[Asterisk-Users] Busy, differences between SIP and Zaptel(bristuff)
Hi all,
a lot of my snoms are being called with this macro:
[macro-ohne-AB]
exten = s,1,DBget(temp=UML/${ARG1})
exten = s,2,Goto(default|${temp}|1)
exten = s,3,Dial(${ARG2},600,g)
exten = s,4,SetVar(PRI_CAUSE=17)
exten = s,5,Hangup
[default]
...
exten = 77,1, Macro(ohne-AB,77,SIP/snom8556)
...
When a call comes over QuadBRI in and the called phone is Busy the caller gets a Busy.
That is fine. When another snom is calling a busy snom, then it gets an
forbidden.
When i change Hangup to Busy the call snom to busy snom is OK. Incoming
ISDN calls get silence, then after 10 seconds an congestion. That is ugly.
What is the right way to make a busy for Incoming calls (QuadBRI) and internal
calls (SIP to SIP).
Best reagards
Thomas
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Re: [Asterisk-Users] Attended call transfer
Does any one know if attended call transfer has been added into the STABLE release of asterisk yet? Any news? I am also looking for #-Transfers for asterisk-stable. Thomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk crashes on special Transfer with MGCP/ATA 186
Hi all, i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco ATA-186 3.1.1 atamgcp We are used to make an special ;) blind transfer like (Flash)Number(Hangup before anyone answers or ring). Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp If one waits until the last one rings, then hangup, everything is fine. If one waits until the last one answers, then hangup, everything is fine, too. Any hints? mgcp debug on: -- Executing AGI(Zap/7-1, nuller.agi) in new stack -- Launched AGI Script /home/kpj/pbx/var/lib/asterisk/agi-bin/nuller.agi -- Accepting call from '01635571857' to '8551' on channel 0/1, span 3 -- AGI Script nuller.agi completed, returning 0 -- Executing Dial(Zap/7-1, MGCP/aaln/[EMAIL PROTECTED]||) in new stack -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: 0, dnd: 0, so: 0, sno: 0 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Called aaln/[EMAIL PROTECTED] -- MGCP/aaln/[EMAIL PROTECTED] is ringing -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP/aaln/[EMAIL PROTECTED] answered Zap/7-1 gw-bzo*CLI mgcp debug on Usage: mgcp debug Enables dumping of MGCP packets for debugging purposes -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf' -- Swapping 1 for 0 on aaln/[EMAIL PROTECTED] -- MGCP Muting 1 on aaln/[EMAIL PROTECTED] -- Started music on hold, class 'default', on Zap/7-1 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '8' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '7' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu' -- Stopped music on hold on Zap/7-1 Oct 15 13:32:58 NOTICE[100377]: chan_mgcp.c:1151 mgcp_fixup: mgcp_fixup(Zap/7-1, Zap/7-1MASQ) -- Swapping 0 for 1 on aaln/[EMAIL PROTECTED] Oct 15 13:32:58 WARNING[14350]: chan_mgcp.c:3033 handle_request: Transfer attempt failed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZapRas problems
Hi I try to use zapras. I am using zaptel-bri-0.0.2 I compiled and patched pppd from ftp://ftp.digium.com/pub/zaptel/misc/ pppd is /usr/sbin/pppd Any idea whats going wrong here? Thomas -- Accepting call from '95' to '8526' on channel 1, span 1 -- AGI Script nuller.agi completed, returning 0 -- Executing ZapRAS(Zap/1-1, debug|64000|noauth|netmask|255.255.255.0|192.168.1.121:192.168.1.122) in new stack -- Starting RAS on Zap/1-1 May 24 16:21:00 WARNING[561180]: app_zapras.c:143 run_ras: wait4 returned -1: No child processes -- RAS on Zap/1-1 terminated with signal 1 == Spawn extension (incoming, 8526, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Budgetone firmware?
Philipp von Klitzing wrote: Hi! I'm still running 1.0.3.81 because I read that once you move up to 1.0.4.x you can't go back again, and my experience isn't *that* crappy. You probably want to start with 1.0.4.26 although also 1.0.4.17 seems to have been relatively stable. Then there is also 1.0.4.39 but it seems to be less popular. Unfortunately many of the firmware files are missing a changelog... Cheers, Philipp Hi, is there any howto available? 1. Where to get firmware-files (cant find anything at grandstream.com) 2. How to configure tftpd (with snom200 my tftp works fine) 3. How to place (naming,config) the files in tftpdir Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with ATA's locking up..
CW_ASN - Gus wrote: You must register with cisco in order to get ata image. I tried, but Cisco (Germany) has no idea how to do this... BTW, my ATAs sometimes cannot make calls. I first have to make a call to one ATA-Extension, wait for the Phone to ring, then i can make calls again. I am using MGCP-Image Version: v2.16.1.ms ata18x (Build 030814a) Thomas PS Has anyone got Transfer with Flash working perfectly? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MWI message not seen on SNOM200
This has been covered before. I think the reason is that asterisk sends notifications from [EMAIL PROTECTED] and pressing the button dials that address. It's not fixed, yet. For this, i have an voicemail-extension for my snoms: exten = asterisk,1,GoTo(8518,1) exten = 8518,1,VoicemailMain2 exten = 8518,2,Hangup Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3x AVM Fritz!Card PCI for a EuroISDN PBX.
