[Asterisk-Users] Cannot start asterisk - CAPI issues
Hi List, I have managed to compile asterisk but I can't start it. What I have done so far as asterisk config is concerned is cut and paste the sample config files from the ONLamp article on Asterisk. http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html When I start asterisk -vvvp I get quite a few messages, notably: chan_capi.c:2603 load_module: Unable to load config capi.conf and: load_modules: Loading module chan_capi.so failed! Now I *do* have some kind of ISDN card in the box which I have not worried about yet (Eicon Diva 2.01 S/T PCI according to lspci) and I understand that CAPI has something to do with ISDN, but I have no clue as why asterisk doesn't start... Did I forget something when I compiled asterisk? Any ideas? Cheers, Jean-Michel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] video conferencing with sip
Hello , Has anyone explored video conferencing on Asterisk with SIP ? I dont want to use H.323 as everything else is SIP based in the set up. I have gone through the lists but there doesnt seem to be any info about video conferencing with SIP. I dont have any users dialing in , but every one is connected through a private WAN IP backbone. The Head office wants to talk to all the branch offices and only the head office video need be displayed .Or if permission is granted , someone else' video may also be seen . I hope you understand what I am trying to get at. Any suggestions or any info about equipment, config , etc. wld be greatly appreciated. Thanks in advance, Shireen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk RELOAD option stability
I was wondering how the reload option in asterisk (asterisk -r -x reload) affects calls in session and other activity like active AGI aplications. I tried it using a single call which i placed to my asterisk box and it didnt get disconnected when i reloaded asterisk. But what about heavy load envoirments with say 50 calls in session. Also does frequent reloads affect the stability of asterisk i mean things does it lead to things like memory leaks -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom IP 500/600
So what? You said you had ssh access. Use ssh forward. Here's another way. Might work for you, I don't know. You could easily setup a secure tunnel (think openvpn) to run your ftp server on locally. That way you could keep all the configs in one place. You could open the tunnel when it's time to update the phones and then have the phones use your local server as the ftp source. You just reboot the phones remotely and close the tunnel once all the phones have updated. With this method, run ntp on both sides so that the timestamp change thingy will work right. Really, there must be a million ways to do this. I think you're trying too hard. John Richard wrote: If the phone is behind a NAT firewall, it would require extra configuration on the firewall. Depending on the circumstance, it is not always be possible to make such a change. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Karl J. Vesterling *Sent:* Saturday, October 30, 2004 6:30 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] polycom IP 500/600 My bad... I thought he was attempting to upload config files for asterisk systems. Yes, an expect script would work just fine... At 11:21 PM 10/30/2004, you wrote: The phone has a web interface. Couldn't you just use an expect script to change it? John Baker Karl J. Vesterling wrote: One could use SCP with certificates for authentication and avoid all the issues with FTP and it's vulnerabilities. At 07:55 PM 10/26/2004, you wrote: Richard wrote: Hi Kristian, I'd like to use ftp because of several advantages it has. For example, ability to change the time stamp and reload the phone. But the default password is a big issue. I'd like to change it but don't want to go to each phone and reset it. Any way to change it? Thanks, I understand why you would want to use FTP (no filename changes). Why is the default password such a big issue? This is a chicken or the egg - how is the phone supposed to know it's new ftp password BEFORE it can get the config file - via FTP!?! -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Best Regards, Karl J. Vesterling *E-Mail:* [EMAIL PROTECTED] *Yahoo Messenger:* karl_vesterling *ICQ: *1548052 *AOL Instant Messenger:* n2vqm *Telephone: Washington DC:* (202) 448-3009 Extension 0 *Annapolis MD:* (240) 524-6706 Extension 0 *Seattle WA:* (360) 516-1822 Extension 0 *Niagara Falls NY:* (716) 286-9175 Extension 0 *Buffalo NY:* (716) 608-1121 Extension 0 *United Kingdom:* 0870 3403428 Extension 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Best Regards, Karl J. Vesterling *E-Mail:* [EMAIL PROTECTED] *Yahoo Messenger:* karl_vesterling *ICQ: *1548052 *AOL Instant Messenger:* n2vqm *Telephone: Washington DC:* (202) 448-3009 Extension 0 *Annapolis** MD**:* (240) 524-6706 Extension 0 *Seattle** WA**:* (360) 516-1822 Extension 0 *Niagara Falls** NY**:* (716) 286-9175 Extension 0 *Buffalo** NY**:* (716) 608-1121 Extension 0 *United Kingdom**:* 0870 3403428 Extension 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Wireless phones connected to VOIP DECT basestation
On Sun, 31 Oct 2004, Remco Barende wrote: I will probably order the base station, it seems like an almost ideal solution to connect phones to a voip pabx. I would not prefer a pci card solution personally, anything connected to the network doesn't cause irq headaches :) On the other hand a pci base station may allow a much lower latency. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] polycom IP 500/600
I think that the topic is sidetracked... My original question is about how to change the default username and password for ftp login. I want to change it, but don't want to punch the keypad manually. I don't think that this can be done via web interface either. Richard -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of John Baker Sent: Saturday, October 30, 2004 9:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] polycom IP 500/600 So what? You said you had ssh access. Use ssh forward. Here's another way. Might work for you, I don't know. You could easily setup a secure tunnel (think openvpn) to run your ftp server on locally. That way you could keep all the configs in one place. You could open the tunnel when it's time to update the phones and then have the phones use your local server as the ftp source. You just reboot the phones remotely and close the tunnel once all the phones have updated. With this method, run ntp on both sides so that the timestamp change thingy will work right. Really, there must be a million ways to do this. I think you're trying too hard. John Richard wrote: If the phone is behind a NAT firewall, it would require extra configuration on the firewall. Depending on the circumstance, it is not always be possible to make such a change. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Karl J. Vesterling *Sent:* Saturday, October 30, 2004 6:30 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] polycom IP 500/600 My bad... I thought he was attempting to upload config files for asterisk systems. Yes, an expect script would work just fine... At 11:21 PM 10/30/2004, you wrote: The phone has a web interface. Couldn't you just use an expect script to change it? John Baker Karl J. Vesterling wrote: One could use SCP with certificates for authentication and avoid all the issues with FTP and it's vulnerabilities. At 07:55 PM 10/26/2004, you wrote: Richard wrote: Hi Kristian, I'd like to use ftp because of several advantages it has. For example, ability to change the time stamp and reload the phone. But the default password is a big issue. I'd like to change it but don't want to go to each phone and reset it. Any way to change it? Thanks, I understand why you would want to use FTP (no filename changes). Why is the default password such a big issue? This is a chicken or the egg - how is the phone supposed to know it's new ftp password BEFORE it can get the config file - via FTP!?! -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Best Regards, Karl J. Vesterling *E-Mail:* [EMAIL PROTECTED] *Yahoo Messenger:* karl_vesterling *ICQ: *1548052 *AOL Instant Messenger:* n2vqm *Telephone: Washington DC:* (202) 448-3009 Extension 0 *Annapolis MD:* (240) 524-6706 Extension 0 *Seattle WA:* (360) 516-1822 Extension 0 *Niagara Falls NY:* (716) 286-9175 Extension 0 *Buffalo NY:* (716) 608-1121 Extension 0 *United Kingdom:* 0870 3403428 Extension 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Best Regards, Karl J. Vesterling *E-Mail:* [EMAIL PROTECTED] *Yahoo Messenger:* karl_vesterling *ICQ: *1548052 *AOL Instant Messenger:* n2vqm *Telephone: Washington DC:* (202) 448-3009 Extension 0 *Annapolis** MD**:* (240) 524-6706 Extension 0 *Seattle** WA**:* (360) 516-1822 Extension 0 *Niagara Falls** NY**:* (716) 286-9175 Extension 0 *Buffalo** NY**:* (716) 608-1121 Extension 0 *United Kingdom**:* 0870 3403428 Extension 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] IAX2 bandwidth efficiency calculations from Farfon
On Sat, 30 Oct 2004 21:15:51 -0400, Steve Kann [EMAIL PROTECTED] wrote: The chart is good, but I think it makes a mistake for iLBC: Isn't iLBC 13.something kbps? Also, since iLBC uses 30ms frames (when used with asterisk, at least), it has slightly lower overhead. Approx 2/3 as much overhead. I had assumed that this was the reason why Wasim used 9kbps for ILBC. 13.5 * 2/3 = 9 But, you are right, there should be a footnote somewhere that says so. (not that I'm a big iLBC fanboy or anything.. -- I still prefer a free codec). Indeed. Also, ILBC is more forgiving on packet loss. G729 sucks with packet loss. In my experience the combination of IAX and ILBC is what makes reliable VoIP possible in third world countries with poor internet infrastructure. Places where SIP+G729 simply does not work. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk RELOAD option stability
On Sun, 31 Oct 2004 08:18:22 +0100, Vikram Rangnekar [EMAIL PROTECTED] wrote: Also does frequent reloads affect the stability of asterisk i mean things does it lead to things like memory leaks Depends on the version of Asterisk you are using and your environment. I have seen frequent reloads crashing certain Asterisk installations. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] video conferencing with sip
Hi, -Original Message- Has anyone explored video conferencing on Asterisk with SIP ? I dont want to use H.323 as everything else is SIP based in the set up. I have gone through the lists but there doesnt seem to be any info about video conferencing with SIP. I dont have any users dialing in , but every one is connected through a private WAN IP backbone. The Head office wants to talk to all the branch offices and only the head office video need be displayed .Or if permission is granted , someone else' video may also be seen . I hope you understand what I am trying to get at. Any suggestions or any info about equipment, config , etc. wld be greatly appreciated. The MeetMe tool does not support video at this time. Asterisk does support videocodecs to be transported over SIP or IAX links, so the only thing that would be required is distribution of video RTP in the MeetMe app, and some way to control that. Hasn't been done as far as I know. It would be great if you could do some research/work in this direction, it's something I'd love to see, but unfortunately I have other things higher on my priorities list. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.711alaw to iLBC
Hi All, I was doing some testing between on extension running SIP at G.711alaw and an IAX extension runing iLBC (also GSM) and found that the voice from the IAX user has a lot of packet loss (very bad voice quality) toward the SIP phone only. From SIP phone to IAX, voice quality is fine. SIP phone is local to * and IAX phone is remote to asterisk. Anyone have any ideas? Could this be a jitter problem, I am not using a jitter buffer in the iax.conf. If I test with two IAX phones both running iLBC the voice quality is fine in both directions. Regards Garry Taylor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modifying CDR data?
I've written a small AGI thing to allow lots of stuff, including diverts. If a call is placed to a diverted number, a new call is initiated from * to that number. Simple. But then, to make billing sane, I need to change the 'dst' in CDR to reflect the number diverted to. How can I do this? I don't think you can change dst from the extension flow just like that (maybe via an app, but that might have alternate consequences) I've done some scripting with entirely different purposes, but it may fit your needs: create an AGI script that is called when a call comes in, use that to store the uniqueid of the call leg into a database. Then check if call diversion is active and log that too. Afterwards, check (i.e. once an hour or whatever is convenient) and match CDR versus your own database. I'm in an AGI script, and I've tried to ForkCDR. This gave me two CDR records (original src,original dst) (original dst,original dst) I want to change the latter to (original dst,diverted dst) ...and I really want to do as much as possible with the stuff available in asterisk. roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.711alaw to iLBC
On Sun, 31 Oct 2004, Garry Taylor wrote: Hi All, I was doing some testing between on extension running SIP at G.711alaw and an IAX extension runing iLBC (also GSM) and found that the voice from the IAX user has a lot of packet loss (very bad voice quality) toward the SIP phone only. From SIP phone to IAX, voice quality is fine. SIP phone is local to * and IAX phone is remote to asterisk. Anyone have any ideas? Could this be a jitter problem, I am not using a jitter buffer in the iax.conf. If I test with two IAX phones both running iLBC the voice quality is fine in both directions. Are you trying to use IAX trunking? If so, try turning it off. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] make transfert and hold with FXS device
Hi,
I'm testing different VOIP hardware with asterisk and try to transfert and
hold a call.
My test with SIPphone (grandstream BT and cisco 7940) and softphone
(sjphone) are ok when I'm using dtmfmode=info.
But with FXS devices (GS Handytone and Vega50 FXS) and very simple phone
(10 digits, #, * and R button), I can't
place the call on hold... and can not make a transfert.
In sip debug mode, I could see the DTMF in the sip messages but if I push
on the 'R' button asterisk hangup the call.
is there a special code,like other PABX, for this functionnality ? for
example : R+1 = hold, R+2 = park...
my sip.conf
;
; SIP Configuration for Asterisk
;
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all; First disallow all codecs
allow=alaw ; Allow codecs in order of preference
allow=ulaw
musicclass=default
language=fr
rtptimeout=60
rtpholdtimeout=300
dtmfmode=info
[6430]
type=friend ; either friend (peer+user), peer or
user
context=TONALITE
host=dynamic
callerid=6430
canreinvite=no ; allow RTP voice traffic to bypass
Asterisk
my extensions.conf
[general]
static=yes
writeprotect=no
[TONALITE]
; Plage VOIP TONALITE
exten = _643X,1,Dial(SIP/${EXTEN},15)
exten = _643X,2,Hangup()
exten = _643X,102,Hangup()
thanks
___
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RE: [Asterisk-Users] G.711alaw to iLBC
IAX extension, ie. firefly. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, 31 October 2004 5:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] G.711alaw to iLBC On Sun, 31 Oct 2004, Garry Taylor wrote: Hi All, I was doing some testing between on extension running SIP at G.711alaw and an IAX extension runing iLBC (also GSM) and found that the voice from the IAX user has a lot of packet loss (very bad voice quality) toward the SIP phone only. From SIP phone to IAX, voice quality is fine. SIP phone is local to * and IAX phone is remote to asterisk. Anyone have any ideas? Could this be a jitter problem, I am not using a jitter buffer in the iax.conf. If I test with two IAX phones both running iLBC the voice quality is fine in both directions. Are you trying to use IAX trunking? If so, try turning it off. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sip CallerPres support?
hi we're interested in CallerPres in chan_sip. what will it take to implement it? roy On Oct 25, 2004, at 4:37 PM, Race Vanderdecken wrote: Roy et All, If someone could expand on CallerPres requirements in chan_sip I can do the work. I have added numerous extras to chan_sip already, RADIUS, new CDRs, Dynamic Dial plans, Find-Me, Follow-Me and such. I am just one programmer, but let me know what needs to be done and I can create the code fairly quickly. Race Vanderdecken Asterisk aT vanderDecken period coM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: 24 October 2004 08:14 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip CallerPres support? hi would it be hard to implement CallerPres support in chan_sip? roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN EDSS1 protocol support
On Fri, 2004-10-29 at 13:41, Maxim Litnitsky wrote: Hi all, I have to implement the following: -- | 10 voice channels |---| Prov E1 | 256 kbit/s for VoIP | Asterisk IP-PBX | | 256 kbit/s for Data (http,mail) - |---| -- Provider gives E1 and on this E1 I will have 10 timeslots for voice, and others for internet. What hardware shall I use? Provider supports EDSS1 ISDN protocol, as I undertood Digium hardware does not support this protcol. I searched google and lists.digium.com and found only this: http://www.redhat.com/archives/fedora-list/2004-October/msg03224.html http://www.mail-archive.com/[EMAIL PROTECTED]/msg30870.html The question: Can I implement all with Asterisk using EDSS1 protcol and how? Give me please a clue!! You didn't define your question good enough. Digium hardware does support EDSS1 (EuroISDN) without problems. However, you didn't say, how your provider let you connect to the internet. You have 30 channels on your E1 (30 timeslots / 64 kbit), not counting the d-channel, which is a total of 2 mbit. Implenting the 10 voice channels is a std. setup, but your provider still needs to tell you, how you access the internet/data part. EDSS1 is only a ISDN signalling protocol, you would probably have to run something like PPP over the lasting 20 channels (or how many your provider has assigned there) to get connectivity. Get better specifications from your provider !!! If your provider indeed is using PPP, then you should have a look at ZapRAS in Asterisk (http://www.voip-info.org/wiki-Asterisk+cmd+ZapRAS) Kind regards, Martin List-Petersen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't install the mfcr2 support correctly
Hi Mr Jack, hi everybody Thank you for your answer for my message titled can't run ztcfg. I tried what you proposed me and the error I told about is not signaled. However I still have problems to get mfc/r2 support running. I refered to the mfcr2 support documentation available in the opencall.org website http://www.opencall.org/installing-mfcr2.html . The digium card I have is the TE410P After installing the zaptel driver using the following commands: make clean make install make config After adding the necessary lines to the zaptel.conf file. The problem is that when I execute the following command : modprobe wct4xxp which is necessary to run ztcfg correctly (as you told me) I get the 4 ports lights of the card off. Yesterday I got them red. And when I connect an E1 to a port I don't get a green light but I get either a red light (or no light at all) if the light was already red or no light if the light was already off. I hope that my message is clear and that someone will help me. _ MSN Hotmail : antivirus et antispam intégrés http://www.msn.fr/newhotmail/Default.asp?Ath=f ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phone that OFFICIALLY support Asterisk
On Thu, 21 Oct 2004 09:39:48 +0900, Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote: Wed, 20 Oct 2004 15:47:59 -0500, Henry Devito [EMAIL PROTECTED] wrote: Where can I buy the act phones? I have now discussed the matter of sample orders and shipments with ACT directly and I have emailed everybody who had contacted me -- or at least I hope so. If there is anybody who is interested in ordering a sample who hasn't been contacted directly -- or somebody coming late to the party -- please email me off-list at: benjamin (at) sunrise-tel (dot) com. We are currently trying to put a combined order for samples together so as to minimise the overheads (bank charges and shipping). rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP test numbers
All you really need is a list of 1-800 numbers in various countries. Most multi-national corporations have a list buried somewhere on their web site. For example: http://www.microsoft.com/resources/howtotell/ww/windows/what.aspx Gilad ;-) -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] moh
My solution to this (as the debian package appears to actually download mpg321 (instead of mpg123) when you install *, was to download mpg123 from the original website and compile/install it myself. http://www.mpg123.de/ mpg123 0.59r is the version im now running (just copied the executable over mpg123 and mpg321 and restarted asterisk (and killed dead looking mpg321 processes) started up astersik, caleld myself and shoved myself on hold, and VOILA, music on hold is working normally and not running 'really' slow Hope this helps! Richard wrote: Hi, I have * 1.0.0. Everything works well except moh. I followed the instruction in http://voip-info.org/wiki-Asterisk+config+musiconhold.conf. I use the default mp3 from *. The problem is that the music is really slow. Seems like it didn't get the right rate to play. Any one having this problem too? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phone that OFFICIALLY support Asterisk
in response to many queries asking for a URL ... http://www.voip-info.org/tiki-index.php?page=ACT%20P104SLD%20IP%20Phone rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom failed registration - Cant figureoutwhats wrong
Thanks John, that worked. I guess that's a pretty common mistake :) Now to build the rest of my config files, that's always fun. On Fri, 29 Oct 2004 19:26:10 -0400, John Bittner [EMAIL PROTECTED] wrote: I just read what I typed... I meant to say put the 614p in the reg.1.address field with out the ip. reg.1.address=614p Sometimes I am dyslexic. John B -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Sent: Friday, October 29, 2004 5:18 PM To: 'Matthew Marlowe'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom failed registration - Cant figureoutwhats wrong Hi, Remove the 614p@ from reg.1.address=[EMAIL PROTECTED] John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Friday, October 29, 2004 5:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom failed registration - Cant figure outwhats wrong Can anyone tell me if the below is wrong for the phone configuration, it keeps failed registration. (I had this working but lost all my tftp config files so I know its a work configuration) 614p is my username password is my password and 10.20.30.3 is the asterisk box Thanks in advance. reg reg.1.displayName=614p reg.1.address=[EMAIL PROTECTED] reg.1.label=614p reg.1.type=private reg.1.thirdPartyName=614p reg.1.auth.userId=614p reg.1.auth.password=password reg.1.server.1.address=10.20.30.3 reg.1.server.1.port=5060 reg.1.server.1.transport= reg.1.server.1.expires=360 reg.1.server.1.register= reg.1.server.1.retryTimeOut= reg.1.server.1.retryMaxCount= reg.1.server.1.expires.lineSeize= reg.1.acd-login-logout=0 reg.1.acd-agent-available=0 -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phone that OFFICIALLY support Asterisk
Most phone manufacturers support Asterisk unless they also provide a PBX product. I have seen postings from snom employees on this list (they even sell their own competing switch) - Original Message - From: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 31, 2004 8:52 AM Subject: Re: [Asterisk-Users] IP Phone that OFFICIALLY support Asterisk in response to many queries asking for a URL ... http://www.voip-info.org/tiki-index.php?page=ACT%20P104SLD%20IP%20Phone rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot start asterisk - CAPI issues
chan_capi.c:2603 load_module: Unable to load config capi.conf You need to create this file /etc/asterisk/capi.conf with the following content : [general] nationalprefix=0 internationalprefix=00 [interfaces] msn=50 incomingmsn=* controller=1 softdtmf=0 accountcode= context=incoming ;echosquelch=1 echocancel=no ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=30 Adjust devices= with the number of B channels supported by your card. For ISDN BRI, it's 2, for PRI, it's 30. Now I *do* have some kind of ISDN card in the box which I have not worried about yet (Eicon Diva 2.01 S/T PCI according to lspci) and I understand that CAPI has something to do with ISDN, but I have no clue as why asterisk doesn't start... You need a kernel support for you card and you also need to load a firmware for some cards. If you have a message like CAPI not installed!, check your kernel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] confusing info from Digium andasteriskdoc aboutsetup of TDM11B
Just put a note that channels may vary do to placement of modules. I think that would be more correct. Also, try a different phone. I had this problem with a cheap cordless once. Give us output from the console. Give me SSH and I will have it working quickly. - Original Message - From: Steve Prior [EMAIL PROTECTED] To: Leif Madsen [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 31, 2004 12:18 AM Subject: Re: [Asterisk-Users] confusing info from Digium andasteriskdoc aboutsetup of TDM11B Looks like it's still incorrect in the first blue paragraph of the section on FXO (it's fixed in the second blue paragraph). Also, the last paragraph of that section twice still calls the channel # 2. Now on to my next confusion... The section on contexts under dislplans mentions a context named [incoming]. This isn't a context that's mentioned anywhere before this and it's not at all clear where it comes from - I'm starting to suspect that some context references belong in the zapatel.conf file. A comment about where the document leaves off. In the beginning the document promises to get to a minimal working set, but it really doesn't go that far. Unless I've missed something, we aren't left with even a complete version of the minimal example extensions.conf file. Something is missing so that I'm not getting a dial tone on the analog phone hooked up to the TDM11B and I have no idea why (can anyone clue me in?) I also tried the: [incoming] exten = s,1,Answer() exten = s,2,Playback(goodbye) exten = s,3,Hangup() example and asterisk didn't appear to see the incoming call and answer the call at all. I'd love for the example files to be complete enough that this example could actually work from either the external POTS line or even better an analog phone hooked to the FXS interface. I think it would be great if attached to the document there was a final version of all of the config files which are known to work with the given configuration. Can you help get me to a dialtone on the internal side or an answer on the external side? Thanks Steve Leif Madsen wrote: On Sat, 30 Oct 2004 12:18:20 -0400, Steve Totaro [EMAIL PROTECTED] wrote: Yes, it should be four unless you care to move the actual module on the card to the second slot. I have fixed this in CVS now. Should be propogated to the website in a few minutes. While we do try and test everything, sometimes things get missed. This is why getting people to test the configurations in Volume-One and report back what does and does not work is important. Thanks for pointing one out! Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't install the mfcr2 support correctly
Dear Khaled, I thing you must read the documentation a little bit more deapely! does zaptel compile ok ? which kernel are you using ? have you configure the zaptel.conf file which parameters are you using for r2 signaling ? refer to this page as guide for starting http://www.asterisk.org/index.php?menu=download read the all page no lights aster wct4xxp means span is not configured! good luck ! Jack - Original Message - From: Abdelghani Khaled [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Sunday, October 31, 2004 12:04 PM Subject: [Asterisk-Users] Can't install the mfcr2 support correctly Hi Mr Jack, hi everybody Thank you for your answer for my message titled can't run ztcfg. I tried what you proposed me and the error I told about is not signaled. However I still have problems to get mfc/r2 support running. I refered to the mfcr2 support documentation available in the opencall.org website http://www.opencall.org/installing-mfcr2.html . The digium card I have is the TE410P After installing the zaptel driver using the following commands: make clean make install make config After adding the necessary lines to the zaptel.conf file. The problem is that when I execute the following command : modprobe wct4xxp which is necessary to run ztcfg correctly (as you told me) I get the 4 ports lights of the card off. Yesterday I got them red. And when I connect an E1 to a port I don't get a green light but I get either a red light (or no light at all) if the light was already red or no light if the light was already off. I hope that my message is clear and that someone will help me. _ MSN Hotmail : antivirus et antispam intégrés http://www.msn.fr/newhotmail/Default.asp?Ath=f ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom failed registration - Cant figureoutwhats wrong
OK, Now I'm confused. It was working but I was using TFTP. I wanted to use FTP so I just copied the config files to the ftp server, changed the login info on the phone to FTP. Now the phone doesnt login via ftp and get the config files but it won't even try to register now. Has anyone ever seen this? On Sun, 31 Oct 2004 09:06:58 -0500, Matthew Marlowe [EMAIL PROTECTED] wrote: Thanks John, that worked. I guess that's a pretty common mistake :) Now to build the rest of my config files, that's always fun. On Fri, 29 Oct 2004 19:26:10 -0400, John Bittner [EMAIL PROTECTED] wrote: I just read what I typed... I meant to say put the 614p in the reg.1.address field with out the ip. reg.1.address=614p Sometimes I am dyslexic. John B -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Sent: Friday, October 29, 2004 5:18 PM To: 'Matthew Marlowe'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom failed registration - Cant figureoutwhats wrong Hi, Remove the 614p@ from reg.1.address=[EMAIL PROTECTED] John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Friday, October 29, 2004 5:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom failed registration - Cant figure outwhats wrong Can anyone tell me if the below is wrong for the phone configuration, it keeps failed registration. (I had this working but lost all my tftp config files so I know its a work configuration) 614p is my username password is my password and 10.20.30.3 is the asterisk box Thanks in advance. reg reg.1.displayName=614p reg.1.address=[EMAIL PROTECTED] reg.1.label=614p reg.1.type=private reg.1.thirdPartyName=614p reg.1.auth.userId=614p reg.1.auth.password=password reg.1.server.1.address=10.20.30.3 reg.1.server.1.port=5060 reg.1.server.1.transport= reg.1.server.1.expires=360 reg.1.server.1.register= reg.1.server.1.retryTimeOut= reg.1.server.1.retryMaxCount= reg.1.server.1.expires.lineSeize= reg.1.acd-login-logout=0 reg.1.acd-agent-available=0 -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MBM -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] video conferencing with sip
I'm prepared to kick off a bounty to get some form of video conference meet me solution going. My specifications would be for a minimum of 4 people in the conference and to have some form of web page control, kick off-join-mute, mute all. Is there some formal way of setting up a bounty on asterisk wiki? I pledge $US250 to begin with however I may increase that should someone show me something fruitful. Anyone else able to/want to kick in some pledges to make this happen. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Florian Overkamp Sent: Sunday, October 31, 2004 4:40 AM To: 'Sayeeda Shireen'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] video conferencing with sip Hi, -Original Message- Has anyone explored video conferencing on Asterisk with SIP ? I dont want to use H.323 as everything else is SIP based in the set up. I have gone through the lists but there doesnt seem to be any info about video conferencing with SIP. I dont have any users dialing in , but every one is connected through a private WAN IP backbone. The Head office wants to talk to all the branch offices and only the head office video need be displayed .Or if permission is granted , someone else' video may also be seen . I hope you understand what I am trying to get at. Any suggestions or any info about equipment, config , etc. wld be greatly appreciated. The MeetMe tool does not support video at this time. Asterisk does support videocodecs to be transported over SIP or IAX links, so the only thing that would be required is distribution of video RTP in the MeetMe app, and some way to control that. Hasn't been done as far as I know. It would be great if you could do some research/work in this direction, it's something I'd love to see, but unfortunately I have other things higher on my priorities list. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot start asterisk - CAPI issues
Thanks for the tip! I'm still having a couple of quirks though... Adjust devices= with the number of B channels supported by your card. For ISDN BRI, it's 2, for PRI, it's 30. Okay, I did that but then I had the exact error you describe below... You need a kernel support for you card and you also need to load a firmware for some cards. If you have a message like CAPI not installed!, check your kernel. Well I do get that message, but I can modprobe capi, see it with lsmod and also see /dev/isdn and /dev/capi20 - so I *assume* the eicon diva card is sort-of-recognized (not sure though... I have zero experience with ISDN cards...) I use the linux 2.6.7 kernel which came with the knoppix distro I've installed on the box. I have checked with make xconfig and the options for isdn support and capi support all seem to be there OK. Thanks to the list I have managed to compile asterisk from CVS head yesterday and my current target is to be able to pick up the sip phone, dial an extension and hear a little music. Very little ambitions for now :-) Anyway, any ideas on what might be wrong with this capi stuff? I suppose I could try and chuck the isdn card but I'll need it later... Cheers, Jean-Michel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN CARD
