similar to: Enable Media Atribute on 180 Ringing

Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks.

Hi Guys, I've setup a asterik box on a trunk with alcatel 4200 pabx. When operator do a call for somedestination terminated by our asterisk he can't transfer this call until called party answer that call. He can't transfer call when it's only ringing. This is a issue of Asterisk or from Alcatel. This PABX have 2 ISDN links. One with asterisk and other with other carrier.

Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. For alcatel users use asterisk lines, should dial 0 to take tone from asterisk. In default configuration in alcatel, if user dial 0 this error occour: !! Unexpected Channel selection 3 -- Extension '' in context 'default' from '' does not exist. Rejecting call on channel 0/31, span 1 In alcatel

Hi guys, It's possible i scheduler in cron some kind of script or application that read asterisk logs and send via e-mail a complete report for pbx activity in specified period ?? I like to see how simultanios calls was made, total time in conversation, averege time of calls, most routes calls, etc.... Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br -------------- next

Hi guys. I'm working on a VOIP service provider. We have two customers running asterisk. Customer A and B. When A call to B everything is ok. When B call to A the call ring but sip messages didn't arrive on asterisk A. In my softswitch i see the invite sip message sended to A. When every other numbers(TDM and SIP) call do A everything is ok. Have any issue in asterisk that can resolve

Hi guys, My asterisk is show me some errors on line registration. This message appear on console: Request to schedule in the past?!?! What it mean ? Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br

Hi guys, I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P. Input calls VOIP Proider ---> Asterisk ---> Alcatel Output Calls VOIP Proider <--- Asterisk <--- Alcatel In alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems: 1. When users dial 2 on phone (alcatel) they don't received a dial tone,

I have an asterisk box registering 100 numbers on a voip provider. Numers are: 2546.1000 to 2546.1099 My problem is that every incoming call arrived to number 2546.1099 that is the last number to register on voip provider. The correct is call arrive in destination number. See this exaple: I call to 2546.1000. -- Executing Dial("SIP/25461099-08738060", "Zap/g1/3000") in new

Hi, My asterisk was working fine but today my calls won't out of my asterisk box. Restarting asterisk i need to wait around 10 min to can run sip show registry command. If i try to run this command before, i receive a error like: no such command. Why this happen ? Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br

I'm registering 5 lines on my asterisk box from one voip provider. Lines; 4040.0000 4040.0001 4040.0002 4040.0003 4040.0004 All lines is registered in 5060 port so when someone call to 4040.0001 the call arrive on asterisk but arrive to last number registered 4040.0004becouse it is listening on same port as all others. How i make each number register in one different port, like

Hi guys, Recentlly i did a asterisk gateway and use it with an alcatel pabx. All is working, i have only two problems. 1. When call incomming to asterisk, it forward to digium card to PABX Alcatel. The user that start the call can't hear the control tone of ring ring ring. Tha calls stay without sound until the called part answer the call. At this point, conversation follow normaly. 2. When

Hi Guys, I Have this peers on my sip.conf [provider-302333-3000] type=friend context=provider secret=xpto username=3023333000 host=sip.provider.com fromuser=3023333000 insecure=very canreinvite=no [provider-302222-3001] type=friend context=provider secret=xpto username=3022223001 host=sip.provider.com fromuser=3022223001 insecure=very canreinvite=no I Have in my sip.conf two extension 3000

Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16

Hi guys, I have a customer with asterisk registering 100 lines from my Voip Provider. In some times during a day we receive this messages: [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request

Hi guys, I configure one Fedora Core Linux 5 for use with asterisk as gateway using Digium TE110P interconected in Alcantel 4100 I've set up it to register 100 voip numbers on my provider. All calls on Alcatel is send to asterisk. In some periods of day i receive this messages on asterisk console: Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer: Peer

Hi, I'm a voip service provider and i'm setting up a asterisk box to register around 100 lines from my central softswitch. This asterisk box will be placed inside a customer and has a digium card to be interconected with customer's pabx. My problem is that when asterisk send register message, my softswitch return with sip 401 and asterisk should send a register message with

Hi, I just compiled and installed Asterisk-1.4.2 along with zaptel-1.4.1 and libpri-1.4.0 on a Debian machine with a TDM400P card. Everything goes ok but when I try to make a call through the ZAP channel get an error message about NO ZAP CHANNEL AVAILABLE. Ztcfg and zttool show the card correctly installed. When I tried to use the debug command ZAP SHOW, it was not present in the CLI. My

Hello I am trying add smbpasswd for user root , my environment is samba integrated with openldap , I found the message down at moment that was creating smbpasswd for root, with command smbpasswd -a root , show this log in my sladp.log ldap_read: want=80, got=80 0000: 17 31 2e 33 2e 36 2e 31 2e 34 2e 31 2e 34 32 30 .1.3.6.1.4.1.420 0010: 33 2e 31 2e 31 31 2e 31 81 36 30 34 80 26 75 69

Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 180 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP

Few days back I installed Asterisk 1.4.2 with Zaptel 1.4.0. All SIP accounts were working fine, today I tried to install a fxs Sangoma A200 card and got the following error. [Apr 12 01:15:17] WARNING[31018]: channel.c:3024 ast_request: No channel type registered for 'Zap' [Apr 12 01:15:17] WARNING[31018]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap'