Re: [Asterisk-Users] {Scanned}
David wrote:
Hello All,
I loaded [EMAIL PROTECTED] I have one X100P card. I try to dail out but get
rejected.
Any help...
Thanks, David
Before someone else answers with a violent reply...
Your question, while reasonable, does not help anyone in helping you. Why don't
you try and provide more details, such as your configuration files, and be a bit
more verbose in explaining your predicament?
Based on your email, i can only provide the following possibilities to your
problem:
1. Your phone line is not connected to the X100P.
2. Asterisk isn't started, or has died for some reason.
3. Your phone line has been disconnected by your phone company.
4. The asterisk box is not powered on.
5. Your Asterisk setup is misconfigured.
6. Your call was rejected because you dialed an invalid phone number.
See what I mean?
Plus, having an email subject of {Scanned} is only going to cause people to
overlook your email.
Flynn
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[Asterisk-Users] NIC irq load balancing
Hi All, I'm developing an outbound call center with 20 agents. My configuration is like this. PRI * NetGear Switch 20 iaxSoftPhone I'm experincing bad voice quality and long delay. I'm thinking about several possibilities. 1. NIC load - All NIC irqs process by CPU0. I tried irabalance, but no effect. #cat /proc/interrupts CPU0 CPU1 0: 158232 385091IO-APIC-edge timer 1: 3 0IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 3: 56271IO-APIC-edge serial 4: 10 0IO-APIC-edge serial 8: 1 0IO-APIC-edge rtc 12:185 0IO-APIC-edge PS/2 Mouse 14: 20312 2667IO-APIC-edge libata 15: 0 0 XT-PIC libata 17: 573544 98848 IO-APIC-level Intel ICH5 18: 947385 0 IO-APIC-level eth0 21: 7908154593392 IO-APIC-level t1xxp NMI: 0 0 LOC: 543220 543219 ERR: 0 MIS: 0 2. NetGear Switch - I'm using FS-526T Switch, which has 24 10/100 ports and 2 Gb sorts. I want to know if this kind of general purpose switch is not suitable for voip. If so, could you recommand one? 3. Server - My server is based on ASUS md, 2 Xeon 2.8G, 1GB ram, 1 sata drive. OS is Redhat9.0 CPU's idle status is 70~100. Regards, Jason __ 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
[Asterisk-Users] specific call transfer
Hi, is it possible to transfer an incomming call to another ext. without answering? I'm not talking about (un)conditional redirection but functionality, when calee can each time decide whether answer the phone or transfer it to any other phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with call pickup
Title: Message I have configured call pickup, and this works fine. Although there are 2 problems, perhaps anyone would know a solution to this; - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. - When a caller that I've answered through Call-Pickup disconnects, my phone does not close the connection but acts like there is still someone on the otherside. (Logging shows dat de Zap/channel has cleared, but not the SIP/channel) I use Asterisk 1.0.2-BRIstuffed-0.2.0-RC2 Any help would be greatly appreciated... Ramon ___ 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] Numbering plan for incoming call CLID on chan_zap (PRI)
On Fri, 7 Jan 2005, Roger Schreiter wrote: whatever dialplan I'm using for outgoing calls via PRI (Digium card, chan_zap), the callerid when receiving calls has no leading zeros, which are normally used to distinguish local, national and international calls in Europe. The number has always the area code in front, but the country code only for foreign calls. This is normal for isdn, the numbers are distinguised by their Type of number and Numbering plan which are sent along with the actual digits. Now I'm looking for any mean to decide, whether the received callerid begins with a country code and thus comes from another country or is domestic. Is there maybe any variable indicating this? Yes, CALLINGTON, but it is broken at the moment in cvs. I have a patch that works for us. Perhaps you can try it as well before I submit it? Contact me off-list if you are interested. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with call pickup
Ramon Peek wrote: - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. This is correct. You are placing a call to *8 which just happens to connect you to caller. As far as your phone is concerned it is talking to someone at *8. ___ 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] OT: TE405P pins and slots
On Thu, 6 Jan 2005, Andrew Kohlsmith wrote: I imagine the Expansion is for more spans -- nothing has been designed for them at this point. Timing is likely for carrying timing across multiple cards, Test for testing and ident is for card order when multiple cards are inserted into one system. The timing port can be really usefull if the drivers can be changed along the lines of http://florz.dyndns.org/zaphfc/. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Signaling / Streaming
Hi When I forward calls from SER (or GNUGK) to Asterisk, the SER ( or GNUGK) are just used for signaling, but the call streaming passes from the endpoint directly to Asterisk, isnt it? Or does the streming passes from the Endpoint to SER and then to the Asterisk? Thanks Joao Pereira ___ 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] off topic - SSH configuration for Digium Support
I've an issue with my TDM4000P card and I will be calling Digium later to ask for their help. Could anyone help me with a basic configuration so they can SSH to me? Thanks John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip protocol question ...
Hi, I'm tryinig to debug SIP call from activex control based on MS RTC (A) to Asterisk (B). I use Etherreal to follow packages and I would like to ask short questions: - Session trace shows following order of packets: A - BInvite B - A100 Trying B - A200 OK, with session description ; repeated 6 times A - B BYE sip: B - A 200 OK - in my newbie logic it seems that B simply disconnects for some reason. In session description there are codec specs. Unfortunately I don't have much docs on this active x control, so don't know how it behaves or whether it works. But anyway, does B anyhow tells reason why it requests disconnection ? Could I somehow from SIP packets gain knowledge about possible cause of disconnection ? Thanks in advance, regards, Robert. ___ 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] Signaling / Streaming
Hi, With Gnugk, make sure the proxy mode is not enabled if you want voice to pass directly from endpoints. Regards Lamine - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 10:21 AM Subject: [Asterisk-Users] Signaling / Streaming Hi When I forward calls from SER (or GNUGK) to Asterisk, the SER ( or GNUGK) are just used for signaling, but the call streaming passes from the endpoint directly to Asterisk, isnt it? Or does the streming passes from the Endpoint to SER and then to the Asterisk? Thanks Joao Pereira ___ 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] Signaling / Streaming
Ok, then I guess the way we use SER and GNUGK to redirect calls to Asterisk makes the diference. If we are using them as proxy, the stream will pass through them, if we dont use proxy, they will be used just for signaling. Joao - Original Message - From: Mamadou Lamine KA [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 10:50 AM Subject: Re: [Asterisk-Users] Signaling / Streaming Hi, With Gnugk, make sure the proxy mode is not enabled if you want voice to pass directly from endpoints. Regards Lamine - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 10:21 AM Subject: [Asterisk-Users] Signaling / Streaming Hi When I forward calls from SER (or GNUGK) to Asterisk, the SER ( or GNUGK) are just used for signaling, but the call streaming passes from the endpoint directly to Asterisk, isnt it? Or does the streming passes from the Endpoint to SER and then to the Asterisk? Thanks Joao Pereira ___ 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] Sip protocol question ...
What control is it ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman Sent: vendredi 7 janvier 2005 11:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sip protocol question ... Hi, I'm tryinig to debug SIP call from activex control based on MS RTC (A) to Asterisk (B). I use Etherreal to follow packages and I would like to ask short questions: - Session trace shows following order of packets: A - BInvite B - A100 Trying B - A200 OK, with session description ; repeated 6 times A - B BYE sip: B - A 200 OK - in my newbie logic it seems that B simply disconnects for some reason. In session description there are codec specs. Unfortunately I don't have much docs on this active x control, so don't know how it behaves or whether it works. But anyway, does B anyhow tells reason why it requests disconnection ? Could I somehow from SIP packets gain knowledge about possible cause of disconnection ? Thanks in advance, regards, Robert. ___ 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] Signaling / Streaming
Yes, This mode is generally used when some endpoints have private addresses behind a NAT while others have public addresses. In this case all the traffic passes through the GK. Take a look at paragraph related to Proxy at http://www.gnugk.org/gnugk-manual-4.html#ss4.2 Lamine - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 11:18 AM Subject: Re: [Asterisk-Users] Signaling / Streaming Ok, then I guess the way we use SER and GNUGK to redirect calls to Asterisk makes the diference. If we are using them as proxy, the stream will pass through them, if we dont use proxy, they will be used just for signaling. Joao - Original Message - From: Mamadou Lamine KA [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 10:50 AM Subject: Re: [Asterisk-Users] Signaling / Streaming Hi, With Gnugk, make sure the proxy mode is not enabled if you want voice to pass directly from endpoints. Regards Lamine - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 10:21 AM Subject: [Asterisk-Users] Signaling / Streaming Hi When I forward calls from SER (or GNUGK) to Asterisk, the SER ( or GNUGK) are just used for signaling, but the call streaming passes from the endpoint directly to Asterisk, isnt it? Or does the streming passes from the Endpoint to SER and then to the Asterisk? Thanks Joao Pereira ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P - Segmentation fault - Help!
Hi all,
I'm trying to install a TDM400P card, and I need some
help.
