Re: [Asterisk-Users] Asterisk and CAS
what about CAS 3 Bit? does * support it? thanks, Paradise Dove On 4/8/05, Steve Underwood [EMAIL PROTECTED] wrote: David Hajek wrote: Hi, is it possible to use Asterisk with T110P and CAS (channel associated signalling)? There are hundreds of CAS protocols. Quite a few currently work with the T110P. Regards, Steve ___ 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 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] Grandstream BT Volume
Guys. Anyway had problems with G BT 100 or 101 volume? Seems the volume is too loud and when talking it makes the voice cut off due to saturation. Anyway to reduce the input voice volume on the phones? ___ 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
Re: [Asterisk-Users] ISDN BRI and signalling
On Fri, 15 Apr 2005, Stefan Gofferje wrote: Bob van der Moezel schrieb: I want to signal BUSY condition to a bristuffed HFC-S ISDN line. However: exten = s,1,Busy has no effect, exten = s,1,Playtones(Busy) is not audable over unanswered line (I live in the Netherlands...) So I currently do: + exten = s,1,Answer + exten = s,2,Playtones(Busy) + exten = s,3,Busy Which obviously is not an ideal solution. Is there a way to get real signalling out a bristuffed HFC-S ? I have the same problem with a Fritz!PCI and chan_capi as external line. I have a HFC-S card in NT mode as internal ISDN. On this card, Busy and Congestion do work. As both drivers came from Junghanns, I suppose, this is a problem with the drivers... In HEAD this is accomplished through the variable PRI_CAUSE and the Hangup application. This is the most generic way. Read more in the wiki or search the mail archives. Additionally the behviour of the Busy and Congestion applications can be changed by setting oob_indication in the zapata config file. Peter ___ 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
Re: [Asterisk-Users] Has anyone had problems with Digium TDM400P and hyperthreading?
On Friday 15 April 2005 07:45, Damian Funnell wrote: Digium have told us that a problem that we are having (with accuracy of zap interface as measured using zttest) may be due to the fact that we have a Xeon processor with hyperthreading and have suggested turning H/T off. Anyone else experienced a problem like this? No too keen about turning H/T off, as we're running the SMP RH kernel and don't really feel like replacing the kernel (and other kernel-specific bits) on the off chance that H/T is actually the problem. AFAIK, HT is not useful if your application does a lot of floating point calculations as there is only one FPU. What happens is that there is a 1-2% overhead when HT is on (for semaphores and such). IF HT does not gain more than this 1-2% you actually end up slowing things down and this is the case for floating point applications (because of only one FPU). IS asterisk floating point heavy ? Paul H -- Paul Hewlett (Linux #359543) Tel: +27 21 852 8812 Cel: +27 72 719 2725 Fax: +27 86 672 0563 -- ___ 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] OctoBRI - unable to specify channel 1
Hi Guys I have installed * with an OctoBRI card. The card laods fine without and errors, ut when I start * I get: == Parsing '/etc/asterisk/zapata.conf': Found Apr 15 09:41:59 WARNING[9893]: chan_zap.c:924 zt_open: Unable to specify channel 1: No such device or address Apr 15 09:41:59 ERROR[9893]: chan_zap.c:6460 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Apr 15 09:41:59 ERROR[9893]: chan_zap.c:10247 setup_zap: Unable to register channel '1-2' Apr 15 09:41:59 WARNING[9893]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Apr 15 09:41:59 WARNING[9893]: loader.c:440 load_modules: Loading module chan_zap.so failed! I have googled for an answer but dont find anything specific. Please can someone just ,let me know where I am going wrong. Thanks Kind Regards Terry Wade Mobile: +27 82 802-5750 Office: +27 11 784-7642 Fax: +27 11 388-0855 Linux is like a Wigwam - No gates, no windows, Apache inside Disclaimer and Confidentiality Warning This message is intended for the addressee only. If you are not the intended recipient of this message, you are notified that any distribution, use of or copying of this communication is strictly prohibited. If you have received the communication in error, please notify the sender immediately. The views and opinions expressed in this message are those of the individual sender of this message and do not necessarily represent the views and opinions of ActiCom. Consequently, ActiCom does not accept responsibility for such views and opinions and this message should not be read as representing the views and opinions of ActiCom without subsequent written confirmation. Each page attached hereto must also be read in conjunction with this disclaimer. ___ 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
Re: [Asterisk-Users] Pre-install questions
Main problem is transcoding so if you're using the same on both sides of a call e.g. G711 then you can run it on a Linksys Router!. If you didn't get replies I suggest you repost with more informative subject. I'm sure some gurus know the answer to your questions. Regards Cameron - Original Message - From: Mike Myers [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, April 12, 2005 10:43 AM Subject: [Asterisk-Users] Pre-install questions Hi. I am considering building an asterisk system for home use, but I am having some trouble understanding a few things after reading the wiki and the various mailing lists. Let me apologize in advance if I have missed something... 1) What are the hardware needs to run asterisk reliably? It appears from the various docs that you don't need much CPU to run asterisk. Assuming I pick a hardware platform with good linux driver support, how much memory do I need? Would an old 700 Mhz P3 be adequate? I have seen examples for much more complicated configurations, but they don't give me much guidance about memory needed and CPU needed for a home installation. 2) Is it possible to take audio line in from a sound card and have the audio piped into a meetme conference? I see from the perl script on the paging and intercom wiki page that it appears you can take audio out (fed to a paging system in this context) and patch it into meetme conference. Can I do the same thing except with audio in? There are two applications for this for me. One I use with my existing nortel venture phone system - I take audio from a PC running replay radio (a PC internet radio recoding system) and patch it into the venture's EFA to make background music available to the phones in the house. That way, my wife can listen to a dr. laura broadcast, or other radio program on any phone in the house. Most of this programs don't seem to stream properly into linux, but in any case, I have an audio source already available and would like to use it. The 2nd application is to take the output of a baby monitor and hook it into a meetme conference (again though line in on a sound card). This would allow my wife and/or I to listen to the monitor from any phone in the house. This would allow us to both listen into the audio from the monitor without having to find the monitor and keep it equipped with fresh batteries, etc... It seems like the ability to feed in line-in could solve both problems for me. 3) Is there any way for Asterisk to take external state from the network to inform it's decision making? For example, we have an enclosed garage, and sometimes we take out 1.5 year old for a drive to get her to go to sleep, and leave her in the car seat in the car in the garage to nap (with the doors open for ventilation!). We leave the door from the garage into the house open so we can hear her. Our alarm system tells our automation system (homeseer), that that door is open, and homeseer can then de-energize the garage doors so if I come home from work early, I don't wake up the baby. What would be great is if Asterisk could detect from homeseer that the garage door was open, and not ring the downstairs phones when the door was open (indicating our baby was sleeping). Similarly if she is sleeping in the nursery upstairs, and that door was closed, asterisk would not ring the phone in that room. Anyone have an idea as to how to effect such behavior? Thanks in advance, and my apologies if this info is available in an existing document. Thanks, Mike __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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 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
Re: [Asterisk-Users] Siemens optiPoint 420 phone and Asterisk
Franz Knipp [EMAIL PROTECTED] wrote: Hi,today I've got two Siemens optiPoint 420 phones and I want to connectthem to an existing Asterisk server.I didn't find any SIP firmware for that phone, according to announcementsit will be released later this year (hopefully soon). The latest firmware for optipoint420 advance SIP seems to be version 4.0.22A, released for HiPath8000. Unfortunately on the Siemens page theonlySIP image that can be downloadedis for OptiPoint400 (www.hipath.de then -download - software/version 2.3.14). For Optipoint410/420 only the HFA version is available. chan_cornet, which would support the proprietary Siemens protocol, is notpart of the CVS tree yet (thanks to Steffen Koepf for writing this). Is Steffen's chan_cornet available for testing? We have * connected to a HiPath4x00 using oh323. Maybe, someone of you can help me getting this phones working withAsterisk by pointing out a good starting point for my investigation andown development (if necessary) ;-)Last but not least, some kind of network diagram to clarify the situation:ISDN NT ---[Siemens PBX]--(S0)--[Asterisk]--(IP)--[optiPoint] What siemens PBX do you use? Does anybody know, if it is worth trying out the optiPoint 400 SIPfirmware on the 410/420 phones? I don't know... maybe it will work... We only have several OptiPoint400 and they work fine. Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard.___ 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] A-Z Termination
Hello I am looking for A-Z termination please send me your prices off-line. Protocols: SIP, IAX Codecs: G723, G729, GSM Regards Mehdi Chouikh Universal Telecom www.unitelexperts.com Tel: +34 902023154 ___ 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] Soekris net4801 usb isdn avm fritz
Hello Asterisk on soekris 4801 + 2,5 hard drive work... It was good to make a small office or home use pbx. But their is a problem with USB isdn fritz card. If sommeone can help me to use chan_capi and usb fritz card... I have dmesg message like this: usb 1-1: capiinit timed out on ep1out fcusb2: USB I/O error, code -110! fcusb2: Firmware does not respond! kcapi: card 1 down. I'm using Fedora Core 2 with kernel 2.6.10-1.771_FC2 (i've test with 2.4.22 with no luck...) Thank's... ___ 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
Re: [Asterisk-Users] cisco 7960 SIP setup
mk111 wrote: I was told that the phone should be able to download the SIP... file once the TFTP address was changed. So far nothing though. Any ideas? have you rebooted the phone after changing the tftp address ? -- Simone Cittadini IT Manager == COMVERT S.R.L. via F.lli Bressan, 21 20126 Milano - ITALY Tel +39.02.27006796(aspetta un beep)105 [EMAIL PROTECTED] http://www.comvert.com ___ 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
Re[2]: [Asterisk-Users] About Audio Latency from PSTN to SIP
Thanks. I tried your suggestion, and it make no use. --- Best regards, Qiao Yuansong mailto: [EMAIL PROTECTED] Friday, April 15, 2005, 10:21:16 AM, you wrote: I'm Andrew. On April 14, 2005 10:01 pm, Qiao Yuansong wrote: My asterisk box and sip phone are not behind a nat, the sip phone and asterisk box are connected by LAN, so the delay is not caused by network congestion, and furthermore, there is no delay from sip to pstn. [sip phone]--LAN--[Asterisk with X100P]--[PSTN] sip to pstn (no delay) pstn to sip (half or one second delay) This doesn't make any sense; the streams are identical. Are different codecs being negotiated when the call origination is one side then the other? put disallow=all allow=ulaw in sip.conf, under [general] and comment out all other allow/disallow lines. Restart asterisk and try again. Something basic is not right. -A. ___ 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 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
Re: [Asterisk-Users] codec introducing huge latency
--- Andrew Kohlsmith [EMAIL PROTECTED] wrote: On April 14, 2005 06:34 pm, chawki hammoud wrote: communications. ulaw is about 80kbps, and gsm about 28-30kbps. I monitored the download and upload data rate during my call using mandrake linux and it gave me 9.3 kb/s using ulaw and 3.1 kb/s for gsm. I think i had a download problem when i used ulaw, while i had enough for upstream all the time, it wasn't the case in the download. It really sounds like you need to review your network. My current isp is one man company and has a lan internet sharing setup. i already placed an order last week with a better isp company and with higher bandwidth and routing. ___ 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 __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ 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 PBX with X100P in India
I'm currently trying to set up an Asterisk PBX system in India. However I'm having trouble configuring the X100P to dial out on the POTS line. Does anyone have any knowledge about this? I know the telephone system is a bit different in India, so would the X100P not be suitable? Is there a change I need to make in the Zaptel.conf or zapata.conf? Because I'm using [EMAIL PROTECTED] 0.6, the extensions.conf is also pretty frustrating... Any help here would be appreciated. ___ 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] OH323 and outgoing calls problem.
Hello,
i have just installed OH323 and configured all outgoing calls from sip
softphones, sip context in extensions files is:
[sip]
exten = _.,1,Dial(OH323/${EXTEN})
this is only one in this context, all softphones uses this context.
After call system trying to cal h :O
It looks:
-- Registered SIP '111' at 195.XXX.XXX.XXX port 5060 expires 1800
-- Saved useragent X-Lite release 1103m for peer 111
-- Executing Dial(SIP/111-3d65, OH323/4812XXX) in new stack
-- H.323 call to 4812XXX with codec(s) XX
-- Called 4812XXX
-- OH323/48122863865-70bc is ringing
-- Hungup 'OH323/4812XXX-70bc'
== Spawn extension (sip, 4812XXX, 1) exited non-zero on 'SIP/111-3d65'
-- Executing Dial(SIP/111-3d65, OH323/h) in new stack
-- H.323 call to h with codec(s) XX
-- Called h
-- Hungup 'OH323/h-1357'
== Spawn extension (sip, h, 1) exited non-zero on 'SIP/111-3d65'
-- H.323 call 'ip$localhost/27188' cleared, reason 1 (Cleared by local user)
-- H.323 call 'ip$localhost/27189' cleared, reason 1 (Cleared by local user)
And what is:
-- Called h
-- Hungup 'OH323/h-1357'
== Spawn extension (sip, h, 1) exited non-zero on 'SIP/111-3d65'
??
On GK displays:
ACF|195.XXX.XXX.XXX:1720|3429_endp|27190|4812XXX:dialedDigits|X:dialedDigits=111:dialedDigits|false;
ARJ|195.XXX.XXX.XXX:1720|h:h323_ID|X:dialedDigits=111:dialedDigits|false|calledPartytRegistered;
DCF|195.XXX.XXX.XXX|3429_endp|27190|normalDrop;
What is the ARJ packet?
The same problem I see in this mail:
http://lists.digium.com/pipermail/asterisk-users/2005-April/098884.html
im using
asterisk-oh323-0.7.2-pre1
openh323-v1_13_5-1
pwlib-v1_6_6-1
Maybe this is configuration problem but there is no other extensions inx sip
context.
Thanks,
Adam
My oh323.conf:
My Oh323.conf:
[general]
listenAddress=ALL
listenPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=yes
h245Tunnelling=yes
h245inSetup=yes
inBandDTMF=yes
jitterMin=20
jitterMax=100
outboundMax=100
inboundMax=100
simultaneousMax=200
wrapLibTraceLevel=9
libTraceLevel=9
libTraceFile=/tmp/oh323_debug.log
gatekeeper=195.XXX.XXX.XXX
gatekeeperTTL=300
userInputMode=TONE
amaFlags=default
accountCode=H323
musionhold=default
context=voip-h323
[register]
alias=ASTERIX
prefix=*
[codecs]
odec=GSM0610
frames=4
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Re: [Asterisk-Users] ISDN BRI and signalling
The problem of Q931 SETUP I have also depends on Junghanns and they do not support in any way the products the sell... - Original Message - From: Stefan Gofferje [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 15, 2005 8:30 AM Subject: Re: [Asterisk-Users] ISDN BRI and signalling Bob van der Moezel schrieb: I want to signal BUSY condition to a bristuffed HFC-S ISDN line. However: exten = s,1,Busy has no effect, exten = s,1,Playtones(Busy) is not audable over unanswered line (I live in the Netherlands...) So I currently do: + exten = s,1,Answer + exten = s,2,Playtones(Busy) + exten = s,3,Busy Which obviously is not an ideal solution. Is there a way to get real signalling out a bristuffed HFC-S ? I have the same problem with a Fritz!PCI and chan_capi as external line. I have a HFC-S card in NT mode as internal ISDN. On this card, Busy and Congestion do work. As both drivers came from Junghanns, I suppose, this is a problem with the drivers... Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ 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] E1 PRI: Unable to set channel to linear mode?
Hi all,
I have some E1 lines plugged into a digium TE410p and all looks happy
but when I try to make an outgoing call I get this error message:
Apr 15 09:28:21 WARNING[9040]: chan_zap.c:3901 zt_read: Unable to set
channel 31 (index 0) to linear mode.
I think linear mode is an audio format - like ulaw / alaw? but I'm
not sure Has anyone come across this?
Thanks very much,
Derek
PS I'm in Ireland with EuroISDN
PPS When I do try to make a test call (and get the error message above)
the called party receives the call and, on answering, hears static (it
must be loud too - it was described as a brain damaging noise by my
poor test subjects :) ).
My /etc/zaptel.conf has:
span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
bchan=1-15,32-46,63-77,94-108
dchan=16,47,78,109
bchan=17-31,48-62,79-93,110-124
And my /etc/asterisk/zapata.conf has:
[channels]
context=PRI-NTL
switchtype=euroisdn
signalling=pri_cpe
group=1
channel = 1-15,17-31
;group=2
channel=32-46,48-62
;group=3
channel=63-77,79-93
;group=4
channel=94-108,110-124
And my test dial line is:
exten = _011353.,1,Dial,Zap/1-1/${EXTEN}
exten = _011353.,2,Busy
exten = _011353.,102,Busy
--
Derek Conniffe
Rivertower Ltd
DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146
Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823
Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180
Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085
Email: [EMAIL PROTECTED]
Web: www.rivertowerhosting.com
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[Asterisk-Users] LiveVoip incoming, no ringback still
I see in the list archives that this problem came up before, but there was no fix for it. Any clues now? Inbound calls from LiveVoip work (I am assuming they will soon fix their packet loss issues at the San Diego pop) except for one thing -- no ringback when the called extension is ringing. My inbound context gives the caller a message and DISA dialtone, DTMF digits decode OK, the extension rings and if the call is answered, it works. If it's not answered, voicemail works. But no ringback tone to the caller. Other incoming SIP and IAX calls get ringback. If I set DIAL_OPTIONS to m, the caller gets music on hold during the ring interval, but setting it back to r yields silence. ___ 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] UDP Sip Data: GS Grandstream - remote office
Hello all, Tried to get remote office working and found out that the GS Budge Tone 100 takes the ip address inside the UDP packet data (SIP) that asterisk writes to.The Asterisk server is currently setup with a ADSL ZyXEL PRESTIGE 600 series router. My isp does dynamic ip assignment - so I cant tell asterisk what to write down in the sip packet as the ip changes on every reboot / day. I Have tried using a sip proxy and this caused only more confusion (sip proxy on same box as asterisk/ sip port forwarding and so on - messes up the LAN configuration). So then I decided whether to use rp-pppoe or debians pppoeconf to make the Eth show the public ip address so that the GS Budge Tone 100 can then take the SIP ip address and then get is routed correctly accross the internet to the asterisk server. Did the whole thing only to realise that I have no idea if the router/modem is configured for bridge. So I spent the lat day and a half trying to get connected with the modem (even in windows XP) and still no connection. So what I am asking is, has anyone had a similar issue. How did you solve it... and if anyone would like to help me just a bit (googled and still nothing) to allow the asterisk box to be awaire of the public ip. -- Kind Regards Etienne ___ 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] *8 nor *8# works for me!
I have put into each phone settings (sip.conf and zapata.conf) in my office: callgroup=1 pickupgroup=1 I cannot pickup any calls from another phone!! What do I miss here? bye Ronald ___ 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] Urgent .... Asterisk - Cisco CCM SIP TRUNK
Hi All, I'm getting a strange problem with asterisk 1.0.6. I've got a SIP Trunk between CCM 4.1 and Asterisk 1.0.6. We are talking about a daily average of 900 calls, and 600 minutes. I'm running in asterisk 3 queues, 2 of them with dynamic members (chan_local), and one with only one static member, and a Voicemail system. Since yerstaday (i wonder why?), and only sometimes, when a call enter a queue, it starts ringing a member phone, and when he tries to pickup the call it jumps to other queue member, and when the other queue member pick up the call, it jumps again to other queue member they are getting crazy until one of thems catchs the call. When it happens, in the next seconds if i try to dial to asterisk, CCM gives-me a busy tone, but nothing reachs asterisk box, i dont now why CCM's gives busy tone without sending the call to asterisk !!! Is this a CCM feature ? All the phones used are cisco (skinny) attached to CCM. The only thing that appears on logs is: file.c:550 in ast_readaudio_callback: Failed to write frame. Thanks in advance Regards João Amaro ___ 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] qos test
Good day all I'm looking for a type of QOS test tool(software) I want to test if a link is good enough for voip and test witch ones will be the best..ens any ideas ___ 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] How to avoid CTL file request for Cisco 7970
Hi All! We buy Cisco 7970 and have a problem. At startup phone requests CTL file and hangs with message Updating CTL. Nothing more happens after it. I tried to place dummy file with the name CTLSEPmac addr.tlv to TFTP root directory. Cisco found it and then message changed to CTL update failed and startup process stops. Some people here told that they can succesfuly register this phone on Asterisk. What can I do to avoid this problem? ___ 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] Empty voicemail attachments?
I have Asterisk newly setup. When a voicemail is left, Asterisk emails out the voicemail as a wav file attachment (in voicemail.conf, append=yes) but the attachment always ends up having a size of 0 bytes and no content to play. Diagnosis steps taken so far: I used Kmail to manually send myself a local email with an attachment and that came through whole. I looked at the wav files stored in the spool directories and they are complete and playable as expected But the content of the file doesn't make it through email. Any ideas anyone can offer as a possible cause of this problem? Thanks a mil voicemail.conf: format=WAV|wav|gsm append=yes [EMAIL PROTECTED] [default] 222 = 1234,Joe,[EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] cannot dial two phones using zap
On Fri, Apr 15, 2005 at 11:04:43AM +0800, Eddie wrote: I do understand how Dial works, but Zap/4 hungup immediately before Zap/3 is answered. Zap/4 doesn't even rings. So the Panasonic extension dialed by Zap/3/206 command will ring and Zap/4/221 will not ring at all, even before ext 206 is picked up? Sorry I didn't mention about this earlier, 206 221 are extensions connected to a Panasonic KX-TD1232 pbx. I missed that in your zapata.conf snipped. I have two extensions 211 212 connected to my TDM400p FXO ports. If you have two extensions numbered 211 212, why are you using 206 and 221 in your Dial command? I would try this: 1. Make sure either extension will ring all by itself. 2. Ring both at the same time, but put them in the other order in the Dial() command and see if that makes a difference. 3. Rather than having: channel = 3,4 try channel = 3 channel = 4 just for fun. 4. I don't know much about that Panasonic PBX, but are you sure calling two lines at the exact same time isn't messing it up? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] Analogue phone transfering
Hi guys, How are you keeping ? I have an analogue phone plugged into a Digium FXS Zap module on my TDM card. The phone works well except that I cannot seem to transfer calls using the "flash" key. I don't seem to get another dialtone as indicated in: http://www.voip-info.org/wiki-Asterisk+tips+zap+transfer Any ideas what I've done wrong ? This is my zapata.conf: [channels]; For analogue phone signalling=fxo_kscontext=defaultchannel=4relaxdtmf=yesthreewaycalling=yestransfer=yesadsi=nousecallerid=norxgain=70.0txgain=50.0 Thanks in advance. Kindest regardsDavid Wilson___D c D a t aTel +27 33 342 7003Fax +27 33 345 4155Cell +27 82 4147413http://www.dcdata.co.za[EMAIL PROTECTED]Powered by Linux, driven by passion ! ___ "Computers are not intelligent. They only think they are." ___ 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 live chat problem
Hello, I'm trying to join the #asterisk channel on irc.freenode.net but when I try to join the channel I receive the following error message: You need to be identified to join that channel Anyone can help me? Thanks Stefano. ___ 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 live chat problem
Hello, I'm trying to join the #asterisk channel on irc.freenode.net but when I try to join the channel I receive the following error message: You need to be identified to join that channel Anyone can help me? Thanks Stefano. ___ 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] IAX2 to IAX2 - one way audio
Hi, everybody. I have schema: H323(cliens)--GNUGK-- | | SIP(cliens)---Asterisk(B)---(IAX2 TRUNK)---Asterisk(A)--(H323 Peer)--PBXPSTN | | SCCP(client)--- ASTERISK(B) - Linux + Asterisk-1.0.6 (IP-addressing INT-192.168.1/24 EXT-62.XXX.XXX.128/25) ASTERISK(A) - FreeBSD + Asterisk CVS-HEAD-04/14/05-21:00:38, Copyright (C) 1999 - 2005 Digium. (IP-addressing INT-192.168.100/24 EXT-62.YYY.YYY.192/27) When I call from PSTN to any clients - call is Ok! When I call from SCCP to any - call is Ok! When I call from H323(clients) to any - call is Ok! When I call from from SIP to PSTN - I have one way audio (PSTN-subscriber can hear SIP-client, at the same time SIP-client doesnt hear PSTN-subscriber). h323.conf ; The NuFone Network's ; Open H.323 driver configuration ; [general] port = 1720 bindaddr = 0.0.0.0 tos=lowdelay amaflags = billing accountcode=H323 disallow=all allow=all ; turns on all installed codecs ; dtmfmode=rfc2833 ; gatekeeper = DISABLE ; AllowGKRouted = no ; context=default ; ; H.323 Alias definitions ; noFastStart = no noH245Tunneling = yes noSilenceSuppression = yes [VIP-101T] type = user host = 192.168.100.55 incominglimit=1 context = to-pstn noFastStart = no noH245Tunneling = yes noSilenceSuppression = yes disallow = all allow = g729 allow = alaw allow=ulaw [avaya] type = peer host = 192.168.100.8 noFastStart = no noH245Tunneling = yes noSilenceSuppression = yes disallow = all allow=alaw iax.conf host A ; Inter-Asterisk eXchange driver definition ; [general] bindport=4569 bindaddr=0.0.0.0 ; amaflags=billing ; accountcode=IAX2 ; language=ru ; bandwidth=low ; allow=all ; jitterbuffer=no ; register = chita:[EMAIL PROTECTED] ; authdebug=no ; tos=lowdelay ; [chara] type=friend host=62.XXX.XXX.129 secret=secret sendani=yes qualify=yes context=to-pstn canreinvite=yes trunk=yes accountcode=CHARA disallow=all allow=g729 host B ; Inter-Asterisk eXchange driver definition ; [general] bindport=4569 bindaddr=0.0.0.0 ; amaflags=billing ; accountcode=IAX2 ; language=ru ; bandwidth=low ; allow=all ; jitterbuffer=no ; register = chara:[EMAIL PROTECTED] ; authdebug=no ; tos=lowdelay ; [chita] type=friend host=62.YYY.YYY.193 secret=secret sendani=yes qualify=yes context=to-pstn canreinvite=yes trunk=yes accountcode=CHITA disallow=all allow=g729 -- Best regards, Andrew Kochetkoff mailto:[EMAIL PROTECTED] ___ 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
RE: [Asterisk-Users] *8 nor *8# works for me!
I use callgroup=2 pickupgroup=2 and it works for me, did you try that? Chris Mason www.anguillaguide.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Friday, April 15, 2005 5:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] *8 nor *8# works for me! I have put into each phone settings (sip.conf and zapata.conf) in my office: callgroup=1 pickupgroup=1 I cannot pickup any calls from another phone!! What do I miss here? bye Ronald ___ 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 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] SIP through firewall is intermittent
We have a PBX in a datacenter in the US and the receptionist is here on Anguilla, her phone and one other, Sipura SPA-841s. are behind a Linksys router doing NAT to an ADSL line. Generally it works fine but occasionally the receptionist cannot hear any voice after the extension rings. Other than portforwarding, which wont work as there is more than one phone, is there a way to improve this setup? Is there a router that would work better? Or maybe a configuration setting that would improve reliability. Chris Mason www.anguillaguide.com ___ 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
RE: [Asterisk-Users] TDM400P Revision question.
My specific issue has to do with ringing on my FXS ports. A Northen Telecom Harmony phone (circa 1983) rings normally but when I connect my newer GE 2.4GHz cordless I never get more than 1/2 ring (it lights up and works fine... just can't get a ring from it). Normally I'd assume that it's a low power issue on the FXS port but with a phone rated at 0.1 REM? I do have some strange voltages though ON-Hook: ~48V DC, 107V AC (this really concerns me...) Off-hook: ~6V DC, ~12VAC (where the hell is this AC component coming from???) Ring: 0V DC, ~45V AC Suffice it to say that electrically this is completely out to lunch... I'd like to throw an oscilloscope on the line to see what's what but I'm having trouble finding one. That on-hook AC is a real problem if the voltmeter is accuate. Couple of things to try 1. Go to the demarc, disconnect the in-house wiring and measure the AC component again (only looking towards the telco's CO). 2. Disconnect asterisk and install an ordinary analog phone. Take the phone off-hook and measure the AC. If the value is very small, then the voltmeter is measuring induced AC on the unterminated wiring. (The phone being off-hook creates the termination.) Put the phone on-hook and measure again. If the value is large, then go looking for the source of the induced AC. Things like wall-warts, fluorescent light ballasts, any device with a transformer in it, electric motors (of some fairly large size), desktop high intensity lamps (with internal transformer), etc, can cause inducedAC if they are within inches of the wiring. Using a scope would be good, but it will only validate the voltmeter results; nothing more. If you're unsure about the quality of the voltmeter, borrow another one from someone and compare the results. ___ 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] VIC2BRI and J4BRI
Hello, the problem has been solved. J4BRI is not compatible with VIC-2BRI unless the VIC-BRI are programmed in point-to-multipoint mode. Nobody knows why ( nor Cisco or Junghanns ) but at least we have a workaround. Eugenio ___ 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] SIP stack pluggable?
Hi all, Is it possible to plug in a SIP stack into Asterisk other than chan_sip*, for example, Vovida? Do you Yahoo!? Yahoo! Small Business - Try our new resources site! ___ 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
Re: [Asterisk-Users] Asterisk live chat problem
I'm trying to join the #asterisk channel on irc.freenode.net but when I try to join the channel I receive the following error message: You need to be identified to join that channel Anyone can help me? Hi Stefano, just select your nickname and then register it: /nick stefano /msg nickserv register password every time you connect you have to identify yourself with /msg nickserv identify password 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
Re: [Asterisk-Users] Empty voicemail attachments?
I found this to happen when a caller is leaving a message * lights the MWI as soon as the message is being recorded. If the called person calls into the * and listens to the message before it is done they here only a partial message and the VM sends an empty attachment. Strange isn't it? I wish * would light the MWI when the message is complete not at the start of recording. - Original Message - From: Andrew C. Brown [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, April 15, 2005 5:14 AM Subject: [Asterisk-Users] Empty voicemail attachments? I have Asterisk newly setup. When a voicemail is left, Asterisk emails out the voicemail as a wav file attachment (in voicemail.conf, append=yes) but the attachment always ends up having a size of 0 bytes and no content to play. Diagnosis steps taken so far: I used Kmail to manually send myself a local email with an attachment and that came through whole. I looked at the wav files stored in the spool directories and they are complete and playable as expected But the content of the file doesn't make it through email. Any ideas anyone can offer as a possible cause of this problem? Thanks a mil voicemail.conf: format=WAV|wav|gsm append=yes [EMAIL PROTECTED] [default] 222 = 1234,Joe,[EMAIL PROTECTED] ___ 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 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] Slack 10 install - THANK YOU - Cisco Reseller Help
Hi Folks -
I'm the newbie who asked the question two or three weeks ago about getting
asterisk up and running on a Slack-10 box.
Received at least 4 responses in about 10 hours with VERY GOOD information
almost everybody gave me an e-mail to send directly too if I had specific
problems. - THANKS - Yeah! - This mailing list WORKS! - And that's
great!!!
Now for my present delima. - Actually this one's been racking my brain since
about March.
I need to find a Cisco Reseller.
A little background. - I live in Japan (retired - or 'retarded' U.S.
Military type) and have been here for the last 20 years. - A lot of my
hardware I can get locally ('Akihabara' for those of you who know Tokyo). -
But for the latest stuff, I need to order it from the States. - One reason
is that local stuff uses Japanese manuals (makes sense) and html setup pages
in Japanese for items like routers, phones, etc. for configuration. - Not
very helpfull.
Last month, I purchased a Cisco-7905G IP Phone from a vendor in the States
paid a pile of money to FedEx to get the thing sent to me.
The phone DID NOT include the SIP firmware. - Since I have a 'BroadVoice'
account, I'm stuck until I can SIP into this phone.
I'd also like to use the 7940/7960 phones but, again, there's the SIP
Firmware problem.
I've been all over the 'wiki' pages have read what's necessary to get this
firmware get it installed. - Basically I need a 'service contract' for
each phone.
Wading through the pages search engines on Cisco's web site is a true
excercise in futility! - A google search is more accurate.
But I've yet to find a Cisco Reseller who knows what I need is willing to
talk about it. - Heck! - Even finding a Cisco Reseller with a valid phone
number where you can talk to a 'PEOPLE' is a feat in itself.
So - If anybody knows a Cisco Reseller, or if you ARE a Cisco Reseller and
know what I need, please contact me. - I'm sure there are others in this ML
that are in the same boat as I or who have been there themselves.
Thanks in advance.
Gary Guthary
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[Asterisk-Users] g729 not work with DTMF and AGI
Hello, I am stuck while attempting to insert DTMF commands from a SIP gateway to an IVR menu running AGI php scrit. If I do: SIP phone -- ulaw -- IVR then dtmf works fine If I do: SIp gateway -- g729 -- IVR then the mneu still works but does not accept DTMF ? I have tried to set dmtfrelax=yes and dtmfmode=info, also tried dtmfinfo=rfc2833 Still did not work, any clues please? ___ 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
Re: [Asterisk-Users] About Audio Latency from PSTN to SIP
Please don't post HTML to the list, and PLEASE TRIM your posts! Maybe I'm getting oversensitive to this lately but the sheer volume of bandwidth wasted due to people not taking 30 seconds to trim replies is staggering! My reply is an example of proper reply trimming; only the essential bits from your post are retained, and everything else is deleted. On April 15, 2005 04:12 am, Qiao Yuansong wrote: put disallow=all allow=ulaw in sip.conf, under [general] and comment out all other allow/disallow lines. Restart asterisk and try again. Something basic is not right. I tried your suggestion, and it make no use. So you have [some_sip_user] type=user disallow=all allow=ulaw context=somecontext in sip.conf for that sip phone? Can you post the output from the sip phone dialing a PSTN number, and then the output from a PSTN incoming call ringing the SIP phone? What version of asterisk? Perhaps you should check out http://www.catb.org/~esr/faqs/smart-questions.html while you're at it. We can't help you if you're not willing to help us. -A. ___ 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] Problems with a SMS-capable Phone on a ZAP Channel / Question about native bridging on digium cards
Hi Everybody, We just changed our setup from a classic PBX connected directly via PRI to the PSTN to a PBX-Asterisk-PRI-PSTN Setup. My question: Does anyone on this list know, how the Digium Cards handle the native bridges between two Zap-Channels? Is it handled on the Digium card or by the Zaptel driver (= in Software?). How is it handled by other cards? I know Aculab does the native bridges in hardware, but what about sangoma or digium? I have a next question: Does anyone has experiences with the T-Com Sinus 701 MMS (i guess this is equivalent to Siemens S440) and sending SMS'es over Asterisk Zap Channels? We have the problem, that this client is no longer able to send SMS via the Asterisk Box. Quite strange, the telephone hangs up just before the SMS Center (01930100) is able to answer... Anyone experienced these problems before? Thanks in advance, Carsten ___ 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
Re: [Asterisk-Users] Slack 10 install - THANK YOU - Cisco ResellerHelp
I can help you. Email me off list. [EMAIL PROTECTED] or [EMAIL PROTECTED]
I am a Cisco Partner.
- Original Message -
From: Gary Guthary [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 15, 2005 6:18 AM
Subject: [Asterisk-Users] Slack 10 install - THANK YOU - Cisco
ResellerHelp
Hi Folks -
I'm the newbie who asked the question two or three weeks ago about
getting
asterisk up and running on a Slack-10 box.
Received at least 4 responses in about 10 hours with VERY GOOD information
almost everybody gave me an e-mail to send directly too if I had specific
problems. - THANKS - Yeah! - This mailing list WORKS! - And that's
great!!!
Now for my present delima. - Actually this one's been racking my brain
since
about March.
I need to find a Cisco Reseller.
A little background. - I live in Japan (retired - or 'retarded' U.S.
Military type) and have been here for the last 20 years. - A lot of my
hardware I can get locally ('Akihabara' for those of you who know
Tokyo). -
But for the latest stuff, I need to order it from the States. - One reason
is that local stuff uses Japanese manuals (makes sense) and html setup
pages
in Japanese for items like routers, phones, etc. for configuration. - Not
very helpfull.
Last month, I purchased a Cisco-7905G IP Phone from a vendor in the States
paid a pile of money to FedEx to get the thing sent to me.
The phone DID NOT include the SIP firmware. - Since I have a 'BroadVoice'
account, I'm stuck until I can SIP into this phone.
I'd also like to use the 7940/7960 phones but, again, there's the SIP
Firmware problem.
I've been all over the 'wiki' pages have read what's necessary to get
this
firmware get it installed. - Basically I need a 'service contract' for
each phone.
Wading through the pages search engines on Cisco's web site is a true
excercise in futility! - A google search is more accurate.
But I've yet to find a Cisco Reseller who knows what I need is willing
to
talk about it. - Heck! - Even finding a Cisco Reseller with a valid phone
number where you can talk to a 'PEOPLE' is a feat in itself.
So - If anybody knows a Cisco Reseller, or if you ARE a Cisco Reseller and
know what I need, please contact me. - I'm sure there are others in this
ML
that are in the same boat as I or who have been there themselves.
Thanks in advance.
Gary Guthary
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[Asterisk-Users] howto forward UAC codec capabilities to the PSTN gw
Hi there, is there any possible way to forward UAC codec capabilities to the PSTN gw w/ Asterisk? Thanks, Pavel ___ 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
Re: [Asterisk-Users] MoH stopped working with cisco 7912/7960
Simone Cittadini wrote: I have asterisk 1.0.6 with cisco 7912/7960 phones (sip) and a isdn card with capi drivers, everything works fine, except for music on hold, even when you transfer a call (which is the most annoying part, since the caller thinks the line is down and hangups). With transfer I don't mean direct transfer (blind), if you directly transfer the call the caller hears the correct type of ringing while he waits for the called to take the call. OK, solved simply upgrading to 1.0.7 -- Simone Cittadini IT Manager == COMVERT S.R.L. via F.lli Bressan, 21 20126 Milano - ITALY Tel +39.02.27006796(aspetta un beep)105 [EMAIL PROTECTED] http://www.comvert.com ___ 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] Re: Cisco 7960s and skinny
understand how to get customized and localized (I'm in Italy) softkeys.
I'm not sure what if anything there is to localize, IIRC chan_sccp
transmits no text to the user except for softkey names, and their you
might be out of luck.
You can customize softkeys replacing labels
static const softkeytypes button_labels [] = {
{ 1, Richiama, sccp_sk_redial },
{ 2, NvChiam, sccp_sk_newcall },
this is working.
I've also downloaded cisco ip telephony locale. Collection of localized
XML labels for 7905 and 7960 phone (tftpboot time)
that works. 7905 now have all the softkeys and all the config sub menus
showing the localized labels/strings.
Booting the phones they are loading XML localized dialtones.
7905 is working but there's an issue about the hold/unhold button. It
stays red after the first call.
the cisco 7905 does register, but it can't receive calls. When you place
an outgoing call it comes back to work and now you can also receive calls.
I'm working on it
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Re: [Asterisk-Users] Asterisk PBX with X100P in India
what kind of problems do u have? can u explain more in detail so we can try helping you? best regards El vie, 15-04-2005 a las 01:29 -0700, Min Hwan Chang escribió: I'm currently trying to set up an Asterisk PBX system in India. However I'm having trouble configuring the X100P to dial out on the POTS line. Does anyone have any knowledge about this? I know the telephone system is a bit different in India, so would the X100P not be suitable? Is there a change I need to make in the Zaptel.conf or zapata.conf? Because I'm using [EMAIL PROTECTED] 0.6, the extensions.conf is also pretty frustrating... Any help here would be appreciated. ___ 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 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
RE: [Asterisk-Users] DTMF does not work with g729 and AGI
First of all I hope you realize you can't have the same context activated at the same time for the same host as * does not support this. So I am just thinking the configuration below are just examples of what you tried. I strongly suggest using dtmfmode=rfc2833 and dtmfmode=info instead of inband. Inband will only work for g711 as there is no compression. Secondly, I would suggest looking at your client and configure the client to match * config. If that does not work, I would capture the data with ethereal and decode the protocol to see what is happening. Most likely problem is with your client. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, April 14, 2005 8:11 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DTMF does not work with g729 and AGI Hello, I have an AGI script that runs a menu at two levels of a tree. If I call the extension from a voip phone with g711, the menu works fine and accepts DTMF no probs. Then, when I Call from a DID, it sends call using SIP and g729 to¨* box. The IVR also starts running, but no DTMF is deteced. I have tried various configs (combinations of dtmfmode=info, dtmfmode=rfc2833 and dtmfrelax=yes, dtmfrelax=no) with no success. Any hint? sip.conf [SS_SIP] type=peer host=XXX.XX.XXX.XX dtmfrelax=no ;dtmfmode=rfc2833 dtmfmode=info context=outbound disallow=all allow=g723.1 allow=g729 [SS_SIP] type=user host=XXX.XX.XXX.XX context=outbound dtmfmode=inband disallow=all allow=g723.1 allow=g729 ___ 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 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
Re: [Asterisk-Users] Slack 10 install - THANK YOU - Cisco ResellerHelp
Henry Devito wrote: I can help you. Email me off list. [EMAIL PROTECTED] or [EMAIL PROTECTED] I am a Cisco Partner. - Original Message - From: Gary Guthary [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, April 15, 2005 6:18 AM Subject: [Asterisk-Users] Slack 10 install - THANK YOU - Cisco ResellerHelp Hi Folks - I'm the newbie who asked the question two or three weeks ago about getting asterisk up and running on a Slack-10 box. Received at least 4 responses in about 10 hours with VERY GOOD information almost everybody gave me an e-mail to send directly too if I had specific problems. - THANKS - Yeah! - This mailing list WORKS! - And that's great!!! [..] Last month, I purchased a Cisco-7905G IP Phone from a vendor in the States paid a pile of money to FedEx to get the thing sent to me. The phone DID NOT include the SIP firmware. - Since I have a 'BroadVoice' account, I'm stuck until I can SIP into this phone. Please follow the instructions from: http://www.mywebcalls.com/rw/ataupgrade.html The page have the latest SIP firmware and instructions to upgrade for the Cisco ATA 186. I'd also like to use the 7940/7960 phones but, again, there's the SIP Firmware problem. I've been all over the 'wiki' pages have read what's necessary to get this firmware get it installed. - Basically I need a 'service contract' for each phone. Wading through the pages search engines on Cisco's web site is a true excercise in futility! - A google search is more accurate. But I've yet to find a Cisco Reseller who knows what I need is willing to talk about it. - Heck! - Even finding a Cisco Reseller with a valid phone number where you can talk to a 'PEOPLE' is a feat in itself. So - If anybody knows a Cisco Reseller, or if you ARE a Cisco Reseller and know what I need, please contact me. - I'm sure there are others in this ML that are in the same boat as I or who have been there themselves. Thanks in advance. Gary Guthary ___ 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 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 -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ 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] Excessive re-registration of Broadvoice account in Asterisk@Home 0.8
I've now had my new [EMAIL PROTECTED] box running for a few days, and one puzzling probem is that Asterisk re-registers my Broadvoice trunk every 20 seconds or so, producing inordinate amounts of logfile data (about 5 lines each time) and presumably unnecessary network traffic, as well. Anyone have any idea what is going on here? The trunk works, by the way, I can receive calls on it -- haven't tried calling out on it, still working on the dialling rules. Below is a snippet of the logfile. Thanks, Wolf Paul Apr 13 15:55:11 DEBUG[28010]: Registration successful Apr 13 15:55:11 DEBUG[28010]: Cancelling timeout 13872 Apr 13 15:55:27 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:55:27 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:55:27 DEBUG[28010]: Scheduled a registration timeout # 13875 Apr 13 15:55:27 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 280: Found Apr 13 15:55:27 DEBUG[28010]: Registration successful Apr 13 15:55:27 DEBUG[28010]: Cancelling timeout 13875 Apr 13 15:55:42 DEBUG[28010]: Manager received command 'Command' Apr 13 15:55:42 DEBUG[28010]: Manager received command 'Command' Apr 13 15:55:43 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:55:43 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:55:43 DEBUG[28010]: Scheduled a registration timeout # 13878 Apr 13 15:55:43 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 281: Found Apr 13 15:55:43 DEBUG[28010]: Registration successful Apr 13 15:55:43 DEBUG[28010]: Cancelling timeout 13878 Apr 13 15:55:59 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:55:59 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:55:59 DEBUG[28010]: Scheduled a registration timeout # 13881 Apr 13 15:55:59 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 282: Found Apr 13 15:55:59 DEBUG[28010]: Registration successful Apr 13 15:55:59 DEBUG[28010]: Cancelling timeout 13881 Apr 13 15:56:15 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:56:15 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:56:15 DEBUG[28010]: Scheduled a registration timeout # 13884 Apr 13 15:56:16 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 283: Found Apr 13 15:56:16 DEBUG[28010]: Registration successful Apr 13 15:56:16 DEBUG[28010]: Cancelling timeout 13884 Apr 13 15:56:32 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:56:32 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:56:32 DEBUG[28010]: Scheduled a registration timeout # 13887 Apr 13 15:56:32 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 284: Found Apr 13 15:56:32 DEBUG[28010]: Registration successful ___ 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
Re: [Asterisk-Users] About Audio Latency from PSTN to SIP
Andrew Kohlsmith wrote: Please don't post HTML to the list, and PLEASE TRIM your posts! Maybe I'm getting oversensitive to this lately but the sheer volume of bandwidth wasted due to people not taking 30 seconds to trim replies is staggering! My reply is an example of proper reply trimming; only the essential bits from your post are retained, and everything else is deleted. [..] Perhaps you should check out http://www.catb.org/~esr/faqs/smart-questions.html while you're at it. We can't help you if you're not willing to help us. The spanish version is located at: http://www.sindominio.net/ayuda/preguntas-inteligentes.html g -A. ___ 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 -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ 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
Re: [Asterisk-Users] Changing IRQ's on TDM
Is the APIC and IO-APIC enabled? Send us 'cat /proc/interrupts' and your /var/log/boot.msg (or your distro's equivalent bootup log). Damian Funnell wrote: Hi all, I've found that a TDM400P card in our * box is sharing IRQ's with two other devices. The server doesn't support assigning IRQ's through the BIOS and the pig only has three PCI slots, so swapping cards between slots hasn't fixed the problem (it just ends up sharing IRQ's with other devices). Any ideas on how we can force the TDM to use a certain IRQ? Plenty of free IRQ's in the box, BIOS just doesn't want to use them. FFF Managed Technology Ltd. 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz ___ 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 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] SIP Message Waiting Notification
We have a asterisk box connected to a softswitch for providing voicemail. When a caller leaves a voice mail we need to send a message waiting notification back to the switch. Has anyone gotten this to work? What is the SIP message I should be looking for? -- Mike ___ 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
RE: [Asterisk-Users] Asterisk became berserk when Internet connection is down and can't register to SIP server.
I think there are a couple of things you can do: 1. Switch the provider to get a stable internet connection ;-) 2. convert your lookups to IP addresses instead of domains. However, if you clients register with address like [EMAIL PROTECTED], then dns will be used to resolve blah.com and then you have a problem. I am not sure if converting to ip addresses is doable on a large scale. 3. monitor your internet connection with another script. If the connection fails then automatically edit * config file to remove your registration with FWD and reload the proper config. 4. configure * with realtime extensions and place peers into mysql db. Then use option 3 to monitor your internet connection and remove the peer on failure. This step does not require reloading config. HTH Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kong Sent: Thursday, April 14, 2005 11:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk became berserk when Internet connection is down and can't register to SIP server. So, any way i can resolve this problem? At 10:55 AM 4/15/2005, you wrote: On 4/14/05, Kong [EMAIL PROTECTED] wrote: Hi, i found a case here, i really don't know is it a bug or something else. i have like 200 ip phones connected to my * server, (ATA's and softphones). and i had it register to SIP service (FWD), so, when my internet connection is down, * is not able to register itself to FWD, never mind that, but it made all the extension berserk. all the client are not able to login to the server. error msg is login timeout, but once i remark the register = :[EMAIL PROTECTED] and restarted the server, immediately * became back to normal. so, i was wondering, is the a bug or something? coz my internet provider is not consistent, sometimes it goes down. thank you. I seem to remember a bug like this that had to do with dns lookups I think. Maybe someone else can remember the exact details and what version it was in. Chris ___ 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 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 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
RE: [Asterisk-Users] Empty voicemail attachments?
I had the same problem on the HEAD version and went to STABLE to resolve it. -Original Message- From: Andrew C. Brown [mailto:[EMAIL PROTECTED] Sent: Friday, April 15, 2005 6:14 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Empty voicemail attachments? I have Asterisk newly setup. When a voicemail is left, Asterisk emails out the voicemail as a wav file attachment (in voicemail.conf, append=yes) but the attachment always ends up having a size of 0 bytes and no content to play. Diagnosis steps taken so far: I used Kmail to manually send myself a local email with an attachment and that came through whole. I looked at the wav files stored in the spool directories and they are complete and playable as expected But the content of the file doesn't make it through email. Any ideas anyone can offer as a possible cause of this problem? Thanks a mil voicemail.conf: format=WAV|wav|gsm append=yes [EMAIL PROTECTED] [default] 222 = 1234,Joe,[EMAIL PROTECTED] ___ 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 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
RE: [Asterisk-Users] Asterisk behind NAT
Try setting externip=(asterisk public ip address) Hth Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, April 15, 2005 12:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk behind NAT Hi List, I've spent hours researching on this topic, found tons of info, so far it doesn't work yet. Here's the scenario Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to send calls to an outside provider. My SIP phones (outside * NAT) are able to register with no problem. The problem comes when I send a call out to my provider I get no audio in either way. My provider sees only my private IP. How can I send my provider the public IP?? I have in my sip.conf: [general] port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) nat=yes externip = x.x.x.x localnet=192.168.1.0/24 (remember I enabled DMZ on the router so all ports are being forwarded to *) Despite of what I do, the externip and the localnet my provider only gets my * private IP. Any tip how to solve this?? Thanks ___ 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 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
Re: [Asterisk-Users] TDM400P Revision question.
