similar to: Cisco Call Manager SIP trunk + Asterisk

Hi, VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/307c10e9/attachment.htm

Hi, I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :(( I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :( This is my configurations files: - sip.conf - [general] port = 5060

No, you don't need a sound card. Do you have ztdummy loaded or zaptel device in your system? Regards, Gus ----- Original Message ----- From: "Chris Hariga" <contact@techselesta.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, October 19, 2003 8:19 PM Subject: [Asterisk-Users] Music on hold... > Hi, > > I need a sound card and mpg123 for music on

Hi, Can U tell me the Vonage ATA 186 settings? I would like to try to have a web interface on my adapter :-)) Best regards, Chris Hariga

Hi, si posible on SIP phones to have the dial tone after 9 like on the FXS card? I set ignorepat => 9 on my extensions.conf... Best regards, Chris HARIGA

Hi, I have Fedora Core 2 running with a T1 card. I try to put the log on db but I get the error: Aug 20 15:17:47 ERROR[262160]: cdr_addon_mysql.c:378 my_load_module: Failed to connect to mysql database asteriskcdrdb on localhost. The database exists and I try with "mysqlaccess localhost asteriskcdrdb" and I get: Access-rights for USER 'localhost', from HOST

Hi all, I want to make a SIP trunk between a Cisco 2811 router and a Asterisk. Both 2811 and Asterisk are working fine (2811 has 1XX and asterisk 2XX). Now I want to configure a trunk so that 2811 users can call * users. I've been reading a lot but I'm still confused. Hope you can help me, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

Hi, I install * and is working fine. I have 3 FXO cards w/ 3 phone lines. All the phones are SIP phones (Grandstream). The SIP phones from the same LAN w/ Asterisk are working but on the external phones (from the Internet) I don?t have sound. All the Grandstream phones from the Internet are register from different locations behind a NAT. All the sip users are register on * but the main issue is

Hi All, Does anyone have a config example for setting up SIP trunking to a CIsco IAD2400 and are willing to share? I've done SIP trunking to Cisco 2600's with PRI's but not to the POTS lines on the IAD's, I'm wondering if that is possible and how to specify the DID on the POTS line config for the IAD. Thanks. JR -- JR Richardson Engineering for the Masses

hello everybody, i want to configure a sip trunk between my system which has asterisk 11.5.1 and a cisco router. this is my scenario: Freepbx-----my system-----cisco-router----Freepbx my system acts like a router. if i set just one codec in dial-peers on cisco router, every thing is ok and i can make a call. but if i set different codecs in a voice class codec and assign it to dial-peers in

hello every body, i want to have h323 trunk between cisco 2800 and asterisk 11.13.1 with ooh323 module. i configured both side and have successful call from cisco to asterisk. but when call comes from asterisk to cisco, my phone rings but no audio is heard and call is disconnected after 5 second. i enable "debug voice rtp" in cisco and see the source address for receiving rtp packets

Hi, I've got a CCM ( Cisco Call Manager ), with a Cisco Unity VM server and about 45 SCCP phones on the ccm, and 200 users on unity. we want to migrate all users to IP Phones to ditch our ancient phone system. I would love to get Linksys-Sipura SPA-941s for the 150 users not on IP phones yet and run sip to an asterisk server, but have their voicemail on Unity. these phones are $150 each,

Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Thanks, Ben Blakely -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/a7e575cc/attachment.htm

It can be done. 1. Setup a new Vm profile on CCM with a mask of XXXX 2. Setup a CTI route point: a. Set the directory number to a pattern. I use *27XX but any pattern that you can send from * is good, ie. 88XXX b. Set the VM profile to the newly created profile c. Set the line to forward all calls to VM 3. Change the dialplan in * to append the extension called to the

Please CC me on answer, as I'm not subscribed on the list Here is the description of the problem. The system description follows. Everytime I read my email using KMail 3.5.6, dovecot hangs up near the end. I get the following in mail.err: IMAP(doudou): file ostream-crlf.c: line 339 (_send_istream): assertion failed: ((size_t)ret <= iov.iov_len) IMAP(doudou): Raw backtrace: imap

Hello all, I've some CTDB issue which I'm not sure where to start... I follow this guide by steve which is nice. The different on what have is that I don't have drbd running... Also I've 4 x GE which is all Connected to a switch with different ip but same subnet This is suppose to load balance the traffic as under samba dns, they all have the same name. I'm only planing 3

Hi I'm having a problem with setting up a kickstart environment based on CentOS 5.2 x86_64, on a Sun X2200 M2 server (both the server and the client in the kickstart environment are Sun X2200 M2 systems): the first attempt to load stage2.img fails with the error screen: "unable to retrieve http://192.168.11.10/source/images/stage2.img". Pressing the "OK" button brings up

Hello! However I'm a newbie in Asterisk/VOIP/CM I would like to make sure that this system design can work: Cisco 2811 Voice Gateway - sip trunk1 - asterisk on linux box - s?p trunk2 - Cisco Call Manager 6.0 There is also a Siemens Hicom old pbx connected with QSIG to the Voice Gateway. I would like to record all calls with mixmon going through Asterisk. Is it possible? Also if there is

Hi! Can anybody help with integration of asterisk with cisco call manager express? I tried to connect them via H323 and Sip, but wasn't successful in it. So can you share any advices or instructions?