[Asterisk-Users] codec trouble?
Hi everyone! Situation: when I call from cell phone to a asterisk-connected phone, all works fine. When I call from the asterisk-connected phone (a Cisco 7960 SIP) to the cell, the connection gets made, but there is no audio going in either way... Asterisk reports the following: Sep 16 08:27:41 WARNING[245775]: chan_sip.c:2679 process_sdp: Insufficient information for SDP (m = '', c = 'IN IP4 123.123.123.123') Sep 16 08:27:47 WARNING[245775]: chan_sip.c:2679 process_sdp: Insufficient information for SDP (m = '', c = 'IN IP4 123.123.123.123') Sep 16 08:27:47 WARNING[360465]: rtp.c:1382 ast_rtp_bridge: codec0 = 268 is not codec1 = 0, cannot native bridge. == Spawn extension (sip, , 1) exited non-zero on 'SIP/105-1559' (123.123.123.123 is the IP of our VoIP-provider, is my cell phone, and 105 is the asterisk-connected phone). Regards, Evert ___ 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] 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] extensions.conf
Good day all I have my extensions sorted out nice but I need some help with more advance config.In short myne looks like this [company1] exten = s,1. plays the message,saying 1 for sales 2 for accounts ens . . . exten = 1,1,Dial(SIP/40615) exten = 1,2,Dial(SIP/403,15) exten = 1,3,Voicemail2(u406) exten = 1,4,Hangup ..ens No what I want it to do is if you press 1 it will dial sip 406 for 15 and 403 for 15s AND then give you a voice that says press 1 for voicemail,2 for mail menu and 3 to ring again? Can someone please help ? Thanks Altus ___ 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] transfering a call
Hello! I am playing around with transfering calls with chan capi. Here is a little example from the readme: --- example: exten = s,1,Answer exten = s,2,capiHOLD exten = s,3,capiECT,55:50 --- However, it only gets to capiHOLD and not further! So when i try to transfer a call, i am stuck in the hold loop. What am i doing wrong? Thanks, Mario ___ 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] extensions.conf
Steven suggested: You can accomplish your intended function by using either Macros, channel variables, or an include. And one more way to do it (just to show how flexible Asterisk is): --- [iax-demo] exten = s,1,Playback(demo-abouttotry) exten = s,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = s,3,Playback(demo-nogo) [some-menu] snip all exten = s... stuff exten = 300,1,Dial(Local/[EMAIL PROTECTED]/n,,g) exten = 300,2,goto(s,3) [some-other-menu] snip all exten = s... stuff exten = 300,1,Dial(Local/[EMAIL PROTECTED]/n,,g) exten = 300,2,Goto(s,1) --- Which, in the given example isn't the most efficient way of doing it! However, if extension 300 is slightly different in [some-menu] and [some-other-menu] and therefore can't be managed by an 'include = extension-300-stuff' or somesuch, then the advantages of a local dial command become apparent. Nick Barnes Senior Consultant. ___ 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] Cisco 79xx + asterisk + some functions Q
It seems that's it not that trivial to replace a common (commercial) PBX and to have instandly all these functions. Anybody experience with Cisco's CallManager and the support of the asked functions ? http://www.asternic.org This is very nice and will help a lot, that's for certain. thank you already for this much appriciated help alex -- visit us at Infosecurity NL - Stand 08.B121 13-14 october 2004 Jaarbeurs - Utrecht Netherlands Free Registration, click here: www.axsguard.com aXs GUARD has completed security and anti-virus checks on this e-mail (http://www.axsguard.com) ___ 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] ftp.digium.com/pub/asterisk/webmin
Hi everyone! Is it safe to use this (old!) webmin module with asterisk 1.0rc2? Regards, Evert ___ 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] Receiving queue urls
Hello list, I'd like to use the Send URL function in the Queue command, so that when an agent answers a call s/he sees the browser opening with a page having pertinent information. Anybody can tell me of a soft phone supporting this feature under Windows? I tried with SJPhone with no success. Thanks l. -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ 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] quality of musiconhold...
Hi everyone! I was wondering... Does the musiconhold quality improve if the mpg123 processes run with a negative priority? If so, is there a way to make them start like that, so I don't have to renice them? Regards, Evert ___ 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] Problems with native h323 channel on Asterisk RC2: No early audio and codec negotiation issues
Hello all, We have been testing Asterisk RC2 with the native H323 channel driver. We followed the instructions with the needed OpenH323 and PWLib versions and everything compiled ok. Operation of the driver seems ok, except from 2 main points: 1) Audio is passed between the two ends of the call only after the call is answered. This was not the case with previous versions of Asterisk (0.9.2 for example), in which audio would begin before the call was answered. Early audio is useful i order to provide the calling user with remote end ringback as well as recorded announcements, etc. 2) The codec capabilities that Asterisk sends seem strange. No matter which codecs we set in the h323.conf file, G711 is the only codec that is sent in the capabilities. In order to use any other codec, we have to enable only the needed codec and disable all others. Again, this problem did not exist in older * versions, like 0.9.2 and it's limiting the capabilities of Asterisk in H323. Has anyone dealt with this problem successfully? Best regards, -- Vlasis Hatzistavrou. ___ 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] H323 - Control Protocol Error (Master slave Determination)
Hi there ! I searched the whole web to find some helping information about H323 Control Protocol, but there is no way to find that information. We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2 + 'asterisk-oh323_1.5 channel driver + wrapper' and configured the dialplan for using our H323 Endpoints which are ip200 Innovaphones. Besides, we also use Gnomemeeting but don't care it's not the problem, I think ! The whole endpoints are registered on an ip400 Gatekeeper which routes every call to asterisk, and asterisk processes the Dialplan and sends the call back to the ip400 and to the correct Endpoint. With this configuration the Endpoints can dial each other above the Gatekeeper and Dial Plan. ;-) well pretty fine - the only damped thing is every call loses connection after 30 sec because of a a H323 control protocol error . this is the asterisk output while phoneing : ### *CLI -- Executing Dial(OH323/R1, OH323/[EMAIL PROTECTED]:1720|15) in new stack -- Called [EMAIL PROTECTED]:1720 -- OH323/L13468 answered OH323/R1 *** [ip$x.x.x.x:2507/1] H.323 CONTROL PROTOCOL ERROR (Capability Exchange) *** [ip$x.x.x.x:2507/1] H.323 CONTROL PROTOCOL ERROR (Master-Slave Determination) *CLI Sep 2 13:57:15 ERROR[294931]: chan_oh323.c:1212 oh323_hangup: OH323/L13468: Failed to hangup channel (timeout). -- Hungup 'OH323/L13468' == Spawn extension (buero, 3020, 1) exited non-zero on 'OH323/R1' -- Hungup 'OH323/R1' *CLI ### I hope you can help me and the whole asterisk community to solve this problem Hopefully, and waiting for response greets alex ___ 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] Cisco 79xx + asterisk + some functions Q
Just some more information regarding the 7914 addon for the 7960 phone. The 7914 requires upgraded firmware to be able to work with a 7960 of firmware 5.x or above. Do not upgrade the firmware of your 7960 above 5.x until you have done your 7914 first as you cannot downgrade the Cisco to a pre 5.X version in order to flash the 7914. I was experimenting with chan_sccp2 as it is claimed to have 7914 support. I got the 7960 working ok with chan_sccp2 but was unable to get the 7914 going and could find no information on getting it working aside from references to it and a screenshot on the sourceforge site. I also found the chan_sccp2 module to be reliable but not robust. For example pressing a speeddial button whilst on a call would bring down *, taking the handset offhook and leaving it offhook would also bring down *. In the end I have gone with the SIP image and am using the Flash Operators Panel which IMHO offers better functionality anyway. It is also cheaper to buy a 15 LCD panel and secondary display adapter, mounting the panel next to the users workstation than it is to buy the 7914 (Which for us cost more than the 7960 itself!) The Flash Operators Panel also has the ability to 'monitor' an extension and then launch a URL when a call comes in. Craig - Original Message - From: Alex Ongena [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, September 16, 2004 3:20 PM Subject: Re: [Asterisk-Users] Cisco 79xx + asterisk + some functions Q It seems that's it not that trivial to replace a common (commercial) PBX and to have instandly all these functions. Anybody experience with Cisco's CallManager and the support of the asked functions ? http://www.asternic.org This is very nice and will help a lot, that's for certain. thank you already for this much appriciated help alex -- visit us at Infosecurity NL - Stand 08.B121 13-14 october 2004 Jaarbeurs - Utrecht Netherlands Free Registration, click here: www.axsguard.com aXs GUARD has completed security and anti-virus checks on this e-mail (http://www.axsguard.com) ___ 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
Re: [Asterisk-Users] Intertex IX66
On Thu, 9 Sep 2004 12:26:56 -0400, Chris HARIGA [EMAIL PROTECTED] wrote: Hi, I have Asterisk w/ 192.168.1.1 and I setup IX66 to be 192.168.1.2 (I'm using pppoe client and dyndns.org on IX66) I setup on Local DNS Server my * box and after that I was able to register my phones from the Internet. I cannot understand my problem with one way sound... what is wrong on my configuration :(( As the IX66 is a sip aware router make sure you have no entries for nat in your sip.conf, and let the ix66 deal with the nat, not * . I hope this helps. Jason ___ 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] Re: IAX to IAX connect question
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] writes: An entry of type peer on the local Asterisk has to be matched with an entry of type user on the remote Asterisk. Likewise, an entry of type user on the local Asterisk has to be matched with an entry of type peer on the remote Asterisk. This is what I do between the two Asterisks I run, as well as for my connection to FWD. It works fine. However, I notice that when an incoming call arrives, it's logged on the Asterisk console as originating at the _peer_, not the _user_. With this in place: [general] register = USER:[EMAIL PROTECTED] [iaxfwd] type=user context=default auth=rsa inkeys=freeworlddialup host=iax2.fwdnet.net [iaxfwd-gw] type=peer username=USER auth=md5 secret=PASS host=iax2.fwdnet.net qualify=yes ...everything looks great. I've got the registry in place, the user is defined to use the specified key for authentication, and the peer is reachable. Calls in either direction work. However, incoming calls are shown as coming from, say, IAX2/[EMAIL PROTECTED]/5. If I remove the host= line from the [iaxfwd] entry, incoming calls fail, so it is obviously _using_ that entry to authenticate FWD. This feels like a bug to me. Or have I misunderstood something? -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ 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] Thoughts on Adding Locking to db.c?
