Re: [asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones
All Polycom phones use the same firmware and bootroms - one reason why the sip.ld is so damn large for them. Thanks Rob. Alleluia! Rob, I will take your word for it - it solves all my worries in deploying different models to the same environment like IP5XX and IP6XX. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones
Can't you just use the same bootrom for all your polycom phones? PaulH On Fri, 2008-03-28 at 15:38 +1100, Lee, John (Sydney) wrote: I have a question about DHCP and boot server supporting more than 1 model of Polycom phones. According to Polycom standards, Polycom phone boots up to get a DHCP address and at the same time getting a boot server string (with username and password) to logon to boot server to download SIP, bootROM and etc. That is okay if there is only one type of phone (that requires a specific SIP and bootROM release). What about if the environment has to support two or more models of Polycom phones? On the boot server side, I can define another home directory like /home/polycom1 and /home/polycom2 to store different SIP and bootROM releases. However, the issue is how different polycom phone model can get a different user account and password to log on to different home directories. I understand the issue here is DHCP and not the boot server but I am a bit of a newbie here. Can anyone help please? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help with voicemail odbc
Dear all, I am still not able to store voicemail into mysql and I am hoping someone can help me out. Here is my voicemail.cof: [general] format = wav attach = yes dbuser=ast dbpass=sqlpass dbhost=localhost dbname=asterisk odbcstorage=asterisk odbctable=voicemessages [default] ; Syntax for new entries looks like this: ; MailboxNumber = password,name,e-mail,pager,options ; (usually, the MailboxNumber is the same as the Extension) 2000 = 1234,Dave Robinson,[EMAIL PROTECTED] 2001 = 1234,Colleen Robinson,[EMAIL PROTECTED] 2002 = 1234,Matthew Robinson,[EMAIL PROTECTED] 2003 = 1234,Lisa Robinson,[EMAIL PROTECTED],,delete=yes Here is my res_odbc.conf [ENV] INFORMIXSERVER = my_special_database INFORMIXDIR = /opt/informix ; All other sections are arbitrary names for database connections. [asterisk] enabled = yes dsn = asterisk username = ast password = sqlpass pooling =no limit = o pre-connect = yes There is no error coming out of asterisk. Can anyone please tell me what could be the problem? Thanks alot for all your help. Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAXy device
I have to chime in here to say that we have had an IAXy for four years and it has given flawless service. Yes, it has no features like DNS but we haven't required this. It's small and easily hidden in a home or soho scenario and I've also used on the same network as the asterisk box or on a network not too far away (like less that 50ms lag). It works fine and has kept on working. All the rest of the complaints are certainly true, but it works and it isn't some kind of useless thing if that's the case. /r On Fri, Mar 28, 2008 at 12:45 AM, Andreas van dem Helge [EMAIL PROTECTED] wrote: It's not bad in the sense of stability (well the original ones are claimed to have overheating issues..). But its that it lacks ANY features. The IAXy has no features at all. Also no security, it MUST be placed behind a firewall, as the configuration doesn't have any sort of security whatsoever. Did I mention it has no features besides DHCP? Not even DNS. Also it's very expensive. I could understand if it was a full-featured device with a webinterface, DNS support 2 Ethernet phone ports I wouldn't complain of the price. But it was released at approx USD $100 at a time when most full-featured adapters sold for a little less, and still sells for $90 today. If they sold them for $40 I wouldn't bash them either.. because honestly thats what they really should be worth. I'd rather use a Grandstream HT than an IAXY honestly. On Thu, Mar 27, 2008 at 3:08 PM, Steve Totaro [EMAIL PROTECTED] wrote: I had a customer using an IAXY (old gen) for an FXO fax machine and it worked almost all the time so it cannot be that bad. Maybe because the fax was very old and did not have high transmit rates. Thanks, Steve Totaro On Thu, Mar 27, 2008 at 2:11 PM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: I guess I've never run asterisk without ANY echo cans :) It's just that the echo was minor enough that MG2 et. al did a fine job. Thanks! Moj Eric Wieling wrote: You will never get latency on a network low enough for echo to be perceived as sidetone (like on analog). If you want to get rid of echo you must cancel echo. Mojo with Horan Company, LLC wrote: Sean Dennis wrote: bilal ghayyad wrote: Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal The device has no echo cancellation and sounds horrible (lots of echo) on about half of the analog phones I tried it on. I wouldn't recommend it unless you absolutely need IAX. It's also very expensive for a 1 port ATA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Echo may be the result of latency on the network. I've not had any echo problems that I remember with my IAXy and I make ten calls a day, five days a week, for the last few years, to all sorts of numbers/areas. I know that this isn't representative of typical business use, but residential use, but I've been using in my business and have never been disappointed :) I will agree that's is fairly expensive, but I WOULD recommend it to people who are on the go often. After setup, it really is plug-n-play IMO. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Call deflection on ISDN PRI in Sweden
Hello List!
We're having trouble making call deflection on ISDN PRI. We would like to
transfer a call to an external extension but keeping the callerid of the caller
so it can be presented to the receiver of the transferred call.
At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware
TE420B. We've ordered the service (CD) from the phone company.
The zapata.conf file inlcludes:
Transfer= yes
Facilityenable=yes
Callerid=asreceived
In extensions.conf we try to transfer a call to an external extension as:
Transfer(ZAP/g0/ ) but that fails with the ${TRANSFERSTATUS} =
UNSUPPORTED.
Ideas anyone? We would really appreciate it!
Kind regards,
Hanna
Hanna Wallin
System Development
Direct: +46 (0)8 736 77 29
Mobile: +46 (0)73 414 13 38
Fax: +46 (0)8 736 77 91
E-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
PocketMobile Communications AB
Wenner-Gren Center
Sveavägen 168, 3 tr
113 46 Stockholm
Nordic web page: www.pocketmobile.se BLOCKED::http://www.pocketmobile.se
International web page: www.pocketmobileworld.com
BLOCKED::http://www.pocketmobileworld.com/
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[asterisk-users] IAX user register problem
hi,
i want to call PC2PC between to IAX client without authentication i
want to allow every user to use PC2PC no any password required. Please
let me know what i have need to do in IAX.conf or any other file to
allow any user to call Pc2Pc.
My IAX.conf
[guest]
type=user
context=default
callerid=Guest IAX User
My extensions.conf
[default]
exten=_.,1,Dial(IAX2/${EXTEN})
exten=_y.,1,Dial(IAX2/${EXTEN})
exten=_a.,1,Dial(IAX2/${EXTEN})
below is my Asterisk console logs which i see after making call.
Mar 28 03:25:43 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:6910 socket_read: Rejected
connect attempt from 203.99.57.80, who was trying to reach
'jaffaradvcommnet@'
Mar 28 03:26:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'j' (from 203.99.57.80)
advcomm6*CLI iax2 show channels
Channel Peer UsernameID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None)203.99.57.80 (None) 4/15232
1/1 0ms -0001ms ms unknow
(None)203.99.57.80 jaffaradvc 5/15233
4/4 0ms -0001ms ms unknow
(None)203.99.57.80 (None) 6/18423
1/1 0ms -0001ms ms unknow
3 active IAX channels
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:27:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'jaffaradvcommnet' (from 203.99.57.80)
i am very thankful if some one help me in this regards,
regards,
Asif
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Re: [asterisk-users] Calling users to the external domain using Asterisk
What you are looking for is something like this piece of code. Adapt it for
your scenario:
[default]
exten = _.,1,NoOp(incomming call from ${CALLERID} to [EMAIL PROTECTED])
exten = _.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
exten = _.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
exten = _.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
exten = _.,5,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
exten = _.,6,GotoIf($[${SIPDOMAIN} = 192.168.1.1]?10)
exten = _.,7,NoOp(@${SIPDOMAIN} is from an external domain, sending to
it...)
exten = _.,8,Dial(SIP/[EMAIL PROTECTED])
exten = _.,9,HangUp()
exten = _.,10,Goto(noturi-default,${EXTEN},1)
exten = h,1,HangUp()
[noturi-default]
;(your dialplan)
Regards,
Ricardo Carvalho.
On Thu, Mar 27, 2008 at 7:47 AM, Aadilkhan Maniyar [EMAIL PROTECTED]
wrote:
Hi All,
I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17 and
using it to make SIP calls.
I have a configuration of Asterisk which serves the users in a particular
domain, say internal.com
I would like to make a SIP call from [EMAIL PROTECTED] to
[EMAIL PROTECTED]
I have added the following lines in extensions.conf
exten = charles,1,Dial(SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED]
)
exten = charles,2,Hangup
Asterisk does a DNS SRV lookup and resolves the external.com to its proper
IP and calls are established.
But the problem with the above configuration is that I have manually added
users that are in the external domain.
Is there any way wherein I can call the users in external.com without
adding them in the extensions.conf?
Any help would be appreciated.
