[asterisk-users] How do I change the rtp packet size in a Cisco 7490 from 10ms to 20ms
Hello, We have an asterisk system with about 40 cisco 7940/7960 phones and a few linksys SPA941. I recently analyzed our network and discovered that the rtp packet size from the cisco phones is 10ms. We want to change the rtp packet size of the Cisco phones from 10ms to 20ms. I know how to do this on linksys phones and sipura ATAs but I cannot figure out how on the 7940/7960s. Is this possible? Does anyone have suggestions as to how I can do achieve this? Any tip or help will be appreciated. Codec: ULAW SIP firmware: 8.2 Thanks. Buki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1 phone 2 voicemail accounts
Hi all, I have a similar issue. I am looking for a way to make 1 phone to subscribe to 2 voicemail accounts on 2 different Asterisk machines on the same LAN linked over IAX2. Management requested that User1 on Asterisk1 should be able to forward a voicemail message to User2 on Asterisk2. All of our users are using GXP2000 SIP phones and the Asterisk servers communicate over IAX2. I will appreciate any help. Thank you and warm regards, Buki On 1/18/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: -- Forwarded message -- From: Philipp Kempgen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thu, 18 Jan 2007 15:36:47 +0100 Subject: Re: [asterisk-users] 1 phone 2 voicemail accounts [EMAIL PROTECTED] wrote: What is the best way to have 1 phone check multiple voicemail accounts. I am using polycom 650 phones, and am wondering if mwi can work when checking multiple accounts. Try [EMAIL PROTECTED],[EMAIL PROTECTED] ; Subscribe to status of multiple mailboxes in sip.conf Best regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] windows mobile 5 softphone for square screen devices
Subject: Re: [asterisk-users] windows mobile 5 softphone for square screen devices Anton Krall a écrit : Guys, anybody has seen or is using some kind of softphone on any square screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they do work on Wm5 but they are designed for standard screens, anybody using anything on square ones? We are using PPCIAX. -- Daniel Hi, I found this from a google search. I have not tried it. https://www.ssldatas.com/globaliptel/(dv4ivf45q5vnz33azj1g4255)/_Pages/NoFrames/PageBuilder.aspx?content=52de526e3499426c875ed35f72ec935fhttps://www.ssldatas.com/globaliptel/%28dv4ivf45q5vnz33azj1g4255%29/_Pages/NoFrames/PageBuilder.aspx?content=52de526e3499426c875ed35f72ec935f You can also find the x-Lite - Soft SIP IP Phone 1.01 in this link below. I used this with PPC 2003 but I have not tried it with my new WM5.0: http://www.pdastreet.com/software/pdas/X-Lite-Soft-SIP-IP-2003-4-13-pdastreet-pdas.html Regards, Buki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 30, Issue 7
On 1/3/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: -- Forwarded message -- From: Dan Austin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 3 Jan 2007 10:33:02 -0800 Subject: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released We've been holding back on this release to coincide with the Asterisk 1.4.0 release. This is mostly a compatibility release, but there are a few new features: * No longer requires register_globals in PHP * Separated code from configuration settings in ./lib/defines.php (hopefully this will make future upgrades easier) * Migrated all database interfaces to PEAR::DB which simplifies the code a bit and opens up the possibility of using other databases to host the scheduling DB (app_cbmysql is still only MySQL, but ODBC is planned/hoped for) * The conference monitoring code now uses the concise output from meetme list, improving the parsing of participant details. * Minor tweaks to improve the cbEnd.php script that enforces the conference duration, plays announcements and populates the conferencing CDRs. * Conference CDR records now store participant duration in seconds instead of a formatted string, allowing for further analysis (the web interface still formats the duration for display purposes) * App_cbmysql is updated to work with Asterisk 1.4.0 * App_cbmysql has it's own build environment now, no longer requiring a Makefile patch, etc... The new release can be found at: http://sourceforge.net/projects/web-meetme/ We do have a volunteer developer who will be maintaining the 2.X.X chain for Asterisk 1.2.X compatibility, so bug fixes and features that are not Asterisk version dependant will still be made available for older installations. Thanks, The Web-MeetMe development team... HI Dan. I have been using the Web Meet-Me 2.1.0 on Asterisk 1.2.12.1 for many months now and I am a big fan and I have been very happy with it. I want to try the v3.0.0 but I would like to know if there are specific steps I need to carry out to upgrade to the v3.0.0 on my current Asterisk 1.2.X? Warm Regards, Buki (Naija Man) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released
Sorry I forgot to change the subject line in my last posting! I have been using the Web Meet-Me 2.1.0 on Asterisk 1.2.12.1 for many months now and I am a big fan and I have been very happy with it. I want to try the v3.0.0 but I would like to know if there are specific steps I need to carry out to upgrade to the v3.0.0 on my current Asterisk 1.2.X? Warm Regards, Buki On 1/7/07, Naija Man [EMAIL PROTECTED] wrote: On 1/3/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: -- Forwarded message -- From: Dan Austin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 3 Jan 2007 10:33:02 -0800 Subject: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released We've been holding back on this release to coincide with the Asterisk 1.4.0 release. This is mostly a compatibility release, but there are a few new features: * No longer requires register_globals in PHP * Separated code from configuration settings in ./lib/defines.php (hopefully this will make future upgrades easier) * Migrated all database interfaces to PEAR::DB which simplifies the code a bit and opens up the possibility of using other databases to host the scheduling DB (app_cbmysql is still only MySQL, but ODBC is planned/hoped for) * The conference monitoring code now uses the concise output from meetme list, improving the parsing of participant details. * Minor tweaks to improve the cbEnd.php script that enforces the conference duration, plays announcements and populates the conferencing CDRs. * Conference CDR records now store participant duration in seconds instead of a formatted string, allowing for further analysis (the web interface still formats the duration for display purposes) * App_cbmysql is updated to work with Asterisk 1.4.0 * App_cbmysql has it's own build environment now, no longer requiring a Makefile patch, etc... The new release can be found at: http://sourceforge.net/projects/web-meetme/ We do have a volunteer developer who will be maintaining the 2.X.X chain for Asterisk 1.2.X compatibility, so bug fixes and features that are not Asterisk version dependant will still be made available for older installations. Thanks, The Web-MeetMe development team... HI Dan. I have been using the Web Meet-Me 2.1.0 on Asterisk 1.2.12.1 for many months now and I am a big fan and I have been very happy with it. I want to try the v3.0.0 but I would like to know if there are specific steps I need to carry out to upgrade to the v3.0.0 on my current Asterisk 1.2.X? Warm Regards, Buki (Naija Man) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to transfer Voicemail messages between 2 Asterisk servers
I have 2 simple asterisk servers linked over IAX. I want to be able transfer voicemail messages from my phone on Asterisk1 to another extension on the remote Asterisk2 by using the option 8 of the VoiceMail menu (transfer to another extension) ie: to transfer my voice mail messages in mailbox of SIP_PHONE1 on Atserisk1 to mailbox of SIP_PHONE2 on Asterisk2. SIP_PHONE1 --Asterisk1 ---IAX2-- Asterisk2 ---SIP_PHONE2 Asterisk1: v1.2.8 Asterisk2: v1.2.12.1 Any help will be appreciated. Thanks. Buki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help needed - Can anyone please explain to me what is causing this - TDM2400P
Hello all. I have two identically configured asterisk servers each with a TDM2422P (with two S400M FXS modules and two X400M FXO modules). *1 works perfectly and the sound quality is great. However, I am having audio quality problem with *2 when making or receiving calls over the PSTN - the sound gets cut off from the earpiece when I start to speak in the microphone and then when I stop talking, I can hear the other party!! I do not have any problems with SIP calls. Also, if I connect a POTs line to Channel 5 - (WCTDM/0/4 FXSKS) and dial the number, it just keeps ring and the ZAP channel does not answer the call, nor does it even show up in the console. If I call out to the PSTN through ZAP 5, All other channels are working ok. I have tried to recompile zaptel but that did not solve the problem. I even went as far as swapping the TDM2400P cards in the two asterisk servers and the problem still persisted, confirming that the digium card is not faulty. I did not notice any conflicting IRQs either. I would really appreciate help in solving this problem. Asterisk1: centOS 4.3 asterisk 1.2.8 zaptel 1.2.6 Asterisk2: CentOS 4.4 Asterisk 1.2.12.1 zaptel 1.2.9.1 Thanks. Naija Man ** [EMAIL PROTECTED] asterisk]# cat /proc/interrupts CPU0 CPU1 0: 225479540 225426528IO-APIC-edge timer 1: 31 34IO-APIC-edge i8042 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 12: 75 3IO-APIC-edge i8042 14:20273802027327IO-APIC-edge ide0 169: 0 0 IO-APIC-level uhci_hcd 185: 0 0 IO-APIC-level ehci_hcd, uhci_hcd 193: 0 0 IO-APIC-level uhci_hcd 201: 0 0 IO-APIC-level uhci_hcd 209: 162967 162431 IO-APIC-level 3ware Storage Controller 217: 225465762 225426370 IO-APIC-level wctdm24xxp 233:9805871 0 PCI-MSI eth0 NMI: 0 0 LOC: 450936535 450936534 ERR: 0 MIS: 0 ** Below is the output of cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM2400P Board 1 IRQ misses: 1 1 WCTDM/0/0 FXSKS (In use) 2 WCTDM/0/1 FXSKS (In use) 3 WCTDM/0/2 FXSKS (In use) 4 WCTDM/0/3 FXSKS (In use) 5 WCTDM/0/4 FXSKS (In use) 6 WCTDM/0/5 FXSKS (In use) 7 WCTDM/0/6 FXSKS (In use) 8 WCTDM/0/7 FXSKS (In use) 9 WCTDM/0/8 FXOKS (In use) 10 WCTDM/0/9 FXOKS (In use) 11 WCTDM/0/10 FXOKS (In use) 12 WCTDM/0/11 FXOKS (In use) 13 WCTDM/0/12 FXOKS (In use) 14 WCTDM/0/13 FXOKS (In use) 15 WCTDM/0/14 FXOKS (In use) 16 WCTDM/0/15 FXOKS (In use) 17 WCTDM/0/16 18 WCTDM/0/17 19 WCTDM/0/18 20 WCTDM/0/19 21 WCTDM/0/20 22 WCTDM/0/21 23 WCTDM/0/22 24 WCTDM/0/23 *** [EMAIL PROTECTED] asterisk]# lsmod Module Size Used by wcusb 23840 0 wctdm 41280 0 wcfxo 16928 0 wcte11xp 30496 0 wct1xxp20640 0 wct4xxp 251328 0 tor2 93600 0 wctdm24xxp 65344 15 zaptel196740 40 wcusb,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2,wctdm24xxp crc_ccitt 6209 1 zaptel ** [EMAIL PROTECTED] asterisk]# cat zapata.conf ; ; Zapata telephony interface ; ; Configuration file [channels] ; usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no callerid=asreceived callwaiting=no usedistinctiveringdetection=no callwaitingcallerid=no threewaycalling=no transfer=no canpark=no cancallforward=no callreturn=no ;callreturn=yes faxdetect=no echocancel=yes echocancelwhenbridged=no callprogress=no busydetect=no ;busydetect=yes musiconhold=default useincomingcalleridonzaptransfer=yes ;busycount=4 ;group=3 context=from-desks signalling=fxo_ks callerid=CORDLESS 1132 channel = 9 group=1 usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no ;usecallingpres=yes useincomingcalleridonzaptransfer=yes rxgain=8.0 txgain=2.0 context=from-pstn signalling=fxs_ks channel = 1-6 ** tel2*CLI zap show status Description Alarms IRQbpviol CRC4 Wildcard TDM2400P Board 1OK 1 0 0 tel2*CLI zap show status * tel2*CLI zap show channels Chan Extension Context Language MusicOnHold pseudofrom-pstn default 1from-pstn default
[asterisk-users] How do I change the rtp packet size in a Cisco 7490 from 10ms to 20ms
Hi thereI am trying to change the rtp packet size of my Cisco 7940 from 10ms to 20ms. Does anyone know how I can do this.Codec: ULAWSIP firmware: 8.2Bootload ID: PC03A300Thanks.Naija Man ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice mail transfer between 2 asterisk servers
Hi,I have 2 simple asterisk servers linked over IAX. I want to know if it will be possible to transfer my voice mail messages in mailbox of SIP_PHONE1 on Atserisk1 to mailbox of SIP_PHONE2 on Asterisk2.SIP_PHONE1 --Asterisk1 ---IAX2-- Asterisk2 ---SIP_PHONE2 Asterisk1: v1.2.10Asterisk2: v1.2.8Thanks.Naija Man ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Generate Random Numbers in dialplan
-- Forwarded message --From:Alexander Lopez [EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate:Sat, 14 Oct 2006 13:04:08 -0400 Subject:RE: [asterisk-users] Re: Generate Random Numbers in dialplanUse the AGI I sent. It looks like the email did not put a CRcorrectly.Run it from the commandline and see if you get output. The AGI works ok for me. You have to insert a carriage return before the second echo. You also have to remove the single quote inserted after the 5. from -5'` to -5` See corrected script below.#!/bin/shRANDNUM=`echo $RANDOM$RANDOM | cut -c1-5`echo SET VARIABLE asteriskrandom $RANDNUM \\\n ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 27, Issue 49
-- Forwarded message --From:Doug Lytle [EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate:Tue, 10 Oct 2006 16:25:11 -0400 Subject:Re: [asterisk-users] How big is *your* dialplan??Steve Murphy wrote: Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? Sounds interesting. Small facility of 60 users:-= 161 extensions (597 priorities) in 59 contexts. =---Ben Franklin quote:Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. Single server stats, 50 user system,-= 238 extensions (870 priorities) in 57 contexts. =-- Buki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How big is *your* dialplan??
-- Forwarded message --From:Doug Lytle [EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date:Tue, 10 Oct 2006 16:25:11 -0400 Subject:Re: [asterisk-users] How big is *your* dialplan??Steve Murphy wrote: Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? Sounds interesting. Small facility of 60 users:-= 161 extensions (597 priorities) in 59 contexts. =---Ben Franklin quote:Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. Single server stats, 50 user system,-= 238 extensions (870 priorities) in 57 contexts. =-- Buki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no callerid from PSTN using TDM2400P
Thanks. My asterisk servers are in California, USA and the service provider is SBC (ATT).
Asterisk 1.2.8zaptel 1.2.6Hardware: digium TDM2422P
I get the following error messages in /var/log/asterisk/messages:
Oct 3 00:34:18 WARNING[16716] chan_zap.c: Ignoring signalling
Oct 3 00:34:18 WARNING[16716] chan_zap.c: Ignoring signalling
Oct 3 00:34:18 WARNING[16716] chan_zap.c: Ignoring signalling
And the following error messages in my Asterisk CLI:
-- Starting simple switch on 'Zap/3-1'
Oct 3 22:53:14 NOTICE[17948]: chan_zap.c:6061 ss_thread: Got event 18 (Ring Begin)...
Oct 3 22:53:16 ERROR[17948]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-22)
Oct 3 22:53:16 WARNING[17948]: chan_zap.c:6091 ss_thread: CallerID feed failed: Success
Oct 3 22:53:16 WARNING[17948]: chan_zap.c:6135 ss_thread: CallerID returned with error on channel 'Zap/3-1
My Configs are:
zapata.conf:
[channels]
;
usecallerid=yes
restrictcid=no
callerid=asreceived
cidsignalling=bell
cidstart=ring
hidecallerid=no
usecallingpres=yes
sendcalleridafter=2
ringtimeout=8000
callwaiting=no
usedistinctiveringdetection=no
callwaitingcallerid=yes
threewaycalling=no
transfer=no
canpark=no
cancallforward=no
callreturn=no
;callreturn=yes
faxdetect=no
echocancel=yes
echocancelwhenbridged=yes
callprogress=yes
busydetect=yes
musiconhold=default
useincomingcalleridonzaptransfer=yes
group=1
context=from-pstn
signalling=fxs_ks
channel = 3
;channel = 1-3
extensions.conf:
[from-pstn]
;
; Inbound calls from PSTN line
exten = s,1,NoOp(TIMESTAMP: ${TIMESTAMP})
exten = s,2,NoOp(CONTEXT: ${CONTEXT})
exten = s,3,NoOp(CALLERIDNUM: ${CALLERID(number)})
exten = s,4,NoOp(CALLERIDNAME: ${CALLERID(name)})
exten = s,n,Goto(main-ivr,start,1)Thanks.-- Forwarded message --
From:Eric \ManxPower\ Wieling [EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comDate:Wed, 04 Oct 2006 19:32:30 -0500Subject:Re: [asterisk-users] no callerid from PSTN using TDM2400PNaija Man wrote: Hello all, Asterisk 1.2.8
zaptel 1.2.6 Hardware: digium TDM2422P I have a fully configured asterisk system with POTS line for PSTN access. I am not receiving the callerid for incoming calls from the PSTN. I get the
following error message. -- Starting simple switch on 'Zap/3-1' Oct 3 22:53:14 NOTICE[17948]: chan_zap.c:6061 ss_thread: Got event 18 (Ring Begin)... Oct 3 22:53:16 ERROR[17948]:
callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-22) Oct 3 22:53:16 WARNING[17948]: chan_zap.c:6091 ss_thread: CallerID feed failed: Success Oct 3 22:53:16 WARNING[17948]: chan_zap.c:6135 ss_thread: CallerID
returned with error on channel 'Zap/3-1It would be helpful to know what country you are in.
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[asterisk-users] no callerid from PSTN using TDM2400P
Hello all,Asterisk 1.2.8zaptel 1.2.6Hardware: digium TDM2422PI have a fully configured asterisk system with POTS line for PSTN access. I am not receiving the callerid for incoming calls from the PSTN. I get the following error message.
-- Starting simple switch on 'Zap/3-1'Oct 3 22:53:14 NOTICE[17948]: chan_zap.c:6061 ss_thread: Got event 18 (Ring Begin)...Oct 3 22:53:16 ERROR[17948]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-22)
Oct 3 22:53:16 WARNING[17948]: chan_zap.c:6091 ss_thread: CallerID feed failed: SuccessOct 3 22:53:16 WARNING[17948]: chan_zap.c:6135 ss_thread: CallerID returned with error on channel 'Zap/3-1My configuration is as below:
***zapata.conf:[channels];usecallerid=yesrestrictcid=nocallerid=asreceivedcidsignalling=bellcidstart=ringhidecallerid=nousecallingpres=yes
sendcalleridafter=2ringtimeout=8000echocancel=yesechocancelwhenbridged=yescallprogress=yesbusydetect=yesmusiconhold=defaultuseincomingcalleridonzaptransfer=yesgroup=1context=from-pstn
signalling=fxs_kschannel = 1-3extensions.conf:[from-pstn];; Inbound calls from PSTN lineexten = s,1,NoOp(TIMESTAMP: ${TIMESTAMP})exten = s,2,NoOp(CONTEXT: ${CONTEXT})
exten = s,3,NoOp(CALLERIDNUM: ${CALLERID(number)})exten = s,4,NoOp(CALLERIDNAME: ${CALLERID(name)})exten = s,n,Goto(main-ivr,start,1)***
The variables $CALLERID(number) and $CALLERID(name) always show up empty when a call is received.Any suggestion will be appreciated.Thanks.
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Re: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN
-- Forwarded message --From:William Piper [EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate:Sat, 30 Sep 2006 22:08:23 -0400 Subject:Re: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LANSure sounds like a firewall issue... if you pinging port 4069 and it is not coming back, that sounds like a firewall problem. Try taking down your iptables and then try see what happens. bp On 9/28/06, Wolfgang_Borgon [EMAIL PROTECTED] wrote: David,Yes, I've also forwarded port 4569 to the server.Since the router is forwarding to the server, I cannot forward it to the client as well -- however, as theclient isn't going out past the LAN, it shouldn'tmatter... unless there's something else going on thatI don't know about.ThanksWolfgang--- David J Carter [EMAIL PROTECTED] wrote: Wolfgang wrote:- I've already sunk several hours into this without any real progress, so I'd really appreciate any helpMy task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569 with iptables, and forwarded 4569 to the server IP.I am not, however, having any luck connecting the softphone to the server. I can telnet, ftp, and http to the server, but not IAX2. Iaxping times out, registration by Idefisk and Firefly also times out. The server fails to see the client as well. Here's a portion of my iax.conf: [client] type=friend username=client secret=** host= 192.168.1.40 context=clientcon and extensions.conf: [clientcon] exten = 2278,1,Dial(IAX2/client)== You say you have 4569 configured in iptables, what about the netgear router? Have you port forwarded 4569 there? DaveIn your iax.conf, instead of...host= 192.168.1.40Use...host=dynamic-Buki - Da Naija Man ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dial-9 (was Extension Numbering)
-- Forwarded message --From:Jay R. Ashworth
[EMAIL PROTECTED]To:asterisk-users@lists.digium.comDate:Sat, 30 Sep 2006 14:35:49 -0400
Subject:[asterisk-users] Dial-9 (was Extension Numbering)
On Sat, Sep 30, 2006 at 01:40:09PM +0100, Gordon Henderson wrote: Here in the UK, I've installed several small systems without a dial-9 for an outside line type thing. The outside line prefix is effectively digit
zero. (which is preserved and dialled on the outgoing zap lines) There is an exception for 999, and I still provide the 9 service too forThis reminds me of something that's bothered me for years, and I'm
curious how people deal with it. This is semi-US specific; don't sayyou weren't warned (or that I'm Americo-centric :-).Using 9 as a dialplan prefix for accessing outside dialtone has one*major* problem: 911.