Rich Adamson wrote: My specific issue has to do with ringing on my FXS ports. A Northen Telecom Harmony phone (circa 1983) rings normally but when I connect my newer GE 2.4GHz cordless I never get more than 1/2 ring (it lights up and works fine... just can't get a ring from it). Normally I'd assume that it's a low power issue on the FXS port but with a phone rated at 0.1 REM? I do have some strange voltages though ON-Hook: ~48V DC, 107V AC (this really concerns me...) Off-hook: ~6V DC, ~12VAC (where the hell is this AC component coming from???) Ring: 0V DC, ~45V AC Suffice it to say that electrically this is completely out to lunch... I'd like to throw an oscilloscope on the line to see what's what but I'm having trouble finding one. That on-hook AC is a real problem if the voltmeter is accuate. Couple of things to try 1. Go to the demarc, disconnect the in-house wiring and measure the AC component again (only looking towards the telco's CO). 2. Disconnect asterisk and install an ordinary analog phone. Take the phone off-hook and measure the AC. If the value is very small, then the voltmeter is measuring induced AC on the unterminated wiring. (The phone being off-hook creates the termination.) Put the phone on-hook and measure again. If the value is large, then go looking for the source of the induced AC. Things like wall-warts, fluorescent light ballasts, any device with a transformer in it, electric motors (of some fairly large size), desktop high intensity lamps (with internal transformer), etc, can cause inducedAC if they are within inches of the wiring. Using a scope would be good, but it will only validate the voltmeter results; nothing more. If you're unsure about the quality of the voltmeter, borrow another one from someone and compare the results. Doesn't anyone use Google anymore? http://lists.digium.com/pipermail/asterisk-users/2005-April/098934.html Also: http://www.google.com/search?hl=enq=site%3Alists.digium.com+boostringerbtnG=Google+Search -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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
Re: [Asterisk-Users] Analogue phone transfering
David Wilson wrote: Hi guys, How are you keeping ? I have an analogue phone plugged into a Digium FXS Zap module on my TDM card. The phone works well except that I cannot seem to transfer calls using the flash key. I don't seem to get another dialtone as indicated in: http://www.voip-info.org/wiki-Asterisk+tips+zap+transfer Any ideas what I've done wrong ? This is my zapata.conf: [channels] ; For analogue phone signalling=fxo_ks context=default channel=4 relaxdtmf=yes threewaycalling=yes transfer=yes adsi=no usecallerid=no rxgain=70.0 txgain=50.0 In zapata.conf you set options and then APPLY the options to a channel. As you can see you are specifying the channel before most of your otions so they are never applied. Move your channel= line AFTER the options you want to set. You might want to remove your rxgain and txgain so you don't blow out your eardrums. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Debugging zaphfc + PBX integration
Howdy - just got meself a cheapy Cologne chip card to play with in the hopes that it will prevail where a £1000 Eicon Diva Server failed... I'm trying to run in NT mode and appear as a trunk to a PBX (it's an Inter-Tel Axxess unit). If I hook a specific port on the BRI board of the PBX directly to a BT ISDN2e wallbox and dial 95000, I get a dialtone and can complete calls. Any time I connect to the Diva Server or the zaphfc card, the handset tells me 'TEST ISDN2 IS UNPLUGGED'... I installed asterisk on a completely virgin box using the install.sh contained in the bristuff 0.2.0-RC7k tarball. /etc/asterisk/zapata.conf [channels] switchtype = euroisdn signalling = bri_net pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context=demo channel = 1-2 /etc/zaptel.conf loadzone=uk defaultzone=uk span=1,1,3,ccs,ami bchan=1-2 dchan=3 Zapata Telephony Interface Registered on major 196 PCI: Enabling device :04:01.0 ( - 0003) ACPI: PCI interrupt :04:01.0[A] - GSI 17 (level, low) - IRQ 185 zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xd002af00 fifo 0xcb3d8000 (0xb3d8000) IRQ 185 HZ 1000 zaphfc: Card 0 configured for NT mode zaphfc: 1 hfc-pci card(s) in this box. Registered tone zone 4 (United Kingdom) zaphfc: card 0 layer 1 state = G2 plugs cable into PBX zaphfc: card 0 layer 1 state = G3 zaphfc: empty HDLC frame or bad CRC received (framelen = 3, stat = 0xfe). zaphfc: empty HDLC frame or bad CRC received (framelen = 40, stat = 0xff). zaphfc: card 0 RX [ 0xfc 0xff 0x3 0xf 0xff 0x6d 0x1 0xff ] 8 bytes zaphfc: card 0 RX [ 0xfc 0xff 0x3 0xf 0xf6 0x3e 0x1 0xff ] 8 bytes zaphfc: card 0 RX [ 0xfc 0xff 0x3 0xf 0x5a 0x1f 0x1 0xff ] 8 bytes I'm told that the RX bytes are the PBX requesting a TEI from the provider (i.e. the * box) but I can't understand why no reply is being given. This is precisely the same problem I had with the Eicon Diva Server card... Is this likely to be as simple as needing the 100 ohm resistors, or can it be fixed with software? Cheers Gavin. ___ 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
Re: [Asterisk-Users] *8 nor *8# works for me!
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Friday, April 15, 2005 5:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] *8 nor *8# works for me! I have put into each phone settings (sip.conf and zapata.conf) in my office: callgroup=1 pickupgroup=1 I cannot pickup any calls from another phone!! What do I miss here? Your SIP phone is eating the *8. You need to look at your SIP phone docs, not Asterisk -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Question on Asterisk CDR / In-Network Calling / MySQL CDR
Hi, I have the asterisk mysql CDR module/patch installed. But I believe that's slightly irrelivant to the question but is included for completeness. How can I determine from a CDR record (csv by default.. or the mysql CDR)... if a call was in-network that is.. from one phone to another?I haven't seen anything other then the DST channel seems to take the format of SIP/xxx or SIP/xxx where x is the extension of the phone. ___ 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 working on FC3+X100P+France Telecom line
I have successfully configured asterisk for my home use on a FC3 server with a X100P card plugged to a France Telecom line (for French user my line is 'partiellement dgrouppe' with tele2). This forum has help me a lot and I would like to thank you all for doing this.
Couple of advices:
- in zapata.conf, you need to have callprogress=no else it does not work with France Telecom line.
- at the beginning asterisk could not read mp3 file though mpg123 was installed. I had to get rid of xmms-mpg123 and mpg321 rpms and to install mpg123-0.59r rpm to make it work.
- to have asterisk working as a service, you need to do make config in both /usr/src/zaptel and /usr/src/asterisk
Enjoy !
Bertrand
FEATURES WORKING SO FAR
- voicemail sent to 2 different emails, one with attachement, the other without
FEATURES NOT YET WORKING
- fax (spandsp)
- slimserver plugin developped by Ian Hailey
Here are my config files:
/etc/zaptel.conf
fxsks = 1
loadzone = fr
defaultzone = fr
/etc/asterisk/zapata.conf -
[channels]
signalling=fxs_ks
language=fr
context=demo
;faxdetect=incoming
;mailbox=0001
usercallerid=no
;callerid=asreceived
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
;relaxdtmf=yes
rxgain=4
txgain=0
immediate=no
busydetect=yes
busycount=3
callprogress=no
;musiconhold=default
channel = 1
/etc/asterisk/extensions.conf-
[general]
static = yes
writeprotect = yes
[globals]
[demo]
; Wait 15 seconds for an answer (pick up the local phone)
exten = s,1,Wait,12
; Answer the phone the s option is to suppress the naming of the box n at the end of
; the greeting
exten = s,2,Answer
exten = s,3,Voicemail(su00010002)
exten = s,4,Hangup
; Did we get a fax?
;exten = fax,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
;exten = fax,2,rxfax(${FAXFILE})
;
;
; Always put these two extensions in any main context that handles
; call processing, for good form and graceful hangups.
;
exten = i,1,Hangup
exten = h,1,Hangup
/etc/asterisk/voicemail.conf-
;
; Voicemail Configuration
;
[general]
; Default formats for writing Voicemail
;format=g723sf|wav49|wav
;format=wav49|gsm|wav
format=gsm|wav
[default]
; limite la duree maxi d'un message xx sec
maxmessage=300
; le message doit durer au moins xx sec
minmessage=5
;permet de recevoir le message dans la boite groupe
;qui achemine les messages vers les boites 0001 0002
;0001 conserve les messages et envoie un email sans attachement
;0002 envoie email avec pice attache et efface le message
= 2477,Group Mailbox,,,delete=yes
0001 = 2477,PutName1Here,PutEmail1Here,,saycid=no|attach=no
0002 = 2477,PutName2Here,PutEmail2Here,,saycid=no|attach=yes|delete=yes
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Re: [Asterisk-Users] TDM400P Revision question.
My specific issue has to do with ringing on my FXS ports. A Northen Telecom Harmony phone (circa 1983) rings normally but when I connect my newer GE 2.4GHz cordless I never get more than 1/2 ring (it lights up and works fine... just can't get a ring from it). Normally I'd assume that it's a low power issue on the FXS port but with a phone rated at 0.1 REM? I do have some strange voltages though ON-Hook: ~48V DC, 107V AC (this really concerns me...) Off-hook: ~6V DC, ~12VAC (where the hell is this AC component coming from???) Ring: 0V DC, ~45V AC Suffice it to say that electrically this is completely out to lunch... I'd like to throw an oscilloscope on the line to see what's what but I'm having trouble finding one. That on-hook AC is a real problem if the voltmeter is accuate. Couple of things to try 1. Go to the demarc, disconnect the in-house wiring and measure the AC component again (only looking towards the telco's CO). 2. Disconnect asterisk and install an ordinary analog phone. Take the phone off-hook and measure the AC. If the value is very small, then the voltmeter is measuring induced AC on the unterminated wiring. (The phone being off-hook creates the termination.) Put the phone on-hook and measure again. If the value is large, then go looking for the source of the induced AC. Things like wall-warts, fluorescent light ballasts, any device with a transformer in it, electric motors (of some fairly large size), desktop high intensity lamps (with internal transformer), etc, can cause inducedAC if they are within inches of the wiring. Using a scope would be good, but it will only validate the voltmeter results; nothing more. If you're unsure about the quality of the voltmeter, borrow another one from someone and compare the results. Doesn't anyone use Google anymore? http://lists.digium.com/pipermail/asterisk-users/2005-April/098934.html Also: http://www.google.com/search?hl=enq=site%3Alists.digium.com+boostringerbtnG=Google+Search Eric, those links have nothing to do with his stated problem. The problem is 105v AC on the pstn line when on-hook and no ringing. ___ 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
Re: [Asterisk-Users] TDM400P Revision question.
Rich Adamson wrote: My specific issue has to do with ringing on my FXS ports. A Northen Telecom Harmony phone (circa 1983) rings normally but when I connect my newer GE 2.4GHz cordless I never get more than 1/2 ring (it lights up and works fine... just can't get a ring from it). Normally I'd assume that it's a low power issue on the FXS port but with a phone rated at 0.1 REM? I do have some strange voltages though ON-Hook: ~48V DC, 107V AC (this really concerns me...) Off-hook: ~6V DC, ~12VAC (where the hell is this AC component coming from???) Ring: 0V DC, ~45V AC Suffice it to say that electrically this is completely out to lunch... I'd like to throw an oscilloscope on the line to see what's what but I'm having trouble finding one. That on-hook AC is a real problem if the voltmeter is accuate. Couple of things to try 1. Go to the demarc, disconnect the in-house wiring and measure the AC component again (only looking towards the telco's CO). 2. Disconnect asterisk and install an ordinary analog phone. Take the phone off-hook and measure the AC. If the value is very small, then the voltmeter is measuring induced AC on the unterminated wiring. (The phone being off-hook creates the termination.) Put the phone on-hook and measure again. If the value is large, then go looking for the source of the induced AC. Things like wall-warts, fluorescent light ballasts, any device with a transformer in it, electric motors (of some fairly large size), desktop high intensity lamps (with internal transformer), etc, can cause inducedAC if they are within inches of the wiring. Using a scope would be good, but it will only validate the voltmeter results; nothing more. If you're unsure about the quality of the voltmeter, borrow another one from someone and compare the results. Doesn't anyone use Google anymore? http://lists.digium.com/pipermail/asterisk-users/2005-April/098934.html Also: http://www.google.com/search?hl=enq=site%3Alists.digium.com+boostringerbtnG=Google+Search Eric, those links have nothing to do with his stated problem. The problem is 105v AC on the pstn line when on-hook and no ringing. The first line of this message says My specific issue has to do with ringing on my FXS ports. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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
RE: [Asterisk-Users] Asterisk behind NAT
I have... Externip=x.x.x.xand nothing... Does not seem to help in anything. Still my provider sees the private IP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev Sent: Friday, April 15, 2005 9:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT Try setting externip=(asterisk public ip address) Hth Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, April 15, 2005 12:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk behind NAT Hi List, I've spent hours researching on this topic, found tons of info, so far it doesn't work yet. Here's the scenario Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to send calls to an outside provider. My SIP phones (outside * NAT) are able to register with no problem. The problem comes when I send a call out to my provider I get no audio in either way. My provider sees only my private IP. How can I send my provider the public IP?? I have in my sip.conf: [general] port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) nat=yes externip = x.x.x.x localnet=192.168.1.0/24 (remember I enabled DMZ on the router so all ports are being forwarded to *) Despite of what I do, the externip and the localnet my provider only gets my * private IP. Any tip how to solve this?? Thanks ___ 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 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 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
Re: [Asterisk-Users] Analogue phone transfering
Hi Eric, Thanks for your reply and guidance. I've tried that but unfortunately am still battling with the same problem. Any other ideas ? Thanks for your help so far. My zapata.conf: [channels] signalling=fxs_ks callprogress=no ;causes problems with calls not being established correctly context=incoming echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=800 ; Asterisk trains to the beginning of the call, number is in milliseconds ;echotraining=yes usecallerid=yes callerid=asreceived callwaiting=no usedistinctiveringdetection=no busydetect=yes busycount=8 adsi=no relaxdtmf=yes faxdetect=incoming channel=1-3 signalling=fxo_ks context=default relaxdtmf=yes ;threewaycalling=yes transfer=yes adsi=no usecallerid=no channel=4 ;rxgain=70.0 ;txgain=50.0 Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 15, 2005 3:21 PM Subject: Re: [Asterisk-Users] Analogue phone transfering David Wilson wrote: Hi guys, How are you keeping ? I have an analogue phone plugged into a Digium FXS Zap module on my TDM card. The phone works well except that I cannot seem to transfer calls using the flash key. I don't seem to get another dialtone as indicated in: http://www.voip-info.org/wiki-Asterisk+tips+zap+transfer Any ideas what I've done wrong ? This is my zapata.conf: [channels] ; For analogue phone signalling=fxo_ks context=default channel=4 relaxdtmf=yes threewaycalling=yes transfer=yes adsi=no usecallerid=no rxgain=70.0 txgain=50.0 In zapata.conf you set options and then APPLY the options to a channel. As you can see you are specifying the channel before most of your otions so they are never applied. Move your channel= line AFTER the options you want to set. You might want to remove your rxgain and txgain so you don't blow out your eardrums. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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 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
RE: [Asterisk-Users] Polycom IP500 phones do not update time fromtime server
Here is that Part: -- TCP_IP netMon tcpIpApp.netMon.enabled=1 tcpIpApp.netMon.period=30/ SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address=10.12.14.33 tcpIpApp.sntp.gmtOffset=-25200 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=4 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1/ port RTP tcpIpApp.port.rtp.filterByIp=1 tcpIpApp.port.rtp.filterByPort=0 tcpIpApp.port.rtp.forceSend= tcpIpApp.port.rtp.mediaPortRangeStart=/ /port /TCP_IP Seshu Kanuri, Seshu (Company IT) wrote: Does anyone know how Polycom 500s will be able to update their time. My setup for a time sync with Public domain Time servers is not successful. Seshu Can you look for the sntp entry in your ipmid.cfg file and post it in it's entirety? Sean NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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
Re: [Asterisk-Users] Who is a QUALITY IAX Termination Provider for 800 DID's?
