similar to: Out bound calls 'you must first dial a 1'

Hi list, I have a call file as following and it works. But, I don't really understand its mechanism. The SIP/voipbuster is a sip trunk which I set up in freePBX with voipbuster account. And 2874 is one of my extension which was assigned to x-lite client. When I place this call file in outgoing folder, it is able to dial out my home phone at 001xxxxxxxxxx. However, the Dst in call logs show

Hello, I'd like to use the AMI interface to originate a call to a context in a dialplan, and handoff the dial control to the context. Whenever I execute the below action, the recipient does ring, but when I answer it dials the recipient again. I believe this is because once answered the system is going to execute the Context/Exten/Prio in the Originate action? Action: Originate Channel:

Greetings all, I am in need of professional insight. I have a 2 node cluster running CentOS, mysql, apache, etc. I have on each system a fiber HBA connected to a fiber SAN. Each system shows the devices sdb and sdc for each of the connections on the HBA. I have sdc1 mounted on both machines as /quorum. When I right to the /quorum from one of the nodes, the file doesn't show up on the

I have an interesting dilemma with sequencing in the dialplan. Up to now, I have assumed that the extensions in the dial plan were tested in the order that they appear in extensions.conf. In other words, I have the following fragment which was designed to dial toll free on the PSTN and all other long distance on VoIP: [longdistance] include => local

Hello folks. I'm new to the list and I have questions about Samba. I have been able to configure Samba 3.x on Solaris 9 with AD authentication for the users. I'm able to mount the shares onto Windows XP clients and able to read the files. Now, if I use a text editor like notepad or GVIM to save an existing file, it saves it. When I try to use Word, Eclipse, or Crimson Editor, it

Sorry for all the newbie questions. I really appreciate everyone's help today. Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm having an issue with SIP extension to extension calling. Any time I dial another extension it goes right into voice mail. My extensions.conf is pretty small and rough but, here's what I have right now. Most of it was taken

I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC. I listed my conigs below I go off hook

[Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups. [Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net' changed from 64.2.142.17 to 64.2.142.29 [Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed Jitterbuf max 600 timeslots And the calls are dropped. I fixed this by turning off enable in dnsmgr.conf My question is: Do you attempt to

we are in process of writing a PHP interface for * conf files. we are parsing the files like INI files but the only prob I have so far is that separate extensions in a context dont have any unique tag that I can capture. This works ok [trunkld] ; ; Long distance context accessed through trunk ; exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:1}) exten => _91NXXNXXXXXX,2,Congestion

I'm trying to create several contexts for extentions with different levels of access to features and I'm wondering how the heck do I include all the contexts so that you can call internal to any extention in another context without giving the features of the higher level context to the lower level context? ie..... [admin] include => local include => longdistance include =>

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

TRUNKMSD1=1 ; MSD digits to strip (usually 1 or 0) TRUNKMSD2=2 ; MSD digits to strip (usually 1 or 0) ; logn distance calls exten => _91NXXNXXXXXX,1,NoOp("Dialing: "${TRUNK}/${EXTEN:${TRUNKMSD1}}) exten => _91NXXNXXXXXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD1}}) exten => _91NXXNXXXXXX,3,Congestion When I dial

Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on

Okay everyone - I'm moving away from using sipura 841 phones. I'm starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to

Is anyone out there using an "AudioCodes MP108 8-Port FXO Analog Gateway (SIP)" with asterisk to support both inbound and outbound calling? If so, I'm interested to hear how it works, and I'd love to see some example confs (both in sip.conf and on the MP108). This product has been recommended to me by a SNOM/Asterisk-friendly distributor, but I would like a second opinion

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

Now that I actually try and google for it, I can't find any dual mode GSM/DECT handsets, only pages telling me that they exist without any actual information!!! Does anyone know of any such handsets? (and even better, ones that are available in Australia) I've searched a few of the major gsm manufacturers (nokia, Panasonic, sonyericsson) but their web sites are absolutely pathetic to the

hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM