RE: [Asterisk-Users] Error in placing call file in directory
On Wed, 2005-03-16 at 16:20, Razza wrote: Chris Blake wrote : -%- If anyone can help I`ll send the call file to you, or is it ok to clutter the list with it ? -%- 'Clutter' the list I'd be interested and at least it is pertinent to * ;o) Howdy Razza and Stefan, thanks for replying Here is my call file === # This is a sample file that can be dumped in #/var/spool/asterisk/outgoing # to generate a call. Obviously, you MUST specify at least a channel in # the same format as you would for the Dial application. Only one # channel name is permitted. Channel: Zap/g4/0117265559 # You may also specify a wait time (default is 45 seconds) for how long # to wait for the channel to be answered, a retry time (default is 5 # mins) for how soon to retry this call, and a maximum number of retries # (default is 0) for how many times to retry this call. MaxRetries: 0 RetryTime: 60 WaitTime: 15 # Once the call is answered, you must provide either an application/data # combination, or a context/extension/priority in which to start the # PBX. Context: ext-local Extension: 200 Priority: 1 = From what I have read so far, this should work, but obviously something is wrong somewhere. I understand the following, correct me if I am wrong : In the above example, Context: ext-local : * will look in extensions.conf for this context, but I do not have it specified there. This context is specified in extensions_additional.conf. However, I do have the #include extensions_additional.conf in extensions.conf, so it should pick it up right ? In any event, I have referenced another context which DOES exist in extensions.conf but I still get the same result. Here is how this context is specified in extensions_additional.conf : [ext-local] exten = 200,1,Macro(exten-vm,200,200) exten = 201,1,Macro(exten-vm,201,201) exten = 202,1,Macro(exten-vm,[EMAIL PROTECTED],202) Also, I notice that although the call is not being made, and I have specified 0 retries in the call file, my log file keeps getting cluttered with these entries following me setting sip debug on via *`s CLI : === =Sip read: OPTIONS sip:192.168.204.95 SIP/2.0 Content-Length: 0 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=29451415291147145415 CSeq: 3895 OPTIONS Max-Forwards: 70 To: sip:192.168.204.95 Via: SIP/2.0/UDP 192.168.204.10;rport;branch=z9hG4bKc0a8cc0a0131c9b142393a2f0af7ccce3064 8 headers, 0 lines Looking for 192.168.204.95 in from-sip-external Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.204.10;branch=z9hG4bKc0a8cc0a0131c9b142393a2f0af7ccce3064 From: sip:[EMAIL PROTECTED];tag=29451415291147145415 To: sip:192.168.204.95;tag=as2d027310 Call-ID: [EMAIL PROTECTED] CSeq: 3895 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:192.168.204.95 Accept: application/sdp Content-Length: 0 to 192.168.204.10:5060 Destroying call '[EMAIL PROTECTED]' = So it appears that it`s destroying the call, and then retrying it again...there are no other active lines in use on this box yet. My permissions on /var/spool/asterisk/outgoing are as follows : drwx--2 asterisk asterisk 4096 Mar 17 11:47 outgoing Still searching google, wiki archives but nothing found yet Any ideas...? -- Chris Blake Cell: 082 775 1492 Work: +27 11 782 0840 Fax : +27 11 782 0841 Mail: [EMAIL PROTECTED] There is no sin but ignorance. -- Christopher Marlowe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to register two SIP phones ( e.g. WindowsMessenger) from different subnet to *
On Thu, 2005-03-17 at 11:34, Alexander Lopez wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohammed Firdosh Nasim Sent: Tuesday, March 15, 2005 11:08 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] How to register two SIP phones ( e.g. WindowsMessenger) from different subnet to * On Sat, 2005-03-12 at 07:42, Luki wrote: Firdosh, there were couple typos on my last email, but that's essentially what I said. There are two ways of doing it -- but neither will work given you current setup. 1) Phone A talks directly to B. 2) Both Phone A and B talk to a common point C. Point C proxies traffic between A and B, because A and B cannot see each other directly. You you can't have both clients on the same subnet, then you need a third subnet C that is reachable from both A and B. Asterisk runs in subnet C and proxies the traffic between A and B. --Luki Hi All, I have a dedicated * server at 172.16.200.150 and my two windows messenger clients are at 172.16.25.X 172.16.15.X. Now the server is visible to both the subnets.Both the users/clients[say msn1 msn2] are configured. Then call is made from one user to another. After the callee receives/accepts the call, neither of users able to hear anything. Sip debug shows 200 OK for the call.Do I have to register= the users, if yes kindly mail the register string. Here are the sip.conf and extensions.conf sip.conf - [msn1] type=friend host=dynamic context=default dtmfmode=inband disallow=all allow=ulaw allow=alaw canreinvite=yes nat=yes [msn2] host=dynamic type=friend context=default dtmfmode=inband disallow=all allow=ulaw allow=alaw canreinvite=yes extensions.conf [default] exten = msn1, 1, Dial(SIP/msn1, 20) exten = msn2, 1, Dial(SIP/msn2, 20) Thanks and regards, Firdosh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users For starters, Get rib of canreinvite=yes, set it to canreinvite=no. This will keep * in the Media path. (You can try msn1 to msn2 directly later) Second, what does the output of 'sip show peers' show?? This is from the CLI prompt on the asterisk server console. I just changed canreinvite=yes to canreinvite=no and its working fine. Thanks a lot for ur suggestion. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Background apps that plays music on hold
Kong wrote: Is there any application that actually work like Background, but instead of playing a specified file, it plays the streaming music from music on hold? the reason i am asking this because i come across a dialplan that goes this way, if a person gets to an extension that is busy, it will playback a message like. the person is current busy, press 1 to leave a message or hold on the line so when this message is finish, it starts to play music on hold. if the user wants to leave a msg now, it can press 1 to do so, else stay on the line still the busy user is done with the call and connects to it. 1. go to the cli 2. type show application waitmusiconhold flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Recording and Archiving
Hi, I need the option to Record certain conversations through * on our help desk.Id like toarchive these for later access.Please can someone point me in the right direction as search has brought up nothing. Thanks Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ser+asterisk - security
Hi there, I'm using ser and asterisktogether. Asterisk for voice mail etc and ser forregistration of the users usig database.I can restrict forwarding callsfrom another sip proxy to ser(using proxy_authorize) but how can I restrict access to asterisk ... Now everyone can forward calls to my asterisk and can place pstn calls. Thanks in advance, Pavel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do you need to recompile the Linux 2.6 kernelfor zaptel modules?
On Thursday 17 March 2005 06:13, Geoff Nordli wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of David Uzzell Sent: Wednesday, March 16, 2005 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Do you need to recompile the Linux 2.6 kernelfor zaptel modules? Geoff Nordli wrote: Hi Everyone. On the Linux 2.6 kernel do I need to recompile the kernel in order to compile the zaptel modules? Using Mandrake 10.1 which runs standard a 2.6 kernel I did not have to recompile the kernel to get them working. cheers, David In order for me to get the zaptel source to compile from cvs I need to: 1) Copy the .config file from the /boot directory to the source directory, run menuconfig, and run make. 2) After I run make I can then compile the zaptel modules. 3) I don't need to install the new kernel, but just run the make command. Hi Guys I am new to this list but a similar problem came up at the asterisk training course I attended last week. When the linux kernel sources are freshly installed, the symbolic links for the kernel header files are not properly set up. All you have to do is type: make prepare in /usr/src/linux to set this all up. You do not have to compile the kernel or anything else. I do not know whether this works for the 2.4 kernel but it does work on Fedora 2 and SuSe 9.1 and later. Paul H -- Paul Hewlett (Linux #359543) Email:`echo [EMAIL PROTECTED] | rev` Tel: +27 21 852 8812 Cel: +27 72 719 2725 Fax: +27 86 672 0563 -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ser+asterisk - security
Pavel Siderov - Hostmates wrote: I can restrict forwarding calls from another sip proxy to ser (using proxy_authorize) but how can I restrict access to asterisk ... Now everyone can forward calls to my asterisk and can place pstn calls. Use iptables on the asterisk machine to only allow SIP traffic from the machine with SER? -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Recording and Archiving
I need the option to Record certain conversations through * on our help desk.Id like to archive these for later access.Please can someone point me in the right direction as search has brought up nothing. Check out the Monitor option. (just search the wiki for monitor) Loads of examples and hints on exactly how to do this. Regards Jonathan -- Jonathan Gill [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN Cards in the USA
Hello Everyone, I am trying to find a single port isdn pci card in the usa for asterisk, but it seems everything is abroad. Does anyone know a good place to find a BRI S/T and U card for north america? Perhaps it could be possible if you get an NT1 box giving you an S0 bus and then using a european card? or perhaps not dunno ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IPSwitchBoard BETA
Hi, I have just published my last few weeks of hard work: IPSwitchBoard BETA. Please let me know what you think and post comments on the Wiki. http://www.voip-info.org/wiki-IPSwitchBoard+BETA I've installed it and tested it, it works great, the idea is great, works simple enough for users to understand :) One problem though: If a phone registered under Monitored Extension has two calls, one Active and one On Hold and if I attempt a transfer via IPSwitchBoard it works fine. After that I Resume an On Hold call and attempt to transfer that call in the same manner, but instead of a call being transferred to an other user IPSwitchBoard initiates a new call from my phone (Monitored one) to the user I wanted to transfer a call to. Is it a bug or a feature ? :) Regards, Ivan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] t.38 support news?
Maybe I've missed it but I'm wondering if there has been any movement towards getting t.38 support into asterisk.. has there been any news? Where is t.38 support at? will it even happen? Steve Underwood is working on it. There's a bounty at http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38%20Bounty Please add some more $$$ there to encourage him a little more :) roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOW-To write an AGI
I tried wiki, but I got too many pages (I think all of them), ...as answer. I want to write an agi. I need a HOW-TO, is there anything available? see the perl agi package from http://asterisk.gnuinter.net/, the agi/agi-test from the asterisk source and http://www.voip-info.org/wiki-Asterisk+AGI roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk t.38 codec negotiation problems
hello list (3rd try as my first post seems to have gone astray in the endless realms of tcp/ip and in my second i accidentially replied to another post not related to the problem), i searched for nearly a week for a solution to this problem, as there is: analog fax machine -» grandstream ata -» asterisk -» sip trunk from provider -» provider gateway to pstn -» analog/isdn fax machine on pstn everything worked out fine until my provider decided to implement t38 into the gateway. now when i send/receive a fax message the gateway tries to connect with t38 and waits for a 415 unsupported media type to receive if the device on the sip trunk is incapable of talking t38, in order to fall back to the old g711 behaviour. but asterisk does not answer with 415 unsupported media type, he answers with his own codec capabilities, and this is ignored by the gateway. so he tries to send with t38 anyway, and that of course fails. call flow: should be: reject t.38, fallback to g.711 PBX GW | | | INVITE F1 | |--| | (100 Trying F2) | |--| | 180 Ringing F3 | |--| | 200 OK F4 | |--| | ACK F5 | |--| | G.711 | |=| | INVITE F6 | |--| | (100 Trying F7) | |--| | 415 Unsupported F8| |--| | INVITE F9 | |--| | (100 Trying F10) | |--| | 200 OK F11 | |--| | ACK F12 | |--| | G.711 | |=| | | F1: m=Audio, PCMA F4: m=Audio, PCMA F6: m=Image, t38 F9: m=Audio, PCMA F11: m=Audio, PCMA is: acknowledge with g.711 on t.38, call rejected by gateway PBX GW | | | INVITE F1 | |--| | (100 Trying F2) | |--| | 180 Ringing F3 | |--| | 200 OK F4 | |--| | ACK F5 | |--| | G.711 | |=| | INVITE F6 | |--| | (100 Trying F7) | |--| | 200 OK F8 | |--| | | F1: m=Audio, PCMA F4: m=Audio, PCMA F6: m=Image, t38 F8: m=Audio, PCMA any thougts on that one? kind regards, michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IPSwitchBoard BETA
I've installed it and tested it, it works great, the idea is great, works simple enough for users to understand :) One problem though: If a phone registered under Monitored Extension has two calls, one Active and one On Hold and if I attempt a transfer via IPSwitchBoard it works fine. After that I Resume an On Hold call and attempt to transfer that call in the same manner, but instead of a call being transferred to an other user IPSwitchBoard initiates a new call from my phone (Monitored one) to the user I wanted to transfer a call to. Is it a bug or a feature ? :) Regards, Ivan Hi Ivan, Bug or feature, I will look at it and try to solve it. Thank you for your interest in IPSwitchBoard tgj ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pictures from the Asterisk Pavilion at Spring VON 2005
On Thu, 10 Mar 2005, Kevin P. Fleming wrote: http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view Enjoy! Anyone have pictures from the Heart show? :( My camera phone just wigged out. I thought I had like 60 pictures right from the stage, but apparently it didn't save anything except for one. If so, please mail me links.. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pictures from the Asterisk Pavilion at Spring VON 2005
On Thu, 10 Mar 2005, Kevin P. Fleming wrote: http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view If you look closely, you'll see me at the booth doing some troubleshooting for Digium during one of my session breaks. We actually setup an IAX2 connection from the main server to the N2Net Gateway server and made several test calls to flesh out a couple of call completion anomalies. VON was a total blast. Can't wait for Boston! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAC Access Bank Manual
Hi, Does anyone have Carrier Access Corporation (CAC) Access Bank I Manual ? Could you please email it to me off list ? We have a FXS channel bank and the framing Error Led is blinking and I have no clue on what could be the problem . Is there command line utilities available in Linux to Troubleshoot T1 connection using Zaptel drivers ? /etc/zaptel.conf = span=1,1,0,esf,b8zs #span=1,1,0,esf,ami #span=1,1,0,d4,b8zs #span=1,1,0,d4,ami #em=1-24 fxols=1-24 loadzone=us defaultzone=us == /etc/asterisk/zapata.conf = [channels] language=us context=default signalling=fxo_ls ;usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes ;threewaycalling=yes transfer=yes ;cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group = 1 channel = 1-24 === dmesg output = Zapata Telephony Interface Registered on major 196 Found TE410P at base address dfcdff80, remapped to d0e23f80 TE410P version c01a009b, burst ON FALC version: 0005, Board ID: 00 Reg 0: 0x0e3c6800 Reg 1: 0x0e3c6000 Reg 2: 0x07fc07fc Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a009b Reg 7: 0x1000 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE410P: Launching card: 0 TE410P: Setting up global serial parameters Found a Wildcard: Wildcard TE410P-Xilinx Registered tone zone 0 (United States / North America) TE410P: Span 1 configured for ESF/B8ZS SPAN 1: Primary Sync Source == -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extension.conf dialplan
hi
any one tell me how to make a dialplan
my extensions.conf
exten = _40,1,Dial(OH323/${EXTEN})
i want to dial to 40 number.
could be any number like 923335224005 or
92512213248
at the moment when i am trying to dial 40923335224005
asterisk is dialing
Executing Dial(OH323/R11429, OH323/40923335224005)
but i want him to dial
Executing Dial(OH323/R11429, OH323/923335224005)
Kamran Ahmad
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RE: [Asterisk-Users] IPSwitchBoard BETA
Bug or feature, I will look at it and try to solve it. One more thing while you are at it: I just installed .NET 2.0 Beta, so I don't know if it's a problem with your app or with new .NET, but after a few minutes of running your app and not doing anything (the app runs idle) a new window appears with a following message: Timer2: Object reference not set to an instance of an object. at IPS.ipdkpbx.astNewChannel(String e) at IPS.ipdkpbx.Timer2_Tick(Object sender, EventArgs e) After I confirm the message everything seems normal but as soon as I try to do any kind of action IPS.exe crashes. Regards, Ivan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extension.conf dialplan
On 02:42, Thu 17 Mar 05, Kamran Ahmad wrote:
hi
any one tell me how to make a dialplan
my extensions.conf
exten = _40,1,Dial(OH323/${EXTEN})
i want to dial to 40 number.
could be any number like 923335224005 or
92512213248
at the moment when i am trying to dial 40923335224005
asterisk is dialing
Executing Dial(OH323/R11429, OH323/40923335224005)
but i want him to dial
Executing Dial(OH323/R11429, OH323/923335224005)
Kamran Ahmad
use this:
exten = _40,1,Dial(OH323/${EXTEN:2})
This will strip the first 2 characters
--
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Two of the most famous products of Berkeley are LSD and BSD. I don't think
that this is a coincidence.
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Re: [Asterisk-Users] extension.conf dialplan
On Thu, 17 Mar 2005 02:42:27 -0800 (PST), Kamran Ahmad [EMAIL PROTECTED]
wrote:
hi
any one tell me how to make a dialplan
my extensions.conf
exten = _40,1,Dial(OH323/${EXTEN})
i want to dial to 40 number.
could be any number like 923335224005 or
92512213248
at the moment when i am trying to dial 40923335224005
asterisk is dialing
Executing Dial(OH323/R11429, OH323/40923335224005)
but i want him to dial
Executing Dial(OH323/R11429, OH323/923335224005)
You need to change your dial comand to this
exten = _40,1,Dial(OH323/${EXTEN:2})
the :2 deletes the first 2 digits and removes the leading 40
Jason
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Re: [Asterisk-Users] extension.conf dialplan
Kamran Ahmad wrote:
hi
any one tell me how to make a dialplan
my extensions.conf
exten = _40,1,Dial(OH323/${EXTEN})
i want to dial to 40 number.
could be any number like 923335224005 or
92512213248
at the moment when i am trying to dial 40923335224005
asterisk is dialing
Executing Dial(OH323/R11429, OH323/40923335224005)
but i want him to dial
Executing Dial(OH323/R11429, OH323/923335224005)
exten = _1,1,Dial(OH323/${EXTEN:1})
exten = _10,1,Dial(OH323/${EXTEN:2})
The first one removes the first number and the second one removes the
first 2 chars. and so on, and so on.
Hope that helps with examples.
David
Kamran Ahmad
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[Asterisk-Users] Compilation problem chan_capi and Eicon Diva 4Bri
Hi *, I want to integrate the Eicon Diva 4Bri Card to Asterisk. Eicon drivers and capi is installed. I use the latest dev version from eicon compiled and installed for my fedora 2 system. I found the chan_capi for asterisk from www.junghanns.net. Also loaded the patch and applied to the chan_capi source tree. I changed the Makefile to include the capi20.h from eicon: INCLUDE=-I$(ASTERISK_HEADER_DIR) -I/usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon make install gives me the following errors: gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -I/usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c In file included from chan_capi.c:35: /usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon/capi20.h:52: error: Syntaxfehler before word /usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon/capi20.h:52: Warnung: kein Semikolon am Ende von »struct« oder »union« /usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon/capi20.h:53: Warnung: type defaults to `int' in declaration of `maxLogicalConnection' /usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon/capi20.h:53: Warnung: data definition has no type or storage class Unfortunately it is german system so also the compiler errors are in german. I realy need help because I am not the r+d expert. Thanks in advance. Kib ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hi there..
Hello Everybody, This is Bharat here. I am on the way of learning Asterisks, and I just wished to know how I go about if got to write dailplans for outbound calls and inbound calls. If you could provide me with a simple example, I could get thru. Waiting for your response Regards Bharat M. Sarvan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hi there..
On 16:51, Thu 17 Mar 05, Bharat M. Sarvan wrote: Hello Everybody, This is Bharat here. I am on the way of learning Asterisks, and I just wished to know how I go about if got to write dailplans for outbound calls and inbound calls. If you could provide me with a simple example, I could get thru. Waiting for your response Grab a comfy chair, lots of your fav. caffeiniated drink, fire up your browser and start reading http://www.voip-info.org -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE110p card with Euro ISDN (Ericsson switch)
I am trying to use TE110p card for Euro ISDN with Ericsson AMS switch. I consistently get one of the following errors: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 or PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 My zaptel.conf file: span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=us defaultzone=us My partner and I have tried to find answer on google, IRC and even digium tech support but no luck so far :-( I observe the same behavior on FC3 as well as RH9, and on different hardware too. One of them is Intel Motherboard with 3 GHz P4 processor (FC3), the other is Supermicro with 2.4GHz Dual Xeon processor (RH9). Surprisingly, a similar machine with Supermicro motherboard works very well with TE405p card. I checked the interrupts etc. and there is no shared interrupt with the card. Hence I suspect the issue is at the signaling level or with the card's driver rather than some interrupts getting lost. I could be completely wrong though. Any help will be greatly appreciated. Thanks, -- jt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hi there..
Bharat M. Sarvan wrote: Hello Everybody, This is Bharat here. I am on the way of learning Asterisks, and I just wished to know how I go about if got to write dailplans for outbound calls and inbound calls. If you could provide me with a simple example, I could get thru. Waiting for your response If you go to http://www.voip-info.org/tiki-index.php and search for extensions you will find exactly what you are in need off. David Regards Bharat M. Sarvan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 79xx 7-4
[EMAIL PROTECTED] wrote: change the sntp_mode: from directedbroadcast (the default) to unicast. This will cause the phone to poll your NTP server. This solved the problem for me. This fixed my problem as well! Thanks Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Error in placing call file in directory [SOLVED]
Greetings *`s, Further to the above thread, the problem has been solved. Stefan was correct in stating that it was a permissions error, but we were only able to catch what permissions the file carried over from its source directory to /var/spool/asterisk/outgoing by completely stopping the * server and cd`ing over to /var/spool, as * was catching the file too quickly and destroying it without giving us a chance to check what perms the file held. Thanks for the assistance Regards -- Chris Blake Cell: 082 775 1492 Work: +27 11 782 0840 Fax : +27 11 782 0841 Mail: [EMAIL PROTECTED] Natural selection won't matter soon, not anywhere as much as concious selection. We will civilize and alter ourselves to suit our ideas of what we can be. Within one more human lifespan, we will have changed ourselves unrecognizably. -- Greg Bear ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco gateways and hairpinning
Hi, Some time I did not touch a cisco. At a previous job, I managed a 53xx If I remembered well, you can define dial-peers at ingress and outgress. The trick is the add a very specific header at ingress and remove it at outgress. Also, by then, not all traffic directions where possible on the 53xx. IP - E1/T1 ok E1/T1 - IP ok E1/T1 - E1/T1 ok IP- IP NOT OK ! I had to loose 2 E1's to move IP traffic back to IP So, I had IP - E1.0 - E1.1 - IP The destination IP could be the same as original (hairpin) or different That was some times ago, only on the 5300 and from memory Regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] -Original Message- From: Steve Blair [mailto:[EMAIL PROTECTED] Sent: mercredi 16 mars 2005 16:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco gateways and hairpinning Hello: Has anyone on this list had to configure hairpinning on a Cisco gateway running IOS 12.2 or 12.3 and using a PRI for connectivity to the PSTN? If so could you tell me how it is done? I'm told this is the source of my call transfer problems and yet I cannot find clear instructions for how the configuration is done. Thanks,Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] t.38 support news?
Hi, I have T.38 over UDPTL with SIP signalling kind of working-ish within Asterisk. I hope to be passing code around for some serious testing by other people in a couple of weeks, or so. Certainly within a month. Once I have it stabilised with UDPTL and SIP I will get it working with IAX. Then I aim to get it working over RTP. I have no plans for T.38 over TCP right now, as it doesn't seem to be in common use. I have no persoanl interest in H.323. If someone wants to extend it to work with H.323 that's fine by me. I am making a PSTN gateway, and extended app_rxfax and app_txfax programs which can transfer via T.38. The bigegst holdup has really been the poor state of T.38 support in current equipment. Few ATAs do it. Fewer do it right. Regards, Steve Matt Hess wrote: Oh, I was just hoping (more like praying to $DEITY) that it'd be in around July or August.. now in a perfect world I'd love for it to be in stable tomorrow (heck, I'd settle for cvs head) but I know that isn't realistic at all. I was just looking for a warm-fuzzy ray of hope. Nathan C. Smith wrote: The developers have talked about it but there are some core components that need to be modified or rewritten to make t.38 integral to asterisk. The way I understand it: They have it in mind, it is a consideration, but don't hold your breath, it won't show up tomorrow. -Nate -Original Message- From: Matt Hess [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 16, 2005 4:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] t.38 support news? Maybe I've missed it but I'm wondering if there has been any movement towards getting t.38 support into asterisk.. has there been any news? Where is t.38 support at? will it even happen? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hong Kong DID
[EMAIL PROTECTED] wrote: Hi there, Anybody on this list knows where I can obtain Hong Kong DID's from ? Cheers, Sahil You get them when you subscribe to a T1 or E1. However, if you want blocks bigger than 200-300 per T1/E1 it is a problem these days, unless you are a telco. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] IPSwitchBoard BETA
I just installed .NET 2.0 Beta, so I don't know if it's a problem with your app or with new .NET, but after a few minutes of running your app and not doing anything (the app runs idle) a new window appears with a following message: Timer2: Object reference not set to an instance of an object. at IPS.ipdkpbx.astNewChannel(String e) at IPS.ipdkpbx.Timer2_Tick(Object sender, EventArgs e) After I confirm the message everything seems normal but as soon as I try to do any kind of action IPS.exe crashes. Regards, Ivan Hi Ivan, That is a bug, I will fix it. Regards tgj ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SUSE 9.2 and Zaptel channels
[EMAIL PROTECTED] is believed to have said: Suse 9.2 uses udev. Look for README.udev in you zaptel source directory and follow the instructions. Regards, Alex Thanks Alex! Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SUSE 9.2 and Zaptel channels
[EMAIL PROTECTED] is believed to have said: I have a fairly current CVS build of asterisk running on SuSE 9.2. You need to get rid of the stuff that gets installed with the system and then install the zaptel stuff. Works fine for me, but I do get warnings about unsupported modules and tainting of the kernel. The wiki has an entry on SuSE: http://www.voip-info.org/wiki-Asterisk+Linux+SuSE Thanks Tim! Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX Registration being lost
[EMAIL PROTECTED] is believed to have said: I've posted this question twice without a single reply. Does that mean no one knows the answer, or no one cares to answer? I've been having an issue with an IAX2 trunk setup in Asterisk. Setup the trunk fine and it registers and works fine. I'm able to make outgoing calls from any extension and I'm happy :). Then my Internet disconnects and it won't register anymore. Simply says Request Sent forever. I can reinstate the trunk by removing it, rebooting the server (maybe a reload would also work) and then setting it up again. It then registers fine again. I'm imagining it has something to do with the fact that my home has a dynamic IP address and it's changing when the connection drops out. Of course it may also be nothing to do with it. I'd appreciate any help anyone can give. Hi there, while I can't help with the aboc problem in the scenario of dynamic IPs. I have whatI think a similar question when using static IP addresses. I have now disabled the registration to my iaxtel number from the production asterisk I am using at work, as it was regularly loosing the registration and thus trying to reregister. The * box is running the stock (aka old!) Asterisk that comes on the SUSE 9.2 cds (Asterisk CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a). My first hypothesis was that the iax2 instability could be a problem in an old release. But I was quite surprised to see the same behaviour on a different test machine (older PowerMac, running Yellow Dog Linux/Asterisk 1.0.6, compiled from sources). After starting asterisk the registration to iaxtel is ok, at first. But if I check at the CLI with iax2 show registry at random after a few minutes, most of the time asterisk is trying to re-register. I am planning to test the same process on the next machine (the mytical mini-ITX/Suse 9.2 for which I pestered the list asking for compilation and zaptel startup help in the past week). BTW: I have the PowerMac at home on a fast link (10 mbit/s) with slowly changning dynamic IPs; the pbx at work is on a slower ADSL link (128/ 640kbit/s) but with static IP address. So, I add the question: who is having no registration problems on a iaxtel link? Thanks, Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hi there..