On Thu, 14 Apr 2005 11:02:48 -0400, Paul wrote: Linn Boyd wrote: I have looked for a quality IAX provider for 800 DID's we currently have two, one is ok and the other is just not of quality, but last night we got an email after a complaint of quality earlier in the day and this is what it said. Remember I never did request a network change, but I just wanted my quality fixed, they have all kinds of contact information and they could have let me know outside of voice mail. I have been trying to call them and trying to email them ever since I found out. I have an 800 number via IAX from Clearpath in Detroit. It has been up continuously for on year, ever since it was provisioned. Getting it setup was a bit pf a pain as Clearpath is a small company, but in a large network operation centre. They're good folks to deal with, just not always the easiest to reach in a hurry. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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
Re: [Asterisk-Users] Call Files to Terminate a call to the dialplan not directly to a channel
You can use (at least in asterisk CVS), this: Channel: Local/[EMAIL PROTECTED] then in extensions.conf [from-internal] exten = 1234,1,Dial(whatever) exten = 1234,2,Dial(otherprov) Not testet though ;) Julian J. M. On 4/14/05, Mystery Glitch [EMAIL PROTECTED] wrote: Can I use the .call files to place a call using the dialplan instead of the channel directly? ---Channel: SIP/[EMAIL PROTECTED] Context: testing Extension: playsample Priority: 1 CallerID: Company 8882650946 WaitTime: 15 ___ 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] Grandstream BT Volume
Guys. Anyway had problems with G BT 100 or 101 volume? Seems the volume is too loud and when talking it makes the voice cut off due to saturation. Anyway to reduce the input voice volume on the phones? ___ 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] Lost DTMF digits
Hi all,
I am writing a program which accepts an incoming call, plays some messages,
accepts some DTMF digits (which make up an outgoing phone number) and then
dials the provided number. However, the problem I am suffering is the loss
of DTMF digits when the program is busy performing other actions. The first
instance is in-between playing soundfiles, e.g.
##
my $ret1 = $AGI-stream_file( '/sound/dialtone', '01' );
my $ret2 = $AGI-stream_file( '/sound/welcome', 01' );
##
In the instance above the program is meant to play a half second dialtone
and then a welcome message. If the user presses a key to start dialling
straight away the playback stops and the digit is returned to $ret.
Unfortunately, if someone presses a key in the half-second in-between
messages the digit is lost.
The second instance is:
##
# grabbing DTMF digits to make outgoing telephone number
my $timeout = 3000; # 3 second timeout
while ( $wait ) {
my $digit = $AGI-wait_for_digit( $timeout );
if( $digit 0 ) { # if digit is input, wait for more digits
$digit -= 48; # making digit decimal from ascii
$outgoing_number .= $digit;
}
else { # stop waiting for more digits
$wait = 0;
}
}
my $ret3 = $AGI-exec( 'dial', Zap/G1/$outgoing_number ); # dial outgoing
number
##
In this instance once the user stops dialling for three seconds the program
assumes the user has entered the complete number and starts dialling. If
however, the user is just slow and has yet to input some digits, any digits
entered while the program is trying to dial are lost. The program will
start listening for more digits if the dial fails but it will still have
lost the digits input during the dial.
If anyone has any idea how to capture the 'lost' DTMF digits while the
program is busy performing other actions I would be much obliged. Even
better would be a way to be constantly listening for DTMF digits in the
background of the program so I could grab the DTMF presses at my leisure
and run commands without fear of 'losing' anything.
Many thanks in advance for your help.
David Farrant
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Re: [Asterisk-Users] Asterisk@Home 0.9 released {Scanned}
Thanks for your hard work. Thanks, David On Thu, 2005-04-14 at 07:15 -0700, [EMAIL PROTECTED] wrote: cool thanks for the update. next time please submit a bug to the [EMAIL PROTECTED] source forge project. Then it will get fixed in the next release. I had no idea this was broken. --- Time Bandit [EMAIL PROTECTED] wrote: More bug fixes. *69 works now. Cisco stuff works. Lots of other fixes. is phpconfig fixed ? when editing a file, it doesn't show the list of sections, it only list Header What needs to be modified : In the function OC_readConfFile around line 131 change : $this-_OC_the_file[] = fgetc($file); to : $this-_OC_the_file[] = fgets($file); I have to manually edit it each time I install it. Thanks for the great work so far hth ___ 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 __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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 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] ilbc codec in Asterisk
Hi all, How can I use ilbc codec in asterisk? been trying to set allow=ilbc at my sip.conf and iax.conf but the call is format to ulaw, though I put allow=ilbc above allow=ulaw. I have been tried for almost two weeks and now seeking experts for help. : ( . need ilbc codec to reduce the total bandwidth used by asterisk. Anyone can share their experience ? Thanks, Stephen ___ 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
Re: [Asterisk-Users] Realtime Friends
My silence comes from the fact that I didn't check my email yesterday. :) It was my impression that if a UA1 registered to server1 and UA2 registered to server2, and both servers were using ARA, then both servers should know how to reach both UAs. I don't have two servers to test this on at the moment. My suggestion would be to take this to the developers list and see if what I theorized above is indeed the expected behavior of ARA. If it is, then we have a bug. If it is not, then we have a feature request. -Matthew Rod Bacon wrote: Matt, can I assume from your silence that you concurr with my thinking that realtime is in fact broken, or is it that I am using it incorrectly? - Original Message - From: Rod Bacon [EMAIL PROTECTED] To: Matthew Boehm [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 13, 2005 9:06 AM Subject: [Asterisk-Users] Realtime Friends Matthew, I got the updates to start working again by ensuring that rtcachefriends=yes. I don't see why this should make a difference, but it does. My understanding was that this parameter only controlled the seeding of the in-memory friends list from the realtime db for purposes of MWI and KeepAlive. I have, however, one remaining issue that I need to resolve. Essentially, I am testing two Asterisk servers (Server1 ans Server2), configured to talk to a common database. I am trying to have calls placed on ANY server routed to SIP UAs registered on ANY OTHER server. Specifically; UA1 registers to Server1. DB is updated correctly. UA2 registers to Server2. DB is updated correctly. I can query the db (using REALTIME LOAD) from either server and see the correct SIP info for either UA. The central dialplan simply routes calls to SIP/UA1 or SIP/UA2. The problem is that Server1 ONLY knows about UA1 and Server2, UA2. The logic seems to be that the lookup in the extensions table (realtime dialplan) happens, then tries to route the call to a SIP registrant that is not in the local (in-memory) friends table. I thought the Server would then go back to the friends realtime table to get the registration info? Is this NOT how it is supposed to work? Should rtcachefriends force the server to update it's friends list on server startup, then at predetermined (configurable?) intervals? Matthew Boehm wrote: (I removed the [] header cause that is what i base my email filter on.) Rod Bacon wrote: I think there's a more sinister bug in play somewhere. The phones are on the same LAN. It was working when I only had a single asterisk server using the database, and seemed to stop when I added a second server. I know this doesn't make any sense... OK. Lemme picture this. You had originally 1 asterisk server and 1 database server. This worked fine with RealTime. Then you added a second asterisk server to connect to this same database server and now the phone won't register with either asterisk server? The SIP registration MUST be ok, because the in-memory database on the server that accepts the registration shows the correct information... the problem is that it doesn't write it to the database. Oh. Weird. But if you turn off the 2nd asterisk server, everything is fine? I think the bug must lie in the update code. When the registration is accepted, the update command is sending nulls to the database for some reason. Yes, this is wierd cause I can't duplicate this. You don't have entries in BOTH sip.conf AND ARA do you? You said the phone does indeed register, it just doesn't update the database using RealTime? Is there any way you can send a full debug output starting slighty before the phone tries to register? have you done a packet sniff to see if asterisk is indeed sending back a 200 OK to the register request? -Matthew -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == -- -- ___ 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-user ___ 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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or
Re: [Asterisk-Users] ilbc codec in Asterisk
On Friday 15 April 2005 15:28, Stephen wrote: Hi all, How can I use ilbc codec in asterisk? been trying to set allow=ilbc at my sip.conf and iax.conf but the call is format to ulaw, though I put allow=ilbc above allow=ulaw. I have been tried for almost two weeks and now seeking experts for help. : ( . need ilbc codec to reduce the total bandwidth used by asterisk. Anyone can share their experience ? Stephen, you would be much much better off paying the tiny sum of $10 for a G.729 license... not only is this a lot more 'industry standard' but the latency is lower as are the conversion times. Best still, the bandwidth usage is even lower than ILBC, I believe :) You can also get a usable implementation at [1] but please be aware this is non-licensed and as such you will be infringing on the intellectual property of patent holders by using it in a live environment. Cheers, Gavin. [1] http://kvin.lv/pub/Linux/Asterisk/ ___ 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
RE: [Asterisk-Users] Dial Macro Arguments
Well, for what it's worth, I hope that some kind of argument feature will be implemented in stable then. Macros are extremely useful, especially with arguments. Regards, Shaun Tierney -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Thursday, April 14, 2005 6:03 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dial Macro Arguments the feature you are talking about is still not commited to stable. at the moment it is only availabe in CVS HEAD. You can try to download the patch and apply it, however I did not succeed in applying it to 1.0.7 so I had to use HEAD. On 4/14/05, Shaun Tierney [EMAIL PROTECTED] wrote: Hello all! I posted a message a while back about a problem I was having in December. I was unable to send arguments to the macro in the dial command. I was told back then to use ^ as the delimiter between the macro name and the arguments and that I had to upgrade to a newer version of Asterisk. Now it appears that this does not work now that I have upgraded to Asterisk 1.0.7. Was this feature removed or replaced? Below is the error message I am receiving and the link to my original message. Apr 14 17:30:53 WARNING[3992]: app_macro.c:90 macro_exec: No such context 'macro-getstartseconds^Zap/3-1' for macro 'getstartseconds^Zap/3-1' http://lists.digium.com/pipermail/asterisk-users/2004-December/075928.html Thanks, Shaun ___ 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 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
Re: [Asterisk-Users] *8 nor *8# works for me!
Eric Wieling wrote: I have put into each phone settings (sip.conf and zapata.conf) in my office: callgroup=1 pickupgroup=1 I cannot pickup any calls from another phone!! What do I miss here? Your SIP phone is eating the *8. You need to look at your SIP phone docs, not Asterisk What am I going to look for, e.g., in a manual for snom 190 and a Budgetone ??? bye Ronald ___ 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
RE: [Asterisk-Users] Slack 10 install - THANK YOU - Cisco Reseller Help
Now for my present delima. - Actually this one's been racking my brain since about March. I need to find a Cisco Reseller. very good and will pre configure the Cisco phones for SIP. http://www.voipsupply.com/home.php --Snip-- Last month, I purchased a Cisco-7905G IP Phone from a vendor in the States paid a pile of money to FedEx to get the thing sent to me. The phone DID NOT include the SIP firmware. - Since I have a 'BroadVoice' account, I'm stuck until I can SIP into this phone. It is a real pain to upgrade to sip on a Cisco. Voip Supply charges an extra $100 to do it for you I'd also like to use the 7940/7960 phones but, again, there's the SIP Firmware problem. I've been all over the 'wiki' pages have read what's necessary to get this firmware get it installed. - Basically I need a 'service contract' for each phone. you need a sip license for each phone. Wading through the pages search engines on Cisco's web site is a true excercise in futility! - A google search is more accurate. But I've yet to find a Cisco Reseller who knows what I need is willing to talk about it. - Heck! - Even finding a Cisco Reseller with a valid phone number where you can talk to a 'PEOPLE' is a feat in itself. So - If anybody knows a Cisco Reseller, or if you ARE a Cisco Reseller and know what I need, please contact me. - I'm sure there are others in this ML that are in the same boat as I or who have been there themselves. You may also want to look at Polycom phones they are less of a headache than the cisco phone. They are both good on quality, look and feel of a business class phone. Cheers, Max ___ 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] sipXphone
Maybe I just woke up too early today. I have SJPhone and X-Lite working perfectly but I cannot for the life of me get sipXphone working properly with Asterisk. Its probably something stupid on my part, but does anyone have a quick setup sheet for it? -Kerry ___ 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] Ring requested on unconfigured channel 0/31 span 1?
