similar to: I forgot to add my email please contact me offline we have around 300, 000 to 1/2 million minutes per month for India and Pakistan .. can ztdummy help trunk mode?

I am getting the following warnings on the CLI when loading Asterisk 1.6.1-rc1: [Feb 12 12:32:34] NOTICE[22261]: timing.c:59 ast_install_timing_functions: Multiple timing modules are loaded. You should only load one. [Feb 12 12:32:34] ERROR[22261]: codec_dahdi.c:398 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory [Feb 12 12:32:33] WARNING[22261]:

Good evening from the CVS of the 2005/05/14 it's impossible to build asterisk* on a redhat 7.3 i get this at compile time chan_sip.c: In function `build_user': chan_sip.c:10007: parse error before `struct' chan_sip.c:10029: `userflags' undeclared (first use in this function) chan_sip.c:10029: (Each undeclared identifier is reported only once chan_sip.c:10029: for each

I have successfully installed and configured the asterisk, the incoming and the outgoing calls are working fine, its a tremendous solution :) Now i want to enable trunking between two asterisk boxes, in the iax.conf i have put: [karachi] ... ... ... trunk=yes ... ... ... everything seems to work fine but when i load asterisk it says: -------------- Feb 17 10:59:14 WARNING[18726]:

Hello All, We are looking for a SIP provider teminating calls in India, Pakistan and Bengladesh. Any one knows a good one? Regards, Cesar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040824/4d5dd4b4/attachment.htm

I signed up for an IAXTEL account and have been trying, unsuccessfully, to get it working. In IAX.CONF I have: [iaxtel_out] type=peer host=iaxtel.com username=USERNAME secret=SECRET auth=rsa inkeys=iaxtel [iaxtel] type=friend context=incoming host=iaxtel.com auth=rsa inkeys=iaxtel However, when I start Asterisk, I get the following warning: [chan_iax2.so] => (Inter Asterisk eXchange

I have setup iax2 between two servers without success. when I launch asterisk with the asterisk -vvvvvgcd command I see serveral wanings listed below. Is this why I cannot make connections?? My question is, how do I setup zaptel timming without any cards if possible? Does anyone have the steps? Thanks for any information. James [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) Apr

Hello All We need to connect our client's offices located in Dubai and Pakistan. Suggest us some economical solution. -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kashif at haditelecom.com MSN: kashif__naeem at hotmail.com Gmail: meet.kashif at gmail.com Skype: kashif.naeem 302 Y Commercial Area,

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Trouble getting chan_sip2 to compile below is what I have done -download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software cp /root/software/chan_sip2s.c /usr/src/asterisk/channels cd /usr/src/asterisk/ patch -p0 acl.c /root/software/acl.c.patch cd /usr/src/asterisk/include/asterisk patch -p0 acl.h /root/software/acl.h.patch - added the follow to /usr/src/asterisk/channels/Makefile

Hello list, scrambling different pieces of info together I've come with the following : I want to have my "register =>" statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:password at sip.provider.net In ext_config

Hi all. Trying to setup H.323 via Asterisk between a PLANET H.323 box and my SIP phones. When calling from the SIP phones, it connects but quickly disconnects citing the following error message: **** --- build_peer +++ build_peer +++ reload_config +++ ooh323_do_reload -- Executing Dial("SIP/yyy-2965", "OOH323/203@xxx") in new stack --- ooh323_request - data

What is this? chan_iax2.c line 4695 (build_peer): Unable to support trunking on peer 'lamas-tigris' without zaptel timing codec_g729b.c Line 413 (load_module): Unable to initialize va stuff: -1 This is why I can't start asterisk in the background -------------- next part -------------- An HTML attachment was scrubbed... URL:

I understood that a timing device (ztdummy if no zaptel hardware is present) was not necessary anymore with linux kernel 2.6. When I enable iax2 trunking I get this warning chan_iax2.c:8908 build_user: Unable to support trunking on user 'xxxxxx' without zaptel timing The linux kernel is 2.6.22-14-386 Can I ignore this message, and is trunking working despite this warning? The ztdummy

Hi All, I have 2 tables 1] user_infos & 2] users class UserInfo has_one :user validates_presence_of :city class User belongs_to :user_info validates_presence_of :first_name i write following code in my create method. @user_info = UserInfo.new(params[:user_info]) @user=@user_info.build_user(:first_name=>'''') if @user_info.save else end now what i want is to

starting 161.1-beta3: chan_iax2.c:10925 build_user: Unable to support trunking on user 'iax-out' without DAHDI timing But I have these "timing" modules: ls /usr/lib/asterisk/modules/res_tim* /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so Do I need to do some magic to get these loaded? modules.conf is set to auto. Is this what

[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'gerrie' Am I correct that when I turn on rtcachefriends in sip.conf, database-changes in my MySQL-DB will not be reflected untill a reload ?? Am I correct that when I turn off qualify in my realtime

Hi, Kerry Garrison from Fonality will be with us live to address the Trixbox so-called "phone home" script issue. I'm going to try to have something about the year "2007 in review" for any and all VOIP and Asterisk-related events, so if anyone wants to report on what they've been doing in 2007, you're welcome to chime in. We're also talking about doing a

I am new to Asterisk and VoIP. I have been given the task of setting up a telephone network in US and India. When customers call the US location, the calls should route to India (using VoIP) and handle there. The Indian location should be able to call Us numbers using the Voip to save money. The solution should be flexible enough to support initial of 5 simultaneous calls with the option to