Re: [asterisk-users] What is the best softphone work with Asterisk
Hi Michael; You tried iaxcomm pro as JIM is complainning from the crashs. PLease advise. Regards Bilal - iaxcomm pro?? On Tue, 24 Jul 2007 19:40:45 -0400, Jim Archer [EMAIL PROTECTED] wrote: I tried several and had very poor luck with each I tried. These included IaxComm, IaxComm Pro, Diax and Firefly II. Also, One other one from I think Germany that had just changed it's name. All of these had issues. I could not get Firefly configured at all to talk to Asterisk. Diax, when the user places a call, just keeps ringing even when the person answered. Both IaxComms would crash. I'm sure there is one out there but I have not found it, although I have not yet tried the SIP soft phones. --On Tuesday, July 24, 2007 2:09 PM -0700 bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. http://new.toolbar.yahoo.com/toolbar/features/norton/index.php ___ --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] What is the best softphone work with Asterisk
Idefisk is now renamed to zoiper . http://www.zoiper.com/ :) On 26/07/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Mittwoch, den 25.07.2007, 12:13 -0700 schrieb bilal ghayyad: Hi BaharatSamaria; Thanks for your kindly email. Are (Xlite and phoner) IAX or SIP? From where I can download them (Xlite and phoner)? I googled for xlite. One of the first matches was a wiki page on voip-info.org, which in turn linked me to the X-Lite manufacturer's homepage. quote CounterPath's X-Lite 3.0 is the market's leading free SIP based softphone available for download. /quote. The first link in the google search list for phoner immediately led me to the phoner homepage, quote - VoIP support for SIP connections Phoner is freeware, so this program can be used and distributed without any restrictions. Distribution has to be free of charge. /quote I think you will have no trouble to find the URIs yourself, probably within about 30 seconds. In doubt you might consult http://www.googleguide.com/ to learn about google. Anselm ___ --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] What is the best softphone work with Asterisk
Am Mittwoch, den 25.07.2007, 12:13 -0700 schrieb bilal ghayyad: Hi BaharatSamaria; Thanks for your kindly email. Are (Xlite and phoner) IAX or SIP? From where I can download them (Xlite and phoner)? I googled for xlite. One of the first matches was a wiki page on voip-info.org, which in turn linked me to the X-Lite manufacturer's homepage. quote CounterPath's X-Lite 3.0 is the market's leading free SIP based softphone available for download. /quote. The first link in the google search list for phoner immediately led me to the phoner homepage, quote - VoIP support for SIP connections Phoner is freeware, so this program can be used and distributed without any restrictions. Distribution has to be free of charge. /quote I think you will have no trouble to find the URIs yourself, probably within about 30 seconds. In doubt you might consult http://www.googleguide.com/ to learn about google. Anselm ___ --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] tdm400p fxs module busy
Dear All
The setup is te110p with an 8 channels PRI to make and receive all calls.
SIP phones throughout the company.
TDM400p with 4 FXS modules to send/receive faxes and make credit card
transactions.
I have an analogue phone on the tdm400p for testing.
I can receive calls to the exten. There is a dialing tone.
However, when I try to make a call I get a busy signal.
Asterisk stated busy then hungup zap/32-1
why wont asterisk supply a resource from the te110p pri card for use by the
tdm400p FXS (fxo signalling)?
configs below:
[EMAIL PROTECTED] etc]# more 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: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS RED
span = 1,0,0,ccs,hdb3,crc4
# termtype: te
bchan=1-8
dchan=16
# Span 2: WCTDM/0 Wildcard TDM400P REV H Board 1
fxoks=32
fxoks=33
fxoks=34
fxoks=35
# Global data
loadzone= uk
defaultzone = uk
[EMAIL PROTECTED] asterisk]# more zapata.conf
[trunkgroups]
[channels]
language=en
internationalprefix = 00
nationalprefix = 0
context=from-pstn
switchtype=euroisdn
pridialplan=local
priindication=outofband
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
group=1
callgroup=0
pickupgroup=0
immediate=no
echotraining=yes
echocancel=yes
echocancelwhenbridged=no
facilityenable=yes
musiconhold=default
overlapdial=yes
immediate=no
txgain=0.0
rxgain=0.0
signalling = pri_cpe
channel = 1-8
faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
signalling = fxo_ks
echocancel=yes
pulsedial=yes
channel=32-35
[EMAIL PROTECTED] asterisk]# more extensions.conf
[general]
static=yes
writeprotect=yes
;
[globals]
OUTBOUND = Zap/g1
FAX1 = Zap/32
FAX2 = Zap/33
STREAMLINE1 = Zap/34
STREAMLINE2 = Zap/35
CUSTSERVE1 = SIP/401 ;Teresa
CUSTSERVE2 = SIP/402 ; Louise
;CUSTSERVE3 = SIP/404 ; Helen
QUAD1 = SIP/451 ; Matt
QUAD2 = SIP/452 ; Johan
CUSTSERVE = CUSTSERVE1CUSTSERVE1
;
FSEXT1 = SIP/400 ; Angela
;FSEXT2 = SIP/403 ; Nigel
FSEXT3 = SIP/410 ; Matt
;
;ELLIS = SIP/411
;QUEENS = SIP/412
;FSSHOPS = ELLISQUEENS
;
QUAD = SIP/450
;
LONDONSOLE1 = SIP/421 ; Zoe
;LONDONSOLE2 = SIP/422 ; Laura
;LONDONSOLE = LONDONSOLE1LONDONSOLE2
;
;PRESS1 = SIP/431 ; Lucy
;PRESS2 = SIP/432 ; Gemma
;PRESSOFFICE = PRESS1PRESS2
;
[macro-oneline]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
;
[macro-oneline1]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup
;
[macro-fax]
exten = s,1,Dial(${ARG1},20,t)
exten = s,3,Hangup
;
[default]
;setupdial out
include = from-pstn
;
;test dialplan
exten = _9xxx,1,SayDigits(${EXTEN:1})
;
;setup the phone extensions
exten = 400,1,Macro(oneline,${FSEXT1})
exten = 401,1,Macro(oneline,${CUSTSERVE1})
exten = 402,1,Macro(oneline,${CUSTSERVE2})
exten = 410,1,Macro(oneline,${FSEXT3})
exten = 421,1,Macro(oneline,${LONDONSOLE1})
exten = 450,1,Macro(oneline,${QUAD})
exten = 451,1,Macro(oneline,${QUAD1})
exten = 452,1,Macro(oneline,${QUAD2})
;
exten = 1000,1,Macro(oneline,${CUSTSERVE})
;exten = 2000,1,Macro(oneline,${FSSHOPS})
;exten = 3000,1,Macro(oneline,${PRESSOFFICE})
;
;record new voice files
Exten = 501,1,Wait(2)
Exten = 501,n,Record(/tmp/asterisk-recording:gsm)
Exten = 501,n,Wait(2)
Exten = 501,n,Playback(/tmp/asterisk-recording)
Exten = 501,n,wait(2)
Exten = 501,n,Hangup
;
;goto voicemail
exten=*98,1,VoiceMailMain([EMAIL PROTECTED])
;
[dialphone]
exten = 90,1,Macro(fax,${FAX1})
;
[from-pstn]
;this is linked to zapata.conf and defines where the ddi points
exten = 00,1,Dial(SIP/401SIP/402,15)
exten = 00,2,Voicemail(1000)
;
exten = 769611,1,Macro(oneline1,${FSEXT1})
exten = 769615,1,Macro(oneline1,${LONDONSOLE1})
;exten = 769616,1,Macro(oneline1,${LONDONSOLE2})
exten = 769636,1,Macro(oneline1,${FSEXT1},${401})
;exten = 769637,1,Macro(oneline1,${NIGEL})
;
exten = _9.,1,Set(CALLERID(number)=00)
exten = _9.,2,Dial(${OUTBOUND}/${EXTEN:1})
exten = _9.,3,Congestion()
exten = _9.,102,Congestion()
;
exten = 999,1,Dial,(${OUTBOUND}/999)
exten = ,1,Dial,(${OUTBOUND}/999)
;
exten = 90,1,Dial(Zap/32,15)
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Re: [asterisk-users] Query
On Thu, Jul 26, 2007 at 05:25:30PM +0530, [EMAIL PROTECTED] wrote: Hi, I am facing problem in configuring D-channel. I did the following configuration for TE-120P card /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf /etc/asterisk/zapata.conf Right? group=1 signalling=pri_cpe switchtype=euroisdn context=incoming channel=1-15,17-31 DIGIUM card is connected through cable to another end.On placing call from other end to asterisk PBX ( through DIGIUM card ) the following error messages is coming on console mode of asterisk Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. == Primary D-Channel on span 1 down Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! What is on the other side? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] What is the best softphone work with Asterisk
On 7/25/07, Jaswinder Singh wrote: Idefisk/zoiper softphone is for IAX2 and it works fine almost everytime . However there is more variety in sip softphones . I think zoiper is much better than other iax2 softphones . Feature wise you are quite right that Zoiper is pretty neat. But Time Bandit's (Marc Charrbonneau) MediaX phone has a tiny memory footprint, you don't even need to install it, just download the exe and execute it. It is quite stable and clear sounding. Just my 2c. -baji. -- ___ --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] Sangoma on Fedora 7 x86_64
Sangoma gives EXCELLENT technical support. I would suggest you try there first. The few problems I have had with installation were addressed promptly and when driver fixes proved necessary, corrected in short order. Also the cards have a 5 year warranty! John Novack Nhadie Ramos wrote: Hi, I'm trying to install asterisk(v1.2.22) with FreepBX(v2.2.3) with a 4-Port FXO Sangoma card A200. I'm using Fedora 7 (x86_64) kernel version 2.6.22.1-27.fc7, but i'm having these errors: $ ztcfg - 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) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) $ lspci -v 02:01.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card Subsystem: NEC Corporation Unknown device 1000 Flags: bus master, medium devsel, latency 64, IRQ 10 Memory at fdde (32-bit, non-prefetchable) [size=64K] Nothing else uses IRQ 10. An error when i installed wanpipe stable version 2.3.4-12, i also get the same error when i used wanpipe 3.1.2 WANPIPE DRIVER COMPILE LOG Thu Jul 26 21:26:33 PHT 2007 --- make -C /lib/modules/2.6.22.1-27.fc7/build SUBDIRS=/usr/local/src/wanpipe-2.3.4-12/kdrvtmp CC=gcc KBUILD_VERBOSE=0 modules make[1]: Entering directory `/usr/src/kernels/2.6.22.1-27.fc7-x86_64' CC [M] /usr/local/src/wanpipe-2.3.4-12/kdrvtmp/sdladrv_src.o In file included from /usr/local/src/wanpipe-2.3.4-12/kdrvtmp/sdladrv_src.c:135: include/linux/wanpipe_common.h: In function 'wan_skb_tail': include/linux/wanpipe_common.h:1017: warning: return makes pointer from integer without a cast include/linux/wanpipe_common.h: In function 'wan_skb_set_raw': include/linux/wanpipe_common.h:1281: error: 'struct sk_buff' has no member named 'mac' include/linux/wanpipe_common.h:1282: error: 'struct sk_buff' has no member named 'nh' include/linux/wanpipe_common.h: In function 'wan_skb_init': include/linux/wanpipe_common.h:1735: warning: assignment makes integer from pointer without a cast make[2]: *** [/usr/local/src/wanpipe-2.3.4-12/kdrvtmp/sdladrv_src.o] Error 1 make[1]: *** [_module_/usr/local/src/wanpipe-2.3.4-12/kdrvtmp] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.22.1-27.fc7-x86_64' make: *** [all] Error 2 That error i really don't understand. Has anybody tried to install Sangoma on Fedora 7? TIA Ronald ___ --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 -- Dog is my co-pilot ___ --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] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected
Do you have any extension in default context of your extensions.conf file to accept incoming calls ? It must be something like; exten = 12345678,1,Answer() exten = 12345678,2,Playback(Welcome) ... 12345678 = The DID number you are calling to reach E1 Idris -Original Message- From: Erick Perez [mailto:[EMAIL PROTECTED] Sent: Thursday, July 26, 2007 7:03 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected Hi, after many issues we finally managed to make our system do outgoing calls with perfect quality. However I cannot detect *any* form of incoming call. when I use an outside phone to call the E1 connected to the sangoma a102, I instantly get a fast busy tone. My /etc/zaptel.conf is: loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:1 bus:4 span: 1] span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 My /etc/asterisk/zapata.conf is: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no #include zapata-auto.conf Zapata-auto.conf has: callerid=asreceived ;Sangoma A102 port 1 [slot:1 bus:4 span: 1] switchtype=euroisdn context=from-pstn group=0 signalling=pri_cpe channel = 1-15,17-31 Note: According to the tech support in the local telco, my E1 should be: E1 PRI, CAS, HDB3, NCRC4, DSS1 However if I configure the card for CAS, it will never connect. My card is currently configured (and makes only outgoing calls) as: E1 PRI, CCS, HDB3,NCRC4 (i have no idea what dss1 is or where it goes) My /etc/wanpipe/wanpipe1.conf is: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 4 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= NCRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES thanks for your help. -- Erick Perez ___ --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] Queue stats
On Thu, 2007-07-26 at 11:06 -0500, Jay Moore wrote: So here is my question: In this format: 1|2|3|4|5|6, 1 - ? 2 - ? 3 - queue in question? 4 - agent answering the queue? 5 - queue event? 6 - queue event info? Is that correct? What are options 1 and 2? Times of some sort I'm guessing, but I'm not entirely sure. The first column is the time, in Unix epoch format (number of seconds since January 1, 1970). This allows you to tell *when* each event happened. The second column is the unique call id of the call in question. This tells you *what call* the event happened on. The third column is the queue name. The fourth column is the queue member or agent. The fifth is the queue event (as described in queuelog.txt). The sixth (and seventh, eighth, etc.) are the event info. Each type of queue event sets different kinds of event info, as described in queuelog.txt. Hopefully that helps clarify things! -- 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] SetCallerPres and Cisco PRI
PS. Check this out: http://bugs.digium.com/print_bug_page.php?bug_id=2471 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Attaching VoiceMails on E-Mails
Hello Marco, On 7/27/07, Marco Mouta [EMAIL PROTECTED] wrote: hi, The VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application uses */usr/sbin/sendmail* to mail voicemail messages to users. This can be any sendmail-compatible MTA. In practice you can use Sendmailhttp://sendmail.org/, Postfix http://postfix.org/, Exim http://www.exim.org/ or any other MTA http://www.voip-info.org/wiki/edit.php?page=Asterisk+voicemail+MTA. It is recommended to use the default one that comes with your distribution. If shall I say I'll use Exim4 here, what do I need to do then? I would say if you just create your own sendmail.sh and place it /usr/sbin/sendmail, asterisk will execute it by default, do not forget to give permissions for asterisk user to execute it. If I'll create this script, what will be its contents then? Currently, the /usr/sbin/sendmail is a symbolic link to /usr/sbin/exim4 for your information. Please advice. GNUbie ___ --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] Default Asterisk Numbers
Hello all, Where can I find the complete list of default Asterisk (telephone) numbers and maybe the other special numbers that are need to be preserve and not use for setting up own dial plan? Thank you. GNUbie ___ --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] Attaching VoiceMails on E-Mails
GNUbie wrote: Hello Tzafrir, On 7/27/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: You need to have a package that provides /usr/sbin/sendmail . While you can get away with using nullmailer or ssmtp (that don't spool mail locally), I would recommend you to install postfix or exim, so a temporary problem won't cause the message to get lost on the way. Oh, I mean, I have Exim installed here but it's not running by default. Should I run it anyway? Yes. You should be able to tell Exim to only listen on the 127.0.0.1 interface. I'm sure you can tell Exim to not listen on any interface, but don't ask me how 8-) You might want to tell Exim to send all e-mail thru your main SMTP server machine (the correct term in most MTA docs is smarthost. ___ --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] Attaching VoiceMails on E-Mails
Hello Tzafrir, On 7/27/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: You need to have a package that provides /usr/sbin/sendmail . While you can get away with using nullmailer or ssmtp (that don't spool mail locally), I would recommend you to install postfix or exim, so a temporary problem won't cause the message to get lost on the way. Oh, I mean, I have Exim installed here but it's not running by default. Should I run it anyway? Please advice. Thank you once again. GNUbie ___ --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] Attaching VoiceMails on E-Mails
On Fri, Jul 27, 2007 at 08:55:01AM +0800, GNUbie wrote: Hello all, I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to send the voicemails as attachment to e-mails and delete the voicemails from my PBX once it has been sent. But, I don't have a running MTA here even on the PBX itself. I just want to send the e-mails to my GMail account from my PBX. Can I just use the mail or mailx command to send the e-mail and attach the voicemail message? You need to have a package that provides /usr/sbin/sendmail . While you can get away with using nullmailer or ssmtp (that don't spool mail locally), I would recommend you to install postfix or exim, so a temporary problem won't cause the message to get lost on the way. Below are snippets of my voicemail.conf and extensions.conf configuration Generally the default voicemail.conf should do. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Attaching VoiceMails on E-Mails
hi,
The
VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use
Sendmailhttp://sendmail.org/,
Postfix http://postfix.org/, Exim http://www.exim.org/ or any
other MTAhttp://www.voip-info.org/wiki/edit.php?page=Asterisk+voicemail+MTA.
It is recommended to use the default one that comes with your distribution.
I would say if you just create your own sendmail.sh and place it
/usr/sbin/sendmail, asterisk will execute it by default, do not forget to
give permissions for asterisk user to execute it.
On 7/27/07, GNUbie [EMAIL PROTECTED] wrote:
Hello all,
I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to
send the voicemails as attachment to e-mails and delete the voicemails from
my PBX once it has been sent. But, I don't have a running MTA here even on
the PBX itself. I just want to send the e-mails to my GMail account from my
PBX. Can I just use the mail or mailx command to send the e-mail and attach
the voicemail message?
Below are snippets of my voicemail.conf and extensions.conf configuration
files. Please advice whatever you think I need to change with my current
configurations.
Thank you in advance.
GNUbie
- - - s n i p - - -
# cat /etc/asterisk/voicemail.conf
[general]
format=wav49
[EMAIL PROTECTED] ; bogus e-mail address
attach=yes
delete=yes
maxmsg=50
maxmessage=180
minmessage=5
maxgreet=60
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
fromstring=The PBX
usedirectory=yes
emaildateformat=%A, %B %d, %Y at %r
sendvoicemail=yes
emailbody=Hi, ${VM_NAME}!\n\nYou have a new voicemail message from
${VM_CALLERID} attached to this e-mail message.\n\nHave a nice day!\n\nThe
PBX
mailcmd=/usr/bin/exim -t ; not sure about this line
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
[default]
101 = 11011,GNUbie,[EMAIL PROTECTED]
# grep 10 /etc/asterisk/extensions.conf
exten = 101,1,Dial(Zap/1,20,rt)
exten = 101,2,VoiceMail(101,u)
exten = 100,1,VoiceMailMain(${CALLERID(num)},s)
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Re: [asterisk-users] Need help with inbound IAX
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 What do you get with: iax2 show registry - -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGqVvRDQNt8rg0Kp4RAj89AJ9XzL/EUCBMG3/qfDkhHMSJggoIWQCggAMu e4tnaTrjg6wRI8kIyOS0Qgo= =AbcT -END PGP 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
[asterisk-users] Attaching VoiceMails on E-Mails
Hello all,
I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to
send the voicemails as attachment to e-mails and delete the voicemails from
my PBX once it has been sent. But, I don't have a running MTA here even on
the PBX itself. I just want to send the e-mails to my GMail account from my
PBX. Can I just use the mail or mailx command to send the e-mail and attach
the voicemail message?
Below are snippets of my voicemail.conf and extensions.conf configuration
files. Please advice whatever you think I need to change with my current
configurations.
Thank you in advance.
GNUbie
- - - s n i p - - -
# cat /etc/asterisk/voicemail.conf
[general]
format=wav49
[EMAIL PROTECTED] ; bogus e-mail address
attach=yes
delete=yes
maxmsg=50
maxmessage=180
minmessage=5
maxgreet=60
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
fromstring=The PBX
usedirectory=yes
emaildateformat=%A, %B %d, %Y at %r
sendvoicemail=yes
emailbody=Hi, ${VM_NAME}!\n\nYou have a new voicemail message from
${VM_CALLERID} attached to this e-mail message.\n\nHave a nice day!\n\nThe
PBX
mailcmd=/usr/bin/exim -t ; not sure about this line
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
[default]
101 = 11011,GNUbie,[EMAIL PROTECTED]
# grep 10 /etc/asterisk/extensions.conf
exten = 101,1,Dial(Zap/1,20,rt)
exten = 101,2,VoiceMail(101,u)
exten = 100,1,VoiceMailMain(${CALLERID(num)},s)
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Re: [asterisk-users] Ring forever
FERNANDO VILLARROEL wrote: Hello list, i need help. My problem is that when I want to call (using ISDN phone or internal SIP client) via the Sip provider a real phone number (ISDN phone or internal SIP Asterisk SIP ), I get a ring tone. When I now decide to hang up (e.g. if nobody answers), the called telephone continues to ring almost forever. -- Call on SIP/nyphone-081a7768 left from hold -- SIP/nyphone-081a7768 answered SIP/2563105-0819cf80 -- Packet2Packet bridging SIP/2563105-0819cf80 and SIP/nyphone-081a7768 nyphone is answering your call and then dialing out to the destination. ___ --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] SetCallerPres and Cisco PRI
Hi Peder, You tried blanking the caller ID field and it didn't work? i.e., exten = ...,n,Set(CALLERID(all)=) It worked for me, although my media gateway was not a Cisco one. Whether SetCallerPres() will work depends entirely on what it accomplishes. Does it just alter the cosmetic From: line, and does the Cisco gateway take stock in that? Or does it tack on the draft privacy headers (Remote-Party-ID) and set privacy to on/full? My gut feeling is that SetCallerPres() applies to calls placed directly out of a PRI interface, not SIP, because presentation is a term typically applied to caller ID in an ISDN, not a SIP context. It is hard to tell whether this intuition is correct because SetCallerPres() is fundamentally implemented in apps/app_setcallerid.c which calls ast_set_callerid() in main/channel.c and appears to apply to a variety of channel types variously. If this doesn't work, try this: http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header A Cisco MGW should support that just fine. Good luck, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Queue Stats
Hello Jay, Sounds like quite a complicated set up. Most queue statistics packages will break your callers down depending on which queue they were actually answered in (or hung up on). If you want your stats listed as if the callers were in a single queue, you can sign up for a FREE OrderlyStats account at http://www.orderlyq.com/orderlystats.html - once you're all done, let us know and we'll show you how OrderlyStats can show these calls as if your three queues were just one. Hope this helps, Matt. Jay wrote: Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller falls into our 'music on hold' queue, a message is played and the caller hears our music on hold while the phones are rung again. 3) After 30 seconds, if the caller is still in our 'moh' queue, they drop out of queue and immediately re-enter the 'moh' queue again until the call is answered or the caller hangs up. How can I find out how many calls are answered out of each queue during certain times (1st shift, 2nd shift, etc...)? Also, I'm curious how I can track how many times a call repeats the 'moh' queue. 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
Re: [asterisk-users] Display IE
Is there more than one display IE in the original ISDN setup message coming from the Telco? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Patricio Sent: Thursday, July 26, 2007 5:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Display IE Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNECT comes from the PSTN containing a Display IE (which has info sent by the telco that is used for billing purposes by the PBX) into span 1 of the asterisk. (The telco is emulating an old billing procedure based on an impulse count. This was commonly used in analog lines.) 2. The asterisk relays the call to span 2 (which is connected to the PBX). 3. The CONNECT that is sent from span 2 to the PBX does not have the Display IE. The asterisk strips this IE from the CONNECT message. This is my problem. Is there a way that i can force the asterisk not to strip the Display IE? Thanks and best regards to all, Óscar Patrício Anthony Francis wrote: Damon Estep wrote: Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is probably being send before the display IE arrives. The display IE is used for CNAM delivery, and should not exceed 15 characters. It is very common to put a message in the display IE that indicates that the CNAM info will be sent in a subsequent Facility IE, and for that you must wait 1 second. If the ISDN setup actually contains information in the display IE, and that is not being captured as the CNAM (callerid(name)) you might need to capture he ISDN messaging to debug it, the telco can usually provide such a trace. I would bet that the display IE contains a information following message, and what you really want is in the facility IE that follows. Very common, as is the Wait(1) workaround. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Patricio Sent: Wednesday, July 25, 2007 6:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Display IE Hi! I have an Asterisk Box that has 2 E1 connections: one to the PSTN and one to a PBX. It is acting as a telephony gateway. I have a problem: the PSTN sends information in the Display IE (in setup, information , etc.messages) that the PBX needs por internal processing. The asterisk does not relay the ISDN frames coming from the PSTN. It regenerates them, and when it does, it ommits the Display IE. Is there a way that i can force the asterisk, not to ignore this IE in the ISDN messages? If anyone can shed some light on this issue, i would be very grateful. Thanks in advance, Best regards, Óscar Patrício ___ --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 Or trace it yourself with: pri intense debug span 1 Make sure you change the 1 to whatever span these calls are coming across. Anthony ___ --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] X100P pass through questions
That is correct. The X100P only detects voltage drop, not polarity reversal. Walter Willis wrote: i am have x100p and not work fine, no detect polarity, and much problems with asterisk 1.2 to up. :S On 7/25/07, Mike Wright [EMAIL PROTECTED] wrote: John Novack wrote: Mike Wright wrote: Just purchased a Motorola Wildcard X100P ... but the button pressed generates no tone; on button release dialtone returns. Sure sounds like polarity reversal. Indeed it was. Punch block in the basement had tip and ring reversed. Probably been that way for thirty years. Amazing that the dsl installer that replaced the inside wiring didn't catch it. For years I've had intermittent problems using IVR. Some phones would work, some would not. Now I know why. Thanks to John Novack and Eric ManxPower Wieling for your help. ___ --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 RTP keepalive packets causing Asterisk warning
Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where the phone did not send rtp keepalives when on mute (resulting in disconnect from tech support hold and concalls) A side effect seems to be that Asterisk pops the following warning on the console... Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP Read too short Grandstream say they are not sure what it is but it should not affect anything. In other words, Don't worry, be happy!. Any thoughts/experience on this? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.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
[asterisk-users] Digium FTP server will be replaced with HTTP server
Some time in the next two weeks, Digium will be shutting down our FTP server, located at ftp.digium.com, and begin using only the existing HTTP server on the same system instead. We have decided to only offer our public downloads over the HTTP protocol, not the FTP protocol, primarily for reasons related to our marketing department :-) The site will still be called ftp.digium.com, but will no longer respond to requests made via the FTP protocol; only the HTTP protocol will be supported. There should be no other user-visible changes when this change is made to the server. -- 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
Re: [asterisk-users] IAX connections broken
Are sites listed by IP or DN. If IP, dumb question but did it change? If DN, can you resolve it from the respective boxea? Dave Bour Desktop Solution Center 905.381.0077 [EMAIL PROTECTED] For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: [EMAIL PROTECTED] [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu Jul 26 10:17:23 2007 Subject: Re: [asterisk-users] IAX connections broken Not likely. #1, I have a public IP on that firewall. #2. If I block 4569 at our firewall, then it goes from closed to stealth. If I forward the port, it goes from stealth to closed. The iaxping tool (http://www.bpvn.com/asterisk/iaxping.zip) has no problems pinging the box from the lan, and our test machine can make an IAX connection to the box. From outside the network, however, it times out. It has to be a NAT problem, but forwarding doesn't appear to be working. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Baji Panchumarti Sent: Thursday, July 26, 2007 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX connections broken what if your internet provider is blocking inbound 4569 ? -- On 7/26/07, Michael Munger wrote: Dear All: I have several boxes that up and running just great, then we changed internet equipment due to a lightning strike, now all my inbound IAX connections (iax2 show peers) have unknown status. If I log into the remote boxes, it says Request sent. The authentications haven't changed at all, and all the iax.conf settings are correct. It looks like a firewall issue, but we've got 4569 TCP UDP forwarded to our Asterisk box. When I use Shields up from GRC.com to test the port, it is showing up as closed rather than open, which normally means the port is open, but the service is not running, yet Asterisk is up and running just fine, and my outbound connections to Voicepulse work fine. I see voicepulse, voicepulse sees me. There is something I am not seeing here. Any thoughts? -Michael ___ ___ --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] Queue stats
Jared Smith wrote: On Thu, 2007-07-26 at 09:37 -0500, Jay Moore wrote: My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. It sounds like you've got quite the queue setup (although I don't quite see why your calls jump out and back into the moh queue). All the of queue statistics you need should be available with careful parsing of the queue log (usually located in /var/log/asterisk/queue_log). You can also trigger custom queue log events from the dialplan by calling the QueueLog() application. In your case, you might want to add a custom queue log entry every time the caller rejoins the moh queue, saying something to the effect of Caller XYZ has rejoined the moh queue for the 10th time or something like that. We had some issues with the announcement message not playing reliably. My fix was to just have them drop out and re-enter the queue. It doesn't seem to have any adverse effects, but if you have any alternative suggestions, I'm more than willing to try them. I've checked my queue log (38megs, yikes) and looked at the queuelog.txt info file for how to parse the lines, but I still have a question. For example, a snippet of my log looks like (line numbers mine): 1) 1185460404|1185460400.334916|queue-ring|NONE|ENTERQUEUE||732 2) 1185460420|1185460400.334916|queue-ring|NONE|EXITWITHTIMEOUT|1 3) 1185460427|1185460400.334916|queue-answer|NONE|ENTERQUEUE||732 4) 1185460448|1185460400.334916|queue-answer|NONE|EXITWITHTIMEOUT|1 5) 1185460454|1185460400.334916|queue-answer|NONE|ENTERQUEUE||732 6) 1185460456|1185460400.334916|queue-answer|SIP/agent3-0a5bc480|CONNECT|2 7) 1185460496|1185460400.334916|queue-answer|SIP/agent3-0a5bc480|COMPLETECALLER|2|40 Here's how I interpret this: 1) Call comes into my ring queue 2) Call exits ring queue due to timeout 3) Call enters answer (moh) queue 4) Call exits answer queue due to timeout 5) Call enters answer queue again 6) Agent 3 picks up the call out of the queue 7) Call ends; caller hangs up So here is my question: In this format: 1|2|3|4|5|6, 1 - ? 2 - ? 3 - queue in question? 4 - agent answering the queue? 5 - queue event? 6 - queue event info? Is that correct? What are options 1 and 2? Times of some sort I'm guessing, but I'm not entirely sure. Thanks for your help, Jay ___ --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] X100P pass through questions
i am have x100p and not work fine, no detect polarity, and much problems with asterisk 1.2 to up. :S On 7/25/07, Mike Wright [EMAIL PROTECTED] wrote: John Novack wrote: Mike Wright wrote: Just purchased a Motorola Wildcard X100P ... but the button pressed generates no tone; on button release dialtone returns. Sure sounds like polarity reversal. Indeed it was. Punch block in the basement had tip and ring reversed. Probably been that way for thirty years. Amazing that the dsl installer that replaced the inside wiring didn't catch it. For years I've had intermittent problems using IVR. Some phones would work, some would not. Now I know why. Thanks to John Novack and Eric ManxPower Wieling for your help. ___ --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] SunRocket / ALLO / etc special offer
On 7/26/07, Matt Hoppes wrote: I would agree... intended to send that to biz, sorry. I see that you also sent it to the biz-list. And if you fail the lie detector test how about agreeing to a full boycott of your service or at least a M.L.D.P. (mailing list death penalty :-) ? -- ___ --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] Asterisk Conference Call
On 7/26/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have asterisk with SNOM SIP phone i want to confrance to my users how to configure confranceing in asterisk meetme.conf is fine but is there any otherway to confranceing If the End device support conference still you can do that ram ___ --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] IAX2 INBAND DTMF?
Matt wrote: I can think of no reason to ever need to do this. You must not peer with Level3, or with anyone who peers with Level3 via IAX :) Why would anyone want to send traffic/calls to Level3? A search of the mailing list archives is all that is needed to know that. 8-) I didn't think that Level3 supported IAX connections. If you are using an ITSP that uses Level3, I would hope the ITSP would be using inband DTMF on SIP for their connection to Level3. In any case, that might be a reason, but I don't know if it is a *good* reason. 8-) ___ --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] IAX connections broken
Not likely. #1, I have a public IP on that firewall. #2. If I block 4569 at our firewall, then it goes from closed to stealth. If I forward the port, it goes from stealth to closed. The iaxping tool (http://www.bpvn.com/asterisk/iaxping.zip) has no problems pinging the box from the lan, and our test machine can make an IAX connection to the box. From outside the network, however, it times out. It has to be a NAT problem, but forwarding doesn't appear to be working. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Baji Panchumarti Sent: Thursday, July 26, 2007 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX connections broken what if your internet provider is blocking inbound 4569 ? -- On 7/26/07, Michael Munger wrote: Dear All: I have several boxes that up and running just great, then we changed internet equipment due to a lightning strike, now all my inbound IAX connections (iax2 show peers) have unknown status. If I log into the remote boxes, it says Request sent. The authentications haven't changed at all, and all the iax.conf settings are correct. It looks like a firewall issue, but we've got 4569 TCP UDP forwarded to our Asterisk box. When I use Shields up from GRC.com to test the port, it is showing up as closed rather than open, which normally means the port is open, but the service is not running, yet Asterisk is up and running just fine, and my outbound connections to Voicepulse work fine. I see voicepulse, voicepulse sees me. There is something I am not seeing here. Any thoughts? -Michael ___ ___ --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] IAX connections broken
what if your internet provider is blocking inbound 4569 ? -- On 7/26/07, Michael Munger wrote: Dear All: I have several boxes that up and running just great, then we changed internet equipment due to a lightning strike, now all my inbound IAX connections (iax2 show peers) have unknown status. If I log into the remote boxes, it says Request sent. The authentications haven't changed at all, and all the iax.conf settings are correct. It looks like a firewall issue, but we've got 4569 TCP UDP forwarded to our Asterisk box. When I use Shields up from GRC.com to test the port, it is showing up as closed rather than open, which normally means the port is open, but the service is not running, yet Asterisk is up and running just fine, and my outbound connections to Voicepulse work fine. I see voicepulse, voicepulse sees me. There is something I am not seeing here. Any thoughts? -Michael ___ ___ --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] Dialtone when automatically picking up.
On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: ;; dialtone in the background isn't there any more ;; dialed '305' ;; everything from here is exactly as expected. OK, I missed this in the first email you sent... Asterisk is playing dialtone *on top* of the background message the first time around? That truly is bizarre. I have no idea what would cause that. -Jared ___ --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] tdm400p fxs module busy
matt,
I just had the same problem... does your CLI report 'unable to
create channel Zap/#'
post the CLI output to help us determine the problem.
daveC
Matt Scott wrote:
Dear All
The
setup is te110p with an 8 channels PRI to make and receive all calls.
SIP phones throughout the company.
TDM400p with 4 FXS modules to send/receive faxes and make credit card
transactions.
I have an analogue phone on the tdm400p for testing.
I can receive calls to the exten. There is a dialing tone.
However, when I try to make a call I get a busy signal.
Asterisk stated busy then hungup zap/32-1
why
wont asterisk supply a resource from the te110p pri card for use by the
tdm400p FXS (fxo signalling)?
configs below:
[EMAIL PROTECTED] etc]# more 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: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS RED
span = 1,0,0,ccs,hdb3,crc4
# termtype: te
bchan=1-8
dchan=16
# Span 2: WCTDM/0 "Wildcard TDM400P REV H Board 1"
fxoks=32
fxoks=33
fxoks=34
fxoks=35
# Global data
loadzone = uk
defaultzone = uk
[EMAIL PROTECTED] asterisk]# more zapata.conf
[trunkgroups]
[channels]
language=en
internationalprefix = 00
nationalprefix = 0
context=from-pstn
switchtype=euroisdn
pridialplan=local
priindication=outofband
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
group=1
callgroup=0
pickupgroup=0
immediate=no
echotraining=yes
echocancel=yes
echocancelwhenbridged=no
facilityenable=yes
musiconhold=default
overlapdial=yes
immediate=no
txgain=0.0
rxgain=0.0
signalling = pri_cpe
channel = 1-8
faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
signalling = fxo_ks
echocancel=yes
pulsedial=yes
channel=32-35
[EMAIL PROTECTED] asterisk]# more extensions.conf
[general]
static=yes
writeprotect=yes
;
[globals]
OUTBOUND = Zap/g1
FAX1 = Zap/32
FAX2 = Zap/33
STREAMLINE1 = Zap/34
STREAMLINE2 = Zap/35
CUSTSERVE1 = SIP/401 ;Teresa
CUSTSERVE2 = SIP/402 ; Louise
;CUSTSERVE3 = SIP/404 ; Helen
QUAD1 = SIP/451 ; Matt
QUAD2 = SIP/452 ; Johan
CUSTSERVE = CUSTSERVE1CUSTSERVE1
;
FSEXT1 = SIP/400 ; Angela
;FSEXT2 = SIP/403 ; Nigel
FSEXT3 = SIP/410 ; Matt
;
;ELLIS = SIP/411
;QUEENS = SIP/412
;FSSHOPS = ELLISQUEENS
;
QUAD = SIP/450
;
LONDONSOLE1 = SIP/421 ; Zoe
;LONDONSOLE2 = SIP/422 ; Laura
;LONDONSOLE = LONDONSOLE1LONDONSOLE2
;
;PRESS1 = SIP/431 ; Lucy
;PRESS2 = SIP/432 ; Gemma
;PRESSOFFICE = PRESS1PRESS2
;
[macro-oneline]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
;
[macro-oneline1]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup
;
[macro-fax]
exten = s,1,Dial(${ARG1},20,t)
exten = s,3,Hangup
;
[default]
;setupdial out
include = from-pstn
;
;test dialplan
exten = _9xxx,1,SayDigits(${EXTEN:1})
;
;setup the phone extensions
exten = 400,1,Macro(oneline,${FSEXT1})
exten = 401,1,Macro(oneline,${CUSTSERVE1})
exten = 402,1,Macro(oneline,${CUSTSERVE2})
exten = 410,1,Macro(oneline,${FSEXT3})
exten = 421,1,Macro(oneline,${LONDONSOLE1})
exten = 450,1,Macro(oneline,${QUAD})
exten = 451,1,Macro(oneline,${QUAD1})
exten = 452,1,Macro(oneline,${QUAD2})
;
exten = 1000,1,Macro(oneline,${CUSTSERVE})
;exten = 2000,1,Macro(oneline,${FSSHOPS})
;exten = 3000,1,Macro(oneline,${PRESSOFFICE})
;
;record new voice files
Exten = 501,1,Wait(2)
Exten = 501,n,Record(/tmp/asterisk-recording:gsm)
Exten = 501,n,Wait(2)
Exten = 501,n,Playback(/tmp/asterisk-recording)
Exten = 501,n,wait(2)
Exten = 501,n,Hangup
;
;goto voicemail
exten=*98,1,VoiceMailMain([EMAIL PROTECTED]})
;
[dialphone]
exten = 90,1,Macro(fax,${FAX1})
;
[from-pstn]
;this is linked to zapata.conf and defines where the ddi points
exten = 00,1,Dial(SIP/401SIP/402,15)
exten = 00,2,Voicemail(1000)
;
exten = 769611,1,Macro(oneline1,${FSEXT1})
exten = 769615,1,Macro(oneline1,${LONDONSOLE1})
;exten = 769616,1,Macro(oneline1,${LONDONSOLE2})
exten = 769636,1,Macro(oneline1,${FSEXT1},${401})
;exten = 769637,1,Macro(oneline1,${NIGEL})
;
exten = _9.,1,Set(CALLERID(number)=00)
exten = _9.,2,Dial(${OUTBOUND}/${EXTEN:1})
exten = _9.,3,Congestion()
exten = _9.,102,Congestion()
;
exten = 999,1,Dial,(${OUTBOUND}/999)
exten = ,1,Dial,(${OUTBOUND}/999)
;
exten = 90,1,Dial(Zap/32,15)
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Checked by AVG Free Edition.