Hi Everyone, I'm having a couple of problems configuring E1 lines into a digium TE405P. The error I'm getting when I ring into one of the numbers is: Apr 15 16:24:45 WARNING[9072]: chan_zap.c:7541 pri_dchannel: Ring requested on unconfigured channel 0/31 span 1 I'm not sure why the channel is unconfigured or what this error means exactly. Does anyone have any ideas? Thanks for any help!, Derek My /etc/zaptel.conf: span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124 dchan=16,47,78,109 My /etc/asterisk/zapata.conf: [trunkgroups] trunkgroup = 1,16,109 spanmap = 1,1,1 spanmap = 2,1,2 spanmap = 3,1,3 spanmap = 4,1,4 [channels] context=PRI-NTL switchtype=euroisdn signalling=pri_cpe group=1 channel=1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124 usecallerid=yes hidecallerid=no callwaiting=no restrictcid=no threewaycalling=no -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.com ___ 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] No Hope with Fax and X100P?
I've read a lot on this board and in the WIKI. Is there no hope of a X100P with SpanDSP accepting incoming fax? Everytime I try it fails to train. Is there something I have missed that could fix the problem? Chris___ 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
Re: [Asterisk-Users] No Hope with Fax and X100P?
Chris, I am using the latest SpanDSP (but also tried with the old one) with X100P without any major problems. Send me your zapata config off line and I will try to help you. Also, what version of SpanDSP/Asterisk do you have? W - Original Message - From: Chris [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 15, 2005 11:27 AM Subject: [Asterisk-Users] No Hope with Fax and X100P? I've read a lot on this board and in the WIKI. Is there no hope of a X100P with SpanDSP accepting incoming fax? Everytime I try it fails to train. Is there something I have missed that could fix the problem? Chris ___ 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 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] Bridging 2 Zap channels
I am running * 1.0.6 with 8 analogue phone lines connected to 2 cards - lspci reveals these as : 03:04.0 Communication controller: Tiger Jet Network Inc. Model 300 128k 03:05.0 Communication controller: Tiger Jet Network Inc. Model 300 128k The wcfxs module is loaded successfully and I have the first 3 lines actually connected. /etc/asterisk/zapata.conf is correct (channels = 1-3) The problem is that under certain circumstances (which I am unable to determine) * bridges 2 of the Zap channels together even though I can see no possible way in the dialplan. This then permanently consumes 2 lines leaving only one available. I have been watching the system for 2 days now and have managed to trap it into this condition twice - the system is only under light load. Can anyone suggest a means of tracking this down via debug commands and suchlike ? Has anyone else seen this and what was the fix ? Paul He -- Paul Hewlett (Linux #359543) Tel: +27 21 852 8812 Cel: +27 72 719 2725 Fax: +27 86 672 0563 -- ___ 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] Maybe not worded right, Answering a call
Can i set the time asterisk takes to answer a call. because it takes at least 10 seconds before it starts any dialplan activity.___ 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] Re: Loop Detection
Daniel Corbe [EMAIL PROTECTED] wrote: Is there any way to turn Loop Detection off or tune the params a bit? I am having an issue with Call Forwarding on my SIP Proxy Server which is causing me great pains. All I can do is sympathize. The same problem occurs when a call comes in through Asterisk, gets sent to SER, then comes back to Asterisk 20 seconds later for voicemail. I have contemplated just commenting out the check in chan_sip.c, but I haven't tried this. Not sure if this might cause other problems. Asterisk has many SIP deficiencies. Asterisk has been built as a monolithic PBX, and it seems to do okay using SIP phones as channels. If you want Asterisk to simply act as a SIP UA, you are going to run into a whole slew of problems. I'm not holding my breath waiting for this to change. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ 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] Re: Fax questions
On Fri, 15 Apr 2005 07:14:03 +0800, Ronald Wiplinger wrote:
There are so many fax information available, so that I am getting confused.
What I hope I can get to work:
Any extension should be able to receive fax, whereby via faxdetect the
fax should be sent to the email address as mentioned in voicemail.conf
Which packages should I install?
How would be the dialplan for this? (as mentione above)
I am using SuSE 9.2 Professional
I use Gentoo, so you'll have to figure out what to install from SuSe
yourself, but I installed the spandsp software, which gives you a nice
software fax modem, then I used this in extensions.conf to switch to
spandsp when a fax tone is detected and email it to my email address:
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(3)
[fax]
exten = 2201,1,Macro(faxreceive)
exten = 2202,1,Macro(faxreceive)
exten = 2203,1,Macro(faxreceive)
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERIDNUM} ${CALLERIDNAME})
NOTE: asterisk automatically jumps to the [fax] context if you are using
faxdetect in your zapata.conf
NOTE2: mailfax is a custom script I wrote. This is what it looks like:
-- START mailfax script --
#!/bin/sh
FAXFILE=$1
RECIPIENT=$2
FAXSENDER=$3
FAXID=`date +%j%H%M%S`
tempfoo=fax
TMPFILE=`mktemp /tmp/${tempfoo}XX`
TMPFILE_A=`mktemp /tmp/${tempfoo}XX`.pdf
/usr/bin/tiff2pdf -p letter ${FAXFILE} ${TMPFILE_A}
metasend -b -t $RECIPIENT -s Fax from $FAXSENDER \
-f ${TMPFILE} -m 'text/plain' -n \
-m 'application/pdf;name=fax'${FAXID}'.pdf' -f ${TMPFILE_A} \
-D 'PDF Fax Document'
rm ${TMPFILE}
rm ${TMPFILE_A}
-- END mailfax script --
I based this config on the excellent information found at the following
website:
http://scottstuff.net/scott/archives/000152.html
But note that I heavily modified the mailfax script from the original
version. Why?
1.) The original used the Perl mime-construct script, which has a lot
of annoying Perl dependencies.
2.) The original didn't send usable MIME attachments to me. Neither
KMail or Evolution could successfully decode the attachment
because the email contained no text/plain part.
My script uses metasend from the metamail package, instead of
mime-construct, and it adds an empty text/plain section at the beginning
of the email so that KMail and Evolution and probably other mail clients
can decode the attachment properly. The trade off is that it has to use
temporary files. Uck. :)
Anyway, the above does everything you need, except the ability to email
the fax to an email specified in voicemail.conf. You'll have to figure
that part out yourself. The above simply emails the message to a static
address. No DIDs or anything.
Also, you'll probably have to change some of the paths for things like
tiff2pdf and metasend.
HTH!
--
Jesse Guardiani, Systems Administrator
WingNET Internet Services,
P.O. Box 2605 // Cleveland, TN 37320-2605
423-559-LINK (v) 423-559-5145 (f)
http://www.wingnet.net
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[Asterisk-Users] about volume in Playback() files
Hi. How can i increment the volume of the files played with Playback or Background() ??? thanks in advance. -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ 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] Outgoing PRI Call Early Media Detection
I am having a problem very similar to several discussed on the list recently. Making outbound calls on a TE410P PRI to several specific outbound numbers is not working for me. Previous posts: problem detecting answer on pri card 800 225 2525 - KLM / NWA reservations http://lists.digium.com/pipermail/asterisk-users/2005-April/098918.html problem detecting answer on pri outcall 800 433 7300 - AA http://lists.digium.com/pipermail/asterisk-users/2005-April/100941.html His PRI trace looks the same as mine! My specific problems are with: 800 789 1331 - Quill Office Supplies 306 239 4885 - Plays a message that this number has been changed These numbers send the audio from their IVR trees back *BEFORE* they give a PRI CONNECT message. They continue to send CALL PROCEEDING messages during the IVR message so that the call is not disconnected. After choosing an option, they then send a CONNECT. This appears to be legal, per this poster's interpretation of ANSI T1.113-1995: http://lists.digium.com/pipermail/asterisk-dev/2004-September/006467.htm l When I call these numbers from a SIP phone, all is well - I hear the message, and am able to choose an IVR selection, even though show channels shows the connection to still be in Dial state. When I call these numbers from another Zap channel, all is very much not well. Because the outgoing channel remains in Dial state, the call is not bridged. I'm using A TE410p, with a Qwest PRI. I've tried changing switchtype from dms100 to national and back. I've tried setting priindication to inband and outofband. But I don't believe the issue is related to those settings anyway. Do other people see this same thing on these numbers? Does Zap not understand this Early Media? Any suggestions? My workaround for now is to Answer() before I Dial() for Zap calls, and extend the dialing timeout on my SIP phones that support early media. Any comments or better solutions? -- Charlie Watts [EMAIL PROTECTED] Mercury Payment Systems - Information Technology 970-385-3187, 800-846-4472 x 3187 ___ 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
Re: [Asterisk-Users] Has anyone had problems with Digium TDM400P and hyperthreading?
[EMAIL PROTECTED] wrote on 04/15/2005 01:45:22 AM: Digium have told us that a problem that we are having (with accuracy of zap interface as measured using zttest) may be due to the fact that we have a Xeon processor with hyperthreading and have suggested turning H/T off. I've never ran Asterisk on an HT-enabled processor. However, I've had too many problems to count with HT and Linux. I turn it off on nearly every server that has it. Then again, most of my servers are not CPU bound and I couldn't care less about the performance. Also, make sure you update your motherboard's BIOS. It's responsible for updating the CPU microcode, and often the BIOS may have newer microcode than your Linux distribution. Anyone else experienced a problem like this? No too keen about turning H/T off, as we're running the SMP RH kernel and don't really feel like replacing the kernel (and other kernel-specific bits) on the off chance that H/T is actually the problem. An SMP kernel should run just fine on a single-processor box. Slower, but fine. At least, it works for me... TIm Massey ___ 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
Re: [Asterisk-Users] cisco 7960 SIP setup
Yes, a few times. All it does is show the following on the screen: Configuring IP, then Configuring CM List then Defaulting Cm to TFTP server, then Opening 66.xx.xx.xx. Then it goes back to the beginning and repeats itself over and over. Mike On Apr 15, 2005, at 1:12 AM, Simone Cittadini wrote: mk111 wrote: I was told that the phone should be able to download the SIP... file once the TFTP address was changed. So far nothing though. Any ideas? have you rebooted the phone after changing the tftp address ? -- Simone Cittadini IT Manager == COMVERT S.R.L. via F.lli Bressan, 21 20126 Milano - ITALY Tel +39.02.27006796(aspetta un beep)105 [EMAIL PROTECTED] http://www.comvert.com ___ 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 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
Re: [Asterisk-Users] cisco 7960 SIP setup
What version of SIP are you trying to load mk111 wrote: Yes, a few times. All it does is show the following on the screen: Configuring IP, then Configuring CM List then Defaulting Cm to TFTP server, then Opening 66.xx.xx.xx. Then it goes back to the beginning and repeats itself over and over. Mike On Apr 15, 2005, at 1:12 AM, Simone Cittadini wrote: mk111 wrote: I was told that the phone should be able to download the SIP... file once the TFTP address was changed. So far nothing though. Any ideas? have you rebooted the phone after changing the tftp address ? -- Simone Cittadini IT Manager == COMVERT S.R.L. via F.lli Bressan, 21 20126 Milano - ITALY Tel +39.02.27006796(aspetta un beep)105 [EMAIL PROTECTED] http://www.comvert.com ___ 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 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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ 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] large analog to asterisk
we are looking at the ability of being able to convert large phone system over to asterisk or if it's possible at all. The building is two sections containing a large office section (with data cabling) and the second section is a hotel with no data cabling. The first section is a no brainer with sip hard and soft phones but the hotel part is where the problem lies. The current count of rooms in the hotel is about 600...that's at a minimum 600 analog connections. Some rooms have 2-3 phones so as a rough number i'm saying 700 total. I see where some people use the Adit 600 to do up to 48 analog connections that trunks over 2 T1 connections back to asterisk but for 700 phones thats 15 Adits with 30 T1'show in the world would you do that?? just several asterisk servers with 2-3 Adits per server? is there any other way? I'm open to suggestions. Thanks.. Shane ___ 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
Re: [Asterisk-Users] Debugging zaphfc + PBX integration
Tim Robinson wrote: Gavin Here is my config: ;NT mode - extension card [channels] nocid=Unavailable withheldcid=Withheld language=en usecallerid=yes callwaiting=yes nationalprefix=0 internationalprefix=00 switchtype = euroisdn signalling = bri_net_ptmp pridialplan=local prilocaldialplan=local echocancel=yes echocancelwhenbridged=no immediate=no overlapdial=yes group = 1 context=cybergear-in channel = 1-2 Don't forget you need a cross-over cable with 100 ohm terminationsa normal ISDN/Cat5e cable will NOT work. This is a soldering iron job. you need to know whether your current ISDN2e line is in Point to Point mode or Point to Multipoint. You then need to set the signalling parameter accordingly. Rgds Tim Gavin Hamill wrote: Howdy - just got meself a cheapy Cologne chip card to play with in the hopes that it will prevail where a £1000 Eicon Diva Server failed... I'm trying to run in NT mode and appear as a trunk to a PBX (it's an Inter-Tel Axxess unit). If I hook a specific port on the BRI board of the PBX directly to a BT ISDN2e wallbox and dial 95000, I get a dialtone and can complete calls. Any time I connect to the Diva Server or the zaphfc card, the handset tells me 'TEST ISDN2 IS UNPLUGGED'... I installed asterisk on a completely virgin box using the install.sh contained in the bristuff 0.2.0-RC7k tarball. /etc/asterisk/zapata.conf [channels] switchtype = euroisdn signalling = bri_net pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context=demo channel = 1-2 /etc/zaptel.conf loadzone=uk defaultzone=uk span=1,1,3,ccs,ami bchan=1-2 dchan=3 Zapata Telephony Interface Registered on major 196 PCI: Enabling device :04:01.0 ( - 0003) ACPI: PCI interrupt :04:01.0[A] - GSI 17 (level, low) - IRQ 185 zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xd002af00 fifo 0xcb3d8000 (0xb3d8000) IRQ 185 HZ 1000 zaphfc: Card 0 configured for NT mode zaphfc: 1 hfc-pci card(s) in this box. Registered tone zone 4 (United Kingdom) zaphfc: card 0 layer 1 state = G2 plugs cable into PBX zaphfc: card 0 layer 1 state = G3 zaphfc: empty HDLC frame or bad CRC received (framelen = 3, stat = 0xfe). zaphfc: empty HDLC frame or bad CRC received (framelen = 40, stat = 0xff). zaphfc: card 0 RX [ 0xfc 0xff 0x3 0xf 0xff 0x6d 0x1 0xff ] 8 bytes zaphfc: card 0 RX [ 0xfc 0xff 0x3 0xf 0xf6 0x3e 0x1 0xff ] 8 bytes zaphfc: card 0 RX [ 0xfc 0xff 0x3 0xf 0x5a 0x1f 0x1 0xff ] 8 bytes I'm told that the RX bytes are the PBX requesting a TEI from the provider (i.e. the * box) but I can't understand why no reply is being given. This is precisely the same problem I had with the Eicon Diva Server card... Is this likely to be as simple as needing the 100 ohm resistors, or can it be fixed with software? Cheers Gavin. ___ 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 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
Re: [Asterisk-Users] Re: Loop Detection
Doug,
Last night I attempted exactly what you just described. I commented
the if { block in chan_sip.c which contains the loop detection code
and tried to place a call. This caused the Asterisk server to dump
core.