We're working on an application in which it appears it would be far more efficient to share data between Asterisk and external applications by simultaneously accessing and updating astdb. While the current asterisk/db.c code uses ast_mutex_lock and unlock pairs to protect the integrity of astdb from multiple Asterisk threads, this of course does nothing to protect astdb from external (i.e. non-Asterisk) apps. Does the list have any thoughts about the advisability (or inadvisability) of modifying db.c to use flock instead of ast_mutex_lock? As an aside I am aware that one could use AGI or an external SQL database for such data sharing; I would just prefer to avoid such overhead or complications in this situation. One could even envision making this a configuration option (i.e. astdb = shared). Thoughts and flames please. George Pajari netVOICE communications www.netvoice.ca www.ip-centrex.ca ___ 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] One Question
Hi friends, Redhat9.0 installed in my machine.Asterisk1.0 installed in that machine. Thereis no mpg123 player. So, I download the mpg123 player and installed it. My sound card is configured correctly. When I tried to check asterisk feature SetMusicOnHold its not working im not able to hear any sounds. But the same config (extensions.conf and musiconhold.conf) is worked for Mepis2003 in another machine. Any one can suggest me Thanks in advance Regards Murali___ 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] No Caller Name sent from Asterisk over National or DMS100 PRI to a Norstar MICS?
I have a PRI link up and running between Asterisk and a Nortel Norstar MICS
v4.1 . I'm having a problem getting the textual Caller Name across the link
from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns
to Ast both elements come through fine. I'm forcing dummy values for testing
using:
exten = s,1,SetCIDName(Test)
exten = s,2,SetCallerID(1234561234)
exten = s,3,Dial(zap/g1/${ARG1},15)
I've tried switchtype=national and dms100 (adjusting accordingly on Ns) with
no change. zapata.conf is currently:
; Norstar #2 (Wharf Road)
context=in-t1nstar
group=1
usecallerid=yes
hidecallerid=no
usecallingpres=no
switchtype=dms100
pridialplan=local
signalling=pri_net
channel = 1-23
The SETUP frame from Ast contains:
Protocol Discriminator: Q.931 (8) len=40
Call Ref: len= 2 (reference 2/0x2) (Originator)
Message type: SETUP (5)
[04 03 80 90 a2]
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: u-Law (34)
[18 03 a1 83 81]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ]
[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) ]
[6c 0c 21 80 31 32 33 34 35 36 31 32 33 34]
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) '1234561234' ]
[70 05 c1 36 31 30 31]
Called Number (len= 7) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '6101' ]
Which doesn't seem to even contain the CIDName... On the other hand, the
SETUP frame from the Ns contains:
Protocol Discriminator: Q.931 (8) len=56
Call Ref: len= 2 (reference 1/0x1) (Originator)
Message type: SETUP (5)
[04 03 80 90 a2]
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: u-Law (34)
[18 03 a1 83 97]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 23 ]
[28 0b b1 53 43 52 44 20 4b 72 69 73 42]
Display (len=11) Charset: 31 [ SCRD KrisB ]
[6c 0c 21 80 36 30 34 38 38 35 36 38 30 38]
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) '6048856808' ]
[70 0c 80 39 36 30 34 38 38 35 36 38 30 38]
Called Number (len=14) [ Ext: 1 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0) '96048856808' ]
Which has the textual ID in the 'Display' element... However I understand
from http://resource.intel.com/telecom/support/tnotes/tnbyos/2000/tn033.htm
that there is no definitive standard for transmitting the name.
So, should even I be expecting Ast to put the name on the wire when it's
originating?
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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Re: [Asterisk-Users] One Question
On Thu, 2004-09-16 at 09:35 +, Murali wrote: Hi friends, Redhat9.0 installed in my machine.Asterisk1.0 installed in that machine. Thereis no mpg123 player. So, I download the mpg123 player and installed it. My sound card is configured correctly. When I tried to check asterisk feature SetMusicOnHold its not working im not able to hear any sounds. But the same config (extensions.conf and musiconhold.conf) is worked for Mepis2003 in another machine. Any one can suggest me Check the version of mpg123 is it 0.59r this is the only version that really works. -- Dave Cotton [EMAIL PROTECTED] ___ 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] One Question
Dave Cotton wrote: On Thu, 2004-09-16 at 09:35 +, Murali wrote: Hi friends, Redhat9.0 installed in my machine.Asterisk1.0 installed in that machine. Thereis no mpg123 player. So, I download the mpg123 player and installed it. My sound card is configured correctly. When I tried to check asterisk feature SetMusicOnHold its not working im not able to hear any sounds. But the same config (extensions.conf and musiconhold.conf) is worked for Mepis2003 in another machine. Any one can suggest me Check the version of mpg123 is it 0.59r this is the only version that really works. What's wrong with 0.59s? That one seems to work fine as well...8-) ___ 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] One Question
On Thu, 2004-09-16 at 12:00 +0200, Evert Meulie wrote: Dave Cotton wrote: On Thu, 2004-09-16 at 09:35 +, Murali wrote: Hi friends, Redhat9.0 installed in my machine.Asterisk1.0 installed in that machine. Thereis no mpg123 player. So, I download the mpg123 player and installed it. My sound card is configured correctly. When I tried to check asterisk feature SetMusicOnHold its not working im not able to hear any sounds. But the same config (extensions.conf and musiconhold.conf) is worked for Mepis2003 in another machine. Any one can suggest me Check the version of mpg123 is it 0.59r this is the only version that really works. What's wrong with 0.59s? That one seems to work fine as well...8-) If you look at the archives you will find this has been discussed at length. 0.59r works for * 0.59s does not. You want MOH to work you use what works with *. -- Dave Cotton [EMAIL PROTECTED] ___ 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] One Question
Dave Cotton wrote: On Thu, 2004-09-16 at 12:00 +0200, Evert Meulie wrote: Dave Cotton wrote: On Thu, 2004-09-16 at 09:35 +, Murali wrote: Hi friends, Redhat9.0 installed in my machine.Asterisk1.0 installed in that machine. Thereis no mpg123 player. So, I download the mpg123 player and installed it. My sound card is configured correctly. When I tried to check asterisk feature SetMusicOnHold its not working im not able to hear any sounds. But the same config (extensions.conf and musiconhold.conf) is worked for Mepis2003 in another machine. Any one can suggest me Check the version of mpg123 is it 0.59r this is the only version that really works. What's wrong with 0.59s? That one seems to work fine as well...8-) If you look at the archives you will find this has been discussed at length. 0.59r works for * 0.59s does not. You want MOH to work you use what works with *. Is it possible to search the archives somewhere online? Downloading all those monthly files in mbox format would be a bit too time-consuming for me... ___ 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] ZAP Hook flash / recall on active zap interface
Hi there, I have a x100p card in an asterisk box. Does anyone know if it's possible to do a hook flash / recall on an active zap channel? On what I'm trying to do... From an ordinary analogue pstn telephone I can call someone, press recall, call someone else, press recall 3 presto we're on a three way chat, with me only using one line - using the telephone company's 3waychat feature... I have tried Flash(Zap/1), and similar commands, however an error is returned Unable to create channel of type 'zap'. I guess * is trying to open another non existent zap interface here... So, does anyone know if / how this is possible using asterisk, with just the one zap interface (x100p card)... grateful for any feedback cheers sophus ___ 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] One Question
On Thu, Sep 16, 2004 at 12:18:00PM +0200, Evert Meulie said: Dave Cotton wrote: If you look at the archives you will find this has been discussed at length. 0.59r works for * 0.59s does not. Is it possible to search the archives somewhere online? Downloading all those monthly files in mbox format would be a bit too time-consuming for me... Google for: site:lists.digium.com mpg123 The first hit seems to be what you are looking for. the site: option limits google to a specific website. ___ 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] Current bristuff error report
Hello, I just noticed an error in the current version of Klaus-Peter Junghanns bristuff package, especially in the HFC module. Everytime I try to unload the HFC module with modprobe -r I got a kernel panic and the complete server hangs up so I need to do a hard reset. Regards, Julian Pawlowski ___ 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] Extension based call forwarding using capiECT
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi! I tried out some more stuff and found out the following: exten = 5,1,Dial(CAPI/279:b0175203,30) instead of exten = 5,1,capiHOLD exten = 5,2,capiECT,279:0175203 seems to work for me. Is that the right way to do it? Thanks in advance for your answers. Benne Am 15.09.2004 um 19:38 schrieb Benjamin Boksa: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I try to get callers forwarded to by mobile phone when they dial a certain digit. In my extensions.conf I have defined the following: [279] exten = s,1,SetLanguage(de) exten = s,2,Wait,5 exten = s,3,BackGround(demo-congrats) exten = s,4,Goto(boksa,#,1) exten = 3,1,VoiceMail,u1 exten = 4,1,VoicemailMain exten = 4,2,Hangup exten = 5,1,capiHOLD exten = 5,2,capiECT,279:017520x exten = t,Goto(boksa,#,2) When I try the setup by dialing 4 while demo-congrats is playing my mobile phone rings and the caller number is spoken, but no connection between the caller and the mobile phone is established. This is the output from asterisk -: == CDR updated on CAPI[contr1/279]/0 -- Executing capiHOLD(CAPI[contr1/279]/0, ) in new stack Sep 15 19:09:05 NOTICE[245776]: app_capiHOLD.c:73 capiHOLD_exec: sent FACILITY_REQ PLCI = 0x101 Sep 15 19:09:05 NOTICE[245776]: app_capiHOLD.c:84 capiHOLD_exec: PLCI = 0x101 is on hold now -- Executing capiECT(CAPI[contr1/279]/0, 279:017520x) in new stack Sep 15 19:09:05 NOTICE[245776]: app_capiECT.c:65 capiECT_exec: ECT to 279:017520x Sep 15 19:09:17 NOTICE[245776]: app_capiECT.c:74 capiECT_exec: call was answered -- Playing 'digits/0' (language 'de') Sep 15 19:09:17 WARNING[245776]: file.c:902 ast_waitstream: Unexpected control subclass '14' -- Playing 'digits/1' (language 'de') -- Playing 'digits/6' (language 'de') -- Playing 'digits/2' (language 'de') -- Playing 'digits/4' (language 'de') -- Playing 'digits/1' (language 'de') -- Playing 'digits/4' (language 'de') -- Playing 'digits/3' (language 'de') -- Playing 'digits/6' (language 'de') -- Playing 'digits/9' (language 'de') -- Playing 'digits/7' (language 'de') Sep 15 19:09:23 NOTICE[245776]: app_capiECT.c:132 capiECT_exec: sent DISCONNECT_B3_REQ NCCI=0x10201 Sep 15 19:09:23 NOTICE[245776]: app_capiECT.c:155 capiECT_exec: onholdPLCI = 257 Sep 15 19:09:23 NOTICE[245776]: app_capiECT.c:177 capiECT_exec: sent FACILITY_REQ PLCI = 0x201 (0x1 0x1) onholdPLCI = 0x101 Sep 15 19:09:23 NOTICE[245776]: app_capiECT.c:178 capiECT_exec: FACILITY_REQ ID=007 #0x01aa LEN=0022 Controller/PLCI/NCCI= 0x101 FacilitySelector= 0x3 FacilityRequestParameter= 06 00 04 01 01 00 00 My MSN is 279 and my mobile phone number is 017520x. What have I done wrong? Is it possible to do that? Thanks a lot for your answers in advance, Benne -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBSH4zR5U9XkJXZKwRApyQAJ9EEsPm6K9t0NrONTb1UX5u1kF2AwCaAz+U cXFpxoWl52ojsEw+cF6e1Qk= =V7wa -END PGP SIGNATURE- ___ 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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBSXeBR5U9XkJXZKwRAuSHAKCHcEzQlrPTDiy7j4vwjpyueJbeFgCeNNrb JrCQcff5uWBgesLT01zSsgs= =lh8J -END PGP SIGNATURE- ___ 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] Audiocodes Mediant 2000
Hi FOlks, I am trying to setup remotely an AudioCodes Mediant 2000 MG Module 2 to work with Asterisk through SIP or H323. But since I don't the product manual, it's being a little hard. Anybody would the manual in PDF(file or URL) to indicate to me? Thanks a lot, Isamar ___ 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] IAX to IAX connect question
I think i got the solution for what i was planing to set. Here is a
ontheway sample (not what i got but its about the same)
Office iax.conf
---
register = 123456:[EMAIL PROTECTED]
jitterbuffer=no
tos=lowdelay
[iaxfwd]
type=user
context=fromiaxfwd
auth=rsa
inkeys=freeworlddialup
diallow=all
allow=ulaw
[myofficename]
type=peer
host=dynamic
auth=rsa
outkeys=myrsa
username=myofficename
context=somecontext
[user01]
type=friend
user=user01
host=dynamic
secret=somepass01
username=user01
context=accesslevel01
[user02]
type=friend
user=user02
host=dynamic
secret=somepass01
username=user01
context=accesslevel01
Office extensions.conf
--
[general]
static=yes
writeprotect=no
[globals]
MYUSER01=IAX2/myofficename:[EMAIL PROTECTED]
MYUSER02=IAX2/myofficename:[EMAIL PROTECTED]
MYOFFICENAMECID=Some name
MYFWDUP=IAX2/123456:[EMAIL PROTECTED]
[extensions]
; set of extensions
; for testing like echotest and others
; or whatever else needed
[fromiaxfwd]
exten = 123456,1,Answer
exten = 123456,2,Dial(${MYUSER01}${MYUSER02},60,r)
exten = 123456,3,Hangup
[toiaxfwd]
exten = _8.,1,SetCallerId,${MYOFFICENAMECID}
exten = _8.,2,Dial(${MYFWDUP}/${EXTEN:1},60,r)
exten = _8.,3,Congestion
[accesslevel01]
include = extensions
ignorepat = 8
include = toiaxfwd
User01 iax.conf
---
register = user01:[EMAIL PROTECTED]
[myofficename]
type=user
context=fromoffice
auth=rsa
inkeys=myrsa
User01 extensions.conf
--
[globals]
MYOFFICE=IAX2/user01:[EMAIL PROTECTED]
FWDCIDNAME=My name01
[extensions]
; my local extensions
[fromoffice]
exten = s,1,goto(extensions,101,1) ; where the zap/1 is located
[toiaxfwd]
exten = _8.,1,SetCallerId,${FWDCIDNAME}
exten = _8.,1,Dial(${MYOFFICE}/${EXTEN},60,r)
exten = _8.,2,Congestion
[localaccess]
; set of local pstn access
[dialaccess]
; where zap/* or local sip phones should point
include = extensions
ignorepat = 8
include = toiaxfwd
ignorepat = 9
include = localaccess
User02 iax.conf
---
register = user02:[EMAIL PROTECTED]
[myofficename]
type=user
context=fromoffice
auth=rsa
inkeys=myrsa
User02 extensions.conf
--
[globals]
MYOFFICE=IAX2/user02:[EMAIL PROTECTED]
FWDCIDNAME=My name02
[extensions]
; my local extensions
[fromoffice]
exten = s,1,goto(extensions,201,1) ; where the zap/1 or sip is located
[toiaxfwd]
exten = _8.,1,SetCallerId,${FWDCIDNAME}
exten = _8.,1,Dial(${MYOFFICE}/${EXTEN},60,r)
exten = _8.,2,Congestion
[localaccess]
; set of local pstn access
[dialaccess]
; where zap/* or local sip phones should point
include = extensions
ignorepat = 8
include = toiaxfwd
ignorepat = 9
include = localaccess
So, in this way, i can keep adding users in the office using only one
context for each user with its own user/pass for validation.
Now, here it comes another thing. When i call from user01 (or home) to FWD,
as soon as it answer it hangsup. There was just a couple times i could do
the FWD echotest or the 411, but not anymore but incoming calls from FWD and
from office works fine. Does anyone see something wrong?
Regards,
Raul Elizondo
FWD# 486533
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[Asterisk-Users] Help with E1 configuration
Hi, I currently have a E100P card installed on my machine and the E1 subscription will be activated pretty soon. However, I have no idea how to configure asterisk to make inbound and outbound call using the E1. Especially for extensions.conf. Below is the configuration I used for zaptel.conf and zapata.conf. Is it possible if someone can verify if the configuration for zaptel and zapata is correct? zaptel.conf --- span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 zapata.conf --- switchtype=euroisdn signalling=pri_cpe group=1 context=default channel=1-15,17-31 I have 1 block of 10 DID numbers that will be subscribed together with E1. I am not able to find any sample for the extensions.conf to do inbound and outbound call. Is it possible for someone could post a sample of how the configuration would look like. Any setting missing for callerid support? PS: I already have an existing asterisk system running on analog ports. This is just an upgrade. Thanks in advanced. Regards, Chin _ Fast. Clear. Easy. The new MSN Search. http://search.msn.com.sg/ ___ 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] Audiocodes Mediant 2000
On Thu, 16 Sep 2004 20:22:57 +0900 (JST), Isamar Maia [EMAIL PROTECTED] wrote: Hi FOlks, I am trying to setup remotely an AudioCodes Mediant 2000 MG Module 2 to work with Asterisk through SIP or H323. But since I don't the product manual, it's being a little hard. Anybody would the manual in PDF(file or URL) to indicate to me? Google found this it may help http://corp.deltathree.com/productsandservices/manuals/bizlink.pdf ___ 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: Re: [Asterisk-Users] Audiocodes Mediant 2000
But since I don't the product manual, it's being a little hard. Anybody would the manual in PDF(file or URL) to indicate to me? Google found this it may help http://corp.deltathree.com/productsandservices/manuals/bizlink.pdf I have seen that already... looking something more objective. I just read that and didn't understand anything Isamar ___ 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] No Caller Name sent from Asterisk over National or DMS100 PRI to a Norstar MICS?
Bill Lohr, et al:
I can say from personal experience that with a PRI in MD (Verizon or
Verizon-CLEC) territory, it is possible to inject CALLER ID NUMBER on a
per call basis regardless of what channel the call originates from. The
callee's PSTN carrier performs a reverse lookup on the NUMBER and displays
whatever name is in the public directory they use for the reverse lookup.
For instance, it is possible to set your Caller ID Number on an outbound
PRI call to 202-456-1414; on the callee's caller ID Display, the name and
number will read THE WHITE HOUSE 202-456-1414. However, I do not
recommend doing this. It is just a colorful example.
I do not believe it is possible to set Caller ID Number on a per-call
basis using anything other than a PRI or other ISDN/SS7 interconnection.
Possibly there are ways it can be tweaked with other types of signalling
but most carriers are probably unwilling/unable to support it. I do not
believe there is any instance where the PSTN will pay any attention at all
to *-set Caller ID Name fields on outbound calls; this app/field is
seemingly only used by non pstn channels, such as SIP.
Setting Caller ID Number dynamically on a per call basis on a POTS,
channelized T1, or other sort of line is definitely not possible.
With Caller ID Name for inbound calls, this is a configurable setting on a
PRI and your provider may or may not be giving you that data. It is
almost always sent on a POTS line. Additionally there are some special
values for the Caller ID Name field that CPE can interpret: O means Out
of Area, P for Private, etc.
As for solving the Caller ID Name problem for outbound calls, I am
somewhat stumped. Presumably, this data is generated, compiled, and
maintained by the ILEC/CLECs involved. In theory, CLECs who issue phone
numbers to their customers should be responsible for the reverse mapping
and sharing of this information, however, as we live in a world where
CLEC's freely trade numbering resources and reverse lookups are not a top
business priority, results may vary considerably. Not to mention most
ILECs couldn't care less about CLEC numbers and what is displayed.
They'd be happy to sabotage that process entirely and undoubtedly that's
what they are doing, whether actively or by default.
Outbound calls from Vonage, which is for the most part PRI based, indicate
a proper Caller ID Number, however the reverse name lookups I have seen
indicate VONAGE as the Caller ID Name. Since Vonage is getting its
lines from various CLECs, somewhere somebody has managed to set the
reverse lookup for their numbers to VONAGE in a public database that
Verizon listens to. If anyone has any real insight or experience with
this process or the applicable databases, I'd love to hear about it.
Dave
I have a PRI link up and running between Asterisk and a Nortel Norstar MICS
v4.1 . I'm having a problem getting the textual Caller Name across the link
from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns
to Ast both elements come through fine. I'm forcing dummy values for testing
using:
exten = s,1,SetCIDName(Test)
exten = s,2,SetCallerID(1234561234)
exten = s,3,Dial(zap/g1/${ARG1},15)
I've tried switchtype=national and dms100 (adjusting accordingly on Ns) with
no change. zapata.conf is currently:
; Norstar #2 (Wharf Road)
context=in-t1nstar
group=1
usecallerid=yes
hidecallerid=no
usecallingpres=no
switchtype=dms100
pridialplan=local
signalling=pri_net
channel = 1-23
The SETUP frame from Ast contains:
Protocol Discriminator: Q.931 (8) len=40
Call Ref: len= 2 (reference 2/0x2) (Originator)
Message type: SETUP (5)
[04 03 80 90 a2]
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: u-Law (34)
[18 03 a1 83 81]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ]
[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) ]
[6c 0c 21 80 31 32 33 34 35 36 31 32 33 34]
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) '1234561234' ]
[70 05 c1 36 31 30 31]
Called Number (len= 7) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '6101' ]
Which doesn't seem to even contain the CIDName... On the other hand, the
SETUP frame from the Ns contains:
Protocol Discriminator: Q.931 (8) len=56
Call Ref: len= 2 (reference 1/0x1) (Originator)
Message type:
Re: [Asterisk-Users] quality of musiconhold...
Current cvs builds of * seem to be spawned with a -20 niceness level automatically. Believe it's coded into res_musiconhold now. Dave Hi everyone! I was wondering... Does the musiconhold quality improve if the mpg123 processes run with a negative priority? If so, is there a way to make them start like that, so I don't have to renice them? Regards, Evert ___ 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 -- = David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales ToadNet - Expect More!410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net ___ 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] Broadvoice BYOD Plans - 3-way and Call Waiting
On Thu, 16 Sep 2004, Ben Wern wrote: I've already run into some trouble with Broadvoice. Broadvoice support tells me that support isn't really available to BYOD plans, which I suppose I understand given the variety of devices out there. I'm hoping that someone on Asterisk-Users has seen the two issues I'm running into and has a suggestion. They don't officially support Asterisk, but when I've called for support the gentleman asked if I was running Asterisk and then gave me some ideas as to what the problem that I was experiencing was related to. The first issue I'm seeing is that incoming caller id shows the number as out of area and the name shows as 147.135.8.129;bvoice I don't have this problem with other incoming SIP providers -- is there some tweak I need to make Asterisk see CID information from Broadvoice? I've not seen this. While I've not connected up a CID capable phone to my phone adapter, the Asterisk debug output clearly shows the proper CID name and CID number when a call comes in. I'm running Asterisk 1.0_RC2 with a Sipura SPA-2000 as my analog phone adapter. Ryan Wilkins ___ 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] No Caller Name sent from Asterisk over Nationalor DMS100 PRI to a Norstar MICS?
