similar to: Asterisk Outbound Calling Warning: Unable To Forward Voice

I am having a problem with SpanDSP. What happens is when I send a fax to SpanDSP the fax message seems to fail in the training phase. I think it's a timing error, however I have no idea about how to rectify the problem. I have included a copy of the log below. I am using a Digium TDM-400P card with 2 x FXO ports and 2 x FXS ports. The fax is connected to one of the FXS ports (Zap3). The

After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from CVS, Asterisk crashes on startup with an apparent MySQL (res_config_register) error: # asterisk -vvvgc > asterisk_startup_error1.log asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: un defined symbol: ast_cust_config_register The log is shown below. I've seen the posts

I am trying to connect audio through HDMI on a config. aplay - l gives: **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: VT1708B Analog [VT1708B Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 So I change my

Hello, I configured a main DC and secondary domain controller successfully. The only problem is that I want the secondary DC to Stand By, so most of the time it is shutdown. The problem is that when the secondary DC is shutdown on the primary DC I receive continously the following error which is filling the logs: Feb 1 14:23:56 saturno samba[3217]: [2014/02/01 14:23:56.021591, 0]

Hello all, the scenario: Carrier ----S2M------ * -----S2M------Siemens | | SIP Clients and many other features With much help from the list, the PRI links are without alarms and inbound calls are working fine (from both: Carrier and Siemens). But I am not able to dial wether outbound nor to the Siemens PBX. I allways get the message: == Everyone is busy

hello, I have been using asterisk now for about 2 years now on a RH8.0 it is our main call gateway. I have on the box 3 T1 TDM cards connected to 2 Rhino channel banks (FXS) and 1 CAC Access bank I (FXO) with so many softphones and ATA 186s. It has been working good till today some few hours ago. i just discovered that there were no dialtone on the phones. Asterisk did not spit out any error, it

We are still having problems getting a Wildcard to work with a German E1 (PMX) interface. When starting asterisk it shows all B-channels starting up successfully (although our carrier told us only the first B-channel starts, if any at all). Incoming calls are not being signaled at all. (They seem to be intercepted by the carrier's switch, as no B-channel is up) Outgoing calls sometimes work,

Hi all, I have to asterisk-1.0.9BRIstuffed-0.2.0n boxes each equipped with a TE410P. Box A is connected with pri1 to the PSTN. Box B is connected with pri1 (cpe) to the Box A at pri2 (net). Now I want Box B to dial out to the PSTN tunneled thru Box A and have it get all ISDN indications in case of call failure, eg. unallocated destination number etc. But currently Box B always gets only

I am getting the following information and asterisk 1.0.7 is not answering the call. Any ideas? jerry ------------------ Sip read: INVITE sip:2828;phone-context=cdp.udp@qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:3173241052;phone-context=+1@qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed To:

I downloaded spandsp0.0.6pre17 I download http://sf.net/projects/agx-ast-addons for app_txfax and found trunk/app_fax to be newer so I used that. spandsp compiled fine. app_fax compiled when loading I get: [Apr 5 08:55:54] ^[[1;31;40mWARNING^[[0;37;40m[7505]: ^[[1;37;40mloader.c^[[0;37;40m:^[[1;37;40m433^[[0;37;40m ^[[1;37;40mload_dynamic_module^[[0;37;40m: Error loading module

All, I am connecting to a CS 1000 nortel PBX. I can call out, I have limited success with call in. I get debug traffic that a call is coming in but I get the message "Unable to create/find channel". I was expecting that incoming calls over the trunk would be handled from my sip definition and goto the nortel context. It is not. Below is the actual incoming call debug information. I am

I'm using Asterisk 1.0.4 with AMP and Broad Voice. I have that with only 5 XTen Lite phones. I'm able to call / etc with internal phones just fine. I can call outside Vonage Numbers, and other BroadVoice Numbers. I have vonage where I live (626) and can call that fine. However, other 626 numbers I get similar errors as below. However, everytime, I try to call cell phones, and or

So I reached the point where my PRI is accepting incoming calls, but I cannot dialout. I must be doing something stupid, but I can't figure it out. The Asterisk box is sitting between the Mitel and the phone company, and has PRI lines to each. Asterisk was built from CVS r1-0 Log for a call from mitel heading outbound: ------------------------- -- Accepting call from '' to

Hi: We just setup the Asterisk and it seems to start ok. We checked the log, and beside the timer warning, there isn't other error message. However, we tried both SIPURA and XLite, but their registration is not accepted (timed out and failed). Could someone tell me what's wrong? [message] Dec 10 01:33:22 WARNING[2649]: Unable to open IAX timing interface: Permission denied Dec 10

I have connected asterisk 1.0.7 with Avaya Definity via E1 with a TE100P Digium Card. Inbound calls are working perfectly and I dont have any problem. But when I try to make an outgoing call with my softphone (xlite) I am getting the following messages. Hungup 'Zap/13-1' Executing Dial("SIP/IZ-bc0a", "Zap/g1/3118") in new stack Called g1/3118 Channel 0/1, span 1 got

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Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

Hi, I?ve strange problem when I?m making a call from SIP (Cisco 7960) to capi (Fritz PCI). When I call a national number, I?m hearing the ringtone when the called party is ringing but when I call an international number, I don?t hear the ringtone and I?ve a silence until the called party answers. Both call are going through the same extension. Here?s 2 log files, one with a national number and

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

Hi All, I have a really strange issue occuring where if I run "show dialplan" or "dialplan show" or "dialplan show parkedcalls", then asterisk dumps core. It only appears to happen with contexts that are created within res_features. I am able to display all my other dialplans, but, every time I try to just do a normal "dialplan show" asterisk core dumps