similar to: Asterisk Log Color Coding

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'm planning to get my Asterisk box out of the LAN, get rid of my router and make the box acts as a Router, Firewall, DHCP Server (with Shorewall). I'll do that to be able to use some SIP clients remotely. Does anyone doing the same with the Asterisk box, is it a good idea, is there any other solution for the SIP emote Clients. Regards. __________________________________

Hi, Is there anyway to pass a variable from one context to another (NOT macro and NOT global) Regards __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

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

What is the best solution? I dont want to have modify firewall's at all or do port fowarding. Ideally I would like a solution that with either a softphone or wireless hardphone one could connect via friends, family, or hotspots without reconfiguring their devices. What are people using? STUN? SER? Thanks in advance! -blake -------------- next part -------------- An HTML attachment was

Hi, We have a Data/Voice service supplied through an integrated T1. Does anyone know if Digium T1 card will support the splitting of the Voice and Data? Regards. __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

Hi, I was wondering if it is possible to connect a skype phone adapter, for example: http://zonetusa.com/DispProduct.asp?ProductID=191 http://www.actiontec.com/products/communications/ipw_usb/index.php http://www.eradian.com/ERadianUS/staticpages/SkytoneRST301Details.htm http://www.dlink.com/products/?pid=466 http://www.usr.com/products/voip/voip-product.asp?sku=USR9620 to a TDM400P, so that

Hi, I noticed that the sound directory is missing from asterisk-1.4.0-beta4.tar.gz. This directory (7 M) witch existed in asterisk-1.4.0-beta3.tar.gz has GSM Core Sounds and some MOH. Does anyone know why it has been removed from the latest beta? Regards. ____________________________________________________________________________________ Sponsored Link Mortgage rates near historic

Hi, I'm looking for a tutorial or installation guide for SER to be used with asterisk to solve the remote SIP agent problem. All the documents available are for large scale installation. Any help is highly appreciated. Regards. __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

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

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

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

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

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

Currently I am having 2 issues with my Avaya 4602 phone: First, the phone registers with my Asterisk server, but when I start dialing I get a busy signal after 4 digits. I specified in the dialplan on the phone to expect 10 digits and that solved that problem, but I still immediately get a busy after the 10th digit. The phone never sends a dial command to asterisk. Second, asterisk is

When trying to make outbound calls I am getting the Warning: File app_dial.c line 313 (wait_for_answer) Unable to forward voice. When making the call it attempts to dial (pounds are actually numbers but replaced to not show numbers we are dialing): Executing Dial("Sip/donas-bd7b", Zap/g1/1##########") in new stack Called g1/1########## Channel 1, span 1 got hangup **Above

Hi, I'm running into issues receiving faxes which, from what I have read, may be caused by frame slips. While I can find many posts saying to investigate it, I can't find any that describe *how* to debug the problem. Tried searching this list as well to no avail. Any pointers would be greatly appreciated. FYI, I'm running wbel, AMP 1.04, spandsp 2pre4. Faxing to a pstn on a

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

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Hi, I am try to configure Asterisk as PBX system with two interfaces as shown below. One interface pointing to the local subnet with a SIP phone and another interface pointing to the external ISP SIP Sever. SJPhone(X.X.141.32)<--------->(Y.Y.47.149)local-intf-|Asterisk|external- intf(Z.Z.247.106)<-------->(w.w.158.26)ISP-SIP-Server----OutsideWorld I am able to setup a call from the