I did a little more research into the subject and found this:
Apparently Asterisk's Call ID behavior is slightly modified if you go
into chan_sip.c and set the variable pedanticsipchecking to 1.
This causes Asterisk to not only compare the Call ID of the call but
the TAG= line in the header.
This behavior is more RFC compliant than the default behavior; however
after enabling this and placing a few test calls I seem to get
extremely long delays in establishing the forwarded leg of my calls
and the RTP stream is not being relayed correctly.
I'm currently at a loss. I am seriously considering replacing this
Asterisk TDM gateway with a Cisco 5350.
-Daniel
On 4/15/05, Doug Meredith [EMAIL PROTECTED] wrote:
Daniel Corbe [EMAIL PROTECTED] wrote:
Is there any way to turn Loop Detection off or tune the params a bit?
I am having an issue with Call Forwarding on my SIP Proxy Server which
is causing me great pains.
All I can do is sympathize. The same problem occurs when a call comes
in through Asterisk, gets sent to SER, then comes back to Asterisk 20
seconds later for voicemail.
I have contemplated just commenting out the check in chan_sip.c, but I
haven't tried this. Not sure if this might cause other problems.
Asterisk has many SIP deficiencies. Asterisk has been built as a
monolithic PBX, and it seems to do okay using SIP phones as channels.
If you want Asterisk to simply act as a SIP UA, you are going to run
into a whole slew of problems. I'm not holding my breath waiting for
this to change.
Doug
--
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com
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RE: [Asterisk-Users] Asterisk behind NAT
Can you show your outbound peer configuration? If you are registering, please include that as well. Thanks Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, April 15, 2005 9:44 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT I have... Externip=x.x.x.xand nothing... Does not seem to help in anything. Still my provider sees the private IP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev Sent: Friday, April 15, 2005 9:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT Try setting externip=(asterisk public ip address) Hth Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, April 15, 2005 12:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk behind NAT Hi List, I've spent hours researching on this topic, found tons of info, so far it doesn't work yet. Here's the scenario Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to send calls to an outside provider. My SIP phones (outside * NAT) are able to register with no problem. The problem comes when I send a call out to my provider I get no audio in either way. My provider sees only my private IP. How can I send my provider the public IP?? I have in my sip.conf: [general] port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) nat=yes externip = x.x.x.x localnet=192.168.1.0/24 (remember I enabled DMZ on the router so all ports are being forwarded to *) Despite of what I do, the externip and the localnet my provider only gets my * private IP. Any tip how to solve this?? Thanks ___ 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 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 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 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] H323 Large Scale
Anyone using H323 on asterisk on a larger scale. For example a few million minutes a month? I would like to hear about your experience good or bad. Thanks, Jon ___ 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
RE: [Asterisk-Users] Debugging zaphfc + PBX integration
If you your board into an ISDN wall socket and it works then you are acting as a terminal so you are in terminal mode. Now, how are you connecting to the PBX? If you are connecting to an ISDN extension on the PBX, then still you have to match the kind of connection, whether it is point-2-point or point-2-multipoint. Then it should work. If you are connecting it to an external line on the PBX, then it will have to be in station mode and also you have to work out p2p or p2mp and also have an ISDN crossover cable (not an ethernet crossover cable). So a lot for you to work out. I have successfully got a zpahfc card working on an extension of our PBX and also a PRI card working on an external PRI line of the PBX, so it should work for you in the end. ___ 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
RE: [Asterisk-Users] Asterisk behind NAT
Do you have any phones connected to your * on the internal subnet? Can they make outbound calls? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev Sent: Friday, April 15, 2005 10:15 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT Can you show your outbound peer configuration? If you are registering, please include that as well. Thanks Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, April 15, 2005 9:44 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT I have... Externip=x.x.x.xand nothing... Does not seem to help in anything. Still my provider sees the private IP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev Sent: Friday, April 15, 2005 9:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT Try setting externip=(asterisk public ip address) Hth Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, April 15, 2005 12:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk behind NAT Hi List, I've spent hours researching on this topic, found tons of info, so far it doesn't work yet. Here's the scenario Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to send calls to an outside provider. My SIP phones (outside * NAT) are able to register with no problem. The problem comes when I send a call out to my provider I get no audio in either way. My provider sees only my private IP. How can I send my provider the public IP?? I have in my sip.conf: [general] port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) nat=yes externip = x.x.x.x localnet=192.168.1.0/24 (remember I enabled DMZ on the router so all ports are being forwarded to *) Despite of what I do, the externip and the localnet my provider only gets my * private IP. Any tip how to solve this?? Thanks ___ 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 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 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 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 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
Re: [Asterisk-Users] H323 Large Scale
Yes. I'm using Asterisk CVS-v1-0-01/24/05 with asterisk-oh323-0.6.5 compiled in on two GW's with two DS1's doing RBS wink in each box which then flip the calls to G711ulaw/H.323 back to an Avaya S8700 where the users are at. Using the Varion quad span cards, and the APIC/IO-APIC kernel enabled, I'm finding that I still can't above about 70 or so ports concurrently active on a dual xeon 2.8 gHz machine before sound quality starts to degrade. This doesn't appear to be a CPU issue, but rather one of missed interrupts either on eth0 or tor2, so I've cut back to 48 and that has been very, very stable. These 48 ports are pretty much off hook all through the business day, so for the month, each box will do a little over 600,000 minutes through it. Additionally, it would appear that oh323-0.6.5 does have some leaks of file sockets going on so after a bunch of connects/disconnects, you can max out your avail fh's. I've resolved this by automagically having * restart itself via CRON each night at 20 to 4a when there isn't any utilization. Aside from the two initial bandaids we had to overcome mentioned above, the solution has been rock solid since that time. On 4/15/05, list [EMAIL PROTECTED] wrote: Anyone using H323 on asterisk on a larger scale. For example a few million minutes a month? I would like to hear about your experience good or bad. Thanks, Jon ___ 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 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
RE: [Asterisk-Users] large analog to asterisk
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shane fowler Sent: Friday, April 15, 2005 10:10 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] large analog to asterisk 600 analog connections. Some rooms have 2-3 phones so as a rough number i'm saying 700 total. I see where some people use the Adit 600 to do up to 48 analog connections that trunks over 2 T1 connections back to asterisk but for 700 phones thats 15 Adits with 30 T1'show in the world would you do that?? just several asterisk servers with 2-3 Adits per server? is there any other way? I'm open to suggestions. Remember that in a hospitality environment, the volume of simultaneous calls is typically quite low, given the number of stations in the system. You could use 600's with the CMG-02 cards to backhaul to asterisk via MGCP. Asterisk's MGCP handling is not as robust as it might be, but it may serve your needs. Another option would be to bank on that high stations:calls ratio. In other words, you'll never need to provide 700 DS0's directly into the PBX. We spec'd a very similar (400 stations) hospitality system recently using a slug of Adtran 624's hanging off of an Adtran 830 equipped with 5 quad T1/PRI cards. Careful planning and dial-plan design can keep most inter-station traffic at the 830, with only those calls requiring trunk or PBX feature access traversing a small number of T1's between the 830 and the PBX (asterisk). -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.11 - Release Date: 04/14/2005 ___ 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
Re: [Asterisk-Users] Hylafax and Asterisk
On 08:57, Fri 15 Apr 05, Andrew Yager wrote: Another option (which I think is just as good) is to use the patches available for chan_capi and set it up to receive faxes. Just search the list for chan_capi and fax. Hi, I stumped upon this couple of hours after my post. We have the capiAnswerFax now as replacement for Hylafax. Too bad it doesn't support sending faxes :( -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ 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
RE: [Asterisk-Users] Asterisk behind NAT
I am not registering, only sending calls, here is the config for the general section and for that provider (gw2). [general] context=default ; Default context for incoming calls recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) realm=asterisk ; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=no; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=3600 ; Default length of incoming/outoing registration disallow=all ;allow=ulaw allow=g729 language=en ; Default language setting for all users/peers ; This may also be set for individual users/peers rtptimeout=300 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity ; when we're on hold (must be rtptimeout) ;progressinband=no ; If we should generate in-band ringing always useragent=Asterisk ; Allows you to change the user agent string nat=yes externip = 1.3.5.7 localnet=192.168.1.0/255.255.255.0 [gw2] type=peer port=5060 host=2.4.6.8 disallow=all defaultip=2.4.6.8 allow=g729 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev Sent: Friday, April 15, 2005 1:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT Can you show your outbound peer configuration? If you are registering, please include that as well. Thanks Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, April 15, 2005 9:44 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT I have... Externip=x.x.x.xand nothing... Does not seem to help in anything. Still my provider sees the private IP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev Sent: Friday, April 15, 2005 9:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT Try setting externip=(asterisk public ip address) Hth Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, April 15, 2005 12:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk behind NAT Hi List, I've spent hours researching on this topic, found tons of info, so far it doesn't work yet. Here's the scenario Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to send calls to an outside provider. My SIP phones (outside * NAT) are able to register with no problem. The problem comes when I send a call out to my provider I get no audio in either way. My provider sees only my private IP. How can I send my provider the public IP?? I have in my sip.conf: [general] port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) nat=yes externip = x.x.x.x localnet=192.168.1.0/24 (remember I enabled DMZ on the router so all ports are being forwarded to *) Despite of what I do, the externip and the localnet my provider only gets my * private IP. Any tip how to solve this?? Thanks ___ 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 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 ___
RE: [Asterisk-Users] Asterisk behind NAT
No phones in local LAN but I can try that, let me do that and I'll get back to you. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Sturtevant Sent: Friday, April 15, 2005 1:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT Do you have any phones connected to your * on the internal subnet? Can they make outbound calls? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev Sent: Friday, April 15, 2005 10:15 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT Can you show your outbound peer configuration? If you are registering, please include that as well. Thanks Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, April 15, 2005 9:44 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT I have... Externip=x.x.x.xand nothing... Does not seem to help in anything. Still my provider sees the private IP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev Sent: Friday, April 15, 2005 9:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT Try setting externip=(asterisk public ip address) Hth Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia Sent: Friday, April 15, 2005 12:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk behind NAT Hi List, I've spent hours researching on this topic, found tons of info, so far it doesn't work yet. Here's the scenario Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to send calls to an outside provider. My SIP phones (outside * NAT) are able to register with no problem. The problem comes when I send a call out to my provider I get no audio in either way. My provider sees only my private IP. How can I send my provider the public IP?? I have in my sip.conf: [general] port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) nat=yes externip = x.x.x.x localnet=192.168.1.0/24 (remember I enabled DMZ on the router so all ports are being forwarded to *) Despite of what I do, the externip and the localnet my provider only gets my * private IP. Any tip how to solve this?? Thanks ___ 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 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 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 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 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 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