You don't *really* want to (and I believe, legally, you can't) requirepeople to dial 9-911. But, this leads you to an alternate problem.If you define 911 in your internal dialplan as a cut-through to dial the
local PSAP over a standard local voice line (and here, I'm assuming youhave some; VoN 911 is a topic I entirely don't want to get into at themoment), then eventually you're going to have either a) a touchtonetm
dial that stutters on it's 1 key, or b) a human who does it, and they'regoing to dial 9-1-800-555-1212, and find themselves talking the EMSinstead of directory assistance... and no one will understand why...
and the EMS people will be mad at *you*.I know that this has been a problem for traditional PBXen for years,and the only solution I've ever been able to see is use 8 as youroutdial prefix... but no one seems to ever do that, even 20 years on.
Is this really not a problem?Cheers,-- jra--Jay R. Ashworth
[EMAIL PROTECTED]Designer Baylink RFC 2100Ashworth AssociatesThe Things I Think'87 e24St Petersburg FL USA
http://baylink.pitas.com +1 727 647 1274That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_
As a habit, I do not force users to dial 9 or any other prefix of any kind to access external lines. You can just check the dialled number and prefix with appropriate digits appropriately. See below. NOTE: THIS IS US-CENTRIC!!
but can be easily made to work for any country.[context for out-bound numbers];;Check for incoming calls…;; Domestic e.164 US numbers go out unchanged.;exten = _1XX,1,Goto(outgoing,${EXTEN},1)
;;; Check for 10 digit NANP
exten = _XX,1,SetVar(PREFIX=1)exten = _XX,2,Goto(outgoing,${PREFIX}${EXTEN},1); Check for 7 digit NANP; Then add 1 and the location area code;exten = _XXX,1,SetVar(PREFIX=1925)
exten = _XXX,2,Goto(outgoing,${PREFIX}${EXTEN},1); Check for emergency numbers;exten = 911,1,Goto(emergency,911,1);; Check for other special numbers and direct to repective contexts.
;;[outgoing]exten = _X.,1,ChanIsAvail(${PSTNCHANNEL})exten = _X.,2,NoOp(AvailChannel=${AVAILCHAN})exten = _X.,3,Set(DialChannel=${CUT(AVAILCHAN,,1)})
exten = _X.,4,Dial(${DialChannel}/${EXTEN},100)exten = _X.,5,Congestionexten = _X.,105,CongestionHope this helps.
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[asterisk-users] Re: Voip Buster - CID
You can try VoipJet (http://www.voipjet.com)A simple configuration in you extensions.conf as below will solve your problem.exten = _X.,1,SetCIDNum(1341212)
exten = _X.,n,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})- Buki
-- Forwarded message --From:Tomislav Parčina [EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comDate:Thu, 28 Sep 2006 08:55:33 +0200Subject:[asterisk-users] Re: Voip Buster - CIDIn article
[EMAIL PROTECTED], [EMAIL PROTECTED] says... There are not many that will allow you to set your own CID even then they
normally ask for proof of the numbers you wish to use.Hi Chris!So, you are saying that I can't set outgoing CID number on Voip Buster? Do you know for any VoIP provider that allows that?
--Tomislav ParčinaLama Computers SplitStinice 12, 21000 SplitTel.: +385(21)495148Mob.: +385(91)1212148SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hrhttp://www.lama.hr
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[asterisk-users] How to detect dial tone on ZAP channel before dialling using TDM2400P
Hello allI have an asterisk box running Asterisk 1.2.8 and I installed a digium TDM2400 with 8 FXO ports. When I amke a call to the PSTN, the zap channel answers, and teh call goes through if a PSTN is connected to the answered port. However, if there is no dial tone in the answered channel, or if no POTS line is connected, the user gets no indication until the call time outs. I want * to be able to detect if there is a dialtone on the channel, before it dials, if not, to send a busy signal or choose another available channel. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
