similar to: caller_id vs cid_number

Hello I just settup a realtime mysql table for sip_peers. All peers (friends) is autenticateing but when i want to initiate a call between them i got the following error. Someone have some ideea? Thank you. ---<Cut Here>--- pbx*CLI>console dial 1014 == Console is full duplex -- Executing [1014@default:1] Dial("OSS/dsp", "SIP/1014|40|t") in new stack

Hello Everyone, We are currently experiencing some higher load on our servers, and since signaling comes into our servers on G729, we would like to implement G729 pass-through. A few questions arise, do we need to convert all the recording to the codec, and what about voicemail? We are also using A2Billing (hope I am not violating any thread rules), and would like to convert all that recording

Hello Everyone, I have disallow=all and allow=g729 set in sip.conf however, it seems that asterisk still thinks it support other codecs: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw). How can I disable gsm,ulaw,alaw..... Thanks in Advance, Nick.

Hello Everyone, We are looking for solutions where the transcoding is abstracted away from our * box (i.e., to the network layer) using some carrier grade gateway, or router. The reason for my post is to know about solutions people have used in the past, and how it fits into their overall architecture. Our transcoding needs consists mainly of u/alaw <-> g729, and gsm would also be good....

Unsubscribe Elvin G. Nodalo -----Original Message----- From: asterisk-users-request at lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit

Unsubscribe Elvin G. Nodalo -----Original Message----- From: asterisk-users-request at lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit

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 Everyone, I have gone through a few really good tutorials from the OpenSIPS site, Asterisk resources etc.. The unanswered question (and final piece of our puzzle) is if it's possible to have a register free environment in an OpenSIPS/Asterisk integration. Most approaches have OpenSIPS relay the UA's REGISTER request to Asterisk which has "host=dynamic" set for the

Hello Everyone, We are getting some rather poor results (relative) with our Asterisk setup. Not sure if we are using the sipp correctly etc.. but nevertheless, is there any documentation that describes how we can get the most our of our Asterisk box. For example when we hit the "too many file" error, and fixing it using ulimit..... Also, is there any way we can allocate sufficient

Anyone here using Level 3 or AT&T wholesale sip terminations services? I would like to know on any minimums they would require? Also, an idea of how competitive the rates are. I am not asking to disclose your custom rate deck, just a "what to expect". Finally, if you guys can PM me contact info to someone from the wholesale department, I would really appreciate it. Kind Regards,

Hello Everyone, I have bumped into the thralling penguin page on linux vs solaris for asterisk. Does the benchmark still hold with the newer versions of kernels? Curious to know of your thoughts. Also, they mentioned running it on Sun Fire x2100, but no benchmarks were given for that. Can increased performance be accomplished simply by changing to Solaris or OpenSolaris? Kind Regards, Nick.

Anyone try this? I saw a post here: http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/94041-setup-of-sangoma-a101-in-my-elastix.html But not sure if it's possible. What I am asking is if there are any T1 cards with virtual functions implemented in their drivers to allow pci-passthrough? Kind Regards, Nick.

Oh heck. It really looks like realtime has been seriously screwed up. When a call comes in to Asterisk, I can see asterisk executing these queries. SELECT * FROM ast_sip_peers WHERE host = '2XX.YYY.142.205' SELECT * FROM ast_sip_peers WHERE name = '2944093' SELECT * FROM ast_sip_peers WHERE name = '2944093' So, the first thing it does is check and see if there are any

Hello Everyone, We are looking to interconnect with a local ILEC over an OC-n transport layer. They basically gave us two options in terms of mapping the SONET to the DS3: * VT1.5s mapping * DS1s mapping The second option is quite clear. We would MUX the connection, and plug the lines into qaud t1 cads etc... The tech mentioned that with the second option we would also need a DACS to convert

I am using VoiceOne http://voiceone.it/ as my management interface. I am not 100% sure when it started, but my CDR is now full of "s" as the DST instead of the actual dialed number. As I understand it - it is because it is being recorded in the CDR while in a macro (as below). Is there any work around so that I can record the actual dialed number? [macro-dialout] exten =

Hello sip show peers does not mark hosts as NAT even though sip.conf and sip_peers table has nat=yes. spitfire*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status voipuser.org/gdsm 216.127.66.119 N 255.255.255.255 5060 Unmonitored 5560/5560 192.168.4.5 D N A 255.255.255.255 5060

Hello, How do I use multiple postgresql databases using res_config_pgsql? I tried creating multiple contexts in res_pgsql.conf, but asterisk is only using the 'general' context. My res_pgsq.conf is [general] ;; Connect to mydb on localhost dbport=5432 dbname=mydb dbuser=pgdbuser requirements=warn [pgwritedb] ;; Connect to mydb2 on another host dbhost=<IP

Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP

hey, i have this problem in the database i have 32 records and the sql filtered 4 records #getting the callerid records depending on the given paramaters and paginate it @callerid_pages = Paginator.new self, CallerId.count, 10, @params[''page''] @caller_ids = CallerId.find(:all, :conditions =>["geotags.firm_id = ? ", @firm_id], :order => " geotags.address1

Hello Osama, and Hisham, At 1330GMT there was some malicious activity coming from your network IP 37.75.210.90. Please act accordingly. Things that may be of use "972599779558" N.