similar to: No ringback even though progressinband=yes is set

Hi, I currently have some systems on AsteriskNOW 1.7 & have been happy with its clean simplicity & reliability. Are many people here using AsteriskNOW 2.0.x? How do you feel about it? Did Digium stick with their previous philosophy of keeping everything very vanilla & making it clean & simple for someone who understands how to manage CentOS, FreePBX, tftp, ntpd, etc. but

1.2? 1.4? 1.6? 1.8? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545

Hello all, I have been running asterisk systems since summer of 2008. I do not claim to be an expert. But I have worked through many issues during this period. I have setup & manage 5 systems, which serve 6 companies total (and of course process calls for all of the people they do business with). I have always been happy with asterisk (well, obviously less happy during the problem times...

Hi, We have clients with various security & compliance requirements. Although not required, it would be ideal to have messages encrypted at rest. We already use SSL/TLS to secure the transmission of most email. However, it would be nice to have them encrypted sitting on our server. Is anyone doing this? I think that ideally, rather than full-disk encryption, we should use an encryption that

Hello. I have a software developer creating a .Net / mono program to use as an AGI script. We are having problems getting it to stream files. From what we can tell, it is talking to asterisk correctly when called from the dial plan. Its stderr output goes to the asterisk console. But asterisk doesn't give any indication that it receives the STREAM FILE command. Asterisk simply quickly

Hi, Does a parameter exist for a queue to delay ringing/sending a caller to all agent phones after the previous call is answered by an agent? My queue ring strategy is set to ringall. I am using Polycom KIRK wireless DECT SIP phones. And it looks like the KIRK wireless server may need a split send to realize all wireless phones are no longer ringing (busy) after 1 call rings & is unanswered,

Hello all, I have a system running AsteriskNOW with asterisk asterisk-1.8.8.1-1_centos5 from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so that blindpreferred=1 (all transfers default as blind transfers). If a customer calls in & we answer & transfer, everything works fine. But if we call out to a customer & then transfer to another internal extension, that

This morning on our newly built server, the following was logged twice: auth: Error: pam(username,127.0.0.1): pam_authenticate() failed: Authentication failure (/etc/pam.d/dovecot missing?) This also happened to be during a time of 100+ imap-login processes, where we were seeing: master: Warning: service(imap-login): process_limit reached, client connections are being dropped The initial error

Hello everyone. I am wondering whether there is a certain technique I should use to identify all log lines in the asterisk/full logfile that are related to a single call. If a user reports that something strange happened with a certain call, I'd like to be able to easily go back and look at the asterisk/full logfile, and look at only the lines that are relevant. I am having some difficulty

Hello, We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to the telco, users mainly use snom 320/300 SIP phones. When dialing to an external phone number with custom ringback music, users reported that they could not hear the music but can only hear the standard ring tone generated by the system. Is there any kind of settings need to allow the ringback music pass to the

Now that we've hashed out some thoughts on the most stable version of asterisk, I'd like to ask the question as to why I should NOT use 1.8? What are specific reasons? For instance a few days back I was speaking with James at Rhino Equipment. He said that he has "no real data" on why I shouldn't use 1.8. They just follow a practice of not jumping on the newest version. But I

Hello! I migrated asterisk 11 to 13 as user of FreePBX 12.0.76.2. As customer of German Telekom, I have three numbers and therefore three trunks - each number is bound to one trunk. After migration, some callers complained about missing ringback tone: they didn't hear any ring tone and where surprised that they suddenly got me anyway :-). The connection afterwards was as expected. Deeper

Whoops, email client auto-filled dev previously. Let's try this again. Michael Maier wrote: <snip> > Ok - but this doesn't seem to answer my main question: > > Why must > > progressinband=never > > be applied especially if asterisk uses it by default? The big difference > between w/ and w/o it is: The default in 13 is "no" which still

Today I found out we are having users w/ problems because: Mar 3 09:57:33 jlgray dovecot: master: Warning: service(imap-login): process_limit reached, client connections are being dropped Mar 3 09:58:42 jlgray dovecot: master: Warning: service(imap-login): process_limit reached, client connections are being dropped Mar 3 10:02:51 jlgray dovecot: master: Warning: service(imap-login):

I have an asterisk system where the asterisk daemon dies typically at least once per day. It is running in the wrapper safe_asterisk, which automatically starts the daemon back up. But we find this unacceptable because when the daemon dies, we usually have active calls drop, and sometimes we have to run asterisk -r -x "module reload" after the daemon starts back up before everything is

This link (http://blogs.digium.com/2008/10/13/asterisknow-15-beta-available-more-coming-soon/ ) seems to indicate that in order to upgrade AsteriskNOW v1.5 from Asterisk 1.4 to 1.6, it's as easy as installing an upgrade package. Does anyone know where to find that upgrade package? If it doesn't yet exist, what is the process for upgrading?

My last problem was nicely solved through this mailing list so hopefully this new problem will have the same happy outcome. My situation is that I have many extensions. Here is a sample: [client-phone](!) type=friend host=dynamic secret=XXXXXXXXXX dtmfmode=auto disallow=all allow=ulaw allow=gsm allow=g723 allow=ilbc subscribemwi=no [4165555555](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxx

1. Subject. 2. asterisk16-1.6.2.6-1_centos5.src.rpm have not asterisk.logrotate in SOURCES 3. for "--without dahdi" diff SPECS/asterisk16-my.spec SPECS/asterisk16.spec 750a750 > %{_libdir}/asterisk/modules/res_timing_dahdi.so 879d878 < %{_libdir}/asterisk/modules/res_timing_dahdi.so

I just put in a Audiocodes Mediant 1000, which seems to be working well except for one annoyance. I am using Polycom 501's and 601',s and if I do a supervised transfer of a PSTN call where I complete the transfer before the 3rd party has answered, the PSTN party hears dead air until the call is answered or goes to voicemail. I'm not really sure where to start my troubleshooting. Any

I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my polycom phones and then it also sends 183-Session Progress. That doesn't seem to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ? Doug.