Thanks,
Aadil
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[asterisk-users] PRI error cause hangup calls
Dear all, When I make a call using my PRI line, all goes well, but suddently the call hangs up. I searched the asterisk logs, and I found that. Write to 55 failed: Unknown error 500 Short write: 0/15 (Unknown error 500) What does this mean? Why this occurs? How could I solve that? Someone could tell me if it was a primary error (the primary shows red alert in all its channels) or it could be a driver or config problem? Thanks in advance. VoipCrazy. Mar 27 14:28:00 DEBUG[20313] chan_zap.c: Got event Alarm(4) on channel 1 (index 0) Mar 27 14:28:00 VERBOSE[20313] logger.c: Write to 55 failed: Unknown error 500 Mar 27 14:28:00 VERBOSE[20313] logger.c: Short write: 0/15 (Unknown error 500) Mar 27 14:28:00 WARNING[20313] chan_zap.c: Detected alarm on channel 1: Red Alarm Mar 27 14:28:00 DEBUG[20313] chan_zap.c: disabled echo cancellation on channel 1 Mar 27 14:28:00 DEBUG[20313] channel.c: Didn't get a frame from channel: Zap/1-1 Mar 27 14:28:00 DEBUG[20313] channel.c: Bridge stops bridging channels SIP/7008-b6a158e0 and Zap/1-1 Mar 27 14:28:00 DEBUG[20313] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1 Mar 27 14:28:00 DEBUG[20313] chan_zap.c: Hangup: channel: 1 index = 0, normal = 36, callwait = -1, thirdcall = -1 Mar 27 14:28:00 DEBUG[20313] chan_zap.c: disabled echo cancellation on channel 1 Mar 27 14:28:00 DEBUG[20313] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Mar 27 14:28:00 DEBUG[20313] chan_zap.c: Updated conferencing on 1, with 0 conference users Mar 27 14:28:00 DEBUG[20313] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/1-1 Mar 27 14:28:00 DEBUG[20313] chan_zap.c: disabled echo cancellation on channel 1 Mar 27 14:28:00 VERBOSE[20313] logger.c: -- Hungup 'Zap/1-1' Mar 27 14:28:00 DEBUG[20313] app_dial.c: Exiting with DIALSTATUS=ANSWER. Mar 27 14:28:00 VERBOSE[20313] logger.c: == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/7008-b6a158e0' in macro 'dialout-trunk' Mar 27 14:28:00 VERBOSE[20313] logger.c: == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/7008-b6a158e0' Mar 27 14:28:00 VERBOSE[20313] logger.c: -- Executing Macro(SIP/7008-b6a158e0, hangupcall) in new stack Mar 27 14:28:00 VERBOSE[20313] logger.c: -- Executing ResetCDR(SIP/7008-b6a158e0, w) in new stack Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 2: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 2 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 3: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 3 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 4: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 4 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 5: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 5 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 6: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 6 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 7: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 7 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 8: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 8 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 9: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 9 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 10: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 10 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 11: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 11 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 12: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 12 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 13: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 13 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 14: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 14 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 15: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 15 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 17: Red Alarm Mar 27 14:28:00 WARNING[5155] chan_zap.c: Unable to disable echo cancellation on channel 17 Mar 27 14:28:00 WARNING[5155] chan_zap.c: Detected alarm on channel 18: Red Alarm Mar 27
[asterisk-users] sip.conf setvar option
Hi, does anybody know about the setvar option in asterisk's sip.conf. I am trying to define it for a peer that's used when making calls using the originate ami call, but it seems to not have any effect. Marcus -- Marcus Hunger - [EMAIL PROTECTED] Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 indigo networks GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5713/2881, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling users to the external domain usingAsterisk
Thanks for the reply Recardo..
I was indeed looking at something like this.
Also I was also looking at Asterisk's SRV lookups. Is there anyway I can
know that a SRV lookup has failed?
Regards,
Aadil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ricardo
Carvalho
Sent: Friday, March 28, 2008 4:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calling users to the external domain
usingAsterisk
What you are looking for is something like this piece of code. Adapt it
for your scenario:
[default]
exten = _.,1,NoOp(incomming call from ${CALLERID} to
[EMAIL PROTECTED])
exten = _.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
exten = _.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
exten = _.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
exten = _.,5,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
exten = _.,6,GotoIf($[${SIPDOMAIN} = 192.168.1.1]?10)
exten = _.,7,NoOp(@${SIPDOMAIN} is from an external domain, sending to
it...)
exten = _.,8,Dial(SIP/[EMAIL PROTECTED])
exten = _.,9,HangUp()
exten = _.,10,Goto(noturi-default,${EXTEN},1)
exten = h,1,HangUp()
[noturi-default]
;(your dialplan)
Regards,
Ricardo Carvalho.
On Thu, Mar 27, 2008 at 7:47 AM, Aadilkhan Maniyar
[EMAIL PROTECTED] wrote:
Hi All,
I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17
and using it to make SIP calls.
I have a configuration of Asterisk which serves the users in a
particular domain, say internal.com
I would like to make a SIP call from [EMAIL PROTECTED] to
[EMAIL PROTECTED]
I have added the following lines in extensions.conf
exten = charles,1,Dial(SIP/[EMAIL PROTECTED]
mailto:SIP/[EMAIL PROTECTED] )
exten = charles,2,Hangup
Asterisk does a DNS SRV lookup and resolves the external.com to its
proper IP and calls are established.
But the problem with the above configuration is that I have manually
added users that are in the external domain.
Is there any way wherein I can call the users in external.com without
adding them in the extensions.conf?
Any help would be appreciated.
Thanks,
Aadil
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Re: [asterisk-users] IAX user register problem
i am getting Registration Refused error when i debug on console.
please tell me how can i registration every user without any username
and password and these user can make calls between each other.
i am very thankful if any body help me in this regards,
advcomm6*CLIiax2 debug
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 3ms SCall: 09398 DCall: 0 [203.99.57.80:47641]
USERNAME: aliadvcommnet
REFRESH : 60
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ
Timestamp: 2ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
CAUSE : Registration Refused
CAUSE CODE : 29
On Fri, Mar 28, 2008 at 3:13 AM, Mian M Asif [EMAIL PROTECTED] wrote:
hi,
i want to call PC2PC between to IAX client without authentication i
want to allow every user to use PC2PC no any password required. Please
let me know what i have need to do in IAX.conf or any other file to
allow any user to call Pc2Pc.
My IAX.conf
[guest]
type=user
context=default
callerid=Guest IAX User
My extensions.conf
[default]
exten=_.,1,Dial(IAX2/${EXTEN})
exten=_y.,1,Dial(IAX2/${EXTEN})
exten=_a.,1,Dial(IAX2/${EXTEN})
below is my Asterisk console logs which i see after making call.
Mar 28 03:25:43 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:6910 socket_read: Rejected
connect attempt from 203.99.57.80, who was trying to reach
'jaffaradvcommnet@'
Mar 28 03:26:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'j' (from 203.99.57.80)
advcomm6*CLI iax2 show channels
Channel Peer UsernameID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None)203.99.57.80 (None) 4/15232
1/1 0ms -0001ms ms unknow
(None)203.99.57.80 jaffaradvc 5/15233
4/4 0ms -0001ms ms unknow
(None)203.99.57.80 (None) 6/18423
1/1 0ms -0001ms ms unknow
3 active IAX channels
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:27:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'jaffaradvcommnet' (from 203.99.57.80)
i am very thankful if some one help me in this regards,
regards,
Asif
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Re: [asterisk-users] problem about voice when using TDM2400p with VPMADT032 echo canceller module
On Wed, 2008-03-26 at 20:45 -0700, Vu AnhTuan wrote: I'm having problem with voice quality on my trixbox using TDM2400B.The trixbox is connected via 20 FXO ports on a TDM2400 with the hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'background crackle/buzz' coming back when they talk. Please contact Digium support, as they'll be better able to help you track down the source of the crackle. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-962+ SPA-932- blf function
On Fri, 2008-03-28 at 02:12 +, John Meksavan wrote: I got the sidecar to subscribed to an extension on the Asterisk server, but the LED state on the SPA-932 never changes even when I am a call with that extension on another VOIP phone- SPA-941. I got the speed dial function to work, but the blf function does not appear to work. I documented the steps to get this working near the bottom of the page at http://www.voip-info.org/wiki/view/SPA-962. In essence, you need to make sure you have dialplan hints, and that those hints are working properly. (This usually involves setting the calllimit setting in sip.conf -- I typically set it to a value of 99.) -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf setvar option
On Fri, 2008-03-28 at 12:30 +0100, Marcus Hunger wrote:
does anybody know about the setvar option in asterisk's sip.conf.
Sure! This is one of my favorite features.
Let's say I have a definition for my phone in sip.conf, and it looks
something like this:
[myphone]
secret=verysecretpassword
type=friend ; a friend is both a user and a peer
host=dynamic ; phone will register to Asterisk
disallow=all
allow=gsm ; first, try to negotiate gsm
allow=ulaw; the try ulaw
setvar=MYVAR=blah
Whenever a call comes into Asterisk from this particular phone, Asterisk
will automatically create a channel variable named MYVAR, and ${MYVAR}
will contain the value blah. I can then use it for whatever purpose I
see fit within my dialplan.
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] Calling users to the external domain usingAsterisk
You can test manually any SRV DNS record using dig, like this:
dig -t SRV _sip._udp.fwd.pulver.com
At the asterisk CLI you can also verify that SRV lookup has been succeeded.
It shows something like this when it does:
parse_srv: SRV mapped to host fwd.pulver.com, port 5060
In your dialplan you can also trigger some Set(CDR(userfield)=SRV call from
${SIPCHANINFO(recvip)}) so that in your mysql CDR table be written which
calls got sent by IP to any SIP URI.
Regards,
Ricardo Carvalho.
On Fri, Mar 28, 2008 at 12:00 PM, Aadilkhan Maniyar [EMAIL PROTECTED]
wrote:
Thanks for the reply Recardo..
I was indeed looking at something like this…
Also I was also looking at Asterisk's SRV lookups… Is there anyway I can
know that a SRV lookup has failed?
Regards,
Aadil
-Original Message-
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Ricardo Carvalho
*Sent:* Friday, March 28, 2008 4:07 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Calling users to the external domain
usingAsterisk
What you are looking for is something like this piece of code. Adapt it
for your scenario:
[default]
exten = _.,1,NoOp(incomming call from ${CALLERID} to
[EMAIL PROTECTED])
exten = _.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
exten = _.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
exten = _.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
exten = _.,5,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
exten = _.,6,GotoIf($[${SIPDOMAIN} = 192.168.1.1]?10)
exten = _.,7,NoOp(@${SIPDOMAIN} is from an external domain, sending to
it...)
exten = _.,8,Dial(SIP/[EMAIL PROTECTED])
exten = _.,9,HangUp()
exten = _.,10,Goto(noturi-default,${EXTEN},1)
exten = h,1,HangUp()
[noturi-default]
;(your dialplan)
Regards,
Ricardo Carvalho.
On Thu, Mar 27, 2008 at 7:47 AM, Aadilkhan Maniyar
[EMAIL PROTECTED] wrote:
Hi All,
I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17 and
using it to make SIP calls.
I have a configuration of Asterisk which serves the users in a particular
domain, say internal.com
I would like to make a SIP call from [EMAIL PROTECTED] to
[EMAIL PROTECTED]
I have added the following lines in extensions.conf
exten = charles,1,Dial(SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED]
)
exten = charles,2,Hangup
Asterisk does a DNS SRV lookup and resolves the external.com to its proper
IP and calls are established.