Im looking for a ISDN card that works under asterisk and supports BRI line. And I just can`t findit. Momently im using card INTERNAL, but Im having problems, asterisk on startup when loading modem fails (i4l driver). Can you please help me, or point to a www address where culd I find some help. Best regards, Bostjan Repnik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom failed registration - Cant figureoutwhats wrong
Ok. Nevermind. For some reason this one phone won't connect to my internal ip of 10.20.30.2 but it's able to connect to the external ip where all of the other phones are able to connect to 10.20.30.2... So that's an internal problem. So the configs do work. On Sun, 31 Oct 2004 09:46:50 -0500, Matthew Marlowe [EMAIL PROTECTED] wrote: OK, Now I'm confused. It was working but I was using TFTP. I wanted to use FTP so I just copied the config files to the ftp server, changed the login info on the phone to FTP. Now the phone doesnt login via ftp and get the config files but it won't even try to register now. Has anyone ever seen this? On Sun, 31 Oct 2004 09:06:58 -0500, Matthew Marlowe [EMAIL PROTECTED] wrote: Thanks John, that worked. I guess that's a pretty common mistake :) Now to build the rest of my config files, that's always fun. On Fri, 29 Oct 2004 19:26:10 -0400, John Bittner [EMAIL PROTECTED] wrote: I just read what I typed... I meant to say put the 614p in the reg.1.address field with out the ip. reg.1.address=614p Sometimes I am dyslexic. John B -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Sent: Friday, October 29, 2004 5:18 PM To: 'Matthew Marlowe'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom failed registration - Cant figureoutwhats wrong Hi, Remove the 614p@ from reg.1.address=[EMAIL PROTECTED] John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Friday, October 29, 2004 5:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom failed registration - Cant figure outwhats wrong Can anyone tell me if the below is wrong for the phone configuration, it keeps failed registration. (I had this working but lost all my tftp config files so I know its a work configuration) 614p is my username password is my password and 10.20.30.3 is the asterisk box Thanks in advance. reg reg.1.displayName=614p reg.1.address=[EMAIL PROTECTED] reg.1.label=614p reg.1.type=private reg.1.thirdPartyName=614p reg.1.auth.userId=614p reg.1.auth.password=password reg.1.server.1.address=10.20.30.3 reg.1.server.1.port=5060 reg.1.server.1.transport= reg.1.server.1.expires=360 reg.1.server.1.register= reg.1.server.1.retryTimeOut= reg.1.server.1.retryMaxCount= reg.1.server.1.expires.lineSeize= reg.1.acd-login-logout=0 reg.1.acd-agent-available=0 -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MBM -- MBM -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot start asterisk - CAPI issues
You need a kernel support for you card and you also need to load a firmware for some cards. If you have a message like CAPI not installed!, check your kernel. I use the linux 2.6.7 kernel which came with the knoppix distro I've installed on the box. I have checked with make xconfig and the options for isdn support and capi support all seem to be there OK. It's not enough, you must compile the correct Eicon driver. Read /usr/src/linux/Documentation/isdn/README.eicon Usually, you also need to load a firmware (with eiconctrl). Check out behind your card, when succesfully started, LEDs are turned on. For old cards, you may try isdn4linux instead CAPI. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk compile error
I get the following error when I try to compile asterisk on my redhat 9 box any ideas? CVS version from October 22, 2004 PIC -c -o pbx_dundi.o pbx_dundi.c pbx_dundi.c:54:18: zlib.h: No such file or directory pbx_dundi.c: In function `update_key': pbx_dundi.c:1313: warning: implicit declaration of function `crc32' pbx_dundi.c: In function `dundi_decrypt': pbx_dundi.c:1369: warning: implicit declaration of function `uncompress' pbx_dundi.c:1369: `Z_OK' undeclared (first use in this function) pbx_dundi.c:1369: (Each undeclared identifier is reported only once pbx_dundi.c:1369: for each function it appears in.) pbx_dundi.c: In function `dundi_encrypt': pbx_dundi.c:1394: warning: implicit declaration of function `compress' pbx_dundi.c:1395: `Z_OK' undeclared (first use in this function) make[1]: *** [pbx_dundi.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/pbx' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk]# -Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zapateller broken in ver 1.0.2?
In a recent upgrade to version * 1.0.2 I have noticed a new behavior in the Zapateller() function. It now produces the 3 tones you get when you hear the were sorry message from the phone company. Anybody notice this New feature? Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Website Design www.cirelle.net ProSpeed High Speed Dial-up - 5 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk RELOAD option stability
+++ Benjamin on Asterisk Mailing Lists [31/10/04 18:11 +0900]: On Sun, 31 Oct 2004 08:18:22 +0100, Vikram Rangnekar [EMAIL PROTECTED] wrote: Also does frequent reloads affect the stability of asterisk i mean things does it lead to things like memory leaks Depends on the version of Asterisk you are using and your environment. I have seen frequent reloads crashing certain Asterisk installations. rgds benjk I'm thinking about a Zap+sip type install servicing say 50 sip connections and say 1 e1/t1 line. normal day to day operations in an office type envoirment. Asterisk version 1.0.2 (stable) . and not using any database connectivity modules like app_data, cdr_mysql etc. What kind of a setup do you experience these crashes under._ -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] norwegian sounds for Asterisk
Does anyone have norwegian sounds (audiopack) for Asterisk? Please send me an url for download if so. (sounds for voicemail too) Best regards LOH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zapateller broken in ver 1.0.2?
Cirelle Enterprises wrote: In a recent upgrade to version * 1.0.2 I have noticed a new behavior in the Zapateller() function. It now produces the 3 tones you get when you hear the were sorry message from the phone company. Anybody notice this New feature? SIT aka Special Information Tone is the three-tone combo you are talking about. Sounds like your installation was broken. fs-2*CLI show application zapateller fs-2*CLI -= Info about application 'Zapateller' =- [Synopsis]: Block telemarketers with SIT [Description]: Zapateller(options): Generates special information tone to block telemarketers from calling you. Returns 0 normally or -1 on hangup. Options is a pipe-delimited list of options. The following options are available: 'answer' causes the line to be answered before playing the tone, 'nocallerid' causes Zapateller to only play the tone if there is no callerid information available. Options should be separated by | characters fs-2*CLI ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic
Does any body have any information about Dialogic MSI board workink with asterisk. Robin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialogic
Robin van Leyden wrote: Does any body have any information about Dialogic MSI board workink with asterisk. According to this document the MSI model is not supported: http://www.asteriskpbx.org/index.php?menu=hardware Keep in mind that the Dialogic drivers for Asterisk are closed source and cost money. Contect Digium for details, of course. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] moh
Thanks Matthew, You are the MAN! It fixed the problem. Richard -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matthew Sent: Sunday, October 31, 2004 3:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] moh My solution to this (as the debian package appears to actually download mpg321 (instead of mpg123) when you install *, was to download mpg123 from the original website and compile/install it myself. http://www.mpg123.de/ mpg123 0.59r is the version im now running (just copied the executable over mpg123 and mpg321 and restarted asterisk (and killed dead looking mpg321 processes) started up astersik, caleld myself and shoved myself on hold, and VOILA, music on hold is working normally and not running 'really' slow Hope this helps! Richard wrote: Hi, I have * 1.0.0. Everything works well except moh. I followed the instruction in http://voip-info.org/wiki-Asterisk+config+musiconhold.conf. I use the default mp3 from *. The problem is that the music is really slow. Seems like it didn't get the right rate to play. Any one having this problem too? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialogic
[EMAIL PROTECTED] wrote: Robin van Leyden wrote: Does any body have any information about Dialogic MSI board workink with asterisk. According to this document the MSI model is not supported: http://www.asteriskpbx.org/index.php? menu=hardware Keep in mind that the Dialogic drivers for Asterisk are closed source and cost money. This is because Digium does not control the relevant APIs - those have to be licensed from Intel. Contect Digium for details, of course. Anyone can contect Intel directly and license an API for Dialogic cards. From there, one can build their own Dialogic driver for Asterisk (or Zapata), and not involve Digium at all. But hiring Digium to do this would be good advice -- given that they know the most about Asterisk, and have already done much of the necessary work -- it is not a requirement. It is important to remember that Digium releases their source code; Intel does not. It's a pretty safe bet that Intel won't be giving Digium permission to release the Dialogic drivers or API under the GPL. Cheers, Jim Van Meggelen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I need help
Hello; I intended to say that certain modifications one brought to the protocol R2 so it can support the E100P degium card and that, for certain country. I work in Algeria, and ISDN protocol doesnt exploited yet, Therefore I will to make tests with the E100P and R2 modified. Can someone help me? Best Regards. Amel MSN Messenger : discutez en direct avec vos amis ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I need help
bonjour je pense vous parler français sinon pour le pb de la carte digium se resume en l'incompatibilité avec le R2, tous simplement par ce que le protocol R2 est sous 8bits or que le EURO ISDN dépasse le s 32 bits, pour cela la seul solution est que vous utilisé une passerel du genre filtre si tu veut, y on a au USA ainsi que allemagne..., tu fait entrez le R2 et il te le transformera en ce que tu veut euro ISDN ,C7/SS7 etc..., c'est le seul moyen à ma connaissance bonne chance On Sun, 31 Oct 2004 21:56:36 +0100, omari amel [EMAIL PROTECTED] wrote: Hello; I intended to say that certain modifications one brought to the protocol R2 so it can support the E100P degium card and that, for certain country. I work in Algeria, and ISDN protocol doesn't exploited yet, Therefore I will to make tests with the E100P and R2 modified. Can someone help me? Best Regards. Amel MSN Messenger : discutez en direct avec vos amis ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- DJAZCALL F.DJEBBAR Directeur Technique 58, Rue Mohamed Khemisti, Oran Tel : +21341333707 Fax: +21341334521 www.djazcall.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pri usage
Title: pri usage Hi, I have a PRI card. Is it a way to get the usage of the channels in real time and keep in log? For example, through mrtg? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN card advise
Hi, I need an advise for a ISDN card for my HomeOffice Asterisk Setup. Currently I started with a couple of x100p for two anolog lines coming from a ISDN NT. Works but on bridged calls the sound quality is bad and distortion, if the call is being routed from the pstn back to pstn on the second line. My setup is very simple. If a call comes from the pstn our internal extension ..rings 4 times in my SIP phone and if no answer goes to my mobile phone using the pstn. Now it´s time to go shooping for a simple ISDN card an I need an advise regarding my simple requirements. Please advise with some options. Thanks Paulo Francisco Paulo AdrianoWaveLIS LDAMobile +351 91 870 87 98Office + 351 21 989 83 34Fax +351 21 989 83 35E-mail : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN CARD
Bostjan Repnik wrote: Im looking for a ISDN card that works under asterisk and supports BRI line. And I just can`t findit. Momently im using card INTERNAL, but Im having problems, asterisk on startup when loading modem fails (i4l driver). Can you please help me, or point to a www address where culd I find some help. http://www.voip-info.org/wiki-Asterisk+Hardware ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] confusing info from Digium andasteriskdoc aboutsetup of TDM11B
Steve, Thank you for testing our document, and for your valuable feedback. We are aware that there is still much work to be done, I and apologize that we have not done a good job of making that clear. I have answered some of your questions below: [EMAIL PROTECTED] wrote: Looks like it's still incorrect in the first blue paragraph of the section on FXO (it's fixed in the second blue paragraph). Also, the last paragraph of that section twice still calls the channel # 2. Now on to my next confusion... The section on contexts under dislplans mentions a context named [incoming]. This isn't a context that's mentioned anywhere before this and it's not at all clear where it comes from - I'm starting to suspect that some context references belong in the zapatel.conf file. Your suspicions are correct (although there is no zapatel.conf file). Specifically, the file where you define context for Zaptel channels is /etc/asterisk/zapata.conf. There are two zap files that are required for Asterisk: /etc/zaptel.conf and /etc/asterisk/zapata.conf /etc/zaptel.conf configures the Linux driver (the interface between the hardware and Linux), whereas /etc/asterisk/zapata.conf defines the Asterisk channel (the mechanism Asterisk creates to communicate with the Zapata telephony interfaces). Hierarchically, it goes something like this: [Asterisk] | [Asterisk Zapata channels (/etc/asterisk/zapata.conf)] | [Linux Zapata driver (/etc/zaptel.conf)] | [TDM Hardware] Technically-speaking, the Zaptel cards do not need Asterisk to work. They are devices under Linux, and could be used by any program designed to work with them. That is why their configuration is stored in the /etc directory. But if Asterisk is to use the Zapata cards, it requires a configuration file to define how it will interact with the hardware; it needs to have those channels defined. That file, being specific to Asterisk, is therefore located in the /etc/asterisk directory. In reality, Asterisk is the only product making use of the Zapata telephony interfaces, so the configuration can seem confusing. But it is very appropriate when one understands the difference between the Zaptel hardware driver, and Zapata channels in Asterisk. A comment about where the document leaves off. In the beginning the document promises to get to a minimal working set, but it really doesn't go that far. Unless I've missed something, we aren't left with even a complete version of the minimal example extensions.conf file. Something is missing With apologies, this is true. That document is still very much at the draft stage. so that I'm not getting a dial tone on the analog phone hooked up to the TDM11B and I have no idea why (can anyone clue me in?) I also tried the: [incoming] exten = s,1,Answer() exten = s,2,Playback(goodbye) exten = s,3,Hangup() example and asterisk didn't appear to see the incoming call and answer the call at all. I'd love for the example files to be complete enough that this example could actually work from either the external POTS line or even better an analog phone hooked to the FXS interface. See below. I've hacked something together which should set you on the right path. I think it would be great if attached to the document there was a final version of all of the config files which are known to work with the given configuration. Excellent suggestion, and one which we will implement. Can you help get me to a dialtone on the internal side or an answer on the external side? I'll do my best: Try this in /etc/zaptel.conf: fxoks=1 fxsks=4 loadzone=us defaultzone=us And for /etc/asterisk/zapata.conf you can try this: ; Zapata telephony interface ; ; Configuration file [channels] ;let's set some parameters context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes caller_id=Zap1 signalling = fxo_ks ;and assign them to a channel channel = 1 ; now we'll change some parameters ; (note that any parameters that have not been ; changed will contunue to apply) context=incoming caller_id=Zap2 signalling = fxs_ks ;and assign them to a channel channel = 4 Finally, you'll want something along these lines in /etc/asterisk/extensions.conf: [general] [default] exten = 6123,1,Dial(Zap/1) exten = 6444,1,Dial(IAX2/[EMAIL PROTECTED]/4569) ; [incoming] exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Background(demo-congrats) include = default This is pretty lean stuff, but it should help to get you going. Regards, Jim Van Meggelen Asterisk Documentation Project Thanks Steve Leif Madsen wrote: On Sat, 30 Oct 2004 12:18:20 -0400, Steve Totaro [EMAIL PROTECTED] wrote: Yes, it should be four unless you care to move the actual module on the
Re: [Asterisk-Users] confusing info from Digium and asteriskdoc aboutsetup of TDM11B
Looks like it's still incorrect in the first blue paragraph of the section on FXO (it's fixed in the second blue paragraph). Also, the last paragraph of that section twice still calls the channel # 2. Hi Leif: I had some similar problems with the docs I found, and struggled for a while. I finally got things all setup and working with the TDM400P card and wrote an article. Maybe it will help you: http://iheavy.com/modules.php?op=modloadname=Newsfile=articlesid=35mode=threadorder=0thold=0 Sean -- Sean Hull iHeavy, Inc. Rockefeller Center, Box 5352 New York, NY 10185 http://www.iheavy.com voice: 646.827.9877 cell: 917.442.3939 fax: 646.827.3434 Sean Hull, founder and senior consultant of Heavyweight Internet Group is the author of O'Reilly and Associates Oracle and Open Source bridging Open Source software and integration with the world's best performing database, Oracle. http://www.oreilly.com/catalog/oracleopen/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] confusing info from Digium andasteriskdoc aboutsetup of TDM11B
Thanks, I'm hoping the result of the grief I'm going through will be a well documented process :-) Here is exactly where I'm at: (power up machine and log in) my exact /etc/zaptel.conf is at: http://home.geekster.com/asterisk/zaptel.conf , but the most important part seems to be - fxoks=1 # FXS(green) module in slot 1 fxsks=4 # FXO (red) module in slot 4 defaultzone=us loadzone=us -- # modprobe zaptel # modprobe wcfxs displays on console: Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Not Installed Module 2: Not Installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) # cat /proc/interrupts CPU0 0: 608425 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 0 XT-PIC usb-uhci 8: 1 XT-PIC rtc 9: 2000 XT-PIC usb-uhci, usb-uhci, ohci1394, eth0 10: 38451 XT-PIC wctdm 11: 0 XT-PIC ehci-hcd 12: 20 XT-PIC PS/2 Mouse 14: 10671 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 0 ERR: 0 # lsmod Module Size Used byNot tainted wcfxs 36192 0 (unused) zaptel179040 0 [wcfxs] autofs 13348 0 (autoclean) (unused) eepro100 22264 1 iptable_filter 2412 0 (autoclean) (unused) ip_tables 14936 1 [iptable_filter] ohci1394 20108 0 (unused) ieee1394 46892 0 [ohci1394] mousedev5524 0 (unused) keybdev 2976 0 (unused) hid22244 0 (unused) input 5888 0 [mousedev keybdev hid] usb-uhci 26188 0 (unused) ehci-hcd 17480 0 (unused) usbcore77024 1 [hid usb-uhci ehci-hcd] ext3 70368 2 jbd52212 2 [ext3] # /sbin/ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 2 channels configured. Is this completely correct so far? It's been suggested that I might need modprobe wcfxo also, but if I do that I get: # modprobe wcfxo /lib/modules/2.4.18-14/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.18-14/misc/wcfxo.o: insmod /lib/modules/2.4.18-14/misc/wcfxo.o failed /lib/modules/2.4.18-14/misc/wcfxo.o: insmod wcfxo failed Is this a problem? Also, at exactly what point should I get a dialtone on the analog phone connected to the FXS (green) module? Thanks Steve Jim Van Meggelen wrote: Steve, Thank you for testing our document, and for your valuable feedback. We are aware that there is still much work to be done, I and apologize that we have not done a good job of making that clear. I have answered some of your questions below: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] video conferencing with sip
dean collins wrote: --SNIPOMATIC-- Is there some formal way of setting up a bounty on asterisk wiki? I pledge $US250 to begin with however I may increase that should someone show me something fruitful. --SNIPOMATIC-- Hi Dean, Just go to http://www.voip-info.org/wiki-Asterisk+bounty, add a page and then enter the details on the page (you could even just paste in the email). -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UDP Fragmentation Problem
Hi everybody, I've got no success to get a friend in Bogota (Colombia) connected to my Asterisk. He has got a ISDN Internet connection and the UDP packets will be fragmented. It seems that the MTU of this connection is round about 400 to 500 Bytes. Therefore most UDP-SIP packages are fragmented. Is Asterisk not able to handle fragmented UDP packages? Is it possible to use SIP over TCP with X-Lite? Or has somebody another hint for me? Regards Bastian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and GnuGK on the same box?
Hello Gang, I'm trying to get asterisk to play with a Lucent iMerge. It seems to that GnuGK talks to it a bit better. So I'm trying to get this: PSTN-iMerge-GnuGK-Asterisk. I'd like to get GnuGK and Asterisk running on the same box. Do they get in each others way? Any tricks to getting them both going and talking to one another on the same box? If they conflicting on ports, I suppse an option is to assign the box two IP and have them listening on two different IPs? Thanks, John -- John Gray [EMAIL PROTECTED] AgoraNet, Inc. (302) 224-2475 102 E. Main Street, Suite 303 (302) 224-2552 (fax) Newark, De 19711http://www.agora-net.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK Asterisk Consultant visiting San Diego
Dear All My business provides Asterisk consultancy in the UK. I am traveling to San Diego / Tijuana from the 4th to the 13th and wondered if there were any fellow Asterisk users who would like to meet for a coffee / drink? Please email me direct ([EMAIL PROTECTED]). Regards John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: AW: [Asterisk-Users] Firefly 1.9.6 released
Robert Berg wrote: We have had some problems registering the firefly with the Asterisk 1.0.2 it seams that IAX version doesn't match? How to solve this? Can you provide a little more information on the problems you're having with registration? Error messages, from either or both the Asterisk and Firefly sides of the connection, would be most useful. Ethereal traces would also be good. Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and GnuGK on the same box?
I tried to do similar thing with avaya definity. I end up doing make asteirsk h323 client to avaya deifnity h323 gateway. It worked for my purpose. if you control over iMerge, this can save a little bit of headache instead of goingthourh gnugk. good luck. On Sun, 31 Oct 2004 17:42:32 -0500, John Gray [EMAIL PROTECTED] wrote: Hello Gang, I'm trying to get asterisk to play with a Lucent iMerge. It seems to that GnuGK talks to it a bit better. So I'm trying to get this: PSTN-iMerge-GnuGK-Asterisk. I'd like to get GnuGK and Asterisk running on the same box. Do they get in each others way? Any tricks to getting them both going and talking to one another on the same box? If they conflicting on ports, I suppse an option is to assign the box two IP and have them listening on two different IPs? Thanks, John -- John Gray [EMAIL PROTECTED] AgoraNet, Inc. (302) 224-2475 102 E. Main Street, Suite 303 (302) 224-2552 (fax) Newark, De 19711http://www.agora-net.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and GnuGK on the same box?
That's what I treid first, but the Lucent iMerge and asterisk don't seem to play well together. I have it so calls from PSTN-iMerge-asterisk ring, but I can't get the call to complete. the iMerge seems to drop the call as soon client answers. I know someone who got it working with gnuGK between the iMerge and asterisk. Thanks, John Jongsuk Lee wrote: I tried to do similar thing with avaya definity. I end up doing make asteirsk h323 client to avaya deifnity h323 gateway. It worked for my purpose. if you control over iMerge, this can save a little bit of headache instead of goingthourh gnugk. good luck. On Sun, 31 Oct 2004 17:42:32 -0500, John Gray [EMAIL PROTECTED] wrote: Hello Gang, I'm trying to get asterisk to play with a Lucent iMerge. It seems to that GnuGK talks to it a bit better. So I'm trying to get this: PSTN-iMerge-GnuGK-Asterisk. I'd like to get GnuGK and Asterisk running on the same box. Do they get in each others way? Any tricks to getting them both going and talking to one another on the same box? If they conflicting on ports, I suppse an option is to assign the box two IP and have them listening on two different IPs? Thanks, John -- John Gray [EMAIL PROTECTED] AgoraNet, Inc. (302) 224-2475 102 E. Main Street, Suite 303 (302) 224-2552 (fax) Newark, De 19711http://www.agora-net.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- John Gray [EMAIL PROTECTED] AgoraNet, Inc. (302) 224-2475 102 E. Main Street, Suite 303 (302) 224-2552 (fax) Newark, De 19711http://www.agora-net.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom IP 500/600
Another idea, not sure if it was stated yet is to just run the ftp server on a private ip address and/or if you are going to have it on a public ip restrict by ip address. I run my ftp server on a private ip which is open to everyone on the private lan and on the public side, for example I only allow one static ip address to connect to it because I only have one outside person with a polycom phone and they have a static ip. Just an idea. On Sat, 30 Oct 2004 22:21:01 -0500, John Baker [EMAIL PROTECTED] wrote: The phone has a web interface. Couldn't you just use an expect script to change it? John Baker Karl J. Vesterling wrote: One could use SCP with certificates for authentication and avoid all the issues with FTP and it's vulnerabilities. At 07:55 PM 10/26/2004, you wrote: Richard wrote: Hi Kristian, I'd like to use ftp because of several advantages it has. For example, ability to change the time stamp and reload the phone. But the default password is a big issue. I'd like to change it but don't want to go to each phone and reset it. Any way to change it? Thanks, I understand why you would want to use FTP (no filename changes). Why is the default password such a big issue? This is a chicken or the egg - how is the phone supposed to know it's new ftp password BEFORE it can get the config file - via FTP!?! -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Best Regards, Karl J. Vesterling *E-Mail:* [EMAIL PROTECTED] *Yahoo Messenger:* karl_vesterling *ICQ: *1548052 *AOL Instant Messenger:* n2vqm *Telephone: Washington DC:* (202) 448-3009 Extension 0 *Annapolis MD:* (240) 524-6706 Extension 0 *Seattle WA:* (360) 516-1822 Extension 0 *Niagara Falls NY:* (716) 286-9175 Extension 0 *Buffalo NY:* (716) 608-1121 Extension 0 *United Kingdom:* 0870 3403428 Extension 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Tool for viewing Message waiting status
Hi all, Is anyone aware of any simple applications to display your message waiting status on screen? All I would like is a little icon in my Windows system tray to tell me I have voice mail - nothing else! I have tried a few of the "status viewers" in the WIKI page on GUIs, but either I can't get them to work or they have more functionality than I want. thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceXML / Asterisk
Dear All Is there anyone out there who is using a VoiceXML system with Asterisk? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] confusing info fromDigium andasteriskdoc aboutsetup of TDM11B
[EMAIL PROTECTED] wrote: Thanks, I'm hoping the result of the grief I'm going through will be a well documented process :-) We're getting there. :-) Here is exactly where I'm at: [snip] - fxoks=1 # FXS(green) module in slot 1 fxsks=4 # FXO (red) module in slot 4 defaultzone=us loadzone=us -- Good so far. # modprobe zaptel # modprobe wcfxs displays on console: Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Not Installed Module 2: Not Installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) PERECT! # cat /proc/interrupts 10: 38451 XT-PIC wctdm Good. It's got its own interrupt (not critical to get it working, but good for performance). # /sbin/ztcfg -vv Zaptel Configuration Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 2 channels configured. Flawless. Is this completely correct so far? It's been suggested that I might need modprobe wcfxo also, but if I do that I get: That's only if you have an X100P # modprobe wcfxo Not required. Is this a problem? Nope. Also, at exactly what point should I get a dialtone on the analog phone connected to the FXS (green) module? You won't get dialtone until Asterisk is running. Did you read ALL of my reply? I gave some examples of all the files you should need to get your dev kit running. Tell me what happens when you implement the zapata.conf and extensions.conf samples I sent. Good luck. Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Embedded Asterisk Paper Complete
Hi all, The journey is complete, at least for this project. http://lists.digium.com/pipermail/asterisk-users/2004-October/067289.html I spent the better part of Halloween putting this together, I hope its useful, enjoy. My ftp server is on the fritz so feel free to post on any other user sites. If you have any difficulties, email me and Ill send the files to you directly. JR ftp://odyssey-tech.net/Embedded_Asterisk.doc ftp://odyssey-tech.net/Embedded_Asterisk.pdf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] confusing info fromDigium andasteriskdoc aboutsetup of TDM11B
Thanks, yes I did read all of your reply (and thank you), and gave those config files a try. Unfortunatly it didn't work, but I'm getting close to wondering if it's a hardware issue. Here is my current situation: Pictures of the card I received are at: http://home.geekster.com/asterisk/tdm11B-front.jpg http://home.geekster.com/asterisk/tdm11B-back.jpg You can see the serial numbers and module positions from there. The /etc/zaptel.conf file I'm using is at: http://home.geekster.com/asterisk/zaptel.conf Copies of the simple config files in my /etc/asterisk directory are at: http://home.geekster.com/asterisk/ Power is connected to the Molex connector on the card. After a power up of my Red Hat 8 system a complete log of the steps I used to load modules and start asterisk are at: http://home.geekster.com/asterisk/log.txt This system was built from a CVS extract from the Digium server last Thursday. The result is that if I pick up the analog phone atatched to port 1 I get nothing but static. If I call the phone number attached to the POTS line plugged into port 4, asterisk does not answer the call. At this point I've blown most of the weekend fighting with this card. I'm pretty frustrated at the moment. Steve Jim Van Meggelen wrote: Did you read ALL of my reply? I gave some examples of all the files you should need to get your dev kit running. Tell me what happens when you implement the zapata.conf and extensions.conf samples I sent. Good luck. Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2_read: I should never be called!