David M. Wilson schrieb: Hi there! I'm currently considering various PBX solutions for our office telephone network, and would very much like to use Asterisk. Currently, my research is incomplete. I have been recommended to use the above cards, but it is unclear from my Googling whether my configuration will work: - 3x Fritz!Card PCI's in one host. - 3x 6 b-channels. - ~20 Budgetone (and some others) handsets. Can anyone answer these questions: - Will the 3 ISDN cards function correctly in one host? - Will running all 3 cards flat out require particularly beefy hardware? - Will the Grandstream phones provide a good equivilant to professional dedicated PBX phones? (assuming a good network) I have read lots about echo problems and so on, is this an issue? Any help in the matter would be very much appreciated. Thanks in advance! You can try http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
WipeOut wrote: Ing. Angel Gomez Garcia wrote: WipeOut wrote: Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. Everyone.. :) Its a known issue.. Later.. OOhh :( Any known workaround ? Not that I know of.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Here with a snom200/SIP and ATA-186/MGCP everything works fine (i dial *8 to pick up a call). -Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Asterisk-Dev] Cisco ATA 186 chan_mgcp Transfer problem
Florian Overkamp schrieb:
Hey, if I press Flash asterisk gets the 'hf' event but does nothing.
What gives ? :-)
We can compare our ATA-configs, because transfering works fine with MGCP
(SIP doesnt).
By the way, I'd think maybe it's not actually transferring but rather
'bridging' through the ATA ?
Maybe you can show some config snippets ?
The call seems (for me) to be bridged by *:
gw-bzo*CLI
gw-bzo*CLI show channels
Channel (ContextExtensionPri ) State Appl. Data
SIP/snom1-baef (default 1 ) Up Bridged Call
CAPI[contr1/8504]/22
CAPI[contr1/8504]/22 (macro-stdexten s3 ) Up Dial
SIP/snom1|20|mt
2 active channel(s)
gw-bzo*CLI
Is this complete?
Thomas
mgcp.conf:
[general]
port = 2727
bindaddr = 0.0.0.0
disallow=all
allow=alaw
inbanddtmf=0
transfer = yes
threewaycalling=yes
musiconhold=1
[192.168.1.25]
transfer = yes
threewaycalling=yes
host = 192.168.1.25
context = default
callerid = Thomas 8504
mailbox = 8504
callgroup = 1
pickupgroup = 1
transfer=1
line = aaln/1
context = default
callerid = Thomas 8506
mailbox = 8506
transfer = 1
line = aaln/2
line = *
sip.conf:
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 192.168.1.2 ; Address to bind to
context = default ; Default for incoming calls
disallow=all
allow=ulaw
allow=alaw
allow=gsm
[snom1]
type=friend
secret=snom1
host=dynamic
defaultip=192.168.1.18
context = default
mailbox = 8501
callerid = Thomas 8501
calleridnum = 8501
callgroup = 1
pickupgroup = 1
capi.conf:
;
; CAPI config
;
;
; Multipoint
[global]
mode=immediate
isdnmode=multipoint
;nationalprefix=00
;internationalprefix=000
[interfaces]
msn=8500,8501,8503,8504,8505,8506,8507,8508,8509,8510,8511,8512,8513,8514,8515,8516,8517,8518
incomingmsn=*
controller=1,2
context=isdn
echosquelch=1
softdtmf=0
rxgain=1
txgain=1
devices=2
extensions.conf:
[general]
static=yes
writeprotect=no
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,DBget(temp=UML/${ARG1})
exten = s,2,Goto(default|${temp}|1)
exten = s,102,Goto(s|3)
exten = s,3,Dial(${ARG2},20,mt)
exten = s,4,Voicemail2(u${ARG1})
exten = s,5,HangUp
exten = s,104,Voicemail2(b${ARG1})
exten = s,105,HangUp
[outgroup]
exten = _X.,1,Dial(CAPI/${CALLERIDNUM}:b${EXTEN},,T)
exten = _X.,2,Dial(CAPI/8501:b${EXTEN},,T)
[asterisk]
include = parkedcalls
exten = 8501,1,Macro(stdexten,8501,SIP/snom1)
exten = 8504,1,Macro(stdexten,8501,MGCP/aaln/[EMAIL PROTECTED])
[isdn]
exten = s,1,AGI(nuller.agi)
; nuller.agi adds a leading zero for incoming calls and jumps to context
; asterisk
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Re: [Asterisk-Users] Call transfer on ATA186
Hi all ATA-Users, after a lot of tests, i found the best (not complete working solution). If you use an an MGCP-Image then 1. CLIP-CallerID works fine (with one Phone Callername-transmission works too) 2. Blind transfer with # works fine 3. Attended transfer (Transfer with consultation?) works with incoming and outgoing calls (with Flash). I never tried H323 because of the mammut sources to compile. With SIP there was no way to get all things working. Sometimes the sound (over ATA) is choppy. Then i have to reboot the ATA and everything is fine again (any hints?). - Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mgcp problems
Pavel Zheltouhov wrote: When I connected over two mgcp channels and sending numerical indication to cisco ata it seems hangup one channel (receving ) and generate 'fast busy' tone. I hack chan_mgcp and my threewaycalling works ok! But why indications are sent after I press hookflash on answering end? Is it possible to do this hack in chan_sip? Threewaycalling is the only thing i am missing with ATA/SIP/Asterisk! -or- does ATA/MGCP/Asterisk complete working (CallerID-transfer, MSG-Waiting-Indicator...)? Can you post a chan_mgcp.diff or your modified chan_mgcp? I really like to use threewaycalling with my ATAs. Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] three way calling and cisco ata 186
Pavel Zheltouhov wrote: Ok, if this is not working with sip or h.323, maybe it does with mgcp ? I tried to get ATA and Asterisk working with MGCP, but nothing worked! Any Howtos available about MGCP/ATA186/Asterisk? Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