But the problem with the above configuration is that I have manually added
users that are in the external domain.
Is there any way wherein I can call the users in external.com without
adding them in the extensions.conf?
Any help would be appreciated.
Thanks,
Aadil
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Re: [asterisk-users] sip.conf setvar option
28 mar 2008 kl. 13.42 skrev Jared Smith:
On Fri, 2008-03-28 at 12:30 +0100, Marcus Hunger wrote:
does anybody know about the setvar option in asterisk's sip.conf.
Sure! This is one of my favorite features.
Let's say I have a definition for my phone in sip.conf, and it looks
something like this:
[myphone]
secret=verysecretpassword
type=friend ; a friend is both a user and a peer
host=dynamic ; phone will register to Asterisk
disallow=all
allow=gsm ; first, try to negotiate gsm
allow=ulaw; the try ulaw
setvar=MYVAR=blah
Whenever a call comes into Asterisk from this particular phone,
Asterisk
will automatically create a channel variable named MYVAR, and ${MYVAR}
will contain the value blah. I can then use it for whatever
purpose I
see fit within my dialplan.
Well, Jared, but that's the reverse. You stripped out this important
part:
am trying to define it for a peer that's used when making calls
using the originate ami call, but it seems to not have any effect.
The important thing with your lesson was that SETVAR is only used on
INCOMING calls from
devices, not outbound calls TO devices. Using ORIGINATE to call a SIP
peer, there's no variables
set from sip.conf.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/ * SIP Masterclass
Orlando FL * April 21-25 2008
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Re: [asterisk-users] sip.conf setvar option
So, wouldn't it be great to enable setvar for outgoing calls too?
On Fri, Mar 28, 2008 at 1:55 PM, Johansson Olle E [EMAIL PROTECTED] wrote:
28 mar 2008 kl. 13.42 skrev Jared Smith:
On Fri, 2008-03-28 at 12:30 +0100, Marcus Hunger wrote:
does anybody know about the setvar option in asterisk's sip.conf.
Sure! This is one of my favorite features.
Let's say I have a definition for my phone in sip.conf, and it looks
something like this:
[myphone]
secret=verysecretpassword
type=friend ; a friend is both a user and a peer
host=dynamic ; phone will register to Asterisk
disallow=all
allow=gsm ; first, try to negotiate gsm
allow=ulaw; the try ulaw
setvar=MYVAR=blah
Whenever a call comes into Asterisk from this particular phone,
Asterisk
will automatically create a channel variable named MYVAR, and ${MYVAR}
will contain the value blah. I can then use it for whatever
purpose I
see fit within my dialplan.
Well, Jared, but that's the reverse. You stripped out this important
part:
am trying to define it for a peer that's used when making calls
using the originate ami call, but it seems to not have any effect.
The important thing with your lesson was that SETVAR is only used on
INCOMING calls from
devices, not outbound calls TO devices. Using ORIGINATE to call a SIP
peer, there's no variables
set from sip.conf.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/ * SIP Masterclass
Orlando FL * April 21-25 2008
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--
Marcus Hunger - [EMAIL PROTECTED]
Telefon: +49 (0)211-63 55 55-61
Telefax: +49 (0)211-63 55 55-22
indigo networks GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5713/2881, Umsatzsteuer-ID: DE219349391
www.sipgate.de - www.sipgate.at - www.sipgate.co.uk
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Re: [asterisk-users] sip.conf setvar option
On Fri, 2008-03-28 at 13:55 +0100, Johansson Olle E wrote: Well, Jared, but that's the reverse. You stripped out this important part: am trying to define it for a peer that's used when making calls using the originate ami call, but it seems to not have any effect. The important thing with your lesson was that SETVAR is only used on INCOMING calls from devices, not outbound calls TO devices. Using ORIGINATE to call a SIP peer, there's no variables set from sip.conf. Absolutely true... and I'll make up for it by pointing out that if you're using the Originate manager command, you can set channel variables by adding the Variable setting to your manager command: Action: Originate Channel: SIP/myphone Context: test Exten: 123 Priority: 1 Async: True ActionID: ThisIsMyVeryOriginalActionID Variable: MYVAR=blah|ANOTHERVAR=baz -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] wrong extension status when call-limit=1 is used
Without call-limit defined, when a sip extension calls another sip extension then show hints shows that both are InUse (as expected). When one of them hangs up, both hints status become Idle (as expected). With call-limit=1 for each SIP extension: the caller is always Idle while the callee is InUse. Is this behavior normal? Doesn't sound right because if, during the latter conversation, another extension calls the caller then it will ring (but shouldn't since call-limit=1 for everyone). The worst case is: if I call from SIP/6010 to SIP/4053, 4053 puts 6010 on hold, 6010 hangs up, then show hints shows that 6010 is Idle but 4053 is Busy and stays like that even if the 4053 softphone re-registers. The only way to clear this Busy state is to restart the asterisk daemon. show channels says that there are 0 active channels and 0 active calls. I am running asterisk 1.2.27. I require call-limit=1 or similar option because I would like the extensions to accept only one call at a time (whether receiving or calling). It can't be done on the client side because the softphones used don't allow that in their configuration (using SJphone). Does call-limit have a known bug (at least for call-limit=1)? Thanks, Vieri Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Two phones fail to agree on codec, asterisk at fault?
Hi list, I am faced by a situation where I am trying to make a softphone and a Siemens C450IP talk to each other. Both are hooked up directly to the same asterisk, in the same IP net. - a softphone runs on 192.168.14.3 - the C450IP is at 192.168.14.30 - asterisk runs on the machine known as 192.168.14.1 I am running Asterisk 1.4.11, backported to Debian Etch by Xorcom. If I set canreinvite=yes for both, everything works. However, I have reason to use canreinvite=no for both. But if I do, then the two phones fail to agree on a codec. So calls are going via an asterisk bridge and the symptoms of my problem are: 1 if C450IP calls softphone, they can talk fine 2 if softphone calls C450IP, voice only goes from C450IP to softphone, not the other way around. I traced this down to the session description protocol, where there is funky stuff going on with the supported codecs each peer announces. Remember, asterisk is between them, and I set disallow=all,allow=ulaw,allow=alaw in [global]. So in situation 1, when the C450IP calls the softphone, these codecs are announced. 0 is ulaw, 8 is alaw, 111 is g726-32, 3 is gsm. C450IP to asterisk: 8, 0 asterisk to softph: 8, 3, 0 softph to asterisk: 8 asterisk to C450IP: 8, 0 They both agree on 8 (alaw) and stuff is working, but it's already curious how asterisk adds the 3 (GSM) in the second line and the 0 (ulaw) in the last. In situation 2, no voice travels from the softphone to the C450IP, and this is the dialog: softph to asterisk: 8, 0, 3 asterisk to C450IP: 0, 8, 111 C450IP to asterisk: 0 asterisk to softph: 3, 0, 8 Again, notice how asterisk basically ignores what it was asked to relay. In the end, the softphone settles for 3 (GSM) but the C450IP chooses 0 (ulaw). Since the softphone has no problem decoding ulaw, it can hear whatever the C450IP transmits, but it returns GSM packets, which the C450IP can't decode, and therefore nothing comes out of that phone. What's going on here? From all I can tell, the clients do the right thing, each selecting the first codec offered by asterisk (which they support), but asterisk is going a bit lala here, isn't it? First of all, why does it even bother with 3 and 111, given how I disallowed them? And second, why does it *dare* to announce more than what is available to the peer to which it relays? -- martin | http://madduck.net/ | http://two.sentenc.es/ there are more things in heaven and earth, horatio, than are dreamt of in your philosophy. -- hamlet spamtraps: [EMAIL PROTECTED] digital_signature_gpg.asc Description: Digital signature (see http://martin-krafft.net/gpg/) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callers in queue passed to agents who accept only one call at a time
--- Atis Lezdins [EMAIL PROTECTED] wrote: On Thu, Mar 27, 2008 at 6:32 PM, Vieri [EMAIL PROTECTED] wrote: I have a queue I configured as strict and a cron script I use to QueueAdd and QueueRemove agents according to my company's requirements. Usually I have 2 or 3 agents at a time and the ring strategy is ringall. These agents use non-open-source Windows softphones that do not let you configure it so that if they're on the phone, a second call will be rejected (agent busy). Instead, it's as if they had call waiting and incoming calls keep popping up while they're conversating with the first caller and they would like to avoid this. I guess the easiest solution would be to find an open-source or free softphone that can be configured to accept only one call at a time (currently using SJphone). Another solution would be if I could tell the Queue() application that if an agent is InUse then don't pass the call. Still another yet more delicate solution would be to have a custom script receive manager events related to the queue which in turn replies with an agi command. For example, whenever an agent answers a call I think that an event such as QueueMemberStatus can be triggered (although I don't know how). If the custom script could receive this event in realtime then it would run an agi command such as QueueRemove(busyagent...). When the agent is free again I suppose the same event is triggered and the custom script can QueueAdd(freeagent...). Could anyone please give me some pointers on this? In queues.conf set ringinuse=no Also make sure that you don't use realtime sip peers (or use rtcachefriends with that). Probably you also need call-limit set to any value in sip.conf Thanks Atis and Rodrigo. However, I can't use ringinuse=no in queues.conf because I'm running 1.2.27 (or is there a backport/patch?). If I use call-limit=1 then I get all sorts of problems (see http://lists.digium.com/pipermail/asterisk-users/2008-March/208558.html ) Besides, call-limit=1 would not allow the agent to do attended transfers. I guess I'm forced to upgrade to 1.4 although there have been several instability issues lately, even on this mailing list. Vieri Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] recommendable softphones / X-Lite / Zoiper for amd64?