Please, see below...
after dmesg command:
[EMAIL PROTECTED] root]# dmesgvia82cxxx: board #1 at
0xD800, IRQ 5Zapata Telephony Interface Registered on major 196PCI:
Found IRQ 3 for device 00:09.0PCI: Sharing IRQ 3 with 00:10.1Freshmaker
version: 71Freshmaker passed register testModule 0: Installed -- AUTO
FXS/DPOModule 1: Installed -- AUTO FXS/DPOModule 2: Installed -- AUTO
FXO (FCC mode)Module 3: Installed -- AUTO FXO (FCC mode)Found a Wildcard
TDM: Wildcard TDM400P REV E/F (4 modules)
after [EMAIL PROTECTED] root]# asterisk -cp
command:
[chan_phone.so] = (Linux Telephony API
Support) == Parsing '/etc/asterisk/phone.conf': Found ==
Registered channel type 'Phone' (Standard Linux Telephony API
Driver)[chan_zap.so] = (Zapata Telephony w/PRI) ==
Parsing '/etc/asterisk/zapata.conf': FoundJan 6 14:57:50
WARNING[-1084944256]: chan_zap.c:665 zt_open: Unable to
specify channel 1: No such device or addressJan 6 14:57:50
ERROR[-1084944256]: chan_zap.c:5340 mkintf: Unable to open channel 1: No such
device or addresshere = 0, tmp-channel = 1, channel = 1Jan 6
14:57:50 ERROR[-1084944256]: chan_zap.c:7377 setup_zap: Unable to register
channel '1-2'Jan 6 14:57:50 WARNING[-1084944256]: loader.c:313
ast_load_resource: chan_zap.so: load_module failed, returning -1 ==
Unregistered channel type 'Tor' == Unregistered channel type
'Zap'Segmentation fault[EMAIL PROTECTED]
root]#
Please, see.conf files
below:
zaptel.conf
fxoks=1-2
fxsks=3-4
loadzone = us
defaultzone=us
zapata.conf
[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=3.5
txgain=3.5
immediate=no
busydetect=yes
busycount=5
callprogress=no
usecallerid=yes
hidecallerid=no
;calleridcallwaiting=yes
callerid=asreceived
musiconhold=default
relaxdtmf=yes
accountcode=pstn_local
amaflags=billing
echotraining=yes
context=fxs ;Context to
FXS ports
group=1
signalling=fxo_ks
channel=1-2
context=fxo ;Context to
FXO ports
group=2
signalling=fxs_ks
channel=3-4
extensions.conf
[fxs]
exten = 100,1,Dial,Zap/1
exten = 100,1,Dial,Zap/2
exten =
_9X.,1,Dial,Zap/3/${EXTEN:1}
[fxo]
exten = s,1,Dial,Zap/4
thanks
César
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Re: [Asterisk-Users] TDM400P - Segmentation fault - Help!
Please stop re-posting the exact same thing over, and over, and over again. Then, while you are sitting thinking about this, wondering why you haven't yet got a response, how about you work out how to switch off HTML emails. Send it in plain text, more people will bother reading it, and responding. On Fri, 2005-01-07 at 10:02 -0300, Csar Davi vila do Nascimento wrote: Hi all, zaptel.conf fxoks=1-2 fxsks=3-4 zapata.conf [channels] signalling=fxo_ks channel=1-2 signalling=fxs_ks channel=3-4 You have your fxo/fxs confused. zaptel.conf and zapata.conf need to be opposite values. Look at the wiki, I am sure there are some examples there. Regards, Adam ___ 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] Sipura SPA-1001 and Tivo Series 1
On Thu, 6 Jan 2005 23:50:40 -0500, David Ishmael wrote: What about when users switch to 100% VoIP? I've been considering getting DirecTV with the HD PVR and I've heard it can't use broadband, in a case like that I would have to route a modem call through VoIP (or is there a better way I'm just not seeing). -Dave I think you'll find that this is much like dealing with fax. Painfull and prone to failure. Since Tivo can be a network aware device that's the more reliable route. FWIW, many series 1 Tivo units blew up their on-boards modems. This is one of the reasons that the TivoNet card to be made. Once the modem was dead you could add a net card or pay Tivo to exchange yours for a remanufactured unit. There is also a wireless TivoNet card which saves you running Cat 5 into your TV room if that's a benefit. Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Thursday, January 06, 2005 11:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Erik Espinoza Subject: Re: [Asterisk-Users] Sipura SPA-1001 and Tivo Series 1 Get the TivoNet card from http://www.9thtee.com/turbonet.htm. I have one in my series 1 Tivo. It's easy to install, and works great. It'll be a lot less hassle then trying to make Tivo use an analog line through your * server. If you consider the cost the FXS port it's cheaper than going through * anyway. Michael On Thu, 6 Jan 2005 19:46:48 -0800, Erik Espinoza wrote: Most digital devices such as modems, fax machines and tivo's can not be used without a lot of changes on VoIP. I've seen success with TiVo when you use a special code to kick it down to 14.4 kbps and use g711ulaw as the codec. I think your best bet is to try to eBay the custom nic for the TiVo series 1. Erik On Thu, 6 Jan 2005 20:39:45 -0500, David Ishmael [EMAIL PROTECTED] wrote: Hi everyone, I just got a Sipura SPA-1001 and have connected my Tivo Series 1 (yes its old). When I do a test call with Tivo, the call always fails (it seems to dial the number but never connects). I can pick up the phone line and hear the Tivo talking. I've tried looking around for anything special I need to do but its still not working. I can connect a phone to the SPA-1001 and can make outgoing calls just fine. I even tried calling the Tivo number and can hear the modem pick up. Has anyone done this? Any help would be greatly appreciated. Thanks, Dave ___ 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] off topic - SSH configuration for Digium Support
On Fri, 7 Jan 2005 10:36:50 +, John Middleton wrote: I've an issue with my TDM4000P card and I will be calling Digium later to ask for their help. Could anyone help me with a basic configuration so they can SSH to me? On your router you'll need to port forward port 22 to your Asterisk server. Persuming that you already have sshd running on your server that's about it. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P - Segmentation fault - Help!
hello, I've tried do it, but nothing happened. Regards Csar - Original Message - From: Adam Goryachev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 9:20 AM Subject: Re: [Asterisk-Users] TDM400P - Segmentation fault - Help! Please stop re-posting the exact same thing over, and over, and over again. Then, while you are sitting thinking about this, wondering why you haven't yet got a response, how about you work out how to switch off HTML emails. Send it in plain text, more people will bother reading it, and responding. On Fri, 2005-01-07 at 10:02 -0300, Csar Davi vila do Nascimento wrote: Hi all, zaptel.conf fxoks=1-2 fxsks=3-4 zapata.conf [channels] signalling=fxo_ks channel=1-2 signalling=fxs_ks channel=3-4 You have your fxo/fxs confused. zaptel.conf and zapata.conf need to be opposite values. Look at the wiki, I am sure there are some examples there. Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P - Segmentation fault - Help!
On January 7, 2005 07:20 am, Adam Goryachev wrote: While I agree with you completely with your comments on HTML posting and repeating the exact same information over and over, your advice on configuration is dead wrong. zaptel.conf fxoks=1-2 fxsks=3-4 zapata.conf [channels] signalling=fxo_ks channel=1-2 signalling=fxs_ks channel=3-4 You have your fxo/fxs confused. zaptel.conf and zapata.conf need to be opposite values. They do?? my zaptel.conf for a TDM430P: fxols=1-3 my zapata.conf for the same: signalling=fxo_ls And his zapata.conf and zaptel.conf look perfectly fine for a 2FXS/2FXO TDM400P (in that channel order). Please, if you're going to give advice on the list at least make an attempt to ensure it's accurate. Cesar - make sure you have /dev/zap and all the files that go with it (i.e. make sure you ran make install in the zaptel directory. Also if you'd tell us what version of Asterisk you're running it would help a lot. Also, Cesar, make sure that you have run ztcfg -vvv and make sure the output is what you're expecting before running asterisk. ztcfg sets up the card so that Asterisk can see it. -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] Sipura SPA-1001 and Tivo Series 1
Yep check out the new generation of set top boxes - all ip based. eg www.akimbo.com just launched at CES yesterday, both Ethernet cat 5 and wireless connections. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Friday, January 07, 2005 7:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Sipura SPA-1001 and Tivo Series 1 On Thu, 6 Jan 2005 23:50:40 -0500, David Ishmael wrote: What about when users switch to 100% VoIP? I've been considering getting DirecTV with the HD PVR and I've heard it can't use broadband, in a case like that I would have to route a modem call through VoIP (or is there a better way I'm just not seeing). -Dave I think you'll find that this is much like dealing with fax. Painfull and prone to failure. Since Tivo can be a network aware device that's the more reliable route. FWIW, many series 1 Tivo units blew up their on-boards modems. This is one of the reasons that the TivoNet card to be made. Once the modem was dead you could add a net card or pay Tivo to exchange yours for a remanufactured unit. There is also a wireless TivoNet card which saves you running Cat 5 into your TV room if that's a benefit. Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Thursday, January 06, 2005 11:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Erik Espinoza Subject: Re: [Asterisk-Users] Sipura SPA-1001 and Tivo Series 1 Get the TivoNet card from http://www.9thtee.com/turbonet.htm. I have one in my series 1 Tivo. It's easy to install, and works great. It'll be a lot less hassle then trying to make Tivo use an analog line through your * