similar to: I can`t send DTMFs through FXO lines - dahdi

Hello everyone, I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my queue interfaces, despite the fact it is free at that time, can you give help? 1. I see many sip channels from that extension: [root at mysweetpbx]# asterisk -rx "*sip show channels*" |grep 648 Peer User/ANR Call ID Seq (Tx/Rx) Format Hold

Hello guys, I have this problem when a call is received in my PBX: (Caller) --> (Redirecting Service) --> (E1 PRI) --> (Asterisk PBX) --> (Internal Phone) Reception works fine, but when conversation finishes and the agent at internal phone hangs up, the call at caller's side is still alive for many seconds until it hangs up. The problem is that Telephone Company is billing me

Hello everyone, I'm trying to send a received fax with mutt, when I try it from the Linux shel it works, but when trying with Asterisk's System command it doesn't. Successful Linux command: echo | mutt -s "New fax" earohuanca at gmail.com -a /tmp/faxes/201306191111.tif Unsuccessful Asterisk Command: same => n,System(mutt -s "New fax" elder.arohuanca at

Hi List, I'm trying to set my Asterisk 1.8.20.1 with TLS on CentOS 5.9, it was easy and straightforward with Debian 6.0.6, but when I introduce this command on CentOS: #./ast_tls_cert -C 10.200.108.17 -O "MyCompany" -d /etc/asterisk/keys/ I got this error message: hostname: Unknown host Same result happens when using server's hostname: #./ast_tls_cert -C ast-centos -O

Dear list, I want to avoid sending calls to a queue when it is full. From the fact that 'maxlen' must be at least 1 (I wish it could be zero but it isn't) I'd like to know if there's a way to do it. Setting the Queue() timeout to a little value is not the most suitable option. I'm using asterisk 1.4.21 but I don't know if there are some options available on release 1.8

Hello everyone, I wonder if there's a product that I can install on my debian-based server to extract CDRs (it'd be better if Excel's downloads are available), also it would be desirable if I can access additional table to update rows (e.g. sip for realtime) Please let me know what you know. Best Regards, Elder D. Arohuanca dCAP Lima - Peru -------------- next part --------------

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:

Hello everyone, I've suffering cut offs after 6 or 7 seconds a call is answered, incoming calls are working fine, but outgoing ones show the gollowing messages when are being dropped: [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical Response) -- See

Hello all, I need the bootrom.ld file to set up some Polycoms I have Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A I've publiched on my FTP files downloaded from http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html (3.2.3 combined and split zips) but my phones are still showing the message: "error, application is

Hello Friends: I've just installed Asterisk 11 on my Linux (debian) server but it is not starting up when trying with "asterisk -vvvvvvvvvvc" and "service asterisk start". Starting process just stop and shows: "Illegal instruction" as final output. Looking at logs I fouind at /var/log/asterisk/messages : [Nov 25 11:09:26] Asterisk 11.6.0 built by root @

Dear all, I wanna know what can I do to get the PBX's clock from -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090428/3218b8b0/attachment.htm

Hello friends: I am facing cutoffs randomly when negotiating calls. The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the remote peer receives a ghost call. Along the atempt I could see six times a messages regarding NAT isuues *[3]* I hope anyone can give me an

Good morning, I'm having suddenly cut-offs and I don`t know why. It's been hapenning since I enabled cdr_odbc/func_odbc in my system. We use func_odbc to register some queue member's events (login, pause, etc.) at an external DB ('remoto' connector) and to uptade local tables at a local DB ('local' connector). Currently we are usind cdr_odbc to Postgresql and cdr_addon

Hello friends, I have been experienced suddenly stops for my Asterisk server, I do not why is it happening. Asterisk's debug messages only tell me I have lacked g729 codec for translation to one peer minutes before the crashes occur [2014-05-27 09:48:30] WARNING[15384][C-0000017c] channel.c: Unable to find a codec translation path from (ulaw) to (g729) [2014-05-27 09:48:30]

Hi everyone, I`ve been having several problems with my current softphone and I`m trying to develop an IAX Client based one. Does anyone know how can I get help or useful resources about it? Specially with Conference function and management of incoming call events to launch an AGI at that time. I?ll be very gratefull for any help you have. Daniel Arohuanca Lagos +51 1 3594122 -------------- next

Hello list, I'm sending calls to a queue in the attended way, that is, *1.* the original call is put on hold, *2.* a second line is open to call the queue, *3.*after an agent is connected the original call is transfered to its final destination. 1. Zap/1-1 <--> SIP/agentA-tag1 2. SIP/agentA-tag2 <--> SIP/agentB-tag 3.

Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello A user seems unable to type DTMF in our Asterisk IVR menu. Can this be due to their phone or PBX that disables DTMFs when a user is off-hook? Thank you.

We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC

Hi, I'm using queue configuration as follows: - queues from* queues.conf* - queue_members from *external Database thru ODBC*, using* Local channels * as interface - sip extensions from *external Database thru ODBC* When a call is sent from queue to an interface (local channel), it is answered but a message appears at the CLI: *[Nov 4 16:56:04] WARNING[13951]: app_queue.c:3014