Version: 7.5.476 / Virus
Re: [asterisk-users] Display IE
Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNECT comes from the PSTN containing a Display IE (which has info sent by the telco that is used for billing purposes by the PBX) into span 1 of the asterisk. (The telco is emulating an old billing procedure based on an impulse count. This was commonly used in analog lines.) 2. The asterisk relays the call to span 2 (which is connected to the PBX). 3. The CONNECT that is sent from span 2 to the PBX does not have the Display IE. The asterisk strips this IE from the CONNECT message. This is my problem. Is there a way that i can force the asterisk not to strip the Display IE? Thanks and best regards to all, Óscar Patrício Anthony Francis wrote: Damon Estep wrote: Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is probably being send before the display IE arrives. The display IE is used for CNAM delivery, and should not exceed 15 characters. It is very common to put a message in the display IE that indicates that the CNAM info will be sent in a subsequent Facility IE, and for that you must wait 1 second. If the ISDN setup actually contains information in the display IE, and that is not being captured as the CNAM (callerid(name)) you might need to capture he ISDN messaging to debug it, the telco can usually provide such a trace. I would bet that the display IE contains a information following message, and what you really want is in the facility IE that follows. Very common, as is the Wait(1) workaround. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Patricio Sent: Wednesday, July 25, 2007 6:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Display IE Hi! I have an Asterisk Box that has 2 E1 connections: one to the PSTN and one to a PBX. It is acting as a telephony gateway. I have a problem: the PSTN sends information in the Display IE (in setup, information , etc.messages) that the PBX needs por internal processing. The asterisk does not relay the ISDN frames coming from the PSTN. It regenerates them, and when it does, it ommits the Display IE. Is there a way that i can force the asterisk, not to ignore this IE in the ISDN messages? If anyone can shed some light on this issue, i would be very grateful. Thanks in advance, Best regards, Óscar Patrício ___ --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 Or trace it yourself with: pri intense debug span 1 Make sure you change the 1 to whatever span these calls are coming across. Anthony ___ --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] Queue stats from the dial plan
Hi guys, Is there any option to retrieve queue stats (particulary am interested in the time of currently longest waiting caller) from the dialplan? Thank, Alex ___ --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] Asterisk Supported Harware Architecture
Hi Saqib, Architecture is depend on what service you want to deliver. Voip is more cheaper then pstn for interoffice connectivity. But consider regulatory issue before using it. visit http://www.voip-info.org/wiki-Asterisk for complete detail. Regards Nasir iqbal On Wed, 2007-07-25 at 22:48 +0500, saqib butt wrote: HI Kindly can anyone plz tell me what will be the broadband architecture for Asterisk, e.g; for a multinational company having offices in different far location. What will the best solution or architecture to setup to go over external PSTN lines accross many locations. Is ISDN is ok or it may need DSL brodband service. kindly guide me about it as i dont know much about establishing asterisk harware/network infrastructure, can u plz forward me to any website for this. THANX ___ --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] strange problem in asterisk + mediant2k setup
Dear all I have asterisk 1.2 with mediant2k i have create SIP Trunk from asterisk 2 mediant and my PRI terminated on Mediant 2000 E1 port now everthing is fine but problem is when i call to somebody outside and he/she disconnect my phone but my asterisk counitine ringing my SIP Snom phone why ??? if i call to outside and mobile or phone would be busy but my IP SNOM Phone give me ruinging means i dont understand problem on mediant side or asterisk outgoing call working fine but only when some one disconnect call i dont get any message like phone is busy or something else but my asterisk phone continue rining - Shape Yahoo! in your own image. Join our Network Research Panel today!___ --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] Problems with new logic being 'n' option to Queue in 1.4.9
I am experiencing a change in behaviour of my Queues in 1.4.9 vs 1.4.8.
I do not pass the 'n' option to any call to Queue() in my dialplan. Yet
since I upgraded to 1.4.9, I have occasionally seen this on my console:
-- Nobody picked up in 2 ms
-- Exiting on time-out cycle
That log message Exiting on time-out cycle is exclusive to the logic in
app_queue meant to deal with the 'n' option. If you don't pass 'n', you
should never see it.
1.4.8 code:
/* exit after 'timeout' cycle if 'n' option
enabled */
if (go_on) {
if (option_verbose 2)
ast_verbose(VERBOSE_PREFIX_3
Exiting on time-out cycle\n);
ast_queue_log(args.queuename,
chan-uniqueid, NONE, EXITWITHTIMEOUT, %d, qe.pos);
record_abandoned(qe);
reason = QUEUE_TIMEOUT;
res = 0;
break;
}
1.4.9 code:
/* exit after 'timeout' cycle if 'n' option enabled
*/
if (go_on = qe.parent-membercount) {
if (option_verbose 2)
ast_verbose(VERBOSE_PREFIX_3
Exiting on time-out cycle\n);
ast_queue_log(args.queuename,
chan-uniqueid, NONE, EXITWITHTIMEOUT, %d, qe.pos);
record_abandoned(qe);
reason = QUEUE_TIMEOUT;
res = 0;
break;
}
In both versions, the variable 'go_on' starts off set to 0, and only gets
set if you pass the 'n' option to Queue(). The manner in which it gets set
differs between 1.4.8 and 1.4.9, but it is only when you pass the 'n'
option, so it shouldn't matter. In my configuration, go_on should always be
zero.
The logic check around go_on is what's worrying me. In 1.4.8, go_on had one
of two values - 0 or 1. If you never passed 'n' to Queue(), it was always
0, so the block of code that takes you back to the dialplan on timeout can
never be executed.
In 1.4.9, if qe.parent-membercount is zero and you didn't pass the 'n'
switch, then you'll exit the queue as if you had timed out, even though you
never passed the 'n' option. I haven't gone through the entire code of
app_queue to see exactly how membercount gets manipulated, but it seems from
my log that these exitwithtimeouts events seem to occur right after an agent
has let their phone ring without picking it up (see the nobody picked up in
2ms message in my example above).
Is it possible for qe.parent-membercount to be set to zero in a queue where
all agents but one are on the phone and that one remaining agent lets their
phone ring without answering it?
--
j.
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Re: [asterisk-users] Default Asterisk Numbers
features.conf On 7/26/07, GNUbie [EMAIL PROTECTED] wrote: Hello all, Where can I find the complete list of default Asterisk (telephone) numbers and maybe the other special numbers that are need to be preserve and not use for setting up own dial plan? Thank you. GNUbie ___ --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 Advice on Asterisk and Linux
Hello Mark, On 7/27/07, Mark Burrows [EMAIL PROTECTED] wrote: Can someone suggest a starting point on learning Linux? First of all, welcome to the community! =) I may consider myself as an experienced systems/network administrator but with Asterisk and telephony, I am still newbie to it. For me, Asterisk and telephony in general is a totally different world. If I may suggest to you, try to check Trixbox http://www.trixbox.org/ because it's a pretty easy to setup VoIP software appliance in one distribution. Although I must admit that I didn't tried installing it and I installed Asterisk instead the-hard-way TM on my Debian Etch http://www.debian.org/ so that I would learn (hopefully) this technology. Anyway, my setup here is only on my home PBX so nothing to worry. If you would like to learn GNU/Linux systems and/or network administration, you will have to choose a distribution to start with. You can choose GNU/Linux distributions at the DistroWatch website http://www.distrowatch.com/. I use Debian GNU/Linux Etch here as my OS. You can also find useful information on GNU/Linux and F/OSS in general at The Linux Documentation Project website http://www.tldp.org/. Your distribution of choice might have a good documentation specially the systems and network administration. Good luck! GNUbie ___ --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] ISDN: Problems starting off
Hi, the first thing I did with Asterisk is listening to `demo-congrats' by Xlite on the same machine. This works perfectly. The config files are those shipped with the package. Now I want to listen to it over ISDN/Capi but I don't succeed. My `capi.conf' is like show in many tutorial on the web. In `extensions.conf' I just added the following lines: [capi-in] exten = 9876543,1,Goto(demo,1000,1) where 9876543 is my MSN without the area prefix. `demo' is the context that plays `demo-congrats'. The debug output I yield ends with (after a pause) DISCONNECT_IND ID=001 #0x0027 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x0 DISCONNECT_RESP ID=001 #0x0027 LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1#02: DISCONNECT_IND on incoming without pbx, doing hangup. CAPI/ISDN1/9876543-2: set channel task to 1 == ISDN1#02: CAPI Hangingup for PLCI=0x101 in state 4 == ISDN1#02: Interface cleanup PLCI=0x101 CAPI devicestate requested for ISDN1/9876543 Seems that the MSN or even `capi-in' cannot be found at all. Could anyone give me a hint what is going wrong here or at least what I have to diagnose next? Thanks in advance. Bertram -- Bertram Scharpf Stuttgart, Deutschland/Germany http://www.bertram-scharpf.de ___ --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] vm-duration announcement missing?
I just saw this on my console:
[Jul 26 11:36:30] WARNING[8667] file.c: File vm-duration does not exist in
any format
[Jul 26 11:36:30] WARNING[8667] file.c: Unable to open vm-duration (format
0x4 (ulaw)): No such file or directory
Thinking I might have lost a file during a fsck or something, I checked -
sure enough, there's no file vm-duration in any format. I downloaded the
current (as of June 14th) core and extra sounds, but it's not in there
either.
1.2.x didn't use this file, but app_voicemail contains reference to it in
1.4.x - as far back as 1.4.0:
if ((!res) (durationm = minduration)) {
res = wait_file2(chan, vms, vm-duration);
[snip stuff about polish syntax]
res = ast_say_number(chan, durationm, AST_DIGIT_ANY,
chan-language, NULL);
res = wait_file2(chan, vms, vm-minutes);
}
Does anyone know where this file can be fetched from, or at least what it's
supposed to say?
Looking back at my logs, there are semi-regular instances of this error
message. In a default setup, it's only used if the message is more than 2
minutes long, which I guess most of my user's VMs aren't.
Thanks
--
j.
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[asterisk-users] Ring forever
Hello list, i need help.
My problem is that when I want to call (using ISDN
phone or internal SIP client) via the Sip provider a
real phone number (ISDN phone or internal SIP
Asterisk SIP ), I get a ring tone. When I
now decide to hang up (e.g. if
nobody answers), the called telephone continues to
ring almost forever.
the sip.conf:
[2563105]
accountcode = 2563105
username = 2563105
secret = 135
callerid = 412563105
context = test
canreinvite = no
dtmfmode = rfc2833
host = dynamic
insecure = very
language = es
nat = yes
qualify = yes
type = friend
disallow=all
allow=g729
[nyphone]
accountcode=nyphone
canreinvite=no
reinvite=yes
dtmfmode=rfc2833
host=72.55.143.XXX
insecure=very
language=es
nat=no
qualify=no
type=peer
disallow=all
allow=g729
My extensions.conf
exten = _00X.,1,dial(sip/${EXTEN:[EMAIL PROTECTED],45)
exten = _00X.,2,hangup
Nyphone is my provider for everyone calls
international.
Fernando Villarroel Noriel.
Chillan
Chile
Sorry my English.
This is log:
SIP Debugging Enabled for IP: 72.55.143.XXX:5060
-- Executing [EMAIL PROTECTED]:1]
Dial(SIP/2563105-0819cf80,
sip/[EMAIL PROTECTED]|45) in new stack
Audio is at 164.77.171.XXX port 16548
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 72.55.143.XXX:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport
From: 2563105
sip:[EMAIL PROTECTED];tag=as726ac50a
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 20 Jul 2007 03:38:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 2475 2475 IN IP4 164.77.171.XXX
s=session
c=IN IP4 164.77.171.XXX
t=0 0
m=audio 16548 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called [EMAIL PROTECTED]
vaca*CLI
--- SIP read from 72.55.143.XXX:5060 ---
SIP/2.0 407 Proxy Authentication Required
CSeq: 102 INVITE
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport
From: 2563105
sip:[EMAIL PROTECTED];tag=as726ac50a
Call-ID:
[EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED]
Contact: sip:72.55.143.XXX:5060;transport=udp
Proxy-Authenticate: DIGEST realm=VoipSwitch,
nonce=118490324119231120007472128429
Content-Length: 0
-
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 72.55.143.XXX:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport
From: 2563105
sip:[EMAIL PROTECTED];tag=as726ac50a
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Audio is at 164.77.171.XXX port 16548
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 72.55.143.XXX:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK3cb0e5aa;rport
From: 2563105
sip:[EMAIL PROTECTED];tag=as726ac50a
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID:
[EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username=test770,
realm=VoipSwitch,
algorithm=MD5, uri=sip:[EMAIL PROTECTED],
nonce=118490324119231120007472128429,
response=413be923621811a639c3b0e83d3a2e74, opaque=
Date: Fri, 20 Jul 2007 03:38:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 2475 2476 IN IP4 164.77.171.XXX
s=session
c=IN IP4 164.77.171.XXX
t=0 0
m=audio 16548 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
vaca*CLI
--- SIP read from 72.55.143.XXX:5060 ---
SIP/2.0 200 OK
CSeq: 103 INVITE
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK3cb0e5aa;rport
From: 2563105
sip:[EMAIL PROTECTED];tag=as726ac50a
Call-ID:
[EMAIL PROTECTED]
To:
sip:[EMAIL PROTECTED];tag=1907470723212675853288937
Contact: sip:72.55.143.XXX:5060;transport=udp
Content-Type: application/sdp
Content-Length: 215
v=0
o=VoipSwitch 9936 9936 IN IP4 72.55.143.XXX
s=VoipSIP
i=Audio Session
c=IN IP4 72.55.143.XXX
t=0 0
m=audio 8936 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (9 headers 10 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 72.55.143.XXX:8936
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100
(g729)/video=0x0
(nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1
[asterisk-users] SetCallerPres and Cisco PRI
Does anybody know if SetCallerPres works on calls via SIP through a Cisco gateway? We are trying to mark outbound calls as anonymous and we set it to prohib, but calls still show outbound callerid. We are SIP from * to the Cisco gateway and then PRI outbound. If we strip the callerid num, then the first number on the PRI gets added as teh callerid, so we can't do that. We need to make it anonymous so that it shows as unknown on the other end. 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] Newbie Advice on Asterisk and Linux
HI All, I’m new to Asterisk and also to Linux. I have a large IVR project that I’m about to embark on. I’m new to programming; new to Linux and new to Asterisk. I think I’m about to climb a steep learning curve. I have an existing IVR which is getting on for nine years old and is no longer supported by my vendor. I intend to replicate the system almost as is and then add additional features and functions. I have been looking for a developer to put together my project and while doing so have done lots of research and spoken to many people. The people who seem to understand my needs have recommended Asterisk. For the last couple of days I’ve been trying to look into Asterisk and learn as much as I can; this has got me excited, motivated and a little confused. Asterisk sounds like a great project and a great community. I think I have as much of an overview as I can. Now I need to set up a Linux system and get Asterisk running on it. I’ve started to read the book Asterisk: The Future Of Telephony and would like to now setup up a hobby computer to do some hands on learning. The book covers Red Hat Linux so I thought I’d look for a ‘Red Had for Dummies’ book. Even that got confusing. There’s Linux Fedora, Enterprise Linux 4 and others. Can someone suggest a starting point on learning Linux? Thanks in advance, Mark No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.20/919 - Release Date: 26/07/2007 9:56 AM ___ --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] Lines Not being Hung UP Major
Btw are the phones behind NAT ?? Also you can try some softphone and make sure that this problem is caused by snom phones or some other factors .. On 25/07/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: I thought it was the fios service but now I realize it's the snom 360! It doesn't hang up random outgoing calls. It seems like it only happens on outbound calls from phones that have been updated to 6.5.12 or 6.5.10. It didn't happen before, but I don't remember what version firmware it was before, maybe 6.2.3 or so. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Arts Sent: Monday, July 16, 2007 5:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lines Not being Hung UP Major Do your SNOM phones sometimes use answer-after:0, and do they have keyboard LEDs subscribed to their own extensions? Do those people hangup calls by puttig down the handset instead of pressing the X key? We are seeing hanging channels in this particular case. Ron Michael J. Liberatore wrote: Hi all, i am having a major asterisk problem. I think it started around 1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360. basically we start getting busy signals, all our 4 line hunt group is busy, i then check the channels and there are open calls that were hung up long ago. i thought it was a zap problem but then i saw the same problem with iax2 calls. its becoming a huge issue because if i dont reboot asterisk several times a day, all our lines get filled up with dead calls. I am now running 1.2.21.1 asterisk with the same problem. Please help. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Need help with inbound IAX
What do you get with: iax2 show registry homer*CLI iax2 show registry Host UsernamePerceived Refresh State 64.85.162.136:456906*** 68.XX.XX.XX:4569 300 Registered is that bad? Patrick ___ --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] Queue stats
Hopefully that helps clarify things! It does immensely. Thanks a ton! Jay ___ --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] Asterisk Vm functionality question
Andrew, Could you elaborate on how you configure the MWI of the mobile device to use asterisk voicemail? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.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] Queue stats
This may not be the best solution for you, but it's the only one I can speak for. We use QueueMetrics for our queue information and reporting. There is a small cost for it, but it is worth every penny. On 7/26/07, Jay Moore [EMAIL PROTECTED] wrote: Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller falls into our 'music on hold' queue, a message is played and the caller hears our music on hold while the phones are rung again. 3) After 30 seconds, if the caller is still in our 'moh' queue, they drop out of queue and immediately re-enter the 'moh' queue again until the call is answered or the caller hangs up. How can I find out how many calls are answered out of each queue during certain times (1st shift, 2nd shift, etc...)? Also, I'm curious how I can track how many times a call repeats the 'moh' queue. Thanks in advance, Jay ___ --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] IAX2 INBAND DTMF?
mailing list archives is all that is needed to know that. 8-) We've never had any issues with L3 and are very happy. I didn't think that Level3 supported IAX connections. If you are using an ITSP that uses Level3, I would hope the ITSP would be using inband DTMF on SIP for their connection to Level3. L3 doesn't support IAX.. but if you peer with a L3 provider using IAX you are still affected. Why would the ITSP need to use inband? They can use RFC to L3. In any case, that might be a reason, but I don't know if it is a *good* reason. 8-) ___ --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] SunRocket / ALLO / etc special offer
I would agree... intended to send that to biz, sorry. On 7/25/07, Anthony Francis [EMAIL PROTECTED] wrote: Matt wrote: If you have been affected by the SunRocket / ALLO folding issue, ChiliTech would like to extend our hand to you to help you in this time. We will transfer your numbers to us for no cost, and will match your SunRocket or ALLO rate. Please contact us at 1-866-678-6858 x 126 or e-mail [EMAIL PROTECTED] We have been around since 2001 serving the Internet community. Matt Hoppes ChiliTech Internet Solutions ___ --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 Using the users group as a marketing platform is kind of a low thing to do don't you think? ___ --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] IAX2 INBAND DTMF?
David Boyd wrote: On Wed, 2007-07-25 at 13:02 -0500, Eric ManxPower Wieling wrote: Short Answer: No. Long Answer: Maybe. If you can get your device to send inband DTMF and tell Asterisk you are using INFO or RFC2833 DTMF, then Asterisk should just pass the DTMF as audio. Then if the call goes via IAX2 it should be inband. This is an ungly hack, should not be supported in any way and if it works just count your blessings. I can think of no reason to ever need to do this. Matt wrote: Is it possible to make Asterisk do inband DTMF over IAX? Snip--- Ok, I am confused. Are you saying that if I use an IAX2 inter machine trunk from one asterisk box to another, and terminate a call over the pstn to a voicemail system or other type of IVR, IAX2 will regenerate the DTMF tones that were originated from the original callers phone? I thought the original posting said that the IAXy device was failing to pass DTMF through to the termination side of the call. What have I missed? I think you missed that the IAXy is not supporting this. The IAXy is not taking the out of band DTMF and converting it back to AUDIO to send to the device connected to the IAXy. ___ --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] IAX connections broken
Dear All: I have several boxes that up and running just great, then we changed internet equipment due to a lightning strike, now all my inbound IAX connections (iax2 show peers) have unknown status. If I log into the remote boxes, it says Request sent. The authentications haven't changed at all, and all the iax.conf settings are correct. It looks like a firewall issue, but we've got 4569 TCP UDP forwarded to our Asterisk box. When I use Shields up from GRC.com to test the port, it is showing up as closed rather than open, which normally means the port is open, but the service is not running, yet Asterisk is up and running just fine, and my outbound connections to Voicepulse work fine. I see voicepulse, voicepulse sees me. There is something I am not seeing here. Any thoughts? -Michael ___ --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] Astribank-8BRI
Thanks for the replies. I decided to go with the USB channel bank. I hope everything will go alright. Lars -- Let's not complicate our relationship by trying to communicate with each other. ___ --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 Conference Call
Dear all I have asterisk with SNOM SIP phone i want to confrance to my users how to configure confranceing in asterisk meetme.conf is fine but is there any otherway to confranceing - Got a little couch potato? Check out fun summer activities for kids.___ --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] Display IE
Oscar Patricio wrote: 3. The CONNECT that is sent from span 2 to the PBX does not have the Display IE. The asterisk strips this IE from the CONNECT message. This is an incorrect statement; 'strips' would imply that Asterisk is just forwarding the CONNECT message from one PRI to the other, but in fact that is not what happens. Asterisk is a multi-protocol telephony platform, and it never *proxies* or directly connects two channels together; instead, the incoming signaling from any channel is converted into Asterisk's internal format, then delivered to the other channel, where it is converted back into that channel's format before being sent out. If there are signaling elements being received that Asterisk does not interpret for its own use or for exposure in the dialplan, then they will be ignored. To get your Display IE to be transferred to the outbound PRI, you'd need to get chan_zap to parse it from the incoming CONNECT message, store it in an Asterisk control frame and then accept that control frame on the outbound side and send a Display IE. Doing this will require code changes :-) -- 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
Re: [asterisk-users] Default Asterisk Numbers
also have a look on http://www.voip-info.org/wiki/view/Asterisk+standard+extensions On Thu, 2007-07-26 at 20:57 -0600, Al lists wrote: features.conf On 7/26/07, GNUbie [EMAIL PROTECTED] wrote: Hello all, Where can I find the complete list of default Asterisk (telephone) numbers and maybe the other special numbers that are need to be preserve and not use for setting up own dial plan? Thank you. GNUbie ___ --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] IAX2 INBAND DTMF?