Hi, Welcome. Read the samples *.conf files (in /etc/asterisk) extension.conf, sip.conf are some good places to start. Read search the wiki. Many info there (also not always very clear) success Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] -Original Message- From: Bharat M. Sarvan [mailto:[EMAIL PROTECTED] Sent: jeudi 17 mars 2005 12:21 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Hi there.. Hello Everybody, This is Bharat here. I am on the way of learning Asterisks, and I just wished to know how I go about if got to write dailplans for outbound calls and inbound calls. If you could provide me with a simple example, I could get thru. Waiting for your response Regards Bharat M. Sarvan image001.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110p card with Euro ISDN (Ericsson switch)
Am Donnerstag 17 März 2005 12:26 schrieb J Thomas: I am trying to use TE110p card for Euro ISDN with Ericsson AMS switch. I consistently get one of the following errors: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 or PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Got the same problem here connecting a Siemens PBX cross via ETSI protocol but so far it seems not to cause any considerable errors for telephony service (still testing setup). Jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hi there..
All the samples are on your system /usr/src/asterisk/configs/ the files have a .sample on them. Also there is allow of information on the Wiki http://www.voip-info.org/wiki-Asterisk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bharat M. Sarvan Sent: Thursday, March 17, 2005 6:21 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Hi there.. Hello Everybody, This is Bharat here. I am on the way of learning Asterisks, and I just wished to know how I go about if got to write dailplans for outbound calls and inbound calls. If you could provide me with a simple example, I could get thru. Waiting for your response Regards Bharat M. Sarvan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: IAX Registration being lost
I've posted this question twice without a single reply. Does that mean no one knows the answer, or no one cares to answer? I've been having an issue with an IAX2 trunk setup in Asterisk. Setup the trunk fine and it registers and works fine. I'm able to make outgoing calls from any extension and I'm happy :). Then my Internet disconnects and it won't register anymore. Simply says Request Sent forever. I can reinstate the trunk by removing it, rebooting the server (maybe a reload would also work) and then setting it up again. It then registers fine again. I'm imagining it has something to do with the fact that my home has a dynamic IP address and it's changing when the connection drops out. Of course it may also be nothing to do with it. I'd appreciate any help anyone can give. Hi there, while I can't help with the aboc problem in the scenario of dynamic IPs. I have whatI think a similar question when using static IP addresses. I have now disabled the registration to my iaxtel number from the production asterisk I am using at work, as it was regularly loosing the registration and thus trying to reregister. The * box is running the stock (aka old!) Asterisk that comes on the SUSE 9.2 cds (Asterisk CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a). My first hypothesis was that the iax2 instability could be a problem in an old release. But I was quite surprised to see the same behaviour on a different test machine (older PowerMac, running Yellow Dog Linux/Asterisk 1.0.6, compiled from sources). After starting asterisk the registration to iaxtel is ok, at first. But if I check at the CLI with iax2 show registry at random after a few minutes, most of the time asterisk is trying to re-register. I am planning to test the same process on the next machine (the mytical mini-ITX/Suse 9.2 for which I pestered the list asking for compilation and zaptel startup help in the past week). BTW: I have the PowerMac at home on a fast link (10 mbit/s) with slowly changning dynamic IPs; the pbx at work is on a slower ADSL link (128/ 640kbit/s) but with static IP address. So, I add the question: who is having no registration problems on a iaxtel link? Iaxtel has not been working correctly for some time (months?), and no one seems to be maintaining it. Try using ethereal, and you'll see that * tries to register but doesn't get a response for at least ten to fifteen seconds, and then the response seems to be a NACK. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Cisco Call Manager
Dear All : We need to use the Conference Room Capability from Asterisk to use it with our IPT Solution which based on Cisco Call Manager.. Also we need to use most of Asterisk features in our IPT Network .. How can I do this ? Any help will be grateful .. Mohamed Farid ,, Telecommunication Security Administrator ,, Notice: This e-mail (including attachments) is confidential and is intended solely for the addressee. Unless you are the addressee, you may not read, copy, use or store this e-mail in any way, or permit others to. If you have received it in error, please contact Mediterranean Smart Cards Company :+202333 1400 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 79xx 7-4
On Thu, 2005-03-17 at 06:33 -0500, Doug Lytle wrote: [EMAIL PROTECTED] wrote: change the sntp_mode: from directedbroadcast (the default) to unicast. This will cause the phone to poll your NTP server. This solved the problem for me. This fixed my problem as well! Thanks Mine too :) Thanks. -- respectfully, Joseph === -= ** = ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAC Access Bank Manual
Carrier Access generally have all of their manuals available for download. You just have to request a free login. they also provide excellent dialin support - also free. If your framing LED is blinking I would double check that both ends of your span are set for ESF. zttool is the tool for working on the cards. On Mar 17, 2005, at 4:40 AM, Vicky Shrestha wrote: Hi, Does anyone have Carrier Access Corporation (CAC) Access Bank I Manual ? Could you please email it to me off list ? We have a FXS channel bank and the framing Error Led is blinking and I have no clue on what could be the problem . Is there command line utilities available in Linux to Troubleshoot T1 connection using Zaptel drivers ? /etc/zaptel.conf = span=1,1,0,esf,b8zs #span=1,1,0,esf,ami #span=1,1,0,d4,b8zs #span=1,1,0,d4,ami #em=1-24 fxols=1-24 loadzone=us defaultzone=us == /etc/asterisk/zapata.conf = [channels] language=us context=default signalling=fxo_ls ;usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes ;threewaycalling=yes transfer=yes ;cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group = 1 channel = 1-24 === dmesg output = Zapata Telephony Interface Registered on major 196 Found TE410P at base address dfcdff80, remapped to d0e23f80 TE410P version c01a009b, burst ON FALC version: 0005, Board ID: 00 Reg 0: 0x0e3c6800 Reg 1: 0x0e3c6000 Reg 2: 0x07fc07fc Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a009b Reg 7: 0x1000 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE410P: Launching card: 0 TE410P: Setting up global serial parameters Found a Wildcard: Wildcard TE410P-Xilinx Registered tone zone 0 (United States / North America) TE410P: Span 1 configured for ESF/B8ZS SPAN 1: Primary Sync Source == -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Quality Detail Record
Hello, I need some help setting up statistics per call. I need to store in a database call quality details such as jitter, packets lost and other informations. Is there any way to do this? I'd really appreciate some links or any other kind of info on this. Thanks, Calin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial multiple extensions, but different variables/timeouts
On Wed, 16 Mar 2005 19:58:55 -0800, Luki [EMAIL PROTECTED] wrote: Anyway, if anyone ever needs this info, they can Google it now :-). Might be a good thing for the wiki too. ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAp channel numbering question
Hi there, Newbie questions on ZAP channel numbering (forgive me if this was asked before): 1. How are channels numbered if I have multiple FXS/FXO cards in the system? Is there a fixed mapping between PCI slot id and the number range allocated for all the ports on that PCI card? 2. Same question for T1/E1 spans. Appreciate your help! Yester ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astguiclient error!
Hello, can anyone using astgui client i have a problem in installation phase everytime i try to create database from MySQL_AST_CREATE_tables.sql it gives error in phone table ERROR 1064 (42000): You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'DBY_server VARCHAR(15), DBY_database VARCHAR(15) default 'asterisk', DBY_user VA' at line 62 i also try manually to create this table but no luck am i missing something ? Thanks In Advance ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Welltech Welgate 3804 FXO Configs
Good Day List, I am looking to see if anyone is willing to share their working configs with me. I would be happy to add to wiki and document steps to get it to work with asterisk. I am looking for both Welgate configs as well as sip.conf and extension.conf snippets. Thanks for your assistance in this matter. ~ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ser+asterisk - security
Hi Andreas, it's impossible to use iptables due to the reason that audio flows through asterisk and users won't be able to communicate w/ *... I've tried that. Regards, Pavel - Original Message - From: Andreas Sikkema [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 17, 2005 10:40 AM Subject: RE: [Asterisk-Users] ser+asterisk - security Pavel Siderov - Hostmates wrote: I can restrict forwarding calls from another sip proxy to ser (using proxy_authorize) but how can I restrict access to asterisk ... Now everyone can forward calls to my asterisk and can place pstn calls. Use iptables on the asterisk machine to only allow SIP traffic from the machine with SER? -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Serusers] ser+asterisk - security
Thanks you ! I'll try! Regards, Pavel - Original Message - From: Rod Bacon To: braincrew.com ; [EMAIL PROTECTED] ; asterisk-users@lists.digium.com Sent: Thursday, March 17, 2005 3:24 AM Subject: Re: [Serusers] ser+asterisk - security Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN). You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified. - Original Message - From: braincrew.com To: [EMAIL PROTECTED] ; asterisk-users@lists.digium.com Sent: Thursday, March 17, 2005 5:00 AM Subject: [Serusers] ser+asterisk - security Hi there, I'm using ser and =sterisktogether. Asterisk for voice mail etc and ser forregistration of the =sers usig database.I can restrict =orwarding callsfrom another sip proxy to ser(using proxy_authorize) =ut how can I restrict access to asterisk ... Now everyone can forward calls to =y asterisk and can place pstn calls. Thanks in advance, Steve ___Serusers mailing list[EMAIL PROTECTED]http://mail.iptel.org/mailman/listinfo/serusers ___Serusers mailing list[EMAIL PROTECTED]http://mail.iptel.org/mailman/listinfo/serusers ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOW-To write an AGI
Roy Sigurd Karlsbakk wrote: I tried wiki, but I got too many pages (I think all of them), ...as answer. I want to write an agi. I need a HOW-TO, is there anything available? see the perl agi package from http://asterisk.gnuinter.net/, the agi/agi-test from the asterisk source and http://www.voip-info.org/wiki-Asterisk+AGI If you have some knowledge of Perl, it's fairly easy... the best thing to do is decide on a sample application and write it from scratch. I have written a few AGIs recently, now I have the daunting task of documenting them and putting them up somewhere :) Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk start problem (automatically)
Hello I am using Astwind under Debian Linux as my first trial of Asterisk. Somehow I managed to restart (!) Asterisk automatically when linux starts. (I did a make install, do you think making asterisk, will do it?) Ok, it is not a big deal, but, unfortunately, it stops with an exit code 127 and restart. Now, it is endless loop. Could somebody tell me the files to modify to stop it? Thanks. Turgut ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Comparing Callmanager to Asterisk
Title: Comparing Callmanager to Asterisk Callmanager does nothing than construct and tear down calls and the actual RTP stream does not flow through the Callmanager but is direct from IP device to IP device. How does this work with Asterisk? I read something that lead me to believe that Asterisk has to process the entire call, is this the case? Blake Parker CCNA Network Engineer Alacare Home Health Hospice, Inc. Email: [EMAIL PROTECTED] Phone: (205) 981-8648 Confidentiality Notice: This e-mail message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the original message. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ser+asterisk - security
[EMAIL PROTECTED] wrote: it's impossible to use iptables due to the reason that audio flows through asterisk and users won't be able to communicate w/ *... I was thinking of just the SIP port. I am assuming that asterisk protects its RTP ports from processing traffic from a third party. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ztdummy - no sound in Asterisk@Home
Hi Im using Asterisk @ Home 0.6 running on VM ware virtual machine. I have no interface card and have configured ztdummy as best I could. I know that the usb timer is installed correctly on the machine.I changed the following files to try getting it working: /usr/src/zaptel/Makefile - I changed the Module to read MODULES = zaptel ztdummy and PRIMARY = torisa then I entered make clean and make install, then recompiled libpri and asterisk, I added a modprobe ztdummy in /etc/rc.d/rc.local to make sure it is present at startup before Asterisk is launched I edited the file etc/init.d/zaptel and changed MODULES and RMODULES to point to ztdummy only. The problem is that after restarting asterisk the sound for commands like Playback and MusicOnHold do not work. But I can still hear ringing tone and call SIP to SIP with good sound. The only way I can get Playback to work after restarting the server is if I recompile zaptel and then make calls without reloading asterisk. Then it works fine. If I restart Asterisk again the sound fails. I have tried modprobe zaptel, modprobe ztdummy and sudo service zaptel start. Please mail me at [EMAIL PROTECTED] if you know how to get sound to work with ztdummy and not have to recompile zaptel every time. Jonathan Berger 082 574 5064 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk+radius
hi Any one give me any hint how to start radius with asterisk. Is there any addon available for asterisk+radius. Please provide me helpfull link which could help me. i am new to radius. regrads kamran __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme2 compilation
That doesn't work. I was trying to do it yesterday, there is a patch that fixes the problem. google for it or if ur too lazy: http://lists.digium.com/pipermail/asterisk-users/2004-August/059709.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_Spy and MOH - Any Status?
Hi List, As most know, Chan_Spy and consequently, the MOH patch that used Chan_Spy disappeared around version 1.0.2 (or so). I know the native MOH patch works well and doesn't require the mpg123, which as proved problematic, at least for me. However, I know of no method to "listen in" or supervise a conversation in real-time. I'm getting requests from users where the support team manager wants to passively "listen-in" so to help their support rep. I did see the post about using Flash Operator Panel and dragging the bridge to a conference group, but that isn't a good option inmy situation. Is there any thought on reviving Chan_Spy? Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pattern Matching?
[EMAIL PROTECTED] wrote:
I need to deploy some quasi-virtual-PBXes, and I'd like to avoid having to
be hands on for each new phone number deployed... so I would like to set
up some administrative extensions that can record greetings... lets say:
[admin]
exten = 8(NXXNXX),1,Record($1|-greeting.gsm)
[incoming]
exten = _(NXXNXX),1,Playback($1|-greeting)
exten = _(NXXNXX),2,Goto($1,1000)
exten = _(NXXNXX),102,Playback(generic-greeting)
[21]
exten = 1000,VoiceMail(2)
[310333]
exten = 1000,VoiceMail(3)
The concept here is like the capture buffer in a Perl regex.
So that if admin dialed 821, it would give them the chance to
record the greeting, which would be put in the 21222-greeting.gsm
file.
If someone called 21, it would play the 21-greeting.gsm
file, if it existed, otherwise if it failed, it would play
generic-greeting.gsm. Then it would change context based in the called
number.
Granted, I'm asking for alot here, but is there any way to approximate
this kind of an advanced configuration with Asterisk?
Steve
Not that difficult. A few things you will need:
${EXTEN} is the current extension dialed
goto statement
You can trim crap off your vars using the ${EXTEN:1} notations. In my
example, I am trimming the front digit off the exten var. If I wanted
to be fancy, I could trim x off the front, and only read for n digits
like this: ${EXTEN:x:n}.
At least, I think I could. Perhaps someone with more recent working
knowledge could confirm that?
It's all the in the wiki. When it's up that is. :)
Sean
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[Asterisk-Users] Using Codec G-726
Hi, What do I need to do to get Asterisk to allow me to use codec G-726? I've already tried allow=all in my sip.conf config.. didn't work... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Possible SPAM] :[Asterisk-Users] about sip, asterisk and cisco ccme
Hi I have a Cisco 2801 with CME and I'm trying to connect it to Asterisk through SIP. Can you please send me the Dial-peer configuration that creates a trunk between the Cisco router and Asterisk. Thank you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Jones Sent: Wednesday, March 16, 2005 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Possible SPAM] :[Asterisk-Users] about sip,asterisk and cisco ccme I am starting to work on a similar solution, but with full call manager rather than CME. I am going to use Asterisk to accept POTS calls through PCI FXO ports (winmodems) and then forward the calls through to call manager via SIP. I don't have my FXO cards yet (waiting for UPS man!!) but I have * talking to the CM through SIP just fine. I am testing with the Cisco softphone, connected as a call manager extension, and using the dial-plan to direct the call to *, and I do successfully get the * voicemail. Why do you want to use h323/skinny rather than SIP? -Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrea Riela Sent: Wednesday, March 16, 2005 5:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Possible SPAM] :[Asterisk-Users] about sip, asterisk and cisco ccme -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, I would create a structure like this: external sip server \ external sip server |-| Asterisk |--| Cisco CME |---| ip phones | external sip server / I would use Asterisk as SIP client for some SIP accounts on external servers ... then register those via H323 (if possible; skynny?) on Cisco CME ... Then I would use Asterisk to add the voicemail feature to Cisco CME. I don't know if that's possible, I'm really newbie on Asterisk, I know only Cisco world, and just a little bit. Any advice will be appreciated. Thanks for your support Regards dott. Andrea Riela -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCOLJtMakHrsrHP9wRAmDfAJ9AgcMf1CmdrLBk4HEdlvWKZiht7QCfcgns GbTX2LxGxO3ZR7iMIPqreJA= =eKlT -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Codec G-726
had the same thin with 729 I had to go disallow=all allow=g279 On Thu, 2005-03-17 at 16:37, Matt wrote: Hi, What do I need to do to get Asterisk to allow me to use codec G-726? I've already tried allow=all in my sip.conf config.. didn't work... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] t.38 support news?
Steve Underwood wrote: The bigegst holdup has really been the poor state of T.38 support in current equipment. Few ATAs do it. Fewer do it right. Care to share which ones do it right? We purchased 2 ATA's all which claim to do T38 and they don't. (Azatel, WorldAxx) -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk+radius
Kamran Ahmad wrote: hi Any one give me any hint how to start radius with asterisk. Is there any addon available for asterisk+radius. Please provide me helpfull link which could help me. i am new to radius. regrads kamran If you are new to radius then I will suggest and highly recommend that you stop now. Instead, do your own billing out of the Asterisk CDRs. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] t.38 support news?
Matthew Boehm wrote: Steve Underwood wrote: The bigegst holdup has really been the poor state of T.38 support in current equipment. Few ATAs do it. Fewer do it right. Care to share which ones do it right? We purchased 2 ATA's all which claim to do T38 and they don't. (Azatel, WorldAxx) -Matthew If I find one that does it right, I'll let you know. :-) Any idea who's chips are in those boxes? Several of the chips are bundled with software, just as many here have found with the PA168 based phones and ATAs. If one box using chip X is good or bad, probably all the others are similar. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Trunking, No connections any more...
