similar to: Cisco Gateway sending call to * without CID Name

Hi All, My research indicates ANI is not really supported with SIP Channels or passed between SIP servers, even with setting function CALLERID(ANI). So the only place this applies is on PRI interfaces, when sending calls out a ZAP PRI you can set the ANI to whatever and CID Number to a different whatever so on the other end of the PRI you will receive the two different values? Is this correct or

Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip >< sip TNT pri >< pri asterisk The TNT is running 11.0.6 and the asterisk servers are running 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to asterisk but not the other way. The call from asterisk to pri to tnt is good, the TNT is passing SIP invite to the

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway

Hi Guys, I have several * servers connected to T1 PRI's from various service providers in multiple locations the US. All the * servers use the same hardware with the same OS and * version. When connected to 5ESS Switches, using the NI2 (national) PRI protocol, the CID name and number come across fine and populate into the * CDR fine. I connected to a DMS-500, NI2 (national) protocol and

Hi All, Just wanted to share a story: We turned up a new customer yesterday evening, typical situation, Cisco 2600 Router with T1 PRI card pointed to the customer's analog PBX with 2 data T1's linked back to our network. The router PRI was configured as a gateway on our CCM 4, like we've done numerous times in the past. The customer needed LD Authorization codes enabled, got

Hi All, Something happened on one of my 1.2.9.1 systems, SIP between * and Cisco Call Manager 4.1, leaving hung or open SIP sessions. No problem now, we found and corrected the problem. But while these hung sessions were increasing to around 480 to 500 sessions, I started getting "too many open files" on the asterisk console and sporadically could not establish new SIP connections.

Hi peeps, I've been struggling with DUNDi for a few days now and I can't seem to make call from Asterisk A to Asterisk B. If I do a "dundi show peers", it finds the other peer but I can't seem to make any calls. Can anybody help me out here. Here's the situation: Machine 1: Debian with Asterisk 1.4.4 --> 192.168.1.103 Machine 2: AsteriskNOW --> 192.168.1.69 The

Hi All, I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet !

Hi All, I'm running 1.2.9.1 and have a sip user setup with accountcode=4444 in the context. lab1*CLI> sip show peer 1234 * Name : 1234 Secret : <Set> MD5Secret : <Not set> Context : sip1004 Subscr.Cont. : <Not set> Language : Accountcode : 4444 AMA flags : Unknown CallingPres : Presentation Allowed, Not Screened Callgroup

I there, I have some trouble to do working caller id name for outgoing calls on the PRI we just installed. Caller id name work on incoming calls only. Caller id number work on incoming and outgoing calls. The provider, Goup Telecom, said that's in what i'm sending. They said that I send the cid name in ascii code and to do it working, I need to send it in hex. So I take some traces

Hi All, I'm running an AGI, calling a perl script the does number lookups to a remote server. I would like to put a timeout in the script. The problem I'm running into is if the DNS server is not responding, the script hangs and waits for 30 seconds before returning to the Asterisk dialplan. I would like a timeout of 1 second, then return. Here is my clean script:

Hi All, Anyone know the sip header to send to a Linksys to resync it's config file? Thanks. JR -- JR Richardson Engineering for the Masses

Hi All, Strange issue, recently I started getting a lot of .lock files in the voicemail /INBOX folder preventing proper access to voicemail. I can delete the .lock files and everything is normal. After searching around, I found some SIP lock file stuff but nothing specific to voicemail. Can someone point me in the right direction to resolve this? I'm runnning 1.2.9 on Debian Sarge.

Hi All, We are using MAX TNT to for some T1 PRI interconnects. I'm seeing the voice channels connect at 56K. Does anyone have the DS0 channels connecting at 64K for voice, if so what is the parameter to select 56k or 64k channels? I'm not having any issues that I know of, just wanted to bounce this off the group for a sanity check. Thanks. JR -- JR Richardson Engineering for the

Questions: 1. Is the wiki DUNDi example and the dundi.conf file too difficult to follow for new users? 2. Does the complexity of the DUNDi setup discourage you from using it or even attempting to configure it? 3. If there was a simple tutorial, step by step guide with easy to setup and test examples, would this encourage more users to investigate and use DUNDi? I'm interested in putting

Hi All, http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr Under example: exten => s,2,Set(CDR(MyFavoriteBand)=Foo Fighters) exten => s,3,Set(CDR(MyFavoriteSong)=Hero) and under description: -userfield: The channel's user specified field. ""-any custom value that you wish to store."" My question is how do you setup more custom fields in the cdr and be

Hi All, Has anyone else noticed that when a sip phone sitting behind a nat registers to asterisk using realtime database, the private IP of the phone is put into the fullcontact field instead of the public contact IP. The database has the correct public IP in the ipaddr field and correct port number in the port field, which is actually what asterisk uses to to contact the device. This

Hi All, I'm running 1.2.9.1 stable. I'm wondering has this patch been applied to stable release or is it still only in CVS. Will this file patch apply correctly to 1.2.9.1 stable? Which file do I patch? I'm guessing app_directory_realtime_1.6.1.patch <http://bugs.digium.com/file_download.php?file_id=4915&type=bug> and config.h.patch

FYI Kristian. http://www.astlinux.org/ Unable to connect to database server This either means that the username and password information in your settings.php file is incorrect or we can't contact the MySQL database server. This could mean your hosting provider's database server is down. The MySQL error was: Can't connect to local MySQL server through socket

> I have a customer that has two buildings that are connected with a > fiber link. We have a single Asterisk server to cover both buildings. > Now the customer went and bought an overhead paging system for the > remote building and they want to integrate it with Asterisk. Is there a > device that can connect over IP or an ATA that has an audio output port? > The buildings