I can think of no reason to ever need to do this. You must not peer with Level3, or with anyone who peers with Level3 via IAX :) ___ --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] Asterisk Vm functionality question
Andrew Kohlsmith wrote: On Thursday 26 July 2007 12:51:06 pm Eric Chamberlain wrote: Could you elaborate on how you configure the MWI of the mobile device to use asterisk voicemail? yes, please explain. SMSing the phone doesn't light MWI, unless you get access to the raw SMSC, as all the email gateways just mangle the message. I suspect people that light the MWI on their Cell phone do not live in the USA. ___ --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] IAX2 INBAND DTMF?
Yes, it is a Blue Digium IAXy. It is on my local LAN , so the Linksys SIP is working fine. It was just a surprising discovery since Digium's owner defined IAX2, specified that there can be no in band DTMF and then Disgium left this out of the IAXy. I believe that they assumed that it would only be used as a station phone. -- -- Steven http://www.glimasoutheast.org Jared Smith [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] On Wed, 2007-07-25 at 17:16 -0400, Steven wrote: My biggest issue with this is that the Iaxys will not generate DTMF tones onto the analog side.. Which type of IAXy do you have? I remember having a problem with this over a year ago with one of the older IAXy boxes (the blue ones), but it seemed to work fine when I swapped it for one of the grey ones. It's been long enough now that I don't remember if I had to do anything else special to get it to work. But I do know for a fact that this elementary school has been using my IAXy to drive their paging system for over a year. (I'm not sure that's much help, but maybe it'll spark someone else's memory.) -- 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 ___ --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] missing digits on PRI
I seem to be missing digits with a PRI. I added dtmf logging in logger.conf This does not happen a-lot but it does happen a number of times over the day. I have watched a few times for calls coming in and the logger only showed me 09 instead of 209. I contacted my provider they checked it out and said they see no impedance issues and everything looks fine on their end. I have installed a handful of T1s at other sites and no issues. The card is a Te205p. I am running 1.4.7.1 asterisk and 1.4.3 zaptel. What might I tweek to get 100% on DTMF. Jerry ___ --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] SunRocket / ALLO / etc special offer
I'll take either Actually now that I have had a chance to think about what I did (sorry bad week here). Yes, I will admit I did patrionize the users list... sorry if I offended anyone. I just figured I'd try to help any SunRocket users out that may not be on the biz list.If you review my history, you'll see I only post business stuff to the biz list. This is an exception. On 7/26/07, Baji Panchumarti [EMAIL PROTECTED] wrote: On 7/26/07, Matt Hoppes wrote: I would agree... intended to send that to biz, sorry. I see that you also sent it to the biz-list. And if you fail the lie detector test how about agreeing to a full boycott of your service or at least a M.L.D.P. (mailing list death penalty :-) ? -- ___ --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] Queue stats
On Thu, 2007-07-26 at 09:37 -0500, Jay Moore wrote: My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. It sounds like you've got quite the queue setup (although I don't quite see why your calls jump out and back into the moh queue). All the of queue statistics you need should be available with careful parsing of the queue log (usually located in /var/log/asterisk/queue_log). You can also trigger custom queue log events from the dialplan by calling the QueueLog() application. In your case, you might want to add a custom queue log entry every time the caller rejoins the moh queue, saying something to the effect of Caller XYZ has rejoined the moh queue for the 10th time or something like that. -- 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] Asterisk Vm functionality question
Yes it's possible. It's also possible to have Asterisk try and find the person in the field and either connect the call or deliver the message. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James R. Stevens Sent: Wednesday, July 25, 2007 10:21 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk Vm functionality question Going over the needs of any PBX that replaces our current system (working toward Asterisk) and have VM functionality question. Currently when someone leaves a voice mail for a sales person (Who is in the field) the system takes the VM and then in turn dials over a POTS line and pages the sales person notifying them of a VM (Does not deliver the message-just notifies) Is this possible with Asterisk? 14 Channel PRI straight into a Sangoma T1/E1 card -- This message has been scanned for viruses and dangerous content by http://www.athensdistributing.com/ Athens Hyperion Scanner, and is believed to be clean. ___ --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 Advice on Asterisk and Linux
Mark, Welcome to the club. Learning Linux can be a daunting task. After working with it for the last decade, I am still learning. My best recommendation is to play with it on a test box, and post questions to a related community forum if you get stuck on something. If you are looking for something more intense and less time-consuming, check your local colleges. The colleges in my area offer several classes on Linux as part of a degree in Network Administration. HTH, John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 7/26/2007 3:08:36 PM HI All, I'm new to Asterisk and also to Linux. I have a large IVR project that I'm about to embark on. I'm new to programming; new to Linux and new to Asterisk. I think I'm about to climb a steep learning curve. I have an existing IVR which is getting on for nine years old and is no longer supported by my vendor. I intend to replicate the system almost as is and then add additional features and functions. I have been looking for a developer to put together my project and while doing so have done lots of research and spoken to many people. The people who seem to understand my needs have recommended Asterisk. For the last couple of days I've been trying to look into Asterisk and learn as much as I can; this has got me excited, motivated and a little confused. Asterisk sounds like a great project and a great community. I think I have as much of an overview as I can. Now I need to set up a Linux system and get Asterisk running on it. I've started to read the book Asterisk: The Future Of Telephony and would like to now setup up a hobby computer to do some hands on learning. The book covers Red Hat Linux so I thought I'd look for a 'Red Had for Dummies' book. Even that got confusing. There's Linux Fedora, Enterprise Linux 4 and others. Can someone suggest a starting point on learning Linux? Thanks in advance, Mark No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.20/919 - Release Date: 26/07/2007 9:56 AM - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --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] vm-duration announcement missing?
James FitzGibbon wrote: Looking back at my logs, there are semi-regular instances of this error message. In a default setup, it's only used if the message is more than 2 minutes long, which I guess most of my user's VMs aren't. This is my fault; we have a pending list of sounds to be recorded and included in the core-sounds package and I've neglected it. I'll put it on my to-do list for tomorrow right now... Sorry :-( -- 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
Re: [asterisk-users] Grandstream RTP keepalive packets causing Asteriskwarning
Grab a network trace (with e.g. Wireshark) and look at the payload type and lengths of the RTP keepalive messages - if you post this information to the list I'm sure someone will comment on what's happening. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: 26 July 2007 19:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Grandstream RTP keepalive packets causing Asteriskwarning Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where the phone did not send rtp keepalives when on mute (resulting in disconnect from tech support hold and concalls) A side effect seems to be that Asterisk pops the following warning on the console... Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP Read too short Grandstream say they are not sure what it is but it should not affect anything. In other words, Don't worry, be happy!. Any thoughts/experience on this? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.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 ___ --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] polycom custom ring tones (slightly OT)
Hi all, Has anyone made up custom ring tones for the Polycom SIP phones? We use different rings for different lines, but the ones it comes with are all very similar. In the interesting of sharing, here's one I made up for paging: PAGE_BEEP se.pat.ringer.13.name=Page Beep se.pat.ringer.13.inst.1.type=chord se.pat.ringer.13.inst.1.value=12 se.pat.ringer.13.inst.1.param=200 se.pat.ringer.13.inst.2.type=chord se.pat.ringer.13.inst.2.value=15 se.pat.ringer.13.inst.2.param=600 se.pat.ringer.13.inst.3.type=branch se.pat.ringer.13.inst.3.value=-2/ Alternatively, since you can use .wav files for ring tones, do people have any recommendations for where to find some distinctive rings? Thanks, -- James Andrewartha Systems Administrator Data Analysis Australia Pty Ltd ___ --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] Asterisk Conference Call
Yes, have them all meet in the cafeteria for brunch. On 7/26/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have asterisk with SNOM SIP phone i want to confrance to my users how to configure confranceing in asterisk meetme.conf is fine but is there any otherway to confranceing Got a little couch potato? Check out fun summer activities for kids. ___ --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] Need help with inbound IAX
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Patrick Buller wrote: What do you get with: iax2 show registry homer*CLI iax2 show registry Host UsernamePerceived Refresh State 64.85.162.136:456906*** 68.XX.XX.XX:4569 300 Registered is that bad? Nope, that's good. It means you have registered to their server no problem. Firstly, which version of Asterisk are you using? If you turn on iax2 debug, and then say call from your cellphone to the DDI you have registered do you get anything at all? - -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGqXG1DQNt8rg0Kp4RAn1rAKC/WwYnvDaqQ9FK3YXmiWEkwkiwUwCfWin8 nciXBwS2Ws+lg/6P8gv5XRI= =7ZZX -END PGP 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
[asterisk-users] Autoreply: Re: Queue stats
On Thu, 2007-07-26 at 09:37 -0500, Jay Moore wrote: My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. It sounds like you've got quite the queue setup (although I don't quite see why your calls jump out and back into the moh queue). All the of queue statistics you need should be available with careful parsing of the queue log (usually located in /var/log/asterisk/queue_log). You can also trigger custom queue log events from the dialplan by calling the QueueLog() application. In your case, you might want to add a custom queue log entry every time the caller rejoins the moh queue, saying something to the effect of Caller XYZ has rejoined the moh queue for the 10th time or something like that. -- 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 . ___ --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] Query
Hi, I am facing problem in configuring D-channel. I did the following configuration for TE-120P card /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf group=1 signalling=pri_cpe switchtype=euroisdn context=incoming channel=1-15,17-31 DIGIUM card is connected through cable to another end.On placing call from other end to asterisk PBX ( through DIGIUM card ) the following error messages is coming on console mode of asterisk Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. == Primary D-Channel on span 1 down Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channelanyway! Can anybody tell me how to overcome this error. Thanx and Regards sanchal ___ --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] Queue stats
Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller falls into our 'music on hold' queue, a message is played and the caller hears our music on hold while the phones are rung again. 3) After 30 seconds, if the caller is still in our 'moh' queue, they drop out of queue and immediately re-enter the 'moh' queue again until the call is answered or the caller hangs up. How can I find out how many calls are answered out of each queue during certain times (1st shift, 2nd shift, etc...)? Also, I'm curious how I can track how many times a call repeats the 'moh' queue. Thanks in advance, Jay ___ --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] Queue stats
Jay, You could try ASTassistant. It has Queue information at a glance. http://www.astassistant.com - Original Message - From: Jay Moore [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 26, 2007 7:37 AM Subject: [asterisk-users] Queue stats Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller falls into our 'music on hold' queue, a message is played and the caller hears our music on hold while the phones are rung again. 3) After 30 seconds, if the caller is still in our 'moh' queue, they drop out of queue and immediately re-enter the 'moh' queue again until the call is answered or the caller hangs up. How can I find out how many calls are answered out of each queue during certain times (1st shift, 2nd shift, etc...)? Also, I'm curious how I can track how many times a call repeats the 'moh' queue. Thanks in advance, Jay ___ --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] Asterisk Vm functionality question
On Thursday 26 July 2007 12:51:06 pm Eric Chamberlain wrote: Could you elaborate on how you configure the MWI of the mobile device to use asterisk voicemail? yes, please explain. SMSing the phone doesn't light MWI, unless you get access to the raw SMSC, as all the email gateways just mangle the message. -A. ___ --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] Dialtone when automatically picking up.
On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: : ;; dialtone in the background isn't there any more : ;; dialed '305' : ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent... Asterisk is playing :dialtone *on top* of the background message the first time around? That :truly is bizarre. I have no idea what would cause that. Correct. I did some more testing, and found that there is some crazy ?cross-talk? going on. If I dial my mobile from that line, two caller-id numbers show up, the one that should, and someone else's line. I'm now thinking this is a problem with the installation of the phone lines, so I'm now hunting down the installer to have him fix it. On the bright side, when I was tracking this down I learned Asterisk fairly well. ;) -- Have you noticed that all you need to grow healthy, vigorous grass is a crack in your sidewalk? ___ --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 Advice on Asterisk and Linux
Mark Burrows wrote: HI All, I’m new to Asterisk and also to Linux. I have a large IVR project that I’m about to embark on. I’m new to programming; new to Linux and new to Asterisk. I think I’m about to climb a steep learning curve. I have an existing IVR which is getting on for nine years old and is no longer supported by my vendor. I intend to replicate the system almost as is and then add additional features and functions. I have been looking for a developer to put together my project and while doing so have done lots of research and spoken to many people. The people who seem to understand my needs have recommended Asterisk. For the last couple of days I’ve been trying to look into Asterisk and learn as much as I can; this has got me excited, motivated and a little confused. Asterisk sounds like a great project and a great community. I think I have as much of an overview as I can. Now I need to set up a Linux system and get Asterisk running on it. I’ve started to read the book Asterisk: The Future Of Telephony and would like to now setup up a hobby computer to do some hands on learning. The book covers Red Hat Linux so I thought I’d look for a ‘Red Had for Dummies’ book. Even that got confusing. There’s Linux Fedora, Enterprise Linux 4 and others. Can someone suggest a starting point on learning Linux? Thanks in advance, Mark No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.20/919 - Release Date: 26/07/2007 9:56 AM ___ --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 http://www.trixbox.org This is one of the many standard configs for aterisk. This uses CentOS 4, Asterisk 1.2, FreePBX 2.2. You can setup a fully working system in about 30 min. Need help you can email me off list. Jonn Taylor ___ --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 inbound IAX
I have just started working with Asterisk and have run into a road block concerning IAX and an inbound DID from callwithus.com. I am getting nowhere and I don't really know how to isolate the problem. The asterisk version is 1.2.7 on ubuntu, sits behind a firewall with iptables. I can connect and make a call to other internal extensions using zoiper and iax. When I try and use the number, I do not see any traffic on the firewall from the provider, so I think it is config string I have put in at callwithus, but I have tried so many things anymore, I am not sure anymore. Any help is much appreciated. Thanks, Patrick I have the following rules on the firewall: -A FORWARD -p udp -m udp --dport 4569 -j ACCEPT -A PREROUTING -i eth2 -p udp -m udp --dport 4569 -j DNAT --to-destination 192.168.1.2 I have similar rules for port 25 and 80 that work. On the asterisk machine, iax.conf looks like: [general] bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no tos=lowdelay autokill=yes register = 45*:[EMAIL PROTECTED] [iaxin] type=friend context=iax-inbound username=iaxin secret=easypass qualify=no host=callwithus.com in extensions.conf: [iax-inbound] exten = s,1,Answer() exten = s,2,Playback(hello-world) exten = s,3,Hangup() This is what callwithus is supposed to forward the call to: IAX/iaxin:[EMAIL PROTECTED]/[EMAIL PROTECTED] ___ --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 custom ring tones (slightly OT)
Hi James, I have one posting for the Cisco7970 ringtone, which you can adapt for the Polycom. It's here: http://www.voipphreak.ca/archives/349 I also have another one I posted for the Polycom Ringtones with a bunch of tunes. It's here: http://www.voipphreak.ca/archives/78 Hope these help :) Thanks, Matt On 27/07/07, James Andrewartha [EMAIL PROTECTED] wrote: Hi all, Has anyone made up custom ring tones for the Polycom SIP phones? We use different rings for different lines, but the ones it comes with are all very similar. In the interesting of sharing, here's one I made up for paging: PAGE_BEEP se.pat.ringer.13.name=Page Beep se.pat.ringer.13.inst.1.type=chord se.pat.ringer.13.inst.1.value=12 se.pat.ringer.13.inst.1.param=200 se.pat.ringer.13.inst.2.type=chord se.pat.ringer.13.inst.2.value=15 se.pat.ringer.13.inst.2.param=600 se.pat.ringer.13.inst.3.type=branch se.pat.ringer.13.inst.3.value=-2/ Alternatively, since you can use .wav files for ring tones, do people have any recommendations for where to find some distinctive rings? Thanks, -- James Andrewartha Systems Administrator Data Analysis Australia Pty Ltd ___ --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] Autoreply: Re: Newbie Advice on Asterisk and Linux
Mark Burrows wrote: HI All, Im new to Asterisk and also to Linux. I have a large IVR project that Im about to embark on. Im new to programming; new to Linux and new to Asterisk. I think Im about to climb a steep learning curve. I have an existing IVR which is getting on for nine years old and is no longer supported by my vendor. I intend to replicate the system almost as is and then add additional features and functions. I have been looking for a developer to put together my project and while doing so have done lots of research and spoken to many people. The people who seem to understand my needs have recommended Asterisk. For the last couple of days Ive been trying to look into Asterisk and learn as much as I can; this has got me excited, motivated and a little confused. Asterisk sounds like a great project and a great community. I think I have as much of an overview as I can. Now I need to set up a Linux system and get Asterisk running on it. Ive started to read the book Asterisk: The Future Of Telephony and would like to now setup up a hobby computer to do some hands on learning. The book covers Red Hat Linux so I thought Id look for a Red Had for Dummies book. Even that got confusing. Theres Linux Fedora, Enterprise Linux 4 and others. Can someone suggest a starting point on learning Linux? Thanks in advance, Mark No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.20/919 - Release Date: 26/07/2007 9:56 AM ___ --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 http://www.trixbox.org This is one of the many standard configs for aterisk. This uses CentOS 4, Asterisk 1.2, FreePBX 2.2. You can setup a fully working system in about 30 min. Need help you can email me off list. Jonn Taylor ___ --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] Autoreply: Queue stats
Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller falls into our 'music on hold' queue, a message is played and the caller hears our music on hold while the phones are rung again. 3) After 30 seconds, if the caller is still in our 'moh' queue, they drop out of queue and immediately re-enter the 'moh' queue again until the call is answered or the caller hangs up. How can I find out how many calls are answered out of each queue during certain times (1st shift, 2nd shift, etc...)? Also, I'm curious how I can track how many times a call repeats the 'moh' queue. Thanks in advance, Jay ___ --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] Asterisk Conference Call
That's actually a good idea. - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 26, 2007 9:23 PM Subject: Re: [asterisk-users] Asterisk Conference Call Yes, have them all meet in the cafeteria for brunch. On 7/26/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have asterisk with SNOM SIP phone i want to confrance to my users how to configure confranceing in asterisk meetme.conf is fine but is there any otherway to confranceing Got a little couch potato? Check out fun summer activities for kids. ___ --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] Query
Hi, I am facing problem in configuring D-channel. I did the following configuration for TE-120P card /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf group=1 signalling=pri_cpe switchtype=euroisdn context=incoming channel=1-15,17-31 DIGIUM card is connected through cable to another end.On placing call from other end to asterisk PBX ( through DIGIUM card ) the following error messages is coming on console mode of asterisk (The OTHER END CONNECTED to DIGIUM is E1 CARD RUNNING APPLICATION) Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. == Primary D-Channel on span 1 down Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! NOTE- The OTHER END CONNECTED to DIGIUM is E1 CARD RUNNING APPLICATION Can anybody tell me how to overcome this error. Thanx and Regards sanchal ___ --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