Hello! I have bin trying to set up trunking between some of my Asterisk boxes but had no luck... I use Asterisk 2.0.6 on SuSE 6.2, but I have had the same problem with erlier releases. I have a working connection and can place multiple calls in both direktions. Than I set trunk=yes on both sides and reload. CLI iax2 show peers shows a (T) and low latency on both sides. Now it is not possible to get a call trough any more:-( I have working timers on both sides, Digium cards or ztDummy. I don't find very much diskussion about problems with trunking accept with the timing. Maybe I have missed something important. Does anyone have an idea??? By the way, I have removed SuSEs precompiled 1.0.0 zaptel drivers and use my own. Thanks in advance! Håkan pgp7mGEOlGQGj.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] echo paid support
I've got echo problems. *** I'm looking for paid support. *** I'll accept free support, but don't mind paying if someone really knows what they are doing. I've read the wiki, etc. Played with the settings in zapata.conf Using V400P PSTN-_T1-_ASTERISK-_BROADVOICE-_PSTNECHO ON CALLED PHONE PSTN-_T1-_ASTERISK-_T1-_PSTNNO ECHO VOIP-_ASTERISK-_T1-_PSTN ECHO ON VOIP PHONE G711 I have another trunk group and different T1's that go to a MAX TNT first: PSTN-_T1-_MAX_TNT-_VOIP-_ASTERISK-_VOIP_PHONE ECHO ON VOIP PHONE g711 PSTN-_T1-_MAX_TNT-_VOIP_G711-_ASTERISK_IAX_GSM-_ASTERISK_IAX_GSM-_VOIP_PHONE_g711 NO ECHO -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_oh323.c ast_oh323_new Internal channel initialization failed [solved]
hi
thanks all who helped me in making this success.
i am using latest asterisk from CVS.
asterisk-oh323-0.7.1,
pwlib-Janus_patch4-src-tar.gz,
openh323-Janus_patch4-src-tar.gz
GnuGatekeeper
it is working asterisk is routing calls to GNUGK
successfully
extensions.conf
exten = _40X,1,Dial(OH323/${EXTEN:2})
Thanks
Kamran Ahmad
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Re: [Asterisk-Users] who have been fabricated their own cards from Tormenta 2 PCI Card?
On Thu, 17 Mar 2005 14:53:27 +0800, XinTai Wang [EMAIL PROTECTED] wrote: who have been fabricated their own cards from Tormenta 2 PCI Card? govarion.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem starting Asterisk - libssl.so.4 cannot be found
you need to get openssl-dev package too for most dependencies problems you need respecitive dev libriaries regards m. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Possible SPAM] :[Asterisk-Users] about sip, asterisk and cisco ccme
Gilbert Abboud wrote: I have a Cisco 2801 with CME and I'm trying to connect it to Asterisk through SIP. Can you please send me the Dial-peer configuration that creates a trunk between the Cisco router and Asterisk. You can try something like this: dial-peer voice 900 voip destination-pattern 9... session protocol sipv2 !(the address of the Asterisk server) session target ipv4:192.168.0.100 !(in Asterisk use dtmfmode=rfc2833) dtmf-relay rtp-nte codec g711ulaw --TWH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] t.38 support news?
Steve Underwood wrote: Matthew Boehm wrote: Steve Underwood wrote: The bigegst holdup has really been the poor state of T.38 support in current equipment. Few ATAs do it. Fewer do it right. Care to share which ones do it right? We purchased 2 ATA's all which claim to do T38 and they don't. (Azatel, WorldAxx) -Matthew If I find one that does it right, I'll let you know. :-) Oh. You said fewer do it right so I took that to mean you had at least one that did it right, the best. Out of the ones you have tested, which ATAs do it best? or the closest to real thing? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Global Intercom on SIP phones
Thank you John, Max Blackmer I would like to create an Intercom extension that will dial a group of extensions which are connected to SIP phones. The SIP phones are setup to auto answer a particular extension assigned to one of the lines in the phone. All phones must answer and broadcast the page message at the same time. Has anyone done this? Or should I install an overhead speaker system using the oss/alsa console as a broadcast. Can the local port be set to auto answer calls? . Yes, it's been done. http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config JT ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] session border control
has session border control been added to asterisk yet? i remember hearing about it, but i haven't been able to find any information on it on wiki. Thanks, daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compilation problem chan_capi and Eicon Diva 4Bri
Upgrade to kernel 2.6.9, there are supposed to be significant bugfixes for CAPI support in 2.6.9. All of my CAPI systems use FC2, 2.6.9. I tried to go 2.6.10 but had problems. Craig - Original Message - From: Kib Eki [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, March 17, 2005 7:02 PM Subject: [Asterisk-Users] Compilation problem chan_capi and Eicon Diva 4Bri Hi *, I want to integrate the Eicon Diva 4Bri Card to Asterisk. Eicon drivers and capi is installed. I use the latest dev version from eicon compiled and installed for my fedora 2 system. I found the chan_capi for asterisk from www.junghanns.net. Also loaded the patch and applied to the chan_capi source tree. I changed the Makefile to include the capi20.h from eicon: INCLUDE=-I$(ASTERISK_HEADER_DIR) -I/usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon make install gives me the following errors: gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -I/usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c In file included from chan_capi.c:35: /usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon/capi20.h:52: error: Syntaxfehler before word /usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon/capi20.h:52: Warnung: kein Semikolon am Ende von »struct« oder »union« /usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon/capi20.h:53: Warnung: type defaults to `int' in declaration of `maxLogicalConnection' /usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon/capi20.h:53: Warnung: data definition has no type or storage class Unfortunately it is german system so also the compiler errors are in german. I realy need help because I am not the r+d expert. Thanks in advance. Kib ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Last guy to get BV working outbound?
I have tried everything to get BV working outbound. All worked fine
until the BV change last week. I called BV and they changed me to sip
gen with a new password. I stripped my Asterisk server to one phone on
Zap/1 until I get this working. The same BV account works fine with a
SPA-3000 so I don't suspect a firewall problem.
Symptoms: Asterisk registers with BV Ok
Incoming calls work
Outbound calls send Invite, receive 100, then 401
Asterisk sends an ACK instead of another Invite with credentials
If anyone knows what specifically makes Asterisk respond to the 401 with
credentials for an authenticated Invite, I'd appreciate it. I can't
seem to find this out.
Thanks in advance,
Brian
Here is my sip.conf:
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP channel to
context = default ; Default context for incoming calls
srvlookup = yes ; Enable DNS SRV lookups on outbound
calls
disallow=all; Disallow all codecs
allow=ulaw ; Allow codecs in order of preference
;
; Configuration for BroadVoice
;
register =
[EMAIL PROTECTED]:pword:[EMAIL PROTECTED]
;
[broadvoice]
type=peer
host=sip.broadvoice.com
secret=pword
fromuser=508XXX
username=508XXX
authuser=508XXX
fromdomain=sip.broadvoice.com
context=incoming
canreinvite=no
dtmfmode=inband
qualify=yes
in extensions.conf:
[default]
exten = _81XX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _81XX,2,Congestion()
exten = _81XX,102,busy()
Other Asterisk info:
*CLI sip show registry
Host Username Refresh State
147.135.0.128:5060508XXX 120 Registered
*CLI
*CLI show version
Asterisk CVS-03/11/05-16:07:49 built by [EMAIL PROTECTED] on a i686
running Linux
*CLI
*CLI Mar 17 10:35:08 NOTICE[-245486672]: chan_sip.c:5047
handle_response: Failed to authenticate on INVITE to 'Analog1
sip:[EMAIL PROTECTED];tag=as212bf17
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[Asterisk-Users] Re: asterisk+radius
i have written app for billing with asterisk. what is the problem in using radius. kamran __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Snom190 intercom
As you can see from the SIP trace below (from the called phone), intercom=true is being appended to the To: header as per requirements. The intercom=true needs to be appended to the request URI, not to the header as a whole -- your To: header should be: To: sip:1011 at 192.168.10.150:2051;line=9avrmhew;intercom=true Mind you, I didn't get the phone to respond to the intercom=true until I added it on the request line as well, so the INVITE line of your request would be: INVITE sip:1011 at 192.168.10.150:2051;line=9avrmhew;intercom=true SIP/2.0 I'm on a Snom 220 with SIP 3.56t, and I'm stuck on the very next step of the process -- answering the phone's challenge to the INVITE request. The wiki indicates that the Snom needs to challenge with realm=snom, but even if I add snom into our internal DNS so that I can set the registrar to snom (that being the only way I can see to change what the phone uses as realm), it still rejects the digest response. Anyone have this working with recent loads of SIP that can shed any light on this? I've email'd snom a few days ago but have yet to get a response. According to their web page, they have a new office as of April 1, and I got a response to a support request (on this very issue) today saying that they'd likely not be able to respond until people are settled into the new offices, so you'll likely have to be patient with them. -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Snom190 intercom
As you can see from the SIP trace below (from the called phone), intercom=true is being appended to the To: header as per requirements. The intercom=true needs to be appended to the request URI, not to the header as a whole -- your To: header should be: To: sip:1011 at 192.168.10.150:2051;line=9avrmhew;intercom=true Mind you, I didn't get the phone to respond to the intercom=true until I added it on the request line as well, so the INVITE line of your request would be: INVITE sip:1011 at 192.168.10.150:2051;line=9avrmhew;intercom=true SIP/2.0 I'm on a Snom 220 with SIP 3.56t, and I'm stuck on the very next step of the process -- answering the phone's challenge to the INVITE request. The wiki indicates that the Snom needs to challenge with realm=snom, but even if I add snom into our internal DNS so that I can set the registrar to snom (that being the only way I can see to change what the phone uses as realm), it still rejects the digest response. Anyone have this working with recent loads of SIP that can shed any light on this? I've email'd snom a few days ago but have yet to get a response. According to their web page, they have a new office as of April 1, and I got a response to a support request (on this very issue) today saying that they'd likely not be able to respond until people are settled into the new offices, so you'll likely have to be patient with them. -- Joshua P. Dady ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk dialplan (and/or VM) via LDAP
