RE: [Asterisk-Users] Netmeeting i can't hear voice
Problem solved. It was NAT. h323 not work behind NATD -Original Message- From: Roman Bessyadovskii Sent: 10 ÓÅÎÔÑÂÒÑ 2004 Ç. 12:59 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Netmeeting i can't hear voice Hi. After a small war with underfined sybol error and conflicts between h323 and oh323 I successfully install h323 channel. Now, I can connect from Netmeeting to SIP and ZAP channels, but I can't here anything. When I call at phone, and try to speak, on another end of line man said, that my voice very low. Microphone volume is maximum... Is there some parameters like rxgain, txgain for h323. Or it is another problem? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unknown IE 40 (cs6, Unknown Information Element)
Hi. I see that message in console. -- Zap/1-1 is ringing !! Unknown IE 40 (cs6, Unknown Information Element) -- Zap/1-1 answered SIP/1016-e34b As see in older messages it some Information send by phone station to my via PRI. But what does IE 40 mean? I cann't find information element 40 (as in know in hex) in documentation. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device
Hi, another problem. I configure TAPI driver for outlook. (https://sourceforge.net/projects/asttapi/) Yesterday all work fine. I configure that Local Phone is Zap/g1/772323 and external call is going to context default. When I call to sip - all work ok. When I call to city (via Zap) local phone ringing, and then I receive next message == Spawn extension (default, 3261090, 1) exited non-zero on 'SIP/1021-260b' -- Executing Dial(Zap/1-1, Zap/g1/989217510935) in new stack -- Called g1/989217510935 -- Zap/2-1 is ringing ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available -- Channel 0/1, span 1 got hangup Sep 15 18:48:54 WARNING[1149062448]: app_dial.c:357 wait_for_answer: Unable to forward frame -- Hungup 'Zap/2-1' == Spawn extension (default, 89217510935, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Here Zap/1-1 is call to LocalPhone, and Zap/2-1 is call to city. What does it mean? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Netmeeting i can't hear voice
Hi. After a small war with underfined sybol error and conflicts between h323 and oh323 I successfully install h323 channel. Now, I can connect from Netmeeting to SIP and ZAP channels, but I can't here anything. When I call at phone, and try to speak, on another end of line man said, that my voice very low. Microphone volume is maximum... Is there some parameters like rxgain, txgain for h323. Or it is another problem? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing
Thanks for help. All works now. Problem was in codecs on different sides Definity: display ds1 1b14 CRC? n Interface Companding: mulaw And when making call via asterisk Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law ^ (35) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 So I can't make call. But incoming call (Definity - Asterisk) works, because asterisk understand ulaw. So, I have once more question. How can I change codec on Digium card on Asterisk side? I configure asterisk and definity with this page (http://www.voip-info.org/wiki-Asterisk+Avaya), and here no one word about what codec asterisk use. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing
May be that information can help... On definity display dialplan analysis Page 1 of 3 DIAL PLAN ANALYSIS TABLE Percent Full: 6 Dialed Total Call Dialed Total Call Dialed Total Call String Length Type String Length Type String Length Type 0 4ext 1 4ext 2 4ext 3 4ext 2073 is extension - i.e. normal digital phone connected to definity on asterisk pri debug span 1 -- Executing Dial(SIP/1015-db22, Zap/g1/2073) in new stack -- Making new call for cr 32785 Protocol Discriminator: Q.931 (8) len=50 Call Ref: len= 2 (reference 17/0x11) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [28 0e b1 52 69 63 6b 20 41 74 72 65 69 64 65 73] Display (len=14) Charset: 31 [ Rick Atreides ] [6c 06 21 81 31 30 31 35] Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '1015' ] [70 05 a1 32 30 37 33] Called Number (len= 7) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '2073' ] -- Called g1/2073 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32785/0x8011) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 d8] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Incompatible destination (88), class = Invalid message (5) ] -- Processing IE 8 (cs0, Cause) -- Channel 2, span 1 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/2-1' == No one is available to answer at this time -Original Message- From: Ken Godee [mailto:[EMAIL PROTECTED] Sent: 30 ÉÀÌÑ 2004 Ç. 20:10 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing Roman Bessyadovskii wrote: Yes, I can make a call on that extension from other definity phone, if you mean it. -Original Message- From: Ken Godee [mailto:[EMAIL PROTECTED] Sent: 30 êáíó 2004 ú. 19:14 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing Roman Bessyadovskii wrote: Hi All. I connect asterisk and definity by manual at www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya. (I just only have E1, not T1 card). I see, that card work (in definity trunk status, and at asterisk Incoming call, from definity is work ok, but when I try outgoing call, I recive -- Executing Dial(SIP/1015-870f, Zap/g1/2073) in new stack -- Called g1/2073 -- Channel 1, span 1 got hangup -- Hungup 'Zap/1-1' == No one is available to answer at this time How fix it? Do you have the Dial Plan set up properly on the Definity side? No, that's not what I mean. You must understand how the Dial plan is used in the Definity. Are all your extensions on the Definity 4 digits starting with a 2? If you do not have first digit 2, length 4, type extension set up in the dial plan, Definity will not know what to do with the four digits your sending into the switch. do a display dialplan and make sure it is set up correctly. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing
Hi All. I connect asterisk and definity by manual at www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya. (I just only have E1, not T1 card). I see, that card work (in definity trunk status, and at asterisk == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 -- B-channel 3 successfully restarted on span 1 -- B-channel 4 successfully restarted on span 1 -- B-channel 5 successfully restarted on span 1 -- B-channel 6 successfully restarted on span 1 -- B-channel 7 successfully restarted on span 1 -- B-channel 8 successfully restarted on span 1 -- B-channel 9 successfully restarted on span 1 -- B-channel 10 successfully restarted on span 1 -- B-channel 11 successfully restarted on span 1 -- B-channel 12 successfully restarted on span 1 -- B-channel 13 successfully restarted on span 1 -- B-channel 14 successfully restarted on span 1 -- B-channel 15 successfully restarted on span 1 (Configured only 15 channels) Incoming call, from definity is work ok, but when I try outgoing call, I recive -- Executing Dial(SIP/1015-870f, Zap/g1/2073) in new stack -- Called g1/2073 -- Channel 1, span 1 got hangup -- Hungup 'Zap/1-1' == No one is available to answer at this time How fix it? Configs: Extensions.conf exten = 1015,1, Dial(SIP/1015,10,t) exten = 93261090,1, Dial(Zap/3/93261090) exten = 2073,1, Dial(Zap/g1/2073) sip.conf [1015] type=friend secret=phone host=dynamic restrictcid=no ; To have the callerid restriced - sent as ANI Zapata.conf [channels] context=default switchtype=national signalling=pri_cpe group=1 channel = 1-15 /etc/zaptel.conf defaultzone=us span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing
Yes, I can make a call on that extension from other definity phone, if you mean it. -Original Message- From: Ken Godee [mailto:[EMAIL PROTECTED] Sent: 30 ÉÀÌÑ 2004 Ç. 19:14 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing Roman Bessyadovskii wrote: Hi All. I connect asterisk and definity by manual at www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya. (I just only have E1, not T1 card). I see, that card work (in definity trunk status, and at asterisk Incoming call, from definity is work ok, but when I try outgoing call, I recive -- Executing Dial(SIP/1015-870f, Zap/g1/2073) in new stack -- Called g1/2073 -- Channel 1, span 1 got hangup -- Hungup 'Zap/1-1' == No one is available to answer at this time How fix it? Do you have the Dial Plan set up properly on the Definity side? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