Hi, I am on amd64 Linux and not really too happy with twinkle, linphone and ekiga. Unfortunately, X-Lite and Zoiper, even though they provide Linux versions (w00t!) have only x86 versions for download. Do you guys know of amd64 versions of those, or can you recommend other softphones that will run on amd64, or which come with source code? Thanks, -- martin | http://madduck.net/ | http://two.sentenc.es/ i wish there was a knob on the tv to turn up the intelligence. there's a knob called 'brightness', but it doesn't seem to work. -- gallagher spamtraps: [EMAIL PROTECTED] digital_signature_gpg.asc Description: Digital signature (see http://martin-krafft.net/gpg/) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf setvar option
28 mar 2008 kl. 14.00 skrev Marcus Hunger: So, wouldn't it be great to enable setvar for outgoing calls too? Well, maybe in the outbound channel then. But that won't help much. mixing the caller's and callee's variables in the INCOMING channel would be messy and only cause issues. But there's another way. Hint hint. Friday afternoon hack. /O ;-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrong extension status when call-limit=1 is used
Remember that if you enable call-limit=1 with a type=friend, you will actually have one inbound call (on the user) and one outbound call (on the peer). Groupcount in the dialplan is propably a better solution to enforce call limits than anything in the SIP channel. It works with all channel drivers too, as an extra benefit. Regards, /Olle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf setvar option
Particularly, I want to set the SIPADDHEADER variable dynamicly for peers with rt-engine. Working around it might be possible, but having the thing working transparently for Dial and Originate would be great. On Fri, Mar 28, 2008 at 2:47 PM, Johansson Olle E [EMAIL PROTECTED] wrote: 28 mar 2008 kl. 14.00 skrev Marcus Hunger: So, wouldn't it be great to enable setvar for outgoing calls too? Well, maybe in the outbound channel then. But that won't help much. mixing the caller's and callee's variables in the INCOMING channel would be messy and only cause issues. But there's another way. Hint hint. Friday afternoon hack. /O ;-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Marcus Hunger - [EMAIL PROTECTED] Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 indigo networks GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5713/2881, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrong extension status when call-limit=1 is used
--- Johansson Olle E [EMAIL PROTECTED] wrote: Remember that if you enable call-limit=1 with a type=friend, you will actually have one inbound call (on the user) and one outbound call (on the peer). Groupcount in the dialplan is propably a better solution to enforce call limits than anything in the SIP channel. It works with all channel drivers too, as an extra benefit. Thanks but suppose the caller is sent to a queue and I want agents to ring only if they are not busy. How could I use groupcount in this case? (in * 1.2) Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to register IAX user without password
hi,
i want to call PC2PC between to IAX client without authentication i
want to allow every user to use PC2PC no any password required. Please
let me know what i have need to do in IAX.conf or any other file to
allow any user to call Pc2Pc.
My IAX.conf
[guest]
type=user
context=default
callerid=Guest IAX User
My extensions.conf
[default]
exten=_.,1,Dial(IAX2/${EXTEN})
exten=_y.,1,Dial(IAX2/${EXTEN})
exten=_a.,1,Dial(IAX2/${EXTEN})
below is my Asterisk console logs which i see after making call.
Mar 28 03:25:43 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:6910 socket_read: Rejected
connect attempt from 203.99.57.80, who was trying to reach
'jaffaradvcommnet@'
Mar 28 03:26:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'j' (from 203.99.57.80)
advcomm6*CLI iax2 show channels
Channel Peer UsernameID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None)203.99.57.80 (None) 4/15232
1/1 0ms -0001ms ms unknow
(None)203.99.57.80 jaffaradvc 5/15233
4/4 0ms -0001ms ms unknow
(None)203.99.57.80 (None) 6/18423
1/1 0ms -0001ms ms unknow
3 active IAX channels
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:27:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'jaffaradvcommnet' (from 203.99.57.80)
i am very thankful if some one help me in this regards,
i am getting Registration Refused error when i debug on console.
please tell me how can i registration every user without any username
and password and these user can make calls between each other.
i am very thankful if any body help me in this regards,
advcomm6*CLIiax2 debug
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 3ms SCall: 09398 DCall: 0 [203.99.57.80:47641]
USERNAME: aliadvcommnet
REFRESH : 60
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ
Timestamp: 2ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
CAUSE : Registration Refused
CAUSE CODE : 29
regards,
Asif
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Re: [asterisk-users] sip.conf setvar option
28 mar 2008 kl. 14.56 skrev Marcus Hunger: Particularly, I want to set the SIPADDHEADER variable dynamicly for peers with rt-engine. Working around it might be possible, but having the thing working transparently for Dial and Originate would be great. That should work today with the unofficial backdoor I implemented. sipaddheader just adds a few channel variables that the outbound channel inherits. If you add them yourself with setvar=_SIPADDHEADER99=X-peeraccountcode: 12345 I think that should work. Out of the box, like magic. This of course only works with calls FROM peers. Have a nice weekend! /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-962+ SPA-932- blf function
John Meksavan wrote: Asterisk Users, I am running Asterisk 1.4.11 on Debian Etch system with the TDM03B wildcard. I recently purchased a SPA-962 and SPA-932- the sidecar for our receptionist. After reading many forum postings on how to configure the side car, I uprgraded the SPA-962 software to 5.1.18(SC) version. I got the sidecar to subscribed to an extension on the Asterisk server, but the LED state on the SPA-932 never changes even when I am a call with that extension on another VOIP phone- SPA-941. I got the speed dial function to work, but the blf function does not appear to work. Did anybody get the blf function to work? What I am doing wrong? Any input would be greatly appreciated. Thanks in advance. Regards, John How well do you know your celebrity gossip? Talk celebrity smackdowns here. http://originals.msn.com/thebigdebate?ocid=T002MSN03N0707A ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users To make it work properly I had to add the following to sip.conf: allowsubscribe=yes notifyringing=yes limitonpeer=yes notifyhold=yes See if that helps. -Sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem about voice when using TDM2400p with VPMADT032 echo canceller module
Vu AnhTuan wrote: hi you, I'm having problem with voice quality on my trixbox using TDM2400B.The trixbox is connected via 20 FXO ports on a TDM2400 with the hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'background crackle/buzz' coming back when they talk. anyone have the same problem? pls help me. thanks a lot. my trixbox and config file: trixbox version 2.4 (Linux kernel 2.6.18, Zaptel 1.4.7) This is definitely a technical support issue. Please contact them about this so that we can help you get it resolved as soon as possible :-) ! Matthew Fredrickson Digium, Inc. zaptel.conf # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCTDM/0 Wildcard TDM2400P Board 1 fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxsks=5 fxsks=6 fxsks=7 fxsks=8 fxsks=9 fxsks=10 fxsks=11 fxsks=12 fxsks=13 fxsks=14 fxsks=15 fxsks=16 fxsks=17 fxsks=18 fxsks=19 fxsks=20 # channel 21, WCTDM, no module. # channel 22, WCTDM, no module. # channel 23, WCTDM, no module. # channel 24, WCTDM, no module. # Global data loadzone = us defaultzone = us zapata.conf -- ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;default ;echotraining=800 ;default rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=0 relaxdtmf=yes ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-channels.conf group=1 ;Include AMP configs #include zapata_additional.conf zapata_additional.conf --- ; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; ; Span 1: WCTDM/0 Wildcard TDM2400P Board 1 ;;; line=1 WCTDM/0/0 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 1 context=default ;;; line=2 WCTDM/0/1 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 2 context=default ;;; line=3 WCTDM/0/2 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 3 context=default ;;; line=4 WCTDM/0/3 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 4 context=default ;;; line=5 WCTDM/0/4 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 5 context=default ;;; line=6 WCTDM/0/5 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 6 context=default ;;; line=7 WCTDM/0/6 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 7 context=default ;;; line=8 WCTDM/0/7 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 8 context=default ;;; line=9 WCTDM/0/8 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 9 context=default ;;; line=10 WCTDM/0/9 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 10 context=default ...more... [IP-PBX ~]# ztcfg -vv -- Zaptel Version: 1.4.7-3259 Echo Canceller: OSLEC Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) Channel 05: FXS Kewlstart (Default) (Slaves: 05) Channel 06: FXS Kewlstart (Default) (Slaves: 06) Channel 07: FXS Kewlstart (Default) (Slaves: 07) Channel 08: FXS Kewlstart (Default) (Slaves: 08) Channel 09: FXS Kewlstart (Default) (Slaves: 09) Channel 10: FXS Kewlstart (Default) (Slaves: 10) Channel 11: FXS Kewlstart (Default) (Slaves: 11) Channel 12: FXS Kewlstart (Default) (Slaves: 12) Channel 13: FXS Kewlstart (Default) (Slaves: 13) Channel 14: FXS Kewlstart (Default) (Slaves: 14) Channel 15: FXS Kewlstart (Default) (Slaves: 15) Channel 16: FXS Kewlstart (Default) (Slaves: 16) Channel 17: FXS Kewlstart (Default) (Slaves: 17) Channel
Re: [asterisk-users] recommendable softphones / X-Lite / Zoiper for amd64?
Contact me at [EMAIL PROTECTED] and ask for a beta for the 64 bit build of zoiper Cheers, Zoa martin f krafft wrote: Hi, I am on amd64 Linux and not really too happy with twinkle, linphone and ekiga. Unfortunately, X-Lite and Zoiper, even though they provide Linux versions (w00t!) have only x86 versions for download. Do you guys know of amd64 versions of those, or can you recommend other softphones that will run on amd64, or which come with source code? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to register IAX user without password
Create a User and a Peer on both the machines for each other.
e.g IAX.conf on PCa
[pca2pcb]
type=peer
host=[IP OF pcb]
username=pca2pcb
serect=pca2pcb12345
qualify=yes
[pcb2pca]
type=user
context=default
auth=md5
secret=pcb2pca12345
deny=0.0.0.0/0.0.0.0
permit=[IP of pcb]
qualify=yes
ON PCb do the reverse in iax.conf
[pcb2pca]
type=peer
host=[IP OF pca]
username=pcb2pca
serect=pcb2pca12345
qualify=yes
[pca2pcb]
type=user
context=default
auth=md5
secret=pca2pcb12345
deny=0.0.0.0/0.0.0.0
permit=[IP of pca]
qualify=yes
NOW in Your extensions.conf you can use as
On PCa
exten=_.,1,Dial(IAX2/pca2pcb/${EXTEN})
exten=_y.,1,Dial(IAX2/pca2pcb/${EXTEN})
exten=_a.,1,Dial(IAX2/pca2pcb/${EXTEN})
and on PCb
exten=_.,1,Dial(IAX2/pcb2pca/${EXTEN})
exten=_y.,1,Dial(IAX2/pcb2pca/${EXTEN})
exten=_a.,1,Dial(IAX2/pcb2pca/${EXTEN})
Let me know if this works.