Daniel, Did you get much progress made on this? I'm new to asterisk - but we are a heavily invested LDAP shop and if I can demo an initial install that pulls telephony configs from LDAP it would really be nifty. I'd be happy to help in any way I can - I'm not much of a developer - but have some good design experience and I could test as well. Let me know... John Dunning Asst Dir Net Tech Svcs Wayne State College___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk dialplan (and/or VM) via LDAP
Am Donnerstag 17 März 2005 17:49 schrieb John B Dunning: Daniel, Did you get much progress made on this? I'm new to asterisk - but we are a heavily invested LDAP shop and if I can demo an initial install that pulls telephony configs from LDAP it would really be nifty. I'd be happy to help in any way I can - I'm not much of a developer - but have some good design experience and I could test as well. Let me know... John Dunning Asst Dir Net Tech Svcs Wayne State College Err? Did you click on the wrong reply to adress? Jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller ID problem
Hi List I've been using Asterisk for quite some time with no major problems, but I've been facing this bug from the beginning and now I want to see if that is fixable. We have a provider who terminates our USA LD traffic and the problem comes when relaying the caller ID I send them from my Asterisk. Here is the weird thing, I send a call with valid caller ID info (areacode+number); my provider gets the call and routes it properly, the end receiver gets the call and does not see the caller ID I sent, they just get 'Unknown Number'. This remains true for some receivers, others using a different telephone company or cellular company do get the caller ID I sent. Examples: Cingular, Verizon do not show my caller ID info Nextel, T-mobile do show my caller ID info Is there something I am not following or not doing it industry standard? Thanks Oswaldo A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: asterisk+radius
Kamran Ahmad wrote: i have written app for billing with asterisk. what is the problem in using radius. kamran Its a pain and redundant. Why run two seperate databases when 1 will do what you need? There is no native radius support for Asterisk. There is an addon, (search the wiki) but the last I heard of it, it was unstable. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Netlogic inbound DID issue
Anyone out there using NetLogic DIDs? And have inbound working? I got
outbound working, but no joy so far with inbound. Here are the relevant
parts from my conf files:
iax.conf
[general]
tos=lowdelay
jitterbuffer=no
register = username:[EMAIL PROTECTED]
[netlogic]
type=friend
host=dynamic
context=sourcekit-main
auth=plaintext
username=
secret=
disallow=all
allow=ulaw
allow=all
extensions.conf
[sourcekit-sip]
exten = 101,1,Dial(SIP/SK-101,20)
exten = 101,2,Voicemail(u101)
exten = 101,102,Voicemail(b101)
exten = 101,103,Hangup
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
[sourcekit-main]
include=sourcekit-sip
exten = +19193233010,1,GoTo(sourcekit-sip,101,1)
exten = _1NXXNXX,1,SetCallerID(9193233010)
exten = _1NXXNXX,2,Dial(IAX2/netlogic/${EXTEN})
exten =
_1NXXNXX,3,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,4,Hangup
[netlogic]
include=sourcekit-main
and, thr debug output from * CLI:
Asterisk Ready.
*CLI iax2 debug
IAX2 Debugging Enabled
*CLI Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX
Subclass: NEW
Timestamp: 00017ms SCall: 00030 DCall: 0 [206.80.70.49:4569]
VERSION : 2
CALLED NUMBER : +19193233010
Unknown IE 045 : Present
CALLING NUMBER : +13362150564
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
LANGUAGE: en
CALLED CONTEXT : netlogic
USERNAME: username
FORMAT : 4
CAPABILITY : 2097151
ADSICPE : 2
DATE TIME : 175199382
Ignoring unknown information element 'Unknown IE' (45) of length 1
Mar 17 12:35:19 NOTICE[21100]: chan_iax2.c:5419 socket_read: Rejected
connect at
tempt from 206.80.70.49, who was trying to reach '[EMAIL PROTECTED]'
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REJECT
Timestamp: 00018ms SCall: 2 DCall: 00030 [206.80.70.49:4569]
CAUSE : No authority found
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[Asterisk-Users] Polycom vs. Cisco IP Phones
Hi all, I am working on building a new VoIP PBX. Looking at the current market for phones it seems my best enterprise options are the Cisco and Polycom phones. I have some experiance with the Cisco 7940G, but the process of flashing the phone with the SIP firmware left a bad taste in my mouth (not to mention the added expense for the phone). What is the general consensis about the polycom IP phones? Are they good? Are they better than Cisco? What do I do for the receptionist's station? Thanks in advance, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MOH patch for bristuffed *
Anybody knows how to patch the music on hold bug on a bristuffed-0.2.0-RC7j 1.0.6-asterisk ? Thanks maxx ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID problem
On Thu, Mar 17, 2005 at 11:59:06AM -0500, Oswaldo Arratia wrote: I send a call with valid caller ID info (areacode+number); my provider gets the call and routes it properly, the end receiver gets the call and does not see the caller ID I sent, they just get 'Unknown Number'. This remains true for some receivers, others using a different telephone company or cellular company do get the caller ID I sent. Examples: Cingular, Verizon do not show my caller ID info Nextel, T-mobile do show my caller ID info When we ran into this with certain carriers, it was because we were using PRI NI-2 to our CLEC, but not setting the isdn numbering type to 'national' (it was set to 'unknown' or something else.) Apparently, some carriers look at that bit before deciding to accept/display the CLID, some don't. Try setting the following in your zaptel.conf: switchtype=national and see if it makes any difference. bill ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID problem
Thanks for the pointer, makes sense. Although I am not using zaptel, I am sending the calls via SIP to my Cisco AS5300 which is connected via PRI to my provider and it happens to be set like NI2. I will test and will let you know. Thanks!!! Oswaldo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Petrisko Sent: Thursday, March 17, 2005 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Caller ID problem On Thu, Mar 17, 2005 at 11:59:06AM -0500, Oswaldo Arratia wrote: I send a call with valid caller ID info (areacode+number); my provider gets the call and routes it properly, the end receiver gets the call and does not see the caller ID I sent, they just get 'Unknown Number'. This remains true for some receivers, others using a different telephone company or cellular company do get the caller ID I sent. Examples: Cingular, Verizon do not show my caller ID info Nextel, T-mobile do show my caller ID info When we ran into this with certain carriers, it was because we were using PRI NI-2 to our CLEC, but not setting the isdn numbering type to 'national' (it was set to 'unknown' or something else.) Apparently, some carriers look at that bit before deciding to accept/display the CLID, some don't. Try setting the following in your zaptel.conf: switchtype=national and see if it makes any difference. bill ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec negociation
Hi, I've got an Asterisk latest CVS head with oh323 installed. There is one thing I can't understand about the codec negociation. I receive calls in G723G729, and send them to another gateway who can handle both codecs too. So all I want to do is just passthrou, for both. It seems that * only try to send with the first of the list, what is fine when it's the good one, but otherwise he complain about being unable to transcode instead of trying the second codec. I hope I've explained well my problem. Could someone explain me a little bit more about the negociation ? Or did someone have the same issue ? I didn't find much info, tried docs google. Thank you. Yves ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: asterisk+radius
Oh this is sad.. I'm familiar with radius.. and was hoping to be able to use asterisk with freeradius to be able to do call accounting and billing.. so you're telling me this is now not a good idea? Am I better off (for now) parsing the csv report each month? On Thu, 17 Mar 2005 11:00:09 -0600, Matthew Boehm [EMAIL PROTECTED] wrote: Kamran Ahmad wrote: i have written app for billing with asterisk. what is the problem in using radius. kamran Its a pain and redundant. Why run two seperate databases when 1 will do what you need? There is no native radius support for Asterisk. There is an addon, (search the wiki) but the last I heard of it, it was unstable. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom vs. Cisco IP Phones
I think Cisco VoIP phones are absolute works of art. The first time I saw one, I wanted them, That being said, I use Polycom IP 500s and I absolutely love them. The speakerphone is excellent, configs are pretty simple once you know what you are doing with them, and the phone is very aesthetically pleasing. Firmware can be updated by placing the .ld files on an FTP server and telling the phone where to get configs. Very simple to do. The compatibility is good for using them with Asterisk with no surprises. My $0.02 Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Max Clark Sent: Thursday, March 17, 2005 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom vs. Cisco IP Phones Hi all, I am working on building a new VoIP PBX. Looking at the current market for phones it seems my best enterprise options are the Cisco and Polycom phones. I have some experiance with the Cisco 7940G, but the process of flashing the phone with the SIP firmware left a bad taste in my mouth (not to mention the added expense for the phone). What is the general consensis about the polycom IP phones? Are they good? Are they better than Cisco? What do I do for the receptionist's station? Thanks in advance, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codec negociation
