similar to: checking if a phone number is UP

Hi, I have IVR and when I press 1, asterisk calls my mobile phone. If my mobile phone is offline, asterisk transfers to asterisk voicemail. I'd like asterisk detects my mobile voicemail and if my mobile voicemail answers, asterisk transfers to asterisk voicemail. For that, I used AMD. So I have problems ! Asterisk detects answering machine everytime! How do I do please ? extensions.conf

Hi, I wonder if I may freely use the default soundfiles that came with asterisk (fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server? Are there any official sources of royalty free music? -- Mvh, Aurimas Skirgaila -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, long time ago, I came up with an optimal configuration set for my environment - good detection and little false positives. Unfortunately some people are always being detected as Answering Machines. I'm not up to re-adjust my precious balance of initial_silence/max_words/... , so I'm thinking about to check if the pickup time is equal to the pickup time when the same phone number was

Hello, I'm having problem connecting my Asterisk 1.4.29.1 with Digium TE420 to providers Cisco 2800 with VWIC-1MFT-E1 card. the same card runs fine with another E1 provider. TE420 led's lite green. Message type: RELEASE COMPLETE (90) < [08 02 80 ac] < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) < Ext: 1 Cause:

I have the following senerio. Call file calls 1st party. When connected give called party option to connect to second party. Issue Dial to second party. Caller answers and the two are bridged together. My issue is that 4 out of 5 calls fail to bridge the audio. Am I missing something or is there some kind of bug? Here is my test dialplan ;Dialer Base Code Files. ;Variables

Hi I came across couple of pointers on the Internet regarding solutions available for providing hosted PBX service. 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty straightforward, but no hosting company wants to use it. 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way

Hello All, I do have asterisk installed for a call centre with aheeva application and i would like to know how to configure the sound for the inbound calls and if there is any possibility for agent to receive a file with the phone number and name of clients: For your information there is no problem related to the outbound call An help would be appreciated Kind Regards Salah. --------------

With asterisk 1.8.7.0 has been running ok for months. Now, this morning, it's crashing. I can restart it, but it crashes after 10+ minutes. It dies like this -- Executing [s at macro-stdexten:2] Dial("SIP/teliax-00000019", "SIP/176,18,rtT") in new stack == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP

Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Thanks in advance! Elder D. Arohuanca dCAP 1497 Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL:

The asterisk server has a permanent IP address, but the provider cannot ensure stable quality traffic for RTP. There is a desire to use an external server, the address of which shall be specified in the SDP, through which flowing media. I use asterisk 13.6 and res_pjsip. Prompt, please: 1. what software can be used on an external RTP proxy? 2. What settings need to be done in pjsip.conf to use

Hi All, Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and asterisk 1.2.14 ? i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but it gave an error - 1.2.14 End - Error Msg WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by 147.120.203.71: No authority found 1.2 END , IAX.conf [trunk14] type=friend host=147.120.203.71 secret=test123

Hello , I would like to tweak my Answeing Machine Detection (AMD) in Asterisk. My current values are AMD(2500|1500|300|5000|120|50|5|256) and we were able to identify approx 25-30 % of all answering machines. Anybody have any suggestion to improve the accuracy of AMD. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Everyone, Unlike going through DIDx, DIDLogic etc.., we have an option of getting DIDs directly from local telco Bell Canada. Currently our SIP Trunk provider assigned a DID to us, and as you know, they just redirect requests it to our PBX. However, when dealing directly with a telco, what equipment will we need? Basically giving us the same capability as a DID provider. If someone can

Hello list, I need a hand to find the best dialplan failover solution when using two SIP Trunks. My reasons to do failover are: a) one of the two providers could be unreachable b) both providers may be UP but one of them could return a SIP error message (maybe caused by DOWN E1s) Googling I found a few possible solutions: 1.

Has anyone been able to configure Asterisk to work over 3G? I bought Nokia Cell Phones just for that purpose and they register fine over WiFi and 3G but the quality is just not good enough and sometimes the call just disconnects. I have Allow as: ilbc gsm ulaw alaw -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667)

I am testing out inbound faxing using res_fax and res_fax_spandsp.so My system answers the call but then sets there on the ReseiveFax line then comes back with an error that it exceeded the maximum retries. How would I go about debugging this? Below is my very simple dialplan code I am using, and the fax show version gives the following as well. FAX For Asterisk Components:

I'm trying to understand why vendors keep making 100Mbps integrated 1-port switches in their hardware SIP phones. Even the recently-announced D40 and D50 Digium phones are limited to 100Mbps. Only the more expensive models (like the D70) can run at 1000Mbps. However, you can't expect a firm with hundreds of extensions to buy the most expensive model... And gigabit speed is important when

I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear list, I am trying to configure the nagios plugin called check_sip. I just read the README file included with the plugin. I follow the readme steps to configure the plugin, without success. In the nagios web interface I can see (No output!) In the status information column. If I run the chech_sip plugin from a linux console, I get /usr/local/nagios/libexec# ./check_sip -u

I am having issues with Blind Transfer on asterisk 1.8 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use