Regards,
Sanjay.
- Original Message -
From: Mian M Asif [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, March 28, 2008 8:04:08 PM (GMT+0530) Asia/Calcutta
Subject: [asterisk-users] how to register IAX user without password
hi,
i want to call PC2PC between to IAX client without authentication i
want to allow every user to use PC2PC no any password required. Please
let me know what i have need to do in IAX.conf or any other file to
allow any user to call Pc2Pc.
My IAX.conf
[guest]
type=user
context=default
callerid=Guest IAX User
My extensions.conf
[default]
exten=_.,1,Dial(IAX2/${EXTEN})
exten=_y.,1,Dial(IAX2/${EXTEN})
exten=_a.,1,Dial(IAX2/${EXTEN})
below is my Asterisk console logs which i see after making call.
Mar 28 03:25:43 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:6910 socket_read: Rejected
connect attempt from 203.99.57.80, who was trying to reach
'jaffaradvcommnet@'
Mar 28 03:26:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'j' (from 203.99.57.80)
advcomm6*CLI iax2 show channels
Channel Peer UsernameID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None)203.99.57.80 (None) 4/15232
1/1 0ms -0001ms ms unknow
(None)203.99.57.80 jaffaradvc 5/15233
4/4 0ms -0001ms ms unknow
(None)203.99.57.80 (None) 6/18423
1/1 0ms -0001ms ms unknow
3 active IAX channels
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:27:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'jaffaradvcommnet' (from 203.99.57.80)
i am very thankful if some one help me in this regards,
i am getting Registration Refused error when i debug on console.
please tell me how can i registration every user without any username
and password and these user can make calls between each other.
i am very thankful if any body help me in this regards,
advcomm6*CLIiax2 debug
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 3ms SCall: 09398 DCall: 0 [203.99.57.80:47641]
USERNAME: aliadvcommnet
REFRESH : 60
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ
Timestamp: 2ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
CAUSE : Registration Refused
CAUSE CODE : 29
regards,
Asif
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[asterisk-users] jingle with Asterisk + PSTN
Hi All I am developing a client that uses libjingle to do xmpp stuff with ejabberd. I can also make audio calls between those clients. What I am trying to archive now is to send calls to pstn using jingle. I was told in the jingle-dev community that asterisk can do that. Is there any way to send jingle audio calls to asterisk and will it understand them ? If yes..can I forward those calls to PSTN ? Thx Any feedback is appreciated. Note: I do not intend to implement SIP in my client ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAXy device
Mojo with Horan Company, LLC wrote: Sean Dennis wrote: bilal ghayyad wrote: Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal The device has no echo cancellation and sounds horrible (lots of echo) on about half of the analog phones I tried it on. I wouldn't recommend it unless you absolutely need IAX. It's also very expensive for a 1 port ATA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Echo may be the result of latency on the network. I've not had any echo problems that I remember with my IAXy and I make ten calls a day, five days a week, for the last few years, to all sorts of numbers/areas. I know that this isn't representative of typical business use, but residential use, but I've been using in my business and have never been disappointed :) I will agree that's is fairly expensive, but I WOULD recommend it to people who are on the go often. After setup, it really is plug-n-play IMO. Just to put out some official word on the matter, the IAXy does indeed have some echo cancellation built in. It has to since it interacts with a phone via a 2 wire to 4 wire conversion with a hybrid. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call deflection on ISDN PRI in Sweden
Hanna Wallin wrote:
Hello List!
We're having trouble making call deflection on ISDN PRI. We would like to
transfer a call to an external extension but keeping the callerid of the
caller so it can be presented to the receiver of the transferred call.
At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware
TE420B. We've ordered the service (CD) from the phone company.
The zapata.conf file inlcludes:
Transfer= yes
Facilityenable=yes
Callerid=asreceived
In extensions.conf we try to transfer a call to an external extension as:
Transfer(ZAP/g0/ ) but that fails with the ${TRANSFERSTATUS} =
UNSUPPORTED.
Ideas anyone? We would really appreciate it!
That supplementary service (CD) is not supported in libpri right now, so
that would be the reason why it doesn't work. The Transfer()
application is for analog lines, IIRC.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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Re: [asterisk-users] SPA-962+ SPA-932- blf function
On the Asterisk CLI show hints Registered Asterisk Dial Plan Hints =- [EMAIL PROTECTED]: SIP/211 State:IdleWatchers 1 - 1 hints registered On the Asterisk CLI sip show subcriptions Peer UserCall IDExtensionLast state TypeMailbox x.x.x.x 218 ad7e0925-24 [EMAIL PROTECTED] Idle dialog-info+xml none 1 active SIP subscription I do have real ip address for my asterisk server under the Peer column. This is the output I get on the Asterisk CLI , when I am in a call with extension 211 (SPA-941). So on my SPA-962 + SPA-932, the LED state remains GREEN, because Asterisk thinks it is in Idle state, which extension 211 is clearly not. Why is that? Best Regards, John Date: Fri, 28 Mar 2008 16:53:35 +1100 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SPA-962+ SPA-932- blf function We have BLF buttons working fine on the SPA932 side-car. What does show hints tell you under Asterisk, and what syntax did you use when configuring the side-car buttons? John Meksavan wrote: Asterisk Users, I am running Asterisk 1.4.11 on Debian Etch system with the TDM03B wildcard. I recently purchased a SPA-962 and SPA-932- the sidecar for our receptionist. After reading many forum postings on how to configure the side car, I uprgraded the SPA-962 software to 5.1.18(SC) version. I got the sidecar to subscribed to an extension on the Asterisk server, but the LED state on the SPA-932 never changes even whenI am a call with that extension on another VOIP phone- SPA-941. I got the speed dial function to work, but the blf function does not appear to work. Did anybody get the blf function to work? What I am doing wrong? Any input would be greatly appreciated. Thanks in advance. Regards, John How well do you know your celebrity gossip? Talk celebrity smackdowns here. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows Live Hotmail is giving away Zunes. http://www.windowslive-hotmail.com/ZuneADay/?locale=en-USocid=TXT_TAGLM_Mobile_Zune_V3___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones
On Fri, Mar 28, 2008 at 12:05 AM, Paul Hales [EMAIL PROTECTED] wrote: Can't you just use the same bootrom for all your polycom phones? PaulH On Fri, 2008-03-28 at 15:38 +1100, Lee, John (Sydney) wrote: I have a question about DHCP and boot server supporting more than 1 model of Polycom phones. According to Polycom standards, Polycom phone boots up to get a DHCP address and at the same time getting a boot server string (with username and password) to logon to boot server to download SIP, bootROM and etc. That is okay if there is only one type of phone (that requires a specific SIP and bootROM release). What about if the environment has to support two or more models of Polycom phones? On the boot server side, I can define another home directory like /home/polycom1 and /home/polycom2 to store different SIP and bootROM releases. However, the issue is how different polycom phone model can get a different user account and password to log on to different home directories. I understand the issue here is DHCP and not the boot server but I am a bit of a newbie here. Can anyone help please? As someone earlier pointed out, different models of polycom phones can be pointed to the same set of configuration files. With the standard ISC dhcpd server, the phones can be told where to look by using a directive like: option tftp-server-name ftp://polycom:[EMAIL PROTECTED]/; This would require a user account on the ftp server like: polycom:x:501:501:Polycom Phone Provisioning:/etc/asterisk/polycom/ftp/:/bin/bash and the configuration files would be placed in the /etc/asterisk/polycom/ftp/ directory. So if you wanted to have separate configurations for certain phones (for upgrade testing, etc., it is easily possible. SJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] recommendable softphones / X-Lite / Zoiper for amd64?
I may be missing something here... but won't a 32bit binary run just fine on a 64bit platform? Would you even see a performance increase or advantage to a 64bit soft phone versus a 32bit version? Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: zoa [EMAIL PROTECTED] To: asterisk users mailing list asterisk-users@lists.digium.com Sent: Friday, March 28, 2008 10:24:47 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] recommendable softphones / X-Lite / Zoiper for amd64? Contact me at [EMAIL PROTECTED] and ask for a beta for the 64 bit build of zoiper Cheers, Zoa martin f krafft wrote: Hi, I am on amd64 Linux and not really too happy with twinkle, linphone and ekiga. Unfortunately, X-Lite and Zoiper, even though they provide Linux versions (w00t!) have only x86 versions for download. Do you guys know of amd64 versions of those, or can you recommend other softphones that will run on amd64, or which come with source code? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-962+ SPA-932- blf function
On Fri, 2008-03-28 at 15:37 +, John Meksavan wrote: So on my SPA-962 + SPA-932, the LED state remains GREEN, because Asterisk thinks it is in Idle state, which extension 211 is clearly not. Why is that? Do you have the call-limit setting in sip.conf for SIP/211? At the Asterisk CLI, type sip show peer 211 and look for a line that looks like: Call limit : 0 If it happens to be set to zero (like in my example above), you don't have call limits enabled. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf setvar option
Ok,
Now I have a friday afternoon patch for the community.
In the branch
http://svn.digium.com/view/asterisk/team/oej/peer-chanvars/
there's an addition to the SIPPEER() dialplan function where you can
retrieve a setvar= channel variable defined in sip.conf for the peer.
The branch is based on 1.4 and the patch will soon be included in the
1.6 trunk.
This way, you can for example add a variable called CELLPHONE with
the peer's cell phone number. If dial(sip/olle) fails, I can now do
dial(zap/${SIPPEER(olle,chanvar[CELLPHONE])})
This is something I came up with a few weeks ago when I created a PBX
based on Asterisk for a company, something that I don't do much, since
I normally use Asterisk in carrier environments with SIP proxys. Doing
this little PBX project was a lot of fun and I learned a lot.
Have a nice weekend!
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Asterisk SIP Masterclass, Orlando Florida April 21-25 2008. Register
today!
