file. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi guys: We have just rented a server and setup a BBS for asterisk discussions at http://bbs.us.xgforce.com feel free to join. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050718/07587c97/attachment.htm

Hello Please help me, i need transfer a call in asterisk to other telco number and free the channel. Can i do with any q931 function?. Thanks a lot Aris... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090528/2bcd93ae/attachment.htm

Hi, does anyone know if its feasible to run asterisk with a PRI card but not run any q931 signalling.. basically push calls down the PRI and tell asterisk in some other way to pickup a particular Zap channel? Steve

- I've just installed 0.6.2, & I would like to see the q931 messages going back & forth. I turned on debugging with "h323 debug toggle", which the README says is "very verbose", but I don't see much. Is there a way for me to see more debugging information, like the "debug isdn q931" of IOS? Or am I missing something? Thanks, Glen IAS

Hi all, I'm looking for a way with any asterisk-version with TE410P (cpe EuroISDN, Q931) for sending an INFORMATION ELEMENT KeypadFacility, eg. *87, during a connected call to the PSTN switch. Are there existing functions in asterisk to generate & send such IE ? If not what existing modules would be best to derive from? TIA, Bruno -------------- next part -------------- A non-text

Hello! I've got a question regarding the Q.931 Setup-field Calling Party Number. It contains five things: Type of number, Number Plan, Presentation and Screening indicators and the actual number. Our provider uses some of those to decide if the numer should be presented or not to the outside world. I've done a crude hack in our GnuGK to always change those so that our numbers are

I there, I have some trouble to do working caller id name for outgoing calls on the PRI we just installed. Caller id name work on incoming calls only. Caller id number work on incoming and outgoing calls. The provider, Goup Telecom, said that's in what i'm sending. They said that I send the cid name in ascii code and to do it working, I need to send it in hex. So I take some traces

I've having problems with a BT 2 span ISDN-30/Digium TE205P asterisk setup with outgoing calls not completing and requiring an Asterisk reset to 'unstick' span 1. Sorry this is a bit long but I'm completely out of my depth :-( This system has been in use for some while and I recently upgraded it to asterisk 1.4.24, zaptel 1.4.11 and libpri 1.4.9. I didn't change

I have a user who's system crashes on pri hangup request. Tried 1.4.19.1 and 1.4.20 as well as the latest libpri no change Progress is as follows...... < Supervisory frame: < SAPI: 00 C/R: 0 EA: 0 < TEI: 000 EA: 1 < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] < N(R): 025 P/F: 1 < 0 bytes of data -- ACKing all packets from 24 to (but not including) 25 -- Since

Dear All, I am trying to setup an ISDN line from local telco on a digium card. The problem I am facing is that I am not getting any caller id from the telco. They say that they have enabled caller id. Please help me out. My zapata.conf -------------------------------------------------------------------------------------------------------------------- [trunkgroups] [channels]

I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a Meridian Option 61C. Calls either way drop with error "Channel 0/23, span 1 got hangup, cause 100". Can anyone offer insight into the cause and solution/workaround? (I tried upgrading to Ast 1.4.13, and upgrading matching zaptel & libpri, put the problem is identical). For testing, I tried a call from the

Hi I have recently upgraded my Asterisk system (from 1.2 to 1.4) and I have started to notice the following messages when I recieve a call on my Zap channel :- [Mar 11 09:20:17] WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 bytes LLC I have a single PRI ISDN 30 link to a Siemens Realitis DX. Here is my zapata.conf :- [channels] echocancel=no echocancelwhenbridged=no rxgain=-5.0

Hello, I am having odd issues with a new PRI based installation. Outbound calls work for all numbers except those that terminate at Sprint! The telco is new to PRI (this is in the Caribbean) and say that Sprint is rejecting the calls, and asked for our PRI settings so they can work with their switch manufacturer. We are using the bare minimum default settings, but I can imagine that a

I have a server running Asterisk 1.4.14, Zaptel 1.4.6, Libpri-1.4.2 on a CentOS 5 server. The server has a single TE110 card connected to a provider called Alestra in Monterrey, Mexico. Since we installed everything we have been having problems dialing certain numbers, those numbers always fail when dialed from Asterisk but if you dial from your cell phone they always go through. I once has a

I have an asterisk with a TE110P configured as T1 which is behind a PSTN gateway. This gateway has an E1 to PSTN and a T1 to asterisk. This T1 is configured as Network and * as CPE. Every call I receive in E1 gateway is directly switched to asterisk using T1. Remember E1 is alaw. Both E1 and T1 have Natural Microsystems boards with a very simple software. When I call to E1 asterisk signalling

I've been working on an asterisk box at work for a few weeks now, things were finally starting to sail smoothly until I hit this head scratcher this morning. It's a rather intricate problem, so bear with me. Heres the scenario. What works: If I call from my sip phone -> sip phone everythings ok If I call from sip phone -> external pots number ok as well If I map one of our

Hi, I have a partial working installation with zaphfc. Incoming call : For incoming call, seems work fine. But the sound is very bad with bounce short crashing sound. Same sound with echo cancel off or on. SDA work fine. Another problem, it's seems that's zaphfc don't reset correctly the line. I have one of my D channel how was busy even after stop communication. Outgoing call :