The owner of the connection to the PSTN (Telco) must insert the NAME portion for Call Display. There is no way around that since its their database the NAME is located in. Someone correct me if I am wrong . Brandon For instance, it is possible to set your Caller ID Number on an outbound PRI call to 202-456-1414; on the callee's caller ID Display, the name and number will read THE WHITE HOUSE 202-456-1414. However, I do not recommend doing this. It is just a colorful example. ___ 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] One Question
On Thursday 16 September 2004 11:18, Evert Meulie wrote: Is it possible to search the archives somewhere online? Downloading all those monthly files in mbox format would be a bit too time-consuming for me... you can read a newgroup feed from www.gmane.org works pretty well. Jon ___ 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] ASTRICON Atlanta Sept 22-24
How many people are going to be attending Astricon in Atlanta Sept 22-24 ? Here is the URL: http://www.astricon.net/ If you are technical and want questions answered I cannot stress how important this first ever conference will be. *No I am not with the conference but, LiveVoip people will be there. For many of you this would be the chance to talk to. Alot of brain power will be there. Introduction to Asterisk John Todd Implementing CLASS features with Asterisk Jerry D. Doty IP network design for VoIP John Brown, Chagres Networks Asterisk dial plan tricks and tips Brian Capouch Who's waiting? Asterisk call queues and agents Francois Lambert Advanced SIP Tutorial Alan Hawrylyshen, SIP Foundry.org Asterisk and the old phone system (PSTN) Paul Mahler, Signate Supporting Asterisk Matthew Fredricksson Performance and Scalability Joachim Vanheuverzwijn (Zoa) Visualising Asterisk - the GUI Jim Thompson Asterisk on FreeBSD Rich Murphey Advanced Asterisk Brian K. West and Josh Roberson ___ 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] No Caller Name sent from Asterisk over Nationalor DMS100 PRI to a Norstar MICS?
The owner of the connection to the PSTN (Telco) must insert the NAME portion for Call Display. There is no way around that since its their database the NAME is located in. Someone correct me if I am wrong . Yes, I think it's fair to say that the ILEC/CLEC to whom the phone number is routed is responsible for publishing reverse name lookup for that number. This is somewhat analogous to in.addr.arpa reverse lookup for DNS. It should be noted that, with LNP, any single phone number can be bound to any carrier, so there is not necessarily any notion of native numbers for a given carrier, etc. That all being said, does anyone have any experience with what databases/mechanisms CLEC's might use to maintain and disseminate these reverse lookups? I can think of some weird scenarios: A) 443-555-1212 is a published number for Consolidated Cheese Corp and is initially serviced by Verizon. B) Consolidated Cheese later ports 410-555-1212 to a facilities-based CLEC-provided PRI C) Consolidated Cheese signs up for a VoIP termination service and wants to set caller ID to 410-555-1212 for its PSTN-bound calls. The VoIP termination service is provided by a different CLEC from (B), possibly in a different geography. So, assuming CLEC in (B) has done its job and published some sort of reverse lookup for 410-555-1212, calls made via (C) should, in theory, correctly display Caller ID Name for the callee's. But how? The only way I can think of for this to work is for a dip to be made for the reverse lookup, presumably via a SS7 request, to the (B) CLEC, even though the call may not originate with, terminate on, or otherwise pass through (B)'s switch network. (B) would only be touched for the reverse name lookup. If this outline is close to reality, is there any notion of distributing these name lookups, a la DNS? What about caching/TTL? Where is this stuff written down? Comments? Dave ___ 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] problem connecting to icallglobe
Anyone has successfully used Asterisk with icallglobe's SIP termination service? I'd been trying to get my Asterisk box to terminate international calls through them. Asterisk seems to register OK, but whenever I send a call to icallglobe's gateway, I always get a '403 forbidden'. What I've done so far: a. register with their gw in sip.conf b. Defined a peer [icallglobe] in sip.conf c. Set secret, username, fromuser, fromdomain etc d. In my extension.conf, I have exten = _0XXX.,1,Dial(SIP/[EMAIL PROTECTED]) e. At the asterisk console, I used 'sip show registry' to make sure the status is 'registered' When I make a call, * will send an INVITE to icallglobe's gateway but it always gets back a '403 Forbidden'. For example, if I dial '0651711' on my phone, I can see * dialing '651711@icallglobe gw IP'. After a while, * complains about getting '403...' from icallglobe gw IP. Did I missed the obvious? Cheers and TIA. Leo ___ 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] No Caller Name sent from Asterisk over Nationalor DMS100 PRI to a Norstar MICS?
If I recall correctly, the National ISDN protocol (NI2, I think) has the capability of forwarding CID NAME to the provider who can then do whatever they want with that information (including simply discard it). On Sep 16, 2004, at 5:29 AM, Brandon Patterson (peering) wrote: The owner of the connection to the PSTN (Telco) must insert the NAME portion for Call Display. There is no way around that since its their database the NAME is located in. Someone correct me if I am wrong . ___ 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] 3 Way Calling on Snom Phones and Asterisk
Has anyone been able to get 3way/Conference working with the snom200 and Asterisk. According to the documentation for the phones the option should come up when you have two lines active on the snom phone. Unfortunately, I don't see this option appear and I am now beginning to wonder if this is a limitation of Asterisk. Does anyone have any suggestions? Any help would be greatly appreciated. Thanks, Brian ___ 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] ZAP Hook flash / recall on active zap interface
The flash application flashes que current ZAP device, it doesn't take a parameter AFAIK. Correct me if I'm wrong. For instance, you could use this for manual call deflection on a POTS line, if the user asks to be transfered to another branch, you could use Flash then Dial the DTMF digits to call the other extension and transfer. Marcelo Pacheco Em Qui 16 Set 2004 07:25, Sophus escreveu: Hi there, I have a x100p card in an asterisk box. Does anyone know if it's possible to do a hook flash / recall on an active zap channel? On what I'm trying to do... From an ordinary analogue pstn telephone I can call someone, press recall, call someone else, press recall 3 presto we're on a three way chat, with me only using one line - using the telephone company's 3waychat feature... I have tried Flash(Zap/1), and similar commands, however an error is returned Unable to create channel of type 'zap'. I guess * is trying to open another non existent zap interface here... So, does anyone know if / how this is possible using asterisk, with just the one zap interface (x100p card)... ___ 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] echo
Good day all We are running x-lite as a sof client and using the zaptel cards Each time I make a call out I get a big echo but when I get a call in there is no echo?Why is this Please Help ___ 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] Language settings Cisco 7960
I use a Cisco 7960 with sccp. I know, that it is possible tho change the language in which informations is displayed on the screen. But I only found informations to do this with Cisco CM. How can I do this with asterisk? regards Jens ___ 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] Help with E1 configuration
zaptel.conf looks good - you may require a ,crc4 at the end of the span line, depending on your provider. I have loadzone=us in mine as well. Change appropriately for yours. zapata.conf also looks good. I would also add the following, before the channel declaration: immediate=no pridialplan=unknown usecallerid=yes Setting immediate=no allows your calls to be answered and routed according to the DID entries you make in your extensions.conf file (the dialplan). Setting it to yes would cause the s extension to be used instead. So you need entries in the dialplan for each DID, under the context [default] which you have defined in zapata. That will take care of inbound calls. Outbound calls: Since you've defined group 1 as including all channels of your PRI, you can use the Dial command and use a g1 instead of a specific Zap channel, to allow asterisk to choose an available channel. All of this is covered on the Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk Good luck with your project! Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of HengWee Chin Sent: Thursday, September 16, 2004 4:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help with E1 configuration Hi, I currently have a E100P card installed on my machine and the E1 subscription will be activated pretty soon. However, I have no idea how to configure asterisk to make inbound and outbound call using the E1. Especially for extensions.conf. Below is the configuration I used for zaptel.conf and zapata.conf. Is it possible if someone can verify if the configuration for zaptel and zapata is correct? zaptel.conf --- span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 zapata.conf --- switchtype=euroisdn signalling=pri_cpe group=1 context=default channel=1-15,17-31 I have 1 block of 10 DID numbers that will be subscribed together with E1. I am not able to find any sample for the extensions.conf to do inbound and outbound call. Is it possible for someone could post a sample of how the configuration would look like. Any setting missing for callerid support? PS: I already have an existing asterisk system running on analog ports. This is just an upgrade. Thanks in advanced. Regards, Chin _ Fast. Clear. Easy. The new MSN Search. http://search.msn.com.sg/ ___ 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] ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration failed
I'm trying to configure Chan_H323 to register with GnuGK... without success... i've failed finding sample configurations. I'd greatly appreciate anyone who can provide sample config of H323.conf and gnugk.ini I am tyring to configure Asterisk as a neighbor in GnuGK. I'm always getting this error on Asterisk. ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration failed. *** And a SecurityDenial error on GnuGK. This is my H323.conf [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g729 allow=G723.1 allow=ulaw ; Allow codecs in order of preference allow=alaw gatekeeper = 66.118.228.198 context=h323 [1005] ; When this line and the context [1004] lines are set type=h323 ; the caller id 1004 is always sent. I don't know why. e164=011005 ; In case, this lines are not set, the GS phones receives context=default ; "Error" as the caller id, and the H323 phone receives ; "asterisk" as the caller-id [1004] type=h323 e164=011004 context=default [asterisk] type=h323 prefix=01 context=h323 This is my entry in GnuGK [RasSrv::Neighbors] asterisk=68.90.233.134;1720;01; I've configured other GKs using this Neighbors section and it doesn't require password. Best regards, Carlos Maynard Jr. [EMAIL PROTECTED] ___ 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] ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration failed
Actually, * it's not a GK, you should configure it as regular Terminal (Not a Gateway)in your GNUGK. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Maynard Sent: Thursday, September 16, 2004 10:56 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration failed I'm trying to configure Chan_H323 to register with GnuGK... without success... i've failed finding sample configurations. I'd greatly appreciate anyone who can provide sample config of H323.conf and gnugk.ini I am tyring to configure Asterisk as a neighbor in GnuGK. I'm always getting this error on Asterisk. ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration failed. *** And a SecurityDenial error on GnuGK. This is my H323.conf [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g729 allow=G723.1 allow=ulaw ; Allow codecs in order of preference allow=alaw gatekeeper = 66.118.228.198 context=h323 [1005] ; When this line and the context [1004] lines are set type=h323 ; the caller id 1004 is always sent. I don't know why. e164=011005 ; In case, this lines are not set, the GS phones receives context=default ; Error as the caller id, and the H323 phone receives ; asterisk as the caller-id [1004] type=h323 e164=011004 context=default [asterisk] type=h323 prefix=01 context=h323 This is my entry in GnuGK [RasSrv::Neighbors] asterisk=68.90.233.134;1720;01; I've configured other GKs using this Neighbors section and it doesn't require password. Best regards, Carlos Maynard Jr. [EMAIL PROTECTED] ___ 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] ${CONTEXT} variable
Hi all,
Is there an equivalent of the ${CONTEXT} variable that represents the
*original* context of the call? i.e. If a call originates in the
'internal' context, no matter where it goes, this alternate version of
${CONTEXT} would never change from saying 'internal'?
I realize I could set this using the dialplan but I just wonder if there
this already exists, and if not, would there be any objection to adding
it? It could be ${CALL_CONTEXT} or ${ORIGINAL_CONTEXT}, or similar.
Thanks,
Chris
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Re: [Asterisk-Users] E3 PCI Cards
Arinze Izukanne wrote: Hi Guys, Does anyone know of E3 PCI cards that work with Asterisk? Arinze No, but if you find an E3 PCI card with nice Linux support there might be people interested in helping to get it working with *. Regards, Steve ___ 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] Uniden UIP-200 Multiple line appearances
Hi - I'm wondering if any has experience with the Uniden UIP-200 phones. The product info says that the 8 led buttons at the top are all programmable. Can they be programmed as separate line appearances (ala Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the phone capable of multiple SIP registrations? Also, the post about these phones at voip-info.org mentions some problems with DHCP and voice prompts getting cut off. Anyone know if these issues have been fixed? Thanks! Noah ___ 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] anyone can see response of a request from other connections?
hi friends i opened 2 connections with asterisk manager API with same user login and sent anOriginate request fromone of the 2connections. now i want to see the response of that command in another connection i hv opened.though i can see the response: originate successfully queued in the connection from which i hv sent the command i m not able to see the response from the other connection . i hv also got the full rights of Read and write in manager.conf means system,call,log,verbose,command,agent,user. any idea? ___ 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] E3 PCI Cards
Steve Underwood wrote: Arinze Izukanne wrote: Hi Guys, Does anyone know of E3 PCI cards that work with Asterisk? Arinze No, but if you find an E3 PCI card with nice Linux support there might be people interested in helping to get it working with *. Doesn't ImageStream have these (E3 and others) cards running in Linux (for their routers Linux-based ?). Still, someone mentioned horse-power AND the 'all eggs in a single E3' problem here... ___ 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] ?
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Re: [Asterisk-Users] No Caller Name sent from Asterisk over National or DMS100 PRI to a Norstar MICS?
There is a bugreport open about * when set as PRI_NET sending the CNAME
field in the DISPLAY IE instead of the FACILITY IE.
Look at bugs.digium.com, I don't rmember the bugreport number.
-Alfred.
Kris Boutilier wrote:
I have a PRI link up and running between Asterisk and a Nortel Norstar MICS
v4.1 . I'm having a problem getting the textual Caller Name across the link
from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns
to Ast both elements come through fine. I'm forcing dummy values for testing
using:
exten = s,1,SetCIDName(Test)
exten = s,2,SetCallerID(1234561234)
exten = s,3,Dial(zap/g1/${ARG1},15)
I've tried switchtype=national and dms100 (adjusting accordingly on Ns) with
no change. zapata.conf is currently:
; Norstar #2 (Wharf Road)
context=in-t1nstar
group=1
usecallerid=yes
hidecallerid=no
usecallingpres=no
switchtype=dms100
pridialplan=local
signalling=pri_net
channel = 1-23
The SETUP frame from Ast contains:
Protocol Discriminator: Q.931 (8) len=40
Call Ref: len= 2 (reference 2/0x2) (Originator)
Message type: SETUP (5)
[04 03 80 90 a2]
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: u-Law (34)
[18 03 a1 83 81]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ]
[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) ]
[6c 0c 21 80 31 32 33 34 35 36 31 32 33 34]
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) '1234561234' ]
[70 05 c1 36 31 30 31]
Called Number (len= 7) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '6101' ]
Which doesn't seem to even contain the CIDName... On the other hand, the
SETUP frame from the Ns contains:
Protocol Discriminator: Q.931 (8) len=56
Call Ref: len= 2 (reference 1/0x1) (Originator)
Message type: SETUP (5)
[04 03 80 90 a2]
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: u-Law (34)
[18 03 a1 83 97]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 23 ]
[28 0b b1 53 43 52 44 20 4b 72 69 73 42]
Display (len=11) Charset: 31 [ SCRD KrisB ]
[6c 0c 21 80 36 30 34 38 38 35 36 38 30 38]
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) '6048856808' ]
[70 0c 80 39 36 30 34 38 38 35 36 38 30 38]
Called Number (len=14) [ Ext: 1 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0) '96048856808' ]
Which has the textual ID in the 'Display' element... However I understand
from http://resource.intel.com/telecom/support/tnotes/tnbyos/2000/tn033.htm
that there is no definitive standard for transmitting the name.
So, should even I be expecting Ast to put the name on the wire when it's
originating?
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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Re: [Asterisk-Users] E3 PCI Cards
On Thu, 2004-09-16 at 09:22, Julio Arruda wrote: Steve Underwood wrote: Arinze Izukanne wrote: Hi Guys, Does anyone know of E3 PCI cards that work with Asterisk? Arinze No, but if you find an E3 PCI card with nice Linux support there might be people interested in helping to get it working with *. Doesn't ImageStream have these (E3 and others) cards running in Linux (for their routers Linux-based ?). Still, someone mentioned horse-power AND the 'all eggs in a single E3' problem here... If you look back at the archives, you will probably find discussion about that card has been here before. There isn't appropriate drivers in linux for telephony. You did mean a telephony interface since you are in asterisk, right? -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Clarification - FAX on local network
Adam Goryachev wrote: On Wed, 2004-09-15 at 04:29, Lee Howard wrote: On 2004.09.14 11:10 Marty Mastera wrote: 2)Packet loss, etc...makes faxing over the internet unreliable I'm not sold on this theory yet. I don't think that it's so much a matter of packet loss (this shouldn't occur regularly), but rather of latency. Transmitting packets over a network, and in particular the internet, can result in latency delays that could, in theory, pose a problem for FoIP, but I've heard of so many people successfully doing FoIP with equipment other than Asterisk (i.e. using Cisco VoIP equipment), that I tend to believe that the reliability factor is more a consequence of SIP or the equipment used (Asterisk and, in my case, a Sipura SP-2000). Actually, I thought it was more related to jitter than latency. Consider that faxing over international PSTN worked reliably back in the bad old days when international calls were sent over satellite (Well, Australia - US anyway). Just my 0.02c... Regards, Adam Latency isn't really an issue. Packet loss can be. Jitter can be. So can other timing issues. Look here http://www.opencall.org/faq/x29.html Regards, Steve ___ 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] E3 PCI Cards
Another promising candidate is Apple's dual G5 (PPC970) Xserve (a 1U server). http://www.apple.com/xserve this one looks as if it might beat the price/performance ratio of a high end Intel server. The Apple G5 Xserv system has a PCI-X interface. Does anyone know what that is and will a T405P or T410P card work? Both systems run LinuxPPC. Does anyone have * running on PPC? ___ 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] Asterisk and SER
Hi All, Can someone help me clear up some stuff? I am about to implement asterisk for a office of about 20 people. I plan on running SIP phones for everyone. (a mix of Cisco Sets and Xlite soft phones) We will place the Asterisk server at a collocation provider and have in connected to the PSTN via 2 PRIs. (digium card) When customers call our 800 number they will be sent to asterisk. When they enter an extension I want asterisk to check if that SIP users is logged in and if not transfer the call back out over PSTN (to a cell phone) Now, here is where things are a little foggy... I want put a local Asterisk server here in the office so that the SIP users connect to it thereby reducing the chatter across the WAN. I would like to have the two Asterisk servers communicate via IAX. Questions: 1. Does this scenario pass muster? Is my thinking logical or does anyone have a better suggestion? 2. Is this possible? Can the remote Asterisk server check to see if the SIP user is logged in to the local Asterisk server before sending the call across the WAN? 3. Should I be using SER vs. another Asterisk server? The problem I see with this is that it doesn't support IAX. I believe that is the preferred method? Am I right? Thanks for all the help from the OSS community. Great software!!! ~chris Christopher Jacob Eye Street Software Program Manager,14151 Newbrook Drive Partner Products Suite 250 301.305.0991Chantilly, VA 20151 www.eyestreet.com ___ 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] E3 PCI Cards
Another promising candidate is Apple's dual G5 (PPC970) Xserve (a 1U server). http://www.apple.com/xserve this one looks as if it might beat the price/performance ratio of a high end Intel server. The Apple G5 Xserv system has a PCI-X interface. Does anyone know what that is and will a T405P or T410P card work? Both systems run LinuxPPC. Does anyone have * running on PPC? Yeah, check out: http://www.voip-info.org/tiki-index.php?page=Asterisk%20MacOSX%20Support Specifically for OS X. There's a download link. The problem still is that no one has written ppc drivers for the Digium cards. As I understand, the only drivers are for GNU/Linux on i386. You wanna write some for the good of the BSD and PPC communities? ;-) ___ 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] spandsp on current cvs?
Steve or anyone... Will spandsp install on the current cvs? Looked like the code at ftp.opencall.org/pub/spandsp was intended to be applied to the old stable release. Anyone know? Rich ___ 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] Broadvoice BYOD Plans - 3-way and Call Waiting
The other issue is that call waiting does not appear to work. The way I'm expecting it to work with Asterisk is to send the second call to me - I'm using SetGroup and CheckGroup within Asterisk to limit my calls to two at a time total. However, if I'm on a phone call (incoming or outgoing), Broadvoice transfers a second call to a person you are calling is busy message -- I don't see any additional SIP traffic to the Asterisk box. You must have call waiting turned off on your comm pilot control panel, go to www.broadvoice.com and log into your control panel and make sure call waiting is turned on. -Chris ___ 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] Re: No Caller Name sent from Asterisk over National or DMS100?
- Original Message - Message: 3 Date: Thu, 16 Sep 2004 07:57:15 -0400 (EDT) From: David Troy [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] No Caller Name sent from Asterisk over National or DMS100 PRI to a Norstar MICS? snip I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: snip everyone remember that we are talking about a private connection here. if i read the original post here correctly the issue is between the * and the Norstar not out to the PSTN. i have been tying NEC's together for 15+ years with a proprietary ISDN protocol that sends station name across the d-channel without any reverse lookup DB. Now that being said I am no expert on d-channel messaging so I can't really answer the question on how/if we can pass the CALLERIDNAME across a private d-channel connection between * and another PBX. Jason Kawakami www.optellabs.com ___ 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] Intertex IX66
Lolll, That's a good one :)) U make my day :) Best regards, Chris HARIGA P.S.: I send my ethereal log to Intertex.se and I hope to fix the problem asap. I will post on the list the solution. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Williams Sent: Thursday, September 16, 2004 4:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Intertex IX66 On Thu, 9 Sep 2004 12:26:56 -0400, Chris HARIGA [EMAIL PROTECTED] wrote: Hi, I have Asterisk w/ 192.168.1.1 and I setup IX66 to be 192.168.1.2 (I'm using pppoe client and dyndns.org on IX66) I setup on Local DNS Server my * box and after that I was able to register my phones from the Internet. I cannot understand my problem with one way sound... what is wrong on my configuration :(( As the IX66 is a sip aware router make sure you have no entries for nat in your sip.conf, and let the ix66 deal with the nat, not * . I hope this helps. Jason ___ 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 smime.p7s Description: S/MIME cryptographic signature ___ 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] how to get caller ID
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. ___ 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] how to get caller ID
vrushank wrote: 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. Have you monitored the console while the line is ringing to verify that asterisk is not seeing the callerid and not paying attention to it? PS: I'm testing a new email client, please forgive me if this message is not in Plain Text. (And someone please let me know!) -- Andrew Thompson http://aktzero.com/ ___ 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] E3 PCI Cards
On Thu, 16 Sep 2004 11:03:48 -0400, Noah Miller [EMAIL PROTECTED] wrote: Does anyone have * running on PPC? http://www.voip-info.org/tiki-index.php?page=Asterisk%20MacOSX%20Support Specifically for OS X. There's a download link. The problem still is that no one has written ppc drivers for the Digium cards. As I understand, the only drivers are for GNU/Linux on i386. That's not entirely correct. The Zaptel drivers work on LinuxPPC. Further, there is some work in progress on Zaptel drivers for BSD and some folks use X100P and TDM400 on FreeBSD already. Since OSX is BSD based, it will eventually benefit from the work done to bring Zaptel to BSD. We have made an Xserve available for Rich Murphey, one of the main contributors to the Asterisk on BSD effort, specifically for him to test things on OSX. What's needed is more contributors to the BSD effort, or so it would seem. Since driver development requires skills that are less common than those required for many other development tasks, there are fewer people who can do it. It also takes more time to move drivers from one platform to another. I think a sponsorship fund could do some good because it might give somebody the ability to work fulltime on drivers for BSD in general and OSX in particular. I believe that it should be possible to raise significant sponsorship funds for drivers (especially for OSX) from end user donations alone. In order to do that, a few people need to come together, think about how to organise this, set up a website, open a kagi and/or paypal account and get the word out. I am discussing this idea at present with some Mac folks who seem to be willing to put a bit of time and effort into this. Anybody who would like to join in on this, please contact me directly. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] Beyond T1
All, This may be a stupid question, but here it is... What interface gives the most density? Do I top out at T1's? For instance, 4 t1's to the Digium Quad span t1 card. Is there an interface available for T3 or DS3? Thanks, Chris ___ 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] call parking forwarding
Hi everbody, I have problem with configuring call parking and forwarding. firstly my setup: I have one asterisk with gnu-gatekeeper on the same PC. As phones I use voip-phones with H323 support. phones are registered on gatekeeper as terminal and asterisk as gateway. I setup features.conf (parking.conf) like: [general] parkext = 700 parkpos = 701-720 context = parkedcalls parkingtime = 45 and include parkedcalls context to extensions.conf but without any success for example: somebody call me from PSTN, and I pick up call on my h323 phone in room #1 Now I want to go to another room (room#2), so I dial #700 (in hope to transfer call to parking queue) At this time I hear tone (one tone for any one keystroke -- I think tones are simulated by phone - not by asterisk) but nothing to happen. Also no records in asterisk logs. Have anybody idea what may be wrong? Another situation: call forward. I have no idea how to do it. There's no any reference in any documentation!? I mean: Somebody call me from PSTN and I pick up this call by my h323 phone. Now I want forward this call to my colleague to another h323 phone. ANY IDEA HOW TO DO IT? Thanks for any help. pgpA1F76idivo.pgp Description: PGP signature ___ 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] What can you do with Asterisk in Brazil following the law
Has anybody any idea what I can do with asterisk following the Brazilian law. I do not have a multimedia license or a telecom license, but I ace asterisk. Are there companies who are looking for asterisk expertise in Rio de Janeiro? Greeting Han ___ 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] ID for outgoing calls from DDI (DID) line
Hi again, in my * I have one ISDN BRI line with DID (DDI) preselection. so in fact I have 100 extensions: +421 33 12 34 56 xx Q: Is in my power -- or in power of * -- to influence which of these extensions will occur in my cellular display? THANKS. pgpfkP2ywWN91.pgp Description: PGP signature ___ 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] Beyond T1
Christopher Jacob wrote: All, This may be a stupid question, but here it is... What interface gives the most density? Do I top out at T1's? For instance, 4 t1's to the Digium Quad span t1 card. Is there an interface available for T3 or DS3? Depending on where you using the circuits, you might try an E1. It uses the same total bandwidth as a T1(I think), but splits the channels at 56K instead of 64K, yielding more channels. (And now I can't remember the number.) I haven't heard of direct DS3 connectivity... Just stretching my imagination a little bit, you might be able to plug a DS3 into a H323 box, and then feed the IP-side of the calls to asterisk -- Andrew Thompson http://aktzero.com/ ___ 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] Beyond T1
Andrew Thompson wrote: Christopher Jacob wrote: All, This may be a stupid question, but here it is... What interface gives the most density? Do I top out at T1's? For instance, 4 t1's to the Digium Quad span t1 card. Is there an interface available for T3 or DS3? Depending on where you using the circuits, you might try an E1. It uses the same total bandwidth as a T1(I think), but splits the channels at 56K instead of 64K, yielding more channels. (And now I can't remember the number.) I haven't heard of direct DS3 connectivity... Just stretching my imagination a little bit, you might be able to plug a DS3 into a H323 box, and then feed the IP-side of the calls to asterisk Actually T1 is 24x64k and E1 is 30x64k - 1.536 megabits/sec -vs- 2.0 if I recall 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
Re: [Asterisk-Users] Beyond T1
You can't run E1 on a circuit designed for T1. T1 is 24 x 64k = 1.5mb; E1 is 30 x 64k = 2mb Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 16 Sep 2004, Andrew Thompson wrote: Christopher Jacob wrote: All, This may be a stupid question, but here it is... What interface gives the most density? Do I top out at T1's? For instance, 4 t1's to the Digium Quad span t1 card. Is there an interface available for T3 or DS3? Depending on where you using the circuits, you might try an E1. It uses the same total bandwidth as a T1(I think), but splits the channels at 56K instead of 64K, yielding more channels. (And now I can't remember the number.) I haven't heard of direct DS3 connectivity... Just stretching my imagination a little bit, you might be able to plug a DS3 into a H323 box, and then feed the IP-side of the calls to asterisk -- Andrew Thompson http://aktzero.com/ ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-09-16%5C8a09dc96117f472aab522092083ad700C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Beyond T1
A T1 is 24 64000bps channels. The 56000bps thing is when robbed bit signalling is used, it steals bits from each voice channel for call signalling, while on the E1 one channel is used for that. When PRI signalling is used each voice channel is the full 64000 bps thing. Marcelo Pacheco Em Qui 16 Set 2004 13:17, Andrew Thompson escreveu: Christopher Jacob wrote: All, This may be a stupid question, but here it is... What interface gives the most density? Do I top out at T1's? For instance, 4 t1's to the Digium Quad span t1 card. Is there an interface available for T3 or DS3? Depending on where you using the circuits, you might try an E1. It uses the same total bandwidth as a T1(I think), but splits the channels at 56K instead of 64K, yielding more channels. (And now I can't remember the number.) I haven't heard of direct DS3 connectivity... Just stretching my imagination a little bit, you might be able to plug a DS3 into a H323 box, and then feed the IP-side of the calls to asterisk ___ 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] Beyond T1
On Thursday 16 September 2004 12:17, Andrew Thompson wrote: Depending on where you using the circuits, you might try an E1. It uses the same total bandwidth as a T1(I think), but splits the channels at 56K instead of 64K, yielding more channels. (And now I can't remember the number.) uh, no. This is definitely NOT correct. T1 is 24 8-bit channels + 1 framing bit sent 8000 times a second. 24*8+1 = 193 bits per frame * 8000 = 1554000bps. E1 is 32 8-bit channels + 1 framing bit sent 8000 times a second. 32*8+1 = 2056000bps. (my E1 knowlege is poor, I hope I am not furthering the misinformation here) In both cases you get 64kbit clean channels unless you're doing robbed-bit (inband) signalling. -A. ___ 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] ${CONTEXT} variable
Christopher L. Wade wrote:
Hi all,
Is there an equivalent of the ${CONTEXT} variable that represents the
*original* context of the call? i.e. If a call originates in the
'internal' context, no matter where it goes, this alternate version of
${CONTEXT} would never change from saying 'internal'?
I realize I could set this using the dialplan but I just wonder if there
this already exists, and if not, would there be any objection to adding
it? It could be ${CALL_CONTEXT} or ${ORIGINAL_CONTEXT}, or similar.
Thinking about this, the name of the variable might be ${DEVICE_CONTEXT}
instead. This seems more in keeping with what I was intending the
variable to represent, which is the 'context=' line from the appropriate
config file.
Thanks,
Chris
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Re: [Asterisk-Users] Beyond T1
Even with the robbed bit thing you get 62666.7 bits/s, since it only steals the LSB every 6 samples. :-) Regards, Steve Marcelo Pacheco wrote: A T1 is 24 64000bps channels. The 56000bps thing is when robbed bit signalling is used, it steals bits from each voice channel for call signalling, while on the E1 one channel is used for that. When PRI signalling is used each voice channel is the full 64000 bps thing. Marcelo Pacheco Em Qui 16 Set 2004 13:17, Andrew Thompson escreveu: Christopher Jacob wrote: All, This may be a stupid question, but here it is... What interface gives the most density? Do I top out at T1's? For instance, 4 t1's to the Digium Quad span t1 card. Is there an interface available for T3 or DS3? Depending on where you using the circuits, you might try an E1. It uses the same total bandwidth as a T1(I think), but splits the channels at 56K instead of 64K, yielding more channels. (And now I can't remember the number.) I haven't heard of direct DS3 connectivity... Just stretching my imagination a little bit, you might be able to plug a DS3 into a H323 box, and then feed the IP-side of the calls to asterisk ___ 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] Transfer and Release of a call out to PSTN
Hi Again All, When using Asterisk with a PRI to the CO is it possible to transfer a call back out and release. In other words, once the call is connected (caller and external 3rd party) Asterisk is removed from the equation thereby freeing the PRI channels. I ask because my scenario is going to require frequent external transfers and I would like to control the PRI costs. Could this be done using SS7? If so, does anyone know if any Asterisk SS7 development is being done? Thanks Chris ___ 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] IAX- FAX
Has anyone had any success using iax for inbound fax into asterisk. I tried this but can seem to get asterisk to listen for fax, is it zap specific ? d ___ 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] Beyond T1
I'm no E1 expert, but as I understand one channel is wasted with framing, so it is as 2048000 bps link, where one 64000 bps channel is wasted with signalling. So there's 31 channels left. If you use EM, FXS or FXO, you could get 31 voice channels, with PRI or MFC/R2D you get 30 voice channels. I now that from the fact that a full E1 with EuroISDN gives you 30 voice channels. An a full E1 with Brazilian R2D also gives you 30 voice channels, as one channel is used for signalling as CAS (Channel Associated Channeling), where each 4 bits is used for each channel. The only situation where you get closer to actual 2mbps out of an E1 channel is when you run SyncPPP, Frame Relay or another bit synchronous protocol on the full trunk/link, where you throw away the channelling and use the whole link as one big synchronous bit pipe. Marcelo Pacheco Em Qui 16 Set 2004 13:26, Andrew Kohlsmith escreveu: On Thursday 16 September 2004 12:17, Andrew Thompson wrote: Depending on where you using the circuits, you might try an E1. It uses the same total bandwidth as a T1(I think), but splits the channels at 56K instead of 64K, yielding more channels. (And now I can't remember the number.) uh, no. This is definitely NOT correct. T1 is 24 8-bit channels + 1 framing bit sent 8000 times a second. 24*8+1 = 193 bits per frame * 8000 = 1554000bps. E1 is 32 8-bit channels + 1 framing bit sent 8000 times a second. 32*8+1 = 2056000bps. (my E1 knowlege is poor, I hope I am not furthering the misinformation here) In both cases you get 64kbit clean channels unless you're doing robbed-bit (inband) signalling. -A. ___ 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] IAX- FAX
D, I have a IAX2 gateway that connects to our remote asterisk gateway that has a PRI. Inbound seems to work without a hitch. Make sure your iax.conf allows ULAW as well, Since fax cannot be compressed. Outbound is a different story. My fax seems to ring thru, but it never seems to establish A carrier. Have you been able to get outbound working? Paul Seniuk -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: September 16, 2004 10:40 AM To: asterisk-users Subject: [Asterisk-Users] IAX- FAX Has anyone had any success using iax for inbound fax into asterisk. I tried this but can seem to get asterisk to listen for fax, is it zap specific ? d ___ 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
Re: [Asterisk-Users] E3 PCI Cards
Steve Underwood wrote: Arinze Izukanne wrote: Hi Guys, Does anyone know of E3 PCI cards that work with Asterisk? Arinze No, but if you find an E3 PCI card with nice Linux support there might be people interested in helping to get it working with *. SBE (side band engineering). -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Unable to dial using SIP using FWD and iConnectHere
Hi.
I cant make SIP calls from asterisk.
When I start asterisk, I get the following message: What does it means??
Asterisk is not behind NAT or Firewall.
--
[chan_sip.so] = (Session Initiation Protocol (SIP))
== Parsing '/etc/asterisk/sip.conf': Found
Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to
get IP address for dhcp--490, SIP disabled
== Registered channel type 'SIP' (Session Initiation Protocol (SIP))
== Registered application 'SIPDtmfMode'
== Parsing '/etc/asterisk/enum.conf': Found
== Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted
Asterisk Ready.
*CLI
I have configured some sample sections in my extensions.conf file to
test asterisk features:
---
[demo]
exten = 1,1,SetCallerID(My Self-Testing)
exten =
1,2,Dial(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${FWDNUMBER})
; call my own FWD number using iax subscription at FWD. I'm receiving
this call in X-Lite soft phone configured at another PC (not at the
asterisk box)
; This woks great! I can dial [EMAIL PROTECTED] from console and get an incoming
call at the X-Lite.
exten = 2,1,Dial(SIP/[EMAIL PROTECTED],r)
; Test for outoging calls using IConnectHere account
exten = 3,1,Dial(SIP/${FWDNUMBER}:[EMAIL PROTECTED]:5060,r)
; Test for outgoing calls using FWD account via SIP
--
These are my peers in sip.conf:
---
[outgoing_sip_iconnect]
; for routing calls outbound to the PSTN numbers via iconnecthere
;
type=peer
username=!!!My Iconnect Number!!!
secret=!!!MyPassword!!!
host=sipauth.deltathree.com
canreinvite=no
qualify=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=G726
[outgoing_sip_fwd]
type=peer
username=!!!My FWD Number
secret=!!!MyPassword
host=fwd.pulver.com
disallow=all
allow=ulaw
allow=G726
-
And this is what I get when I type dial [EMAIL PROTECTED] or [EMAIL PROTECTED]... messages
are the same except for server IP address, which are indeed the right
ones (I've cheked that out making pings).
*CLI dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp, SIP/[EMAIL PROTECTED]) in new stack
*CLI Sep 16 11:26:23 WARNING[278544]: chan_sip.c:590 __sip_xmit:
sip_xmit of 0x8109224 (len 726) to 65.39.205.114 returned -1: Bad file
descriptor
-- Called [EMAIL PROTECTED]
Sep 16 11:26:24 WARNING[245775]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x8109224 (len 726) to 65.39.205.114 returned -1: Bad file descriptor
Sep 16 11:26:25 WARNING[245775]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x8109224 (len 726) to 65.39.205.114 returned -1: Bad file descriptor
Sep 16 11:26:26 WARNING[245775]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x8109224 (len 726) to 65.39.205.114 returned -1: Bad file descriptor
Sep 16 11:26:27 WARNING[245775]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x8109224 (len 726) to 65.39.205.114 returned -1: Bad file descriptor
Sep 16 11:26:28 WARNING[245775]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x8109224 (len 726) to 65.39.205.114 returned -1: Bad file descriptor
Sep 16 11:26:29 WARNING[245775]: chan_sip.c:673 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Critical Request)
Sep 16 11:26:29 WARNING[278544]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x8106c7c (len 364) to 65.39.205.114 returned -1: Bad file descriptor
== No one is available to answer at this time
-- Executing Wait(OSS/dsp, 1) in new stack
Sep 16 11:26:30 WARNING[245775]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x8106c7c (len 364) to 65.39.205.114 returned -1: Bad file descriptor
Sep 16 11:26:31 WARNING[245775]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x8106c7c (len 364) to 65.39.205.114 returned -1: Bad file descriptor
Sep 16 11:26:32 WARNING[245775]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x8106c7c (len 364) to 65.39.205.114 returned -1: Bad file descriptor
Sep 16 11:26:33 WARNING[245775]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x8106c7c (len 364) to 65.39.205.114 returned -1: Bad file descriptor
Sep 16 11:26:34 WARNING[245775]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x8106c7c (len 364) to 65.39.205.114 returned -1: Bad file descriptor
Sep 16 11:26:35 WARNING[245775]: chan_sip.c:673 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Non-critical Request)
-- Timeout on OSS/dsp
== CDR updated on OSS/dsp
-- Executing Goto(OSS/dsp, #|1) in new stack
-- Goto (demo,#,1)
-- Executing Playback(OSS/dsp, demo-thanks) in new stack
Console call has been answered
-- Playing 'demo-thanks' (language 'en')
-- Executing Hangup(OSS/dsp, ) in new stack
== Spawn extension (demo, #, 2) exited non-zero on 'OSS/dsp'
Hangup on console
*CLI
---
---
avast! Antivirus: Outbound message clean.
Virus Database (VPS): 0438-2, 16/09/2004
Tested on: 16/09/2004 19:18:45
avast!
Re: [Asterisk-Users] E3 PCI Cards
Bob Knight wrote: Steve Underwood wrote: Arinze Izukanne wrote: Hi Guys, Does anyone know of E3 PCI cards that work with Asterisk? Arinze No, but if you find an E3 PCI card with nice Linux support there might be people interested in helping to get it working with *. SBE (side band engineering). I don't know if any of their cards are really suitable for telephony, but they don't appear to do any E3 cards in PCI form. They have E3 mezzanine cards for cPCI. Regards, Steve ___ 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] Beyond T1
Well, you might be better off at that scale to use a cisco as5850 or equiv with SER and Asterisk. I might not work so well with 672 calls going thru 1 asterisk box. ds3 - Cisco as5850 - Asterisk (Possible multiple depending on actual config and use) - Original Message - From: Marcelo Pacheco [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, September 16, 2004 12:52 PM Subject: Re: [Asterisk-Users] Beyond T1 I'm no E1 expert, but as I understand one channel is wasted with framing, so it is as 2048000 bps link, where one 64000 bps channel is wasted with signalling. So there's 31 channels left. If you use EM, FXS or FXO, you could get 31 voice channels, with PRI or MFC/R2D you get 30 voice channels. I now that from the fact that a full E1 with EuroISDN gives you 30 voice channels. An a full E1 with Brazilian R2D also gives you 30 voice channels, as one channel is used for signalling as CAS (Channel Associated Channeling), where each 4 bits is used for each channel. The only situation where you get closer to actual 2mbps out of an E1 channel is when you run SyncPPP, Frame Relay or another bit synchronous protocol on the full trunk/link, where you throw away the channelling and use the whole link as one big synchronous bit pipe. Marcelo Pacheco Em Qui 16 Set 2004 13:26, Andrew Kohlsmith escreveu: On Thursday 16 September 2004 12:17, Andrew Thompson wrote: Depending on where you using the circuits, you might try an E1. It uses the same total bandwidth as a T1(I think), but splits the channels at 56K instead of 64K, yielding more channels. (And now I can't remember the number.) uh, no. This is definitely NOT correct. T1 is 24 8-bit channels + 1 framing bit sent 8000 times a second. 24*8+1 = 193 bits per frame * 8000 = 1554000bps. E1 is 32 8-bit channels + 1 framing bit sent 8000 times a second. 32*8+1 = 2056000bps. (my E1 knowlege is poor, I hope I am not furthering the misinformation here) In both cases you get 64kbit clean channels unless you're doing robbed-bit (inband) signalling. -A. ___ 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
Re: [Asterisk-Users] Unable to dial using SIP using FWD and iConnectHere
Rodolfo Grave wrote: Hi. I cant make SIP calls from asterisk. When I start asterisk, I get the following message: What does it means?? Asterisk is not behind NAT or Firewall. -- [chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to get IP address for dhcp--490, SIP disabled What is dhcp--490? Is that the name of your linux machine? Do your linux box get its IP via DHCP from your provider? Do a reverse lookup on your linux boxes IP and see if it comes up as dhcp--490.yourcarrier.tld or something like that. If so, try pinging dhcp--490 and also the reverse lookup address. You may have to add the yourcarrier.tld to your lookup file(can't remember the name right now) so that your dns lookups automatically attempt to search for dhcp--490.yourcarrier.tld before they fail out as unknown. -- Andrew Thompson http://aktzero.com/ ___ 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] Unable to dial using SIP using FWD and iConnectHere
If it's what Andrew is talking about, then add the hostname to /etc/hosts. On Thursday 16 September 2004 05:27 pm, Andrew Thompson wrote: Rodolfo Grave wrote: Hi. I cant make SIP calls from asterisk. When I start asterisk, I get the following message: What does it means?? Asterisk is not behind NAT or Firewall. -- [chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to get IP address for dhcp--490, SIP disabled What is dhcp--490? Is that the name of your linux machine? Do your linux box get its IP via DHCP from your provider? Do a reverse lookup on your linux boxes IP and see if it comes up as dhcp--490.yourcarrier.tld or something like that. If so, try pinging dhcp--490 and also the reverse lookup address. You may have to add the yourcarrier.tld to your lookup file(can't remember the name right now) so that your dns lookups automatically attempt to search for dhcp--490.yourcarrier.tld before they fail out as unknown. -- -- Heritage Communications Corporation Melbourne, FL USA 32935 http://www.hcc.net ___ 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] IAX2 only asterisk scalability
Would anybody have any numbers on how large a box would be required to convert 100 or 200 SIP calls to IAX2, without transcoding, echo cancel, .. Or a setup with individual IAX2 calls coming on one side, and trunking being used to 1 or more remote boxes on the other side, to improve bandwidth usage ? It doesn't matter if you don't have a test done for exactly 100 or 200 calls, I'm just looking for with configuration 'A', I was able to switch 'x' concurrent calls before having quality problems, or system load going thru the roof. I'm seriously thinking about developing a trunking VPN utility that would alow me to add trunking outside asterisk's code, so I can keep jitter buffer. I'm much better coding in 'C' from ground up then changing existing code. It would know IAX2 packet format and take packets between the local host and each remote one and bufffer them for say 50ms (configurable) adding all subsequent packets to the first one, flushing that macro packet, then decoding on the other side, much like a VPN tunneling protocol. I already have my own VPN that does almost exactly that, except I'd like it to know much more about IAX2 packets, in order to compress that better. Regards, Marcelo Pacheco ___ 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] Transfer and Release of a call out to PSTN
On Thu, 16 Sep 2004 12:41:47 -0400, Christopher Jacob [EMAIL PROTECTED] wrote: When using Asterisk with a PRI to the CO is it possible to transfer a call back out and release. In other words, once the call is connected (caller and external 3rd party) Asterisk is removed from the equation thereby freeing the PRI channels. I ask because my scenario is going to require frequent external transfers and I would like to control the PRI costs. You're looking for a feature called Take Back and Transfer, TBT for short. It works by the telco always monitoring the trunks for DTMF from your end, for example, the TBT code might be *8. You would send *8,12125551212 down the line and the telco will pull the call back and send it to the number specified, in this example direct service for Manhattan. The last time I looked into it it was a specialized service and had a higher per minute rate than conventional termination. sl ___ 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] ID for outgoing calls from DDI (DID) line
On Thu, 16 Sep 2004, Maros RAJNOCH wrote: in my * I have one ISDN BRI line with DID (DDI) preselection. so in fact I have 100 extensions: +421 33 12 34 56 xx Q: Is in my power -- or in power of * -- to influence which of these extensions will occur in my cellular display? I guess you mean you cant to control which of your assigned ddi extensions show up as callerId to the remote party when calling our from your asterisk pbx? That is possible in principle, provided your pstn provider lets you. You also have to agree the the format (TON and numbering plan) possibly the number of digits to send. Then asterisk will pass whatever is set with the SetCallerId() application in the dialplan or set in the sip.conf etc files for each internal extension. As an example, we send three digits which just happens to match our internal extensions so we do not have to fiddle with SetCallerId. In zaptel.conf we have prilocaldialplan=unknwon since that is what our pstn provider wants. Talking to your pstn provider may save you a lot of trial and error. Peter ___ 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] Static noise and server locked when using two 4FXO tdm400p pci cards
Hello all We have tested for a mounth or two an asterisk PBX using one T1 channel bank with 24 fxs and one TDM400P digium card with 4 FXO modules. This worked with minor problems, the most notorious being some sporadic static noice or failure in the first FXO module on the wildcard. Now we have a client with 12 pstn lines and 48 extensions and we are trying to deploy an Asterisk PBX server using two(x24)channel banks (Access Bank 1) an three TDM400P pci cards with 4 FXO each. But when we use more that one TDM400P card, after some random number of calls, one of the cards starts to give a loud static noise when calling from inside in all their channels and if we keep trying to use the lines the server gets frozen. Restarting Asterisk don't solves the problem and the only way of recovering the channels is to reload the zaptel modules (if the system is not locked yet). We have seen some similar problems reports in the list, and some people telling they asked to digium support, but not a real solution. Does anybody knows if is this a major hardware problem with Digium TDM cards and zaptel driver or if there is some way of fixing this? Thanks very much Luis ___ 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] rxfax application
attempting to get asterisk pbx to receive inbound faxes
have defined the necessary extension as per technote;
default]
; Answer the line and listen
exten = s,1,Answer
; Dial an extension, let asterisk give a ringtone
exten = s,2,Dial(IAX2/3987,40,r)
; Hangup if nobody picked up within 40 seconds
exten = s,3,Hangup
; Did we get a fax?
exten = fax,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = fax,2,rxfax(${FAXFILE})
and can see the SetVar command being executed but then the console echos;
pbx_extension_helper: No application 'rxfax' for extension (pstn-incoming,fax,2)
the asterisk installation is very much the 'basic' install as per the installation
guide so i would guess this does not have this 'rxfax' application
if this is correct can anyone assist with how i get the rxfax to run ??
GT
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Re: [Asterisk-Users] Unable to dial using SIP using FWD and iConnectHere
Hi and thanks. I added the entry in the /etc/hosts file and it is working now... I also had to add more parameters at the peers definition: authname, username Now.. The problem with this solution is that my hostname and my ip changes everytime I reset my box (at least)... how can I solve this? Can't I just say asterisk to use the eth0 IP?? Thanks a lot. RODOLFO Brian Wilkins wrote: If it's what Andrew is talking about, then add the hostname to /etc/hosts. On Thursday 16 September 2004 05:27 pm, Andrew Thompson wrote: Rodolfo Grave wrote: Hi. I cant make SIP calls from asterisk. When I start asterisk, I get the following message: What does it means?? Asterisk is not behind NAT or Firewall. -- [chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to get IP address for dhcp--490, SIP disabled What is dhcp--490? Is that the name of your linux machine? Do your linux box get its IP via DHCP from your provider? Do a reverse lookup on your linux boxes IP and see if it comes up as dhcp--490.yourcarrier.tld or something like that. If so, try pinging dhcp--490 and also the reverse lookup address. You may have to add the yourcarrier.tld to your lookup file(can't remember the name right now) so that your dns lookups automatically attempt to search for dhcp--490.yourcarrier.tld before they fail out as unknown. --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 0438-2, 16/09/2004 Tested on: 16/09/2004 20:24:42 avast! - copyright (c) 2000-2004 ALWIL Software. http://www.avast.com ___ 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] ?
vrushank wrote: ! p.s. maybe set your time/date correct ___ 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] Unable to dial using SIP using FWD and iConnectHere
Rodolfo Grave wrote: Hi and thanks. I added the entry in the /etc/hosts file and it is working now... I also had to add more parameters at the peers definition: authname, username Now.. The problem with this solution is that my hostname and my ip changes everytime I reset my box (at least)... how can I solve this? Can't I just say asterisk to use the eth0 IP?? snip Brian Wilkins wrote: If it's what Andrew is talking about, then add the hostname to /etc/hosts. I did a quick search and found this link: http://www.faqs.org/docs/securing/chap9sec91.html /etc/resolv.conf is the file I believe you should edit. I believe that you want to add your providers domain.tld to this file so that your dns lookups automatically looks up your_dhcp_name.providerdomain.tld when you run a query for your_dhcp_name. Even if your_dhcp_name changes, having the domain suffix in /etc/resolv.conf should allow the query to complete successfully. Hope this helps. -- Andrew Thompson http://aktzero.com/ ___ 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] Transfer and Release of a call out to PSTN
You're looking for a feature called Take Back and Transfer, TBT for short. I thought it was Two B-Channel Transfer? It works by the telco always monitoring the trunks for DTMF from your end, for example, the TBT code might be *8. You would send *8,12125551212 down the line and the telco will pull the call back and send it to the number specified, I guess there's more than one way to skin a cat - All the implementations I've read use PRI signalling to make another outbound call, then sends a command message with the CRNs of both calls and they are then bridged in the CO switch, leaving both PRI channels free to take additional calls. Paul ___ 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] Uniden UIP-200 Multiple line appearances
Noah Miller wrote: Hi - I'm wondering if any has experience with the Uniden UIP-200 phones. The product info says that the 8 led buttons at the top are all programmable. Can they be programmed as separate line appearances (ala Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the phone capable of multiple SIP registrations? No. They can only be programmed for speed-dials, DND, and Mute. Not sure if multiple line appearances is in the product roadmap - i doubt it. Also, the post about these phones at voip-info.org mentions some problems with DHCP and voice prompts getting cut off. Anyone know if these issues have been fixed? DHCP hasn't been an issue for quite some time now. Works perfectly. Voice prompt clipping has not been resolved. I've been unable to get Uniden to acknowledge that the problem even exists (even after providing them a detailed test-case). This is the probably _the_ most annoying thing about UIP200 phones - if you have the problem, please report it to Uniden support ... they give me the impression that I've been the only one to complain about this. ___ 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] Earthlink Releases SIP Based P2P File-Sharing App
This is BAAAD! Now even SIP get's tainted... http://slashdot.org/articles/04/09/16/1317247.shtml?tid=95 _ Surf the net and talk on the phone with Xtra JetStream @ http://xtra.co.nz/jetstream ___ 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] Earthlink Releases SIP Based P2P File-Sharing App
On Thursday 16 September 2004 14:42, Andreas Anderson wrote: This is BAAAD! Now even SIP get's tainted... SIP's already tainted... nasty-ass protocol, that. :-) -A. ___ 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] IAX- FAX
We suffer the same from with outbound using a mediatrix sip/fx box The connected fax machine dials and during handshake drops the call. The Iax link is set to use ULAW Im trying to get asterisk to handle inbound natively, i.e asterisk answer listens and dumps into a file on the linux box, I read voip-info but can get it to work. Have you got a config I can read over? Thanks d -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 16 September 2004 18:13 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] IAX- FAX D, I have a IAX2 gateway that connects to our remote asterisk gateway that has a PRI. Inbound seems to work without a hitch. Make sure your iax.conf allows ULAW as well, Since fax cannot be compressed. Outbound is a different story. My fax seems to ring thru, but it never seems to establish A carrier. Have you been able to get outbound working? Paul Seniuk -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: September 16, 2004 10:40 AM To: asterisk-users Subject: [Asterisk-Users] IAX- FAX Has anyone had any success using iax for inbound fax into asterisk. I tried this but can seem to get asterisk to listen for fax, is it zap specific ? d ___ 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 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] Extensions Submenus
Hi, How can I create a submenus in extensions.conf. For example: 1 for english, 2 for polish and then again depending which option was selected: 1 support 2 sales 3 other 0 operator Thanks Bart, ___ 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