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Re: [asterisk-users] recommendable softphones / X-Lite / Zoiper for amd64?
also sprach Tim Nelson [EMAIL PROTECTED] [2008.03.28.1637 +0100]: I may be missing something here... but won't a 32bit binary run just fine on a 64bit platform? Would you even see a performance increase or advantage to a 64bit soft phone versus a 32bit version? Not if all the libraries have been compiled for 64bit. Sure, I can run an entire 32bit system on 64bit hardware thanks to backward compatibility, but I actually run a 64bit machine with native 64bit code. And no, this is not for performance reasons and there wouldn't be any benefits. I just can't run 32bit software. -- martin | http://madduck.net/ | http://two.sentenc.es/ never trust an operating system for which you do not have the source. -- source unknown spamtraps: [EMAIL PROTECTED] digital_signature_gpg.asc Description: Digital signature (see http://martin-krafft.net/gpg/) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream BLF and Call-limit
I am trying to get BLF working on Grandstream phones with 1.2.27. I actually have it working, but I found a very strange issue and I am wondering if anybody knows what the problem is. Here is the scenario. If I have 3 phones, A, B and C. A monitors presence of B and C. Right now, if I call from B to C, B goes solid red and C flashes red, which is correct. If I add call-limit to the sip config for those phones, which the Grandstream docs show to do, and I then call from B to C. The presence for B never changes and C just goes solid red (even during ringing). The reverse holds true if I call from C to B. B shows solid red and C doesn't change from green. Any idea? If I remove call-limit on the sip.conf entries, it all goes back to working fine. I tried 2, 9 and 99 on the call-limit and they all have the same issues. I can't imagine why call-limit causes hints to stop updating correctly. Peder ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] overlap calls from NT-BRI timeout problem
Hi, Iam getting calls from an POTS system on an NT port. Multiport BRI card running bristuff 0.3. From time to time the recognized number is incomplete and dial failed. Is there any way to increase timeout waiting for called numbers? Because dialed numbers can be from 3 to 13 digits there is no way to regocnize the completeness of the number. Other option switch to en-block dialing is not possible because of bad documentation of the old POTS system. I hear the old telephone provider can validate numbers and so they can avoid such problems. I guess I do not get access to this POTS club information. best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on Dynamic Queue and Agent
Dear Asterisk-User friends, After realtime queues are defined, how does it work with the agents? There seems to be no db table for agents. If I can't define agents for the realtime queues in the db, how can agent login/logoff be done? Thanks alot for your help. Thanks, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-962+ SPA-932- blf function
Thanks for you guys help. The status LED on the sidecar takes an awfully look time to change from GREEN to RED and vice versa. Some times, it would reguire up to 15-20 minutes at beginning or ending the call on the extension. What would cause the delay? Is it my network? Best Regards, John Date: Fri, 28 Mar 2008 09:05:02 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SPA-962+ SPA-932- blf function John Meksavan wrote: Asterisk Users, I am running Asterisk 1.4.11 on Debian Etch system with the TDM03B wildcard. I recently purchased a SPA-962 and SPA-932- the sidecar for our receptionist. After reading many forum postings on how to configure the side car, I uprgraded the SPA-962 software to 5.1.18(SC) version. I got the sidecar to subscribed to an extension on the Asterisk server, but the LED state on the SPA-932 never changes even when I am a call with that extension on another VOIP phone- SPA-941. I got the speed dial function to work, but the blf function does not appear to work. Did anybody get the blf function to work? What I am doing wrong? Any input would be greatly appreciated. Thanks in advance. Regards, John How well do you know your celebrity gossip? Talk celebrity smackdowns here. http://originals.msn.com/thebigdebate?ocid=T002MSN03N0707A ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users To make it work properly I had to add the following to sip.conf: allowsubscribe=yes notifyringing=yes limitonpeer=yes notifyhold=yes See if that helps. -Sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ How well do you know your celebrity gossip? http://originals.msn.com/thebigdebate?ocid=T002MSN03N0707A___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on Dynamic Queue and Agent
mark morreny wrote: Dear Asterisk-User friends, After realtime queues are defined, how does it work with the agents? There seems to be no db table for agents. If I can't define agents for the realtime queues in the db, how can agent login/logoff be done? Thanks alot for your help. Thanks, Mark There is a table for dynamic realtime queue members, called queue_members by default. If you are using Asterisk 1.4, this table should have a column for the queue to which that member belongs, the interface on which the member receives calls, the queue member's name, the member's penalty, and a boolean column for determining if the member is paused. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail custom greeting
Hi, I have a wav file recording that i want to use on my voicemail, how can i set this up? thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] More info on my previous dynamic queue question
Hi, Sorry to resend the same question. This mail is just to add a few bits of details: When I tried to join the support queue, I get L RealTime: Retrieve SQL: SELECT * FROM queue_member_table WHERE interface LIKE '%' AND queue_name = 'Support' ORDER BY interface [Mar 29 10:01:52] WARNING[6203]: app_queue.c:3939 queue_exec: Unable to join queue 'Support' In show queue. it looks like the queue is set up fine: *CLI show queue Support has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: 102 (realtime) (Invalid) has taken no calls yet 101 (realtime) (Invalid) has taken no calls yet No Callers Comp-Sales has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Agent/1001 (Unavailable) has taken no calls yet Agent/1002 (Unavailable) has taken no calls yet Agent/1003 (Unavailable) has taken no calls yet No Callers What is the problem? Is this due to the (invalide) status? I did not do the AgentCallbackLogin cuz I don't know how to get it to work with realtime queue ( there is no realtime agent ). Could anyone please help me out? Your help will be greatly appreciated. Thanks, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on Dynamic Queue and Agent
Dear Mark, I did also populate members to the queue_member_table. The output of show queue also tells me that Asterisk read the member info too. When I tried to access the queue, it saidUnable to join queue 'Support' What do you think may have gone wrong? Also, how would I be able to add a login/logoff function for the members in the queue? I could not get agentcallbacklogin to work with realtime queue. Does it work? Thank you so much for your help. Thanks, Mark On Sat, Mar 29, 2008 at 1:49 AM, Mark Michelson [EMAIL PROTECTED] wrote: mark morreny wrote: Dear Asterisk-User friends, After realtime queues are defined, how does it work with the agents? There seems to be no db table for agents. If I can't define agents for the realtime queues in the db, how can agent login/logoff be done? Thanks alot for your help. Thanks, Mark There is a table for dynamic realtime queue members, called queue_members by default. If you are using Asterisk 1.4, this table should have a column for the queue to which that member belongs, the interface on which the member receives calls, the queue member's name, the member's penalty, and a boolean column for determining if the member is paused. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on Dynamic Queue and Agent
Dear Mark, Here is my queue_member table, is this how it should look? mysql SELECT * FROM queue_member_table WHERE interface LIKE '%' AND queue_name = 'Support' ORDER BY interface - ; +--+++---+-++ | uniqueid | membername | queue_name | interface | penalty | paused | +--+++---+-++ |3 | 101| Support| Agent/101 |NULL | NULL | |4 | 102| Support| Agent/102 |NULL | NULL | +--+++---+-++ Many thanks, Mark On Sat, Mar 29, 2008 at 2:19 AM, mark morreny [EMAIL PROTECTED] wrote: Dear Mark, I did also populate members to the queue_member_table. The output of show queue also tells me that Asterisk read the member info too. When I tried to access the queue, it saidUnable to join queue 'Support' What do you think may have gone wrong? Also, how would I be able to add a login/logoff function for the members in the queue? I could not get agentcallbacklogin to work with realtime queue. Does it work? Thank you so much for your help. Thanks, Mark On Sat, Mar 29, 2008 at 1:49 AM, Mark Michelson [EMAIL PROTECTED] wrote: mark morreny wrote: Dear Asterisk-User friends, After realtime queues are defined, how does it work with the agents? There seems to be no db table for agents. If I can't define agents for the realtime queues in the db, how can agent login/logoff be done? Thanks alot for your help. Thanks, Mark There is a table for dynamic realtime queue members, called queue_members by default. If you are using Asterisk 1.4, this table should have a column for the queue to which that member belongs, the interface on which the member receives calls, the queue member's name, the member's penalty, and a boolean column for determining if the member is paused. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Star Wars Echo Sound
Actually, its just the opposite... The call is okay for a few seconds, then the odd echo kicks in. When the training isn't turned on, it takes 20 seconds to so to kick the echo. With the training on, it works great except for this bug. Several of the people using the same * system but different phone stations are not seeing this problem. I saw someone else believed it was a softphone issue. Is it possible that its not a sangoma problem, but rather a polycom 501 issue? I just want to start putting the grind to the correct people. Rob Chris Earle wrote: I wanna say that's the echotraining taking effect. What it does is try to cause some echo so it can dynamically reconfigure the levels on the fly -- right at the start of the call. I know this happens with digium cards -- not sure if the Sangoma cards behave the exact same way. It's only at the start of the call right? once that occurs, the EC is kicked in and everything is fine? -- Chris Earle System Solutions Specialist, Network Technologies Division CBL Data Recovery w: http://www.cbltech.com Rob Schall [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] We have a location that is having a really odd issue. We have a sangoma POTs card. We are running software echo cancellation with the card (through asterisk) to try to eliminate some major echoing problems. I've turned on both EC and echotrain, which seemed to have gotten rid of the echo for the most part. However, we are now running into an issue where the outside caller hears a star wars type of sound. I expierenced this myself when talking to them. By this, I mean you hear a few words from them, then a few seconds lagging behind, you'll hear a muffled (darth vader) version of the same thing. Has anyone seen this? Thanks, Rob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones
Paul Hales wrote: Can't you just use the same bootrom for all your polycom phones? To elaborate in case it isn't obvious from above: Even if you needed different config files or even SIP applications by phone, you don't have to go to separate DHCP entries by phone. The MACADDESS.cfg file points to everything *except* the bootrom, so as long as you can share the same bootrom for all your phones, you can set one particular phone to use the new latest-and-greatest SIP application and related config files by just changing it's particular MAC.cfg file. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7971
Matthew Gibson wrote: What are you trying to do? I run a 7970 here with SIP. Get it to work ;) I can get the phone to register but something via way of NAT (I'm not using it) is getting in the way. I was hoping to find an example SEPxxx.xml file from someone using the 7971. Firmware is 8.3.3 -- J. Oquendo SGFA #579 (FW+VPN v4.1) SGFE #574 (FW+VPN v4.1) wget -qO - www.infiltrated.net/sig|perl http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x3AC173DB smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 650
No, you can keep dialing and make your call if you wish, or you can answer the call. -- Scott Plante, CTO Insight Systems, Inc. (+1) 404 873 0058 x104 [EMAIL PROTECTED] http://zyross.com Brent Torrenga wrote: List, Question about the Polycom 650: when dialing the digits for a phone number, and an incoming call comes in, does the phone prevent you from completing your outgoing call until the phone stops ringing, like a Cisco 79X0 does? --Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to register IAX user without password for any user
Dear Sanjay,
Sorry sanjay i miss to explain completely. My PC2PC mean is
Dialer2Dialer i want to allow call between Dialer with out any
registry and authentication through IAX.
so i need to setup Asterisk accept calls from any user and users can
call to each other without any password and registration.
please help how can i configure Asterisk using IAX in this regards.
thanks,
Asif
Message: 9
Date: Fri, 28 Mar 2008 20:54:51 +0530 (IST)
From: [EMAIL PROTECTED]
Subject: Re: [asterisk-users] how to register IAX user without
password
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=utf-8
Create a User and a Peer on both the machines for each other.
e.g IAX.conf on PCa
[pca2pcb]
type=peer
host=[IP OF pcb]
username=pca2pcb
serect=pca2pcb12345
qualify=yes
[pcb2pca]
type=user
context=default
auth=md5
secret=pcb2pca12345
deny=0.0.0.0/0.0.0.0
permit=[IP of pcb]
qualify=yes
ON PCb do the reverse in iax.conf
[pcb2pca]
type=peer
host=[IP OF pca]
username=pcb2pca
serect=pcb2pca12345
qualify=yes
[pca2pcb]
type=user
context=default
auth=md5
secret=pca2pcb12345
deny=0.0.0.0/0.0.0.0
permit=[IP of pca]
qualify=yes
NOW in Your extensions.conf you can use as
On PCa
exten=_.,1,Dial(IAX2/pca2pcb/${EXTEN})
exten=_y.,1,Dial(IAX2/pca2pcb/${EXTEN})
exten=_a.,1,Dial(IAX2/pca2pcb/${EXTEN})
and on PCb
exten=_.,1,Dial(IAX2/pcb2pca/${EXTEN})
exten=_y.,1,Dial(IAX2/pcb2pca/${EXTEN})
exten=_a.,1,Dial(IAX2/pcb2pca/${EXTEN})
Let me know if this works.
Regards,
Sanjay.
hi,
i want to call PC2PC between to IAX client without authentication i
want to allow every user to use PC2PC no any password required. Please
let me know what i have need to do in IAX.conf or any other file to
allow any user to call Pc2Pc.
My IAX.conf
[guest]
type=user
context=default
callerid=Guest IAX User
My extensions.conf
[default]
exten=_.,1,Dial(IAX2/${EXTEN})
exten=_y.,1,Dial(IAX2/${EXTEN})
exten=_a.,1,Dial(IAX2/${EXTEN})
below is my Asterisk console logs which i see after making call.
Mar 28 03:25:43 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:6910 socket_read: Rejected
connect attempt from 203.99.57.80, who was trying to reach
'jaffaradvcommnet@'
Mar 28 03:26:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'j' (from 203.99.57.80)
advcomm6*CLI iax2 show channels
Channel Peer UsernameID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None)203.99.57.80 (None) 4/15232
1/1 0ms -0001ms ms unknow
(None)203.99.57.80 jaffaradvc 5/15233
4/4 0ms -0001ms ms unknow
(None)203.99.57.80 (None) 6/18423
1/1 0ms -0001ms ms unknow
3 active IAX channels
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:27:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'jaffaradvcommnet' (from 203.99.57.80)
i am very thankful if some one help me in this regards,
i am getting Registration Refused error when i debug on console.
please tell me how can i registration every user without any username
and password and these user can make calls between each other.
i am very thankful if any body help me in this regards,
advcomm6*CLIiax2 debug
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 3ms SCall: 09398 DCall: 0 [203.99.57.80:47641]
USERNAME: aliadvcommnet
REFRESH : 60
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ
Timestamp: 2ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
CAUSE : Registration Refused
CAUSE CODE : 29
regards,
Asif
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Re: [asterisk-users] Newbie Polycom: DND answered as on the phone instead of not available
There is a sip.cfg entry divert.dnd.x.contact that is supposed to be where the call goes if DND is enabled. You could presumably set that to * plus the extention to go to the extension's voicemail, or to some other dialplan to play whatever you want, though I haven't tried it. Lee, John (Sydney) wrote: I am using Polycom IP600 phone. If I call a phone which has DND (do not disturb) enabled, the message to the caller will be The person on extension ... is on the phone, please leave a message Is there a way to pick the person ... not available message instead? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: DND answered as on the phone instead of not available
What is your extensions.conf setup? that has alot to do with it (I strongly suggest you use macros.) What SIP NNN code does the phone return when DND? On Mon, Mar 17, 2008 at 2:00 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: I am using Polycom IP600 phone. If I call a phone which has DND (do not disturb) enabled, the message to the caller will be The person on extension ... is on the phone, please leave a message Is there a way to pick the person ... not available message instead? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call deflection on ISDN PRI in Sweden
*CLI show application Transfer
-= Info about application 'Transfer' =-
[Synopsis]
Transfer caller to remote extension
[Description]
Transfer([Tech/]dest[|options]): Requests the remote caller be transferred
to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only
an incoming call with the same channel technology will be transfered.
Note that for SIP, if you transfer before call is setup, a 302 redirect
SIP message will be returned to the caller.
The result of the application will be reported in the TRANSFERSTATUS
channel variable:
SUCCESS Transfer succeeded
FAILURE Transfer failed
*** UNSUPPORTED Transfer unsupported by channel driver ***
So what you need to do is use app_dial instead of app_transfer.
Everything else should be able to remain the same.
On Fri, Mar 28, 2008 at 4:25 AM, Hanna Wallin
[EMAIL PROTECTED] wrote:
Hello List!
We're having trouble making call deflection on ISDN PRI. We would like to
transfer a call to an external extension but keeping the callerid of the
caller so it can be presented to the receiver of the transferred call.
At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware
TE420B. We've ordered the service (CD) from the phone company.
The zapata.conf file inlcludes:
Transfer= yes
Facilityenable=yes
Callerid=asreceived
In extensions.conf we try to transfer a call to an external extension as:
Transfer(ZAP/g0/ ) but that fails with the ${TRANSFERSTATUS} =
UNSUPPORTED.
Ideas anyone? We would really appreciate it!
Kind regards,
Hanna
Hanna Wallin
System Development
Direct: +46 (0)8 736 77 29
Mobile: +46 (0)73 414 13 38
Fax: +46 (0)8 736 77 91
E-mail: [EMAIL PROTECTED]
PocketMobile Communications AB
Wenner-Gren Center
Sveavägen 168, 3 tr
113 46 Stockholm
Nordic web page: www.pocketmobile.se
International web page: www.pocketmobileworld.com
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Re: [asterisk-users] Star Wars Echo Sound
If the problem is specific to certian inspections I would verify the LAN segments involved in connecting those devices. Rob Schall wrote: Actually, its just the opposite... The call is okay for a few seconds, then the odd echo kicks in. When the training isn't turned on, it takes 20 seconds to so to kick the echo. With the training on, it works great except for this bug. Several of the people using the same * system but different phone stations are not seeing this problem. I saw someone else believed it was a softphone issue. Is it possible that its not a sangoma problem, but rather a polycom 501 issue? I just want to start putting the grind to the correct people. Rob Chris Earle wrote: I wanna say that's the echotraining taking effect. What it does is try to cause some echo so it can dynamically reconfigure the levels on the fly -- right at the start of the call. I know this happens with digium cards -- not sure if the Sangoma cards behave the exact same way. It's only at the start of the call right? once that occurs, the EC is kicked in and everything is fine? -- Chris Earle System Solutions Specialist, Network Technologies Division CBL Data Recovery w: http://www.cbltech.com Rob Schall [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] We have a location that is having a really odd issue. We have a sangoma POTs card. We are running software echo cancellation with the card (through asterisk) to try to eliminate some major echoing problems. I've turned on both EC and echotrain, which seemed to have gotten rid of the echo for the most part. However, we are now running into an issue where the outside caller hears a star wars type of sound. I expierenced this myself when talking to them. By this, I mean you hear a few words from them, then a few seconds lagging behind, you'll hear a muffled (darth vader) version of the same thing. Has anyone seen this? Thanks, Rob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have any kind of day you want, Anthony Francis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two phones fail to agree on codec, asterisk at fault?
With canreinvite=no you are forcing asterisk to remain in the call path. As long as Asterisk is in the call path, it is supposed to be transcoding the calls, so it doesn't care what the compatible codecs are between then endpoints. Each leg of the call is phone-asterisk so asterisk negotiates a compatible codec set with each phone. If there is a codec difference between two legs of a call, it should be transcoding between them, unless you have that disabled somehow. (A quick google and I don't see how to disable transcoding apart from limiting codecs.) Now the other issue here is why Asterisk is offering GSM to the softphone and g726 to the C450IP. Try setting the allow and disallow settings for each channel rather than in Global. I tend to set things like codecs on a per-device basis rather than in global. Global settings have a bad habit of being overridden. Good luck, Brent martin f krafft wrote: Hi list, I am faced by a situation where I am trying to make a softphone and a Siemens C450IP talk to each other. Both are hooked up directly to the same asterisk, in the same IP net. - a softphone runs on 192.168.14.3 - the C450IP is at 192.168.14.30 - asterisk runs on the machine known as 192.168.14.1 I am running Asterisk 1.4.11, backported to Debian Etch by Xorcom. If I set canreinvite=yes for both, everything works. However, I have reason to use canreinvite=no for both. But if I do, then the two phones fail to agree on a codec. So calls are going via an asterisk bridge and the symptoms of my problem are: 1 if C450IP calls softphone, they can talk fine 2 if softphone calls C450IP, voice only goes from C450IP to softphone, not the other way around. I traced this down to the session description protocol, where there is funky stuff going on with the supported codecs each peer announces. Remember, asterisk is between them, and I set disallow=all,allow=ulaw,allow=alaw in [global]. So in situation 1, when the C450IP calls the softphone, these codecs are announced. 0 is ulaw, 8 is alaw, 111 is g726-32, 3 is gsm. C450IP to asterisk: 8, 0 asterisk to softph: 8, 3, 0 softph to asterisk: 8 asterisk to C450IP: 8, 0 They both agree on 8 (alaw) and stuff is working, but it's already curious how asterisk adds the 3 (GSM) in the second line and the 0 (ulaw) in the last. In situation 2, no voice travels from the softphone to the C450IP, and this is the dialog: softph to asterisk: 8, 0, 3 asterisk to C450IP: 0, 8, 111 C450IP to asterisk: 0 asterisk to softph: 3, 0, 8 Again, notice how asterisk basically ignores what it was asked to relay. In the end, the softphone settles for 3 (GSM) but the C450IP chooses 0 (ulaw). Since the softphone has no problem decoding ulaw, it can hear whatever the C450IP transmits, but it returns GSM packets, which the C450IP can't decode, and therefore nothing comes out of that phone. What's going on here? From all I can tell, the clients do the right thing, each selecting the first codec offered by asterisk (which they support), but asterisk is going a bit lala here, isn't it? First of all, why does it even bother with 3 and 111, given how I disallowed them? And second, why does it *dare* to announce more than what is available to the peer to which it relays? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Star Wars Echo Sound
They are all connected directly to the same switch which asterisk also connects into. Its a small office (6 people). Rob Anthony Francis wrote: If the problem is specific to certian inspections I would verify the LAN segments involved in connecting those devices. Rob Schall wrote: Actually, its just the opposite... The call is okay for a few seconds, then the odd echo kicks in. When the training isn't turned on, it takes 20 seconds to so to kick the echo. With the training on, it works great except for this bug. Several of the people using the same * system but different phone stations are not seeing this problem. I saw someone else believed it was a softphone issue. Is it possible that its not a sangoma problem, but rather a polycom 501 issue? I just want to start putting the grind to the correct people. Rob Chris Earle wrote: I wanna say that's the echotraining taking effect. What it does is try to cause some echo so it can dynamically reconfigure the levels on the fly -- right at the start of the call. I know this happens with digium cards -- not sure if the Sangoma cards behave the exact same way. It's only at the start of the call right? once that occurs, the EC is kicked in and everything is fine? -- Chris Earle System Solutions Specialist, Network Technologies Division CBL Data Recovery w: http://www.cbltech.com Rob Schall [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] We have a location that is having a really odd issue. We have a sangoma POTs card. We are running software echo cancellation with the card (through asterisk) to try to eliminate some major echoing problems. I've turned on both EC and echotrain, which seemed to have gotten rid of the echo for the most part. However, we are now running into an issue where the outside caller hears a star wars type of sound. I expierenced this myself when talking to them. By this, I mean you hear a few words from them, then a few seconds lagging behind, you'll hear a muffled (darth vader) version of the same thing. Has anyone seen this? Thanks, Rob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New Tutorial: Asterisk on EPIA VIA C3
Hello list, after spending the best part of an afternoon trying to build Asterisk on an old EPIA VIA C3, I thought that writing a tutorial would make life easier for future compilers: http://astrecipes.net/index.php?n=356 I had never compiled Asterisk for a different architecture, and I'm pretty disappointed at how complex it is - building Zaptel, Libpri and Asterisk requires discovering three different procedures, and even passing the required architecture to the autoconfig module was not enough for a clean build - libpthread and libresolv would not link, so you have to add them manually. Aybody got an idea of who should be notified of this immediate problem, apart for the time-wasteful general compilation procedure? Thanks l. -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Application registration on Asterisk 1.4 and 1.6?
Hi Kevin, I need I little bit of help again. I have installed in my PC for testing the last available version of asterisk for testings. And I am using easyeclipse with cdt plugin to create a C project and compile the app_skel.c source file from the asterisk-1.4.18.1. (GCC 4.1.3) I noticed some compilation problems: - load_module and unload_module can't be static. - AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, Skeleton (sample) Application) at the bottom need to be commented. error: ‘AST_MODULE’ doesn't be declared here (not a function). This is rare, because if do Ctrl + Click over AST_MODULE_INFO_STANDARD, the eclipse brings me to module.h file where is the #define clause. Making the necessary changes it compile but when I try to load the module in the CLI it returns this message [Mar 28 18:03:47] WARNING[18665]: loader.c:376 load_dynamic_module: Module 'app_skel.so' did not register itself during load [Mar 28 18:03:47] WARNING[18665]: loader.c:649 load_resource: Module 'app_skel.so' could not be loaded. Any Ideas? best regards. Mensaje original De: Kevin P. Fleming [EMAIL PROTECTED] Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Enviado: martes 25 de marzo de 2008, 13:59:13 Asunto: Re: [asterisk-users] Application registration on Asterisk 1.4 and 1.6? jonas boering wrote: Hi Kevin I've just arrived from my holidays I have reviewed my emails and saw that for some reason most part of my last message appears to be cut off. Continuing with the previous discussion, can you provide an example skeleton code of how the new registration way works on asterisk 1.4 and 1.6? Of course... it's already present in the tree, amazingly it is even called 'app_skel.c'. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tarjeta de crédito Yahoo! de Banco Supervielle. Solicitá tu nueva Tarjeta de crédito. De tu PC directo a tu casa. www.tuprimeratarjeta.com.ar ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail custom greeting
I have a wav file recording that i want to use on my voicemail, how can i set this up? You could play that file before sending the person to your voicemail and pass the s option to it Type show application voicemail on asterisk CLI to see the options. hth ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.19-rc4 and 1.6.0-beta7 Now Available
The Asterisk.org development team has released Asterisk versions 1.4.19-rc4 and 1.6.0-beta7. These releases contain significant bug fixes over the previous pre-releases of 1.4.19 and 1.6.0. We would like to thank everyone for all of the help with pre-release testing. Unless anything new comes up, 1.4.19 will be released at the beginning of next week. Both releases are available for download from http://downloads.digium.com/. Thank you for your support! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail custom greeting
Mark Quitoriano wrote: Hi, I have a wav file recording that i want to use on my voicemail, how can i set this up? thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You could save it to your asterisk voicemail directory, which is often something like: /var/spool/asterisk/voicemail/your_context/your_voicemailbox_number The files used are unavail.*, busy.*, and greet.* -- Asterisk will choose the easiest-to-deal-with sound format when playing the files, so that's why there's threeish of each (WAV, wav, and gsm on my box). In my experience, I just delete the two extra ones and asterisk just makes-do with what it's got :) Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two phones fail to agree on codec, asterisk at fault?
martin f krafft wrote: What's going on here? From all I can tell, the clients do the right thing, each selecting the first codec offered by asterisk (which they support), but asterisk is going a bit lala here, isn't it I think Brent's on to it there -- as he suggested, get your allow= and disallow= statements in each [peer], rather than in [global] ;) Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-962+ SPA-932- blf function
Have you set a call limit for each SIP peer? This is now required as of version 1.4. It took me a while to figure out all the issues when migrating to 1.4. John Meksavan wrote: Thanks for you guys help. The status LED on the sidecar takes an awfully look time to change from GREEN to RED and vice versa. Some times, it would reguire up to 15-20 minutes at beginning or ending the call on the extension. What would cause the delay? Is it my network? Best Regards, John Date: Fri, 28 Mar 2008 09:05:02 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SPA-962+ SPA-932- blf function John Meksavan wrote: Asterisk Users, I am running Asterisk 1.4.11 on Debian Etch system with the TDM03B wildcard. I recently purchased a SPA-962 and SPA-932- the sidecar for our receptionist. After reading many forum postings on how to configure the side car, I uprgraded the SPA-962 software to 5.1.18(SC) version. I got the sidecar to subscribed to an extension on the Asterisk server, but the LED state on the SPA-932 never changes even when I am a call with that extension on another VOIP phone- SPA-941. I got the speed dial function to work, but the blf function does not appear to work. Did anybody get the blf function to work? What I am doing wrong? Any input would be greatly appreciated. Thanks in advance. Regards, John How well do you know your celebrity gossip? Talk celebrity smackdowns here. http://originals.msn.com/thebigdebate?ocid=T002MSN03N0707A ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users To make it work properly I had to add the following to sip.conf: allowsubscribe=yes notifyringing=yes limitonpeer=yes notifyhold=yes See if that helps. -Sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users How well do you know your celebrity gossip? Talk celebrity smackdowns here. http://originals.msn.com/thebigdebate?ocid=T002MSN03N0707A ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Finding iaxy's (iaxies?)
According to http://kb.digium.com/entry/12/ The Iaxy will respond to pings on port . You can ping your broadcast IP on your network and listen with tcpdump on your network on port which will show the Iaxy responding and what IP address it is coming from. Ex. ping 192.168.1.255 tcpdump -i eth0 udp port Before I get my karma whacked again, does this work for anybody? 1) Shouldn't ping 192.168.1.255 be ping -b 192.168.1.255 2) Aren't pings ICMP and thus invisible when tcpdump is looking for UDP? 3) How do you set a port on an ICMP ping? 4) How do YOU find an Iaxy on your network? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail custom greeting
On Sat, Mar 29, 2008 at 7:26 AM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: You could save it to your asterisk voicemail directory, which is often something like: /var/spool/asterisk/voicemail/your_context/your_voicemailbox_number The files used are unavail.*, busy.*, and greet.* -- Asterisk will choose the easiest-to-deal-with sound format when playing the files, so that's why there's threeish of each (WAV, wav, and gsm on my box). In my experience, I just delete the two extra ones and asterisk just makes-do with what it's got :) i can't see any unavail.* or busy.* wav or gsm files. can i just create one and put it there as unavail. and busy. ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
