similar to: setting caller id number and usingsip type=peerfor incomming calles.

> > To get around this i updated CVS HEAD and changed the sip entity from > > type=user to type=peer (yes peer!) (type=friend works too but im making > > a point) the client now must register to set his outbound caller*ID > Number. > > Yes, that is normal. SIP has difficulty separating the remote party > identification from the authentication identification

Just to bug you all (feel free to rant at me), a client wants to set his caller*ID number for outbound calls though us to PSTN. the client is using SIP to us, he can set the caller*ID name fine. if he sets his caller*ID number to anything other than his account number (8440101), the call is dropped into the default context (and then hung up by our dial plan). To get around this i

> I would like to have all SIP phones to work on prepaid basis > and without need to dial any access number, instead I would > like to use the phone as normal dialing only the destination > number, for example 00464090510. I use the AccountCode for authentication. This is how, for example: exten => _00XX.,1,DeadAGI(astcc.agi,${ACCOUNTCODE},011${EXTEN:2}) > Once the call is

[cross-posted to -biz and -users since it could fall into either category] Interesting new product that has been introduced that I think some would be interested in here (at least, those users in the United States and perhaps Canada): CNAM delivery via IP lookup. The problem: inbound calls on many PRI connections, and also over many VoIP providers, do not include caller name. This means that

Can someone help me, im using latest CVS, asterisk and cdr_mysql, when I make a bridge call (using .call files in outgoing/) I always get 'billsec=12' in the cdr, both mysql and Master file even if the call lasted longer, watching the Master file while making a call I see it updated at 12 seconds even while im still 'in' the DIAL app and the call continues on just fine. Iv looked

I am working for Accudata Technologies. We provide CNAM via http request or raw TCP/IP connection. We would like to provide the same capability to Asterisk. I installed Asterisk on Fedora 2.0 and did reading about AGI and AGI application at http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI, http://www.voip-info.org/wiki-Asterisk+Manager+API, and The Asterisk Handbook Version2 but still

Hi Folks, I've been looking around and found some references of some Caller ID patches (Mantis bug#9) for X100P and TDM400 for Netherlands, Sweden and UK. It's been quite hard to understand what has finally been incorporated to the distribution (if anything) or which patches must be applied in witch snapshot of the repository. I've tried some different approaches but nothing worked

Has any one found any issues (bug?) with ver 1.2.5 as CallerID generated on an outgoing FXS port (to the handset) fails when UK tones are used, with a message 'Didn't finish Caller-ID spill. Cancelling.' Any tips on getting this running ? Looked at Mantis, but only known bugs seem to relate to XP100 cards - not the TDM card. Thanks

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > brett-asterisk@worldcall.net > Sent: Tuesday, November 30, 2004 2:53 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] National (US) callerid name > resolution for yourasterisk box >

----------------------------------------------------- "Yeah, we rocked the vote all right. Those little bastards betrayed us again." - Hunter S. Thompson on the 2004 election. ---------- Forwarded message ---------- Date: Tue, 22 Mar 2005 19:16:09 -0800 (PST) From: Matt Klein <mklein@nmedia.net> To: Asterisk Users Mailing List - Non-Commercial Discussion

V I R U S A L E R T Our viruschecker found the 'W32/Sobig-E' virus(es) in your email to the following recipient(s): -> lizzi@cnam.fr Please check your system for viruses, or ask your system administrator to do so. For your reference, here are the headers from your email: ------------------------- BEGIN HEADERS ----------------------------- Received: from MOTSE-V0BGS5WDD

FreeCNAM.org is providing a free CNAM API for Open Source PBX users. This API queries a private CNAM database, and returns standard 15-Character CNAM results. Any entry not already in the database will be queued for investigation, and added to the database as soon as information is located. This system has access to several CNAM backends, and is not a party to any use-limiting or no-caching

Hi All, I've been in contact with a CNAM (caller name) directory provider. Currently they offer 2 products. One is CNAM via SS7 and the other is Directory Assistance data via http/xml. I am very interested in getting the CNAM data via http/xml (or DNS TXT). I suggested this to the sales rep and he told me that I'm the second person in less than a month who has asked for this. What

Hi all, I've installed mantis.noarch and mantis-config-httpd.noarch packages. The problem is I always get the same error in Apache:. You don't have permission to access /mantis on this server. It's curious 'cause: * I work in localhost mode and I've running drupal site and phpMyAdmin without problems. So, Apache works fine. * I've created mantis ddbb according the docs

Hi everyone, I am trying to figure out the behavior of trustrpid Basically its not behaving the way I expected it to or maybe I am missing a configuration option or something else. When a call from a phone is sent to the * box it has the following sip headers: From: "From Phone" <sip:1001 at 10.0.0.29>;tag=4bf4bb4e11e92476. Remote-Party-ID: "Cloutier"

Hi, Some efforts were put in having a customized version of MantisBT default visual style for The CentOS Bugs website. The work was done for mantis-1.1.8-5.el5 package from epel repository and is available for you to download at https://projects.centos.org/svn/artwork/trunk/Identity/Webenv/Themes/Default/Mantis/1.1.8-5.el5/. In this location there is a README file describing how to install

Hello, I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work! I placed this in my peer: (sip.conf) sendrpid=yes trustrpid=yes or sendrpid=yes trustrpid=no (and restarted Asterisk) and the line "Remote-Party-ID" does not appear in my sip debug! Please help me, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone have a MAX/APX with working ingress PRI calling name? I recently acquired a MAX TNT on the cheap and it's integrating fine except for one thing. In the 11.0.0 release notes, it is stated that ISDN calling name will, if present and permitted by presentation flags, be added to the From: and Remote-Party-ID: headers of the INVITE. I'm not able to make this happen. Pcap

On June 5, 2011, we migrated from Mantis to Jira as the issue tracker for Asterisk [1]. We temporarily left Mantis running in read-only mode to smooth the transition. At 15 months, temporary has turned into semi-permanent. As a part of other infrastructure changes we are making to the community services, we will finally shut down Mantis for good. We will update our DNS servers on the morning of

On June 5, 2011, we migrated from Mantis to Jira as the issue tracker for Asterisk [1]. We temporarily left Mantis running in read-only mode to smooth the transition. At 15 months, temporary has turned into semi-permanent. As a part of other infrastructure changes we are making to the community services, we will finally shut down Mantis for good. We will update our DNS servers on the morning of