Title: Message All -- System FreeBSD 5.2, Dell PowerEdge 2450 Asterisk installed from ports (1.0.1) Only using IAX2 (VoicePulse) and SIP (clients) Oct 31 18:34:29 NOTICE[165595136]: chan_iax2.c:2442 iax2_read: I should never be called!Oct 31 18:34:30 NOTICE[165595136]: chan_iax2.c:2442 iax2_read: I should never be called! I get a lot of those on each call that comes into the system. There is virtually no documentation on this error in google... any ideas? Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] goto() results in invalid extension
Hello,
Trying to rewrite my dialplan, and it is a little complex. But my
extensions.conf redirection works, but the referred to contexts result
in invalid extension Please help... I have the extension set to 's'
currently, but originally it was 6044. The change didn't make any
difference. Still receive the invalid extension message.
Michael
[main]
; 6044 main office line.
exten = 6044,1,GotoifTime(08:30-11:59|mon-fri|*|*?officehours,s,1)
exten = 6044,3,GotoifTime(12:00-12:59|mon-fri|*|*?lunch,s,1)
exten = 6044,2,GotoifTime(13:00-16:29|mon-fri|*|*?officehours,s,1)
exten = 6044,4,Goto(afterhours,1)
[officehours]
exten =s,2,Dial(${RECEPTION},15,r)
exten =s,3,Dial(${STAFF},10,r)
exten =s,4,Answer
exten =s,5,NoOp,${CALLERID}
exten =s,10,ResponseTimeout(5)
exten =s,16,Background(thankyouwmfm)
exten =s,17,Background(911)
exten =s,18,Background(mdorhospital)
exten =s,19,Background(nooneavail2answer)
exten =s,20,Background(appointmentdesk)
exten =s,21,Background(press1)
exten =s,22,Background(nursemessage)
exten =s,23,Background(press2)
exten =s,24,Goto(s,10)
include = menu
[lunch]
exten =s,1,Answer
exten =s,2,ResponseTimeout(5)
exten =s,6,Background(thankyouwmfm)
exten =s,7,Background(911)
exten =s,8,Background(mdorhospital)
exten =s,9,Background(closed4lunch)
exten =s,10,Background(reopenatoneoclk)
exten =s,11,Background(pleasecallbackatthattime)
exten =s,12,Goto(s,2)
include = menu-after-hours
[afterhours]
exten =s,3,Answer
exten =s,4,NoOp,${CALLERID}
exten =s,5,ResponseTimeout(5)
exten =s,6,Background(thankyouwmfm)
exten =s,7,Background(911)
exten =s,9,Background(nowclosed)
exten =s,8,Background(mdorhospital)
exten =s,10,Background(patientoptions)
exten =s,11,Background(appointmentdesk)
exten =s,12,Background(press1)
exten =s,13,Background(nursemessage)
exten =s,14,Background(press2)
exten =s,15,Background(4hoursOfop)
exten =s,16,Background(press3)
exten =s,17,Background(physicianoncall)
exten =s,18,Background(press4)
exten =s,20,Goto(s,5)
include = menu-after-hours
[on-call]
exten =s,1,ResponseTimeout(5)
exten =s,2,Playback(oncallmdline)
exten =s,3,Playback(nonurgentmatters)
exten =s,4,Playback(mdfee10)
exten =s,5,Playback(feewaived)
exten =s,6,Playback(voicemailphysoncall)
exten =s,7,Background(speakoncallmd)
exten =s,8,Background(press9)
exten =s,9,Background(otherwise)
exten =s,10,Background(press3)
exten =s,11,Background(return2nurse)
exten =s,12,Goto(s,1)
include = menu
;---
; Menu System.
;---
[menu] ; menu used when people are supposed to be here.
exten =1,1,Macro(sipexten,100,10)
exten =1,2,Voicemail(u100)
exten =1,3,Hangup
exten =2,1,Macro(sipexten,110,10)
exten =2,2,Voicemail(u110)
exten =2,3,Hangup
exten =3,1,Playback(hoursofop)
exten =3,2,Goto(main,s,1)
exten =4,1,Goto(on-call,s,1)
exten =9,1,Playback(pbx-transfer)
exten =9,2,Dial(${ONCALL})
exten =9,3,Hangup
include = invalid
[menu-after-hours] ; when the office is likely empty.
;exten =1,1,Macro(sipexten,100,10)
exten =1,2,Voicemail(u100)
exten =1,3,Hangup
;exten =2,1,Macro(sipexten,110,10)
exten =2,2,Voicemail(u110)
exten =2,3,Hangup
exten =3,1,Playback(hoursofop)
exten =3,2,Goto(main,1)
exten =4,1,Goto(on-call,s,1)
exten =9,1,Playback(pbx-transfer)
exten =9,2,Dial(${ONCALL})
exten =9,3,Hangup
include = invalid
___
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Re: [Asterisk-Users] Embedded Asterisk Paper Complete
Could you email me the PDF I am having PASV FTp problems. I have the same setup. Out of interest which case are you using. I looked at the CF adaptor you used, but not sure if the Morex 3677 case I am using is high enough. Kilburn JR Richardson wrote: Hi all, The journey is complete, at least for this project. http://lists.digium.com/pipermail/asterisk-users/2004-October/067289.html I spent the better part of Halloween putting this together, I hope its useful, enjoy. My ftp server is on the fritz so feel free to post on any other user sites. If you have any difficulties, email me and Ill send the files to you directly. JR ftp://odyssey-tech.net/Embedded_Asterisk.doc ftp://odyssey-tech.net/Embedded_Asterisk.pdf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Embedded Asterisk Paper Complete
There is an embedded space in the PDF filename that appears to be causing ftp to choke. . . FYI. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Embedded Asterisk Paper Complete
files mirrored on voip-info.org here: http://www.voip-info.org/tiki-index.php?page=Asterisk+embedded+systems Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: JR Richardson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 31, 2004 4:30 PM Subject: [Asterisk-Users] Embedded Asterisk Paper Complete Hi all, The journey is complete, at least for this project. http://lists.digium.com/pipermail/asterisk-users/2004-October/067289.html I spent the better part of Halloween putting this together, I hope it's useful, enjoy. My ftp server is on the fritz so feel free to post on any other user sites. If you have any difficulties, email me and I'll send the files to you directly. JR ftp://odyssey-tech.net/Embedded_Asterisk.doc ftp://odyssey-tech.net/Embedded_Asterisk.pdf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UDP Fragmentation Problem
On October 31, 2004 05:36 pm, Bastian Schern wrote: I've got no success to get a friend in Bogota (Colombia) connected to my Asterisk. He has got a ISDN Internet connection and the UDP packets will be fragmented. It seems that the MTU of this connection is round about 400 to 500 Bytes. Therefore most UDP-SIP packages are fragmented. Is Asterisk not able to handle fragmented UDP packages? Is it possible to use SIP over TCP with X-Lite? Or has somebody another hint for me? As far as I am aware there is no such thing as a fragmented UDP packet; each packet is sent out on its own, there is no coherency between UDP packets like there is with TCP packets. I could be very wrong here, it's been a late night with the kids. :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for a service provider
I am new to this VoIP thing and I am looking for a good service provider for VoIP. I realize that this is a hardware/software list, but figured that if you are all talking about the equipment, then you have to know some business class service providers. Shane Flynn IT Administrator Visible School ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UDP Fragmentation Problem
Andrew Kohlsmith wrote: On October 31, 2004 05:36 pm, Bastian Schern wrote: I've got no success to get a friend in Bogota (Colombia) connected to my Asterisk. He has got a ISDN Internet connection and the UDP packets will be fragmented. It seems that the MTU of this connection is round about 400 to 500 Bytes. Therefore most UDP-SIP packages are fragmented. Is Asterisk not able to handle fragmented UDP packages? Is it possible to use SIP over TCP with X-Lite? Or has somebody another hint for me? As far as I am aware there is no such thing as a fragmented UDP packet; each packet is sent out on its own, there is no coherency between UDP packets like there is with TCP packets. I could be very wrong here, it's been a late night with the kids. :-) Packet fragmentation is at the IP layer, so UDP will have fragmented packets too. But... the OS should handle that and Asterisk shouldn't find out - it's a all or none policy, so it should receive the whole packet at once or nothing. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for a service provider
Hi Shane, This type of a request is really meant for asterisk-biz. However, if you contact me off-list I will forward you our A-Z Wholesale Termination Rate-Card. Cheers, Sahil Quoting Shane Flynn [EMAIL PROTECTED]: I am new to this VoIP thing and I am looking for a good service provider for VoIP. I realize that this is a hardware/software list, but figured that if you are all talking about the equipment, then you have to know some business class service providers. Shane Flynn IT Administrator Visible School ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UDP Fragmentation Problem
Bastian Schern wrote: Hi everybody, I've got no success to get a friend in Bogota (Colombia) connected to my Asterisk. He has got a ISDN Internet connection and the UDP packets will be fragmented. It seems that the MTU of this connection is round about 400 to 500 Bytes. Therefore most UDP-SIP packages are fragmented. Is Asterisk not able to handle fragmented UDP packages? Is it possible to use SIP over TCP with X-Lite? Or has somebody another hint for me? Fragmentation should not matter for the end-point (the source or destination of the UDP datagram), since the IP stack itself should take care of the reassembly.. Butit is quite weird they have such a small MTU. Many websites that have problems with Path MTU discovery would be broken by that (dumb websites, but still, way too many...). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialogic
On Sun, 2004-10-31 at 15:26 -0500, Jim Van Meggelen wrote: [EMAIL PROTECTED] wrote: Robin van Leyden wrote: Does any body have any information about Dialogic MSI board workink with asterisk. According to this document the MSI model is not supported: http://www.asteriskpbx.org/index.php? menu=hardware Keep in mind that the Dialogic drivers for Asterisk are closed source and cost money. This is because Digium does not control the relevant APIs - those have to be licensed from Intel. Contect Digium for details, of course. Anyone can contect Intel directly and license an API for Dialogic cards. From there, one can build their own Dialogic driver for Asterisk (or Zapata), and not involve Digium at all. But hiring Digium to do this would be good advice -- given that they know the most about Asterisk, and have already done much of the necessary work -- it is not a requirement. It is important to remember that Digium releases their source code; Intel does not. It's a pretty safe bet that Intel won't be giving Digium permission to release the Dialogic drivers or API under the GPL. Exactly, That is why Digium charges money for the closed source channel driver to connect to the Dialogic drivers. Digium had to sign a NDA to get the driver information. Downside of Dialogic hardware is you won't be as free to upgrade at will to the newest CVS versions of asterisk and you will be using a less tested channel driver. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Amount of time asterisk take to pickup incoming call on ZAP interface
Hi, is it possible to change the amount of time it takes asterisk to pickup an incoming call on a zaptel interface? cheers Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] record all calls
Good day all I want to record all call on my zapvpbinternal channels. I had a look on the net and and found astGUIclient,I want something easy and simple that will save it in date/user files. Please advice Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] moh
Just an FYI: If you are *EVER* unsure that mpg123 is correctly installed (correct verison etc), you can enter the asterisk source tree, and type 'make mpg123' (without quotes), and mpg123 v0.59r will be download ed, unpacked, and built for you, and then a simple make install will install asterisk AND mpg123 in one smooth motion. -josh Richard wrote: Thanks Matthew, You are the MAN! It fixed the problem. Richard -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matthew Sent: Sunday, October 31, 2004 3:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] moh My solution to this (as the debian package appears to actually download mpg321 (instead of mpg123) when you install *, was to download mpg123 from the original website and compile/install it myself. http://www.mpg123.de/ mpg123 0.59r is the version im now running (just copied the executable over mpg123 and mpg321 and restarted asterisk (and killed dead looking mpg321 processes) started up astersik, caleld myself and shoved myself on hold, and VOILA, music on hold is working normally and not running 'really' slow Hope this helps! Richard wrote: Hi, I have * 1.0.0. Everything works well except moh. I followed the instruction in http://voip-info.org/wiki-Asterisk+config+musiconhold.conf. I use the default mp3 from *. The problem is that the music is really slow. Seems like it didn't get the right rate to play. Any one having this problem too? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inbound numbers question
I'm a newbie here. I have a general question that can help drive how exactly I'm going to get started. Say I have a single inbound number (1-800-my-number). When a call is connected on that number, and another call comes in, will asterisk answer it, will a call waiting signal be triggered, or will a busy signal occur? I'm thinking about this for several reasons. 1) for pbx, I want to make sure all inbound calls are picked up. 2) for conferencing, I want to be able to just give someone a single number to call every time (for regularly scheduled meetings). As I understand it with traditional telco equipment, you'd have a t1 or a group of lines, and the inbound call would get answered as long as there was an open line available. (it would rotate through the available lines) If this is still the case with voip in the asterisk world, does this mean I'd have to have a set of inbound lines? And if so, could you point me to the wiki and call me names, or send a link to documentation about how to set up this rollover behavior? The way I'm planning on things is to set up my server in a data center with high bandwidth availability (all voip, no pots or telco t1). Inbound calls will be for voicemail, routed to my landline or cell phone, or conference calls. This might sound like a stupid question, but I'm wondering if setting up asterisk is the way to go, or if I should use a virtual pbx service/conf. call provider. I prefer asterisk because I'd have control over everything (and I'm a geek who is addicted to OSS). If I have to maintain 10 or more dial-in numbers at a cost of $10/month/line, it might not make sense though. (figuring max conf. call of 6 visitors, and 2 inbound calls that might be routed to land lines) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Amount of time asterisk take to pickup incoming call on ZAP interface
On Mon, 1 Nov 2004 15:36:41 +1100, Sophus [EMAIL PROTECTED] wrote: Hi, is it possible to change the amount of time it takes asterisk to pickup an incoming call on a zaptel interface? I presume you are talking about an analog FXO port here. The reason why it takes Asterisk a while before it picks up is that it will try to read the caller ID information which it is programmed to expect alongside the first few rings and consequently it won't pickup any earlier. If you don't need caller ID detection or if you are in a country that has a different way to send caller ID than the US, you may want to turn caller ID detection off in the Zaptel driver. Check /etc/asterisk/zapata.conf rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inbound numbers question
Lister Account wrote: I'm a newbie here. I have a general question that can help drive how exactly I'm going to get started. Say I have a single inbound number (1-800-my-number). When a call is connected on that number, and another call comes in, will asterisk answer it, will a call waiting signal be triggered, or will a busy signal occur? Asterisk will answer the call, as long as there are available lines. I'm thinking about this for several reasons. 1) for pbx, I want to make sure all inbound calls are picked up. 2) for conferencing, I want to be able to just give someone a single number to call every time (for regularly scheduled meetings). As I understand it with traditional telco equipment, you'd have a t1 or a group of lines, and the inbound call would get answered as long as there was an open line available. (it would rotate through the available lines) If this is still the case with voip in the asterisk world, does this mean I'd have to have a set of inbound lines? And if so, could you point me to the wiki and call me names, or send a link to documentation about how to set up this rollover behavior? What you point out would be correct. Plug the T1 line into one of Digium's T1 cards, configure your dialplan appropriately and you're all set. The way I'm planning on things is to set up my server in a data center with high bandwidth availability (all voip, no pots or telco t1). Inbound calls will be for voicemail, routed to my landline or cell phone, or conference calls. This might sound like a stupid question, but I'm wondering if setting up asterisk is the way to go, or if I should use a virtual pbx service/conf. call provider. I prefer asterisk because I'd have control over everything (and I'm a geek who is addicted to OSS). If I have to maintain 10 or more dial-in numbers at a cost of $10/month/line, it might not make sense though. (figuring max conf. call of 6 visitors, and 2 inbound calls that might be routed to land lines) All these can be done with *, just depends on how you've got your dialplan configured. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Amount of time asterisk take to pickup incoming call on ZAP interface
Sophus wrote: Hi, is it possible to change the amount of time it takes asterisk to pickup an incoming call on a zaptel interface? The command exten = s,1,Wait,5 would tell asterisk to wait 5 seconds before picking up the line. More info here: http://www.voip-info.org/wiki-Asterisk+cmd+Wait Flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linux and Windows
Asterisk is working only in Linux? Can not work in Windows 2000? Please advise. Regards Bilal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linux and Windows
I saw something on the Digium site a few days ago that Asterisk was available for MS based platforms. Its called AstWind. http://www.digium.com/index.php?menu=astwind Cheers, Sahil Quoting Bilal Ghayad [EMAIL PROTECTED]: Asterisk is working only in Linux? Can not work in Windows 2000? Please advise. Regards Bilal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linux and Windows
On Fri, 1 Nov 2002 09:46:46 +0300, Bilal Ghayad [EMAIL PROTECTED] wrote: Asterisk is working only in Linux? Can not work in Windows 2000? You can have Asterisk on any operating system you like, as long as it is a proper operating system that actually deserves the name, that is to say a system that belongs to the Unix family. Unfortunately for you, Windoze is just about the only system that hasn't been allowed to come home into the world of Unix, a leftover legacy system from the last century, due to the infinite wisdom of Messieurs Gates and Ballmer. However, there is a workaround you can use. You can run a Linux kernel called CoLinux inside Windoze and then run Asterisk inside that Linux Kernel. There is even a package that installs everything for you, it's called Astwind. search the Wiki at http://www.voip-info.org for Astwind for more info in this. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] goto() results in invalid extension
[EMAIL PROTECTED] wrote:
[main]
; 6044 main office line.
exten = 6044,1,GotoifTime(08:30-11:59|mon-fri|*|*?officehours,s,1)
exten = 6044,3,GotoifTime(12:00-12:59|mon-fri|*|*?lunch,s,1)
exten = 6044,2,GotoifTime(13:00-16:29|mon-fri|*|*?officehours,s,1)
exten = 6044,4,Goto(afterhours,1)
Your numbering sequence is incorrect, spot the difference:
exten = 6044,1,GotoifTime(08:30-11:59|mon-fri|*|*?officehours,s,1)
exten = 6044,2,GotoifTime(13:00-16:29|mon-fri|*|*?officehours,s,1)
exten = 6044,3,GotoifTime(12:00-12:59|mon-fri|*|*?lunch,s,1)
exten = 6044,4,Goto(afterhours,1)
snip
[afterhours]
exten =s,3,Answer
exten =s,4,NoOp,${CALLERID}
exten =s,5,ResponseTimeout(5)
exten =s,6,Background(thankyouwmfm)
There's nowhere to go with (afterhours,1). I'd try to Goto(afterhours,s,3)
--
Andreas SikkemaRits tele.com
Scheepmakersstraat 11 3011 VH Rotterdam
t: +31 (0)10 2245544f: +31 (0)10 2245540
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