similar to: How to recognize the Telco provider on outgoing calls only by sounds?

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 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 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

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 @

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

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 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 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]

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

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

Hi, I've just installed DAHDI at two PBXs as follows: *PBX-1 PBX-2* FXO ------------- FXS When I try to send calls from PBX-1 to PBX-2 I just receive the message: "Starting simple switch on 'DAHDI/1-1" It seems like if PBX-1 couldn't send DTMFs, but when I set immediate=yes at chan_dahdi.conf at PBX-2 dialplan is executed at PBX-2 but nothing is heard at

Hi again, Thanks Ted and Marc its works. But of course after pulling in in some real life data I discoverd one hitch. Often there are missing intervals. For example: from <- c(0, 1.2, 4.0, 4.2, 5.0, 25.0, 30.1, 45) to <- c(1.2, 4.0, 4.2, 5.0, 25, 30.1, 36.2, 50) intensity <- c(0, 1, 3, 2, 1, 0, 2, 5) barplot(intensity, width = -(to - from), space = 0, horiz = TRUE, ylim =

I have a existencial problem. There are some provider that offer the service of bandwith asimetric as download/upload link as for example 512/256. but most of them offer not exclusive this amount of transmission or reception capacity. They usually offer the service with more users as a ratio of 1/10. How I can design a appropriate diagrama with htb where the amount of bandwith could vary.?

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:

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

hi every body. Im new in this program. Im traying to install R in linux suse10.0 in two following form: a) with the file R-2.5.0.tar.gz b) and the rpm file : R-base-2.5.0-2.1.i586.rpm ****** In the first case a) when i uncompressed and type: linux:/opt/R/R-2.5.0 # ./configure the followind message is showed linux:/opt/R/R-2.5.0 # ./configure checking build system type...

Hello. I would like to know if writing a new front-end for a widely-used programming language (in my case, Oberon) is a valid proposal for GSoC. By the way, I'm porting the original Wirth's Oberon compiler and I think that making available a translator for this language with an industrial-quality backend would make it a more interesting choice for many projects. Best regards. Walter

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I am a newcomer to CentOS and been lurking the list for a week or so. This morning I installed CentOS 4.1 on a Compaq Evo (Pentium IV, 512MB RAM) After the initial boot just after installing, the console if constantly filled with "eth1: mismatched read page pointers 0 vs ff". This message appears at least every 40 seconds. The same

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, About a week ago I was forced to migrate my main (production) email server from RH9 to Centos4.1. The installation has dovecot-0.99.11 + postfix-2.1.5 and >600 mbox accessed email accounts. Number 1. Some users accessing thru squirremail (installed in my webserver) can read their INBOX at /var/spool/mail directory but when trying to