If you don't want to proxy the media through * the put this setting: canreinvite=yes this will allow the 2 end points to connect directly for the RTP bypassing you. otherwise I have noticed the same when I try to proxy I have to make sure everyone is using the same codec or it doesn't work well. .o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yves Sent: Thursday, 17 March, 2005 12:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Codec negociation Hi, I've got an Asterisk latest CVS head with oh323 installed. There is one thing I can't understand about the codec negociation. I receive calls in G723G729, and send them to another gateway who can handle both codecs too. So all I want to do is just passthrou, for both. It seems that * only try to send with the first of the list, what is fine when it's the good one, but otherwise he complain about being unable to transcode instead of trying the second codec. I hope I've explained well my problem. Could someone explain me a little bit more about the negociation ? Or did someone have the same issue ? I didn't find much info, tried docs google. Thank you. Yves ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom vs. Cisco IP Phones
The lack of full SIP suport and the cost of Ciscos license plus the added base cost of their phones moved us away from Cisco and over to Polycom. They have been working extremely well. Software updates are free and the update process is relatively simple. I have found the IP600 is a great desk phone. I do use some IP300s for break rooms etc, but do not like them on a desk. Not sure the rquirements for your receptionist. I have found that the IP600 does have most everything required to function properly. If you do have an office without DID and a lot of traffic then you may want to look at the tools to display status on her computer. I do have a Snom inhouse for testing when I get a chance. Their quality however is not as good as either Cisco or Polycom. On Mar 17, 2005, at 11:03 AM, Max Clark wrote: Hi all, I am working on building a new VoIP PBX. Looking at the current market for phones it seems my best enterprise options are the Cisco and Polycom phones. I have some experiance with the Cisco 7940G, but the process of flashing the phone with the SIP firmware left a bad taste in my mouth (not to mention the added expense for the phone). What is the general consensis about the polycom IP phones? Are they good? Are they better than Cisco? What do I do for the receptionist's station? Thanks in advance, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Redhat 9 Music on hold
I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines connected via TE405P. Everything works great, except MOH. I added an exten with MusicOnHold(30), and it plays just fine. Conferences have music when no one is in. I have SIP phones. When I place a call on hold, the CLI give no indication the call is on hold. I have set musiconhold(default) everywhere, removed it from everywhere, nothing seems to help. I am using 59r of MPG123, and do not have MPG321 installed. I did a 'make mpg123' from asterisk, make no difference. Please help!! This is driving me crazy! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: PC sound hardware for voice recording
What would be a minimum sound card/microphone combo for good voice quality recording on a budget? This would be for * voice prompts. Would a soundblaster live and a good mic do the job? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID problem - SOLVED
I did it and it worked. The problem was the national plan! Thank you very much for your tip. For those who run into this, here is the configuration of the voice port in a Cisco AS5300 series: interface Serial2:23 no ip address isdn switch-type primary-ni isdn incoming-voice modem isdn map address .. plan isdn type national isdn T310 6 isdn send-alerting isdn negotiate-bchan no cdp enable -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Petrisko Sent: Thursday, March 17, 2005 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Caller ID problem On Thu, Mar 17, 2005 at 11:59:06AM -0500, Oswaldo Arratia wrote: I send a call with valid caller ID info (areacode+number); my provider gets the call and routes it properly, the end receiver gets the call and does not see the caller ID I sent, they just get 'Unknown Number'. This remains true for some receivers, others using a different telephone company or cellular company do get the caller ID I sent. Examples: Cingular, Verizon do not show my caller ID info Nextel, T-mobile do show my caller ID info When we ran into this with certain carriers, it was because we were using PRI NI-2 to our CLEC, but not setting the isdn numbering type to 'national' (it was set to 'unknown' or something else.) Apparently, some carriers look at that bit before deciding to accept/display the CLID, some don't. Try setting the following in your zaptel.conf: switchtype=national and see if it makes any difference. bill ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Polycom IP 300/500 Conferencing Behavior
On Fri, 21 Jan 2005, Greg Boehnlein wrote: Hello, I've got a mixture of SPIP 300 and 500 phones in production for various clients. I've got the XML settings configured for local conferencing, but I'm not seeing the expected behavior from the phone when I attempt to conference two calls together. According to the manual, while talking to the first party, you simply hit Conference, dial the second party and then Conference to join them. This is supposed to put the first party on Hold until you bridge them together with the second press of the Conference button. That is all fine and well, but it doesn't quite work the way that the manual describes. Instead of joining the two calls together when the Conference key is pressed for the second time, the first party is taken off hold and hears dead silence. The only way to correctly join the parties is to hit the Hold and then Resume soft key, at which point all three parties can talk to each other. As an illustration Conf - Dial - Conf doesn't work. However, Conf - Dial - Conf - Hold - Resume DOES work. I'm running 1.3.4 firmware on all the phones, and I can't for the life of me figure out what is causing this problem. It is very likely some misconfiguration in the XML files, but I can't find it. Anyone have any suggestions? Hello, I just thought I would follow-up on this post and mention that somwhere between Jan 21st and today, the Conferencing issue that I described below is no longer an issue. Normally, I wouldn't include the entire quoted context of the message for bandwidth reasons, but in this case, since the topic is nearly 2 months old, I figured it would be helpful to keep things consistent and on-thread. I'm not sure what may have been fixed. I.E. I don't know if it was a patch to chan_sip in stable or what. All I can tell you is that I haven't made a single change to either sip.conf or my XML config files, since the original posting in January. However, I have updated asterisk several times from the 1.0 branch. I'm happy that this is fixed, but I am going to do a little more reasearch to see if I can get it to fail again by backdating chan_sip and incrementing it forward. This should be interesting. :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec negociation
I read about this option. But does it work on a h323 channel ? (inAccessnetwork's one) Brian C. Fertig wrote: If you don't want to proxy the media through * the put this setting: canreinvite=yes this will allow the 2 end points to connect directly for the RTP bypassing you. otherwise I have noticed the same when I try to proxy I have to make sure everyone is using the same codec or it doesn't work well. .o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yves Sent: Thursday, 17 March, 2005 12:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Codec negociation Hi, I've got an Asterisk latest CVS head with oh323 installed. There is one thing I can't understand about the codec negociation. I receive calls in G723G729, and send them to another gateway who can handle both codecs too. So all I want to do is just passthrou, for both. It seems that * only try to send with the first of the list, what is fine when it's the good one, but otherwise he complain about being unable to transcode instead of trying the second codec. I hope I've explained well my problem. Could someone explain me a little bit more about the negociation ? Or did someone have the same issue ? I didn't find much info, tried docs google. Thank you. Yves ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime Problem = Segmentation faults
Hi: I had asterisk with RealTime database working perfectly in a RH 9.0 machine. I used the sip cache so I even had MWI working. The problem is that I decided to move to Fedora Core 3. I installed the lastets cvs version of asterisk and the RealTime addon from asterisk-addons. I at first had the problems with the kernel and the zaptel driver but all that was solved with the configuration from the Asterisk Wiki. Then when I moved my configuration to the new asterisk server and configured the RealTime addon it falls in a Segmentation fault. If I do not load the res_config_mysql.so (edited at modules.conf) then asterisks runs without any problem. But if I load the module from boot or from the asterisk command load res_config_mysql.so then I get the Segmentation fault again. I'm not sure what the problem is. Is it a Fedora Core 3 problem, or an Asterisk latest version problem? I don't think it is a configuration problem because I just used the same configuration I had before. The only diferences may be the OS and probably the asterisk version that is only one week newer than the one I was running in the old asterisk server, so I'm probably even running the same version of asterisk in both machines. Any advise? Someone else have a similar configuration working with Fedora Core 3? Thanks in advance, -- Jose R. Ortiz Ubarri (CHEO), CS System Administrator / Programmer High Performance Computing facility - UPR Email: [EMAIL PROTECTED]|[EMAIL PROTECTED] Phone: 787-758-3054 Fax: 787-758-3058 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Polycom vs. Cisco IP Phones
Not sure the rquirements for your receptionist. I have found that the IP600 does have most everything required to function properly. If you do have an office without DID and a lot of traffic then you may want to look at the tools to display status on her computer. I do have a Snom inhouse for testing when I get a chance. Their quality however is not as good as either Cisco or Polycom. My experience is that the Cisco and Polycom phones are both about in terms audio quality and useability. Neither one does exactly what I'd expect with respect to multiple lines. They both take a little extra setup in this regard, but you can read the wiki for that stuff. Snoms do exactly what I'd expect for a multiple line phone, are very easy to setup, but the audio quality and usability do not compare favorably with either Cisco or Polycom. Between Cisco and Polycom, I went with Polycom just because of cost. The Polycom units are MUCH less expensive (since this is not the biz list, ask me privately about my reseller that is cheaper than others you'll find). On the other hand, Polycom VoIP phones are NOT supported by the company. The only way I've gotten support for them is from this list. Cisco definitely supports all their products quite nicely (for a fee). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
