similar to: Dead air on 7960 sip at start of call.

Hi folks, Recently we assigned our users agent id's and switched to having them use agentcallbacklogin instead of just ringing the phones directly.... it's been going well for the most part. Before we gave the users agent id's we had their sip configuration set to incominglimit=1 for one line on their phone, to prevent call waiting beeps. If they wanted to pick up while already on a

Hi, I've looked through the list archives, bug tracker and cvs changelogs and can't see anything that refers to the particular problem i've seen recently. I'm running CVS-D2004.06.09.14.00.00-06/24/04-00:43:55 which I realise is not exactly recent, but I wanted to find out more before I updated. We've had asterisk running for our office using 7960's and a TE410P for a

Hi, I have installed the latest version of asterisk(1.4.0-beta3), and built app_rxfax/txfax. I'm using spandsp from here, http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20061207.tar.gz Everything builds ok. I had to manually apply the patch from the site so configure would spot spandsp libraries. However, when I try dialing my virtual fax extension (either from a phone or

Hi, Anyone used this service, any comments on reliability/support? Thanks John

Does anyone know of a BYOD provider that terminates calls to NCFA numbers (UK 'national rate'). I enjoyed broadvoices unlimited to those numbers, but this is getting silly now, it doesnt work and no answer if after switching to a new provider it will ever work. Can anyone suggest an alternative provider that serves NCFA? -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605

Hi all My * implementation is working brilliantly with only one small fault left to kill. I'm using IAXTalk from Telappliant for my incoming/outgoing calls to the pstn network; if I set my codec to GSM everything works great - no pauses but quality is a bit poor. If it set the codec to alaw (I think I'm using the correct one - I'm in the UK) I get intermittent pauses on the call.

Hi everyone I'm interested in using the Telappliant/voip-talk offering as an alternative to my DDI analog problem. (see [Asterisk-Users] Multiple DDI & Hunting on Analog Lines (UK) for details) Does anyone on the list have any recent comments on reliability etc? I would really appricated some positive and negative comments. Cheers Matt

Hello folks, i upgraded to the actual CVS head from yesterday (27.5.) but can not get incoming SIP calls from my provider (sipgate). If someone calls my number, my asterisk responds with the following error: May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request: Failed to authenticate user "<CallerID>" <sip:<CallerID>@217.10.66.11>;tag=as38e9693c I

Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723

I went on a service call yesterday to find an asterisk system with a T100P card on a Qwest PRI and a TDM40B card connected to fax machines. The TDM40B LEDs were not lit, and the system did not respond to keyboard input. However, calls were being processed for the PRI and 7960 phones. I replaced the TDM40B card with a new one, and the system now seems to be ok. But I'm wondering, why

Hello Everybody, I was going thru the C code of Asterisk. Does anybody know how does one go about modifying the C code of Asterisk? Please do reply. Regards, Bharat M. Sarvan EZZI BPO Pvt Ltd., PUNE. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Ive just installed a TDM4000P with 4 fxos. The zaptel config is fine, zttest comes back with configured. If i call a line when zttest it shows on the display,and then goes when the line drops. In * when a call comes in, it follows my dialplan and answers the call according to the log, but IT DOESN'T actually pick up the call, i.e. it continues ringing. I'm using KS signalling, and

Hello Everybody, I am trying to configure Asterisk to listen into a database which is created in PostgreSQL. Whenever asterisk starts up, it is unable to connect to the pg database and gives the following error: [cdr_pgsql.so] => (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:284 my_load_module: cdr_pgsql:

Hi everybody Hope I picked the right mailing list. If not, please tell me. We've got a problem with call forwardings. It's exactly the same problem as described in bug 12708, which is resolved by now. Situation: Caller -> asterisk -> call forward to mobile (packet2packet bridge) Quote from original bug reporter: 'One issue that we have noticed repeatedly is that there is a

Hello, I'm running Asterisk 1.4.26.2, DAHDI 2.2.0.2, and libpri 1.4.10.2 on Ubuntu with a Digium TE122 connected to my PRI. I have an extension setup so that managers can monitor calls: ---- # grep 9900 /etc/asterisk/extensions.conf exten => 9900,1,DAHDIScan() exten => 9900,2,Hangup ---- For whatever reason, this no longer works. It's not often used, so I don't even know when

I'm starting on a bit of a long term project (for myself) to implement dead air detection with an icecast stream. Basically at the location where we have icecast implemented it's a radio studio and it'll be good if we have the system automatically detect if the dj output is dead. To which I am firstly posting on here to find out if anyone's already implemented such a system and

Hi All, I have an asterisk box on my DMZ, and I'm using a PF for my firewall, I can make a call but some reasons I have a dead air. Any Ideas? below are my rules... ext_if = "bce0" int_if = "bce1" altitude = "172.16.1.0/24" #### machines #### vbox = "172.16.1.1" uci = "172.16.1.4" voices = "203.172.x.1" ipc =

I have csvde dump from active directory I process on my postfix mta. It takes output like this: "CN=Curtis xxx,OU=Domain Users,OU=xxx xxx,DC=xxx-xxx,DC=local",X400:c=US\;a= \;p=xxx xxx xxx\;o=Exchange\;s=xxx\;g=xxx\;;SMTP:Cxxx at xxx-xxx.com and should return a relay_recipient map in the form of: Cxxx at xxx-xxx.com OK The command I am using is: cat $1 | tr -d \" | tr , \\n|

hi, I am preparing my package for R 2.13 build and check gives no warnings just OK's However when running R CMD INSTALL it gives me (nfortunately it is in german) ** help *** installing help indices ** building package indices ... ** testing if installed package can be loaded Fehler: '\U' ohne Hex-Ziffern in der Zeichenkette beginnend mit "C:\U" genutzt Ausführung

On Mon, Jul 01, 2019 at 10:25:17AM +0200, Christoph Hellwig wrote: > And I've demonstrated that I can't send patch series.. While this > has all the right patches, it also has the extra patches already > in the hmm tree, and four extra patches I wanted to send once > this series is merged. I'll give up for now, please use the git > url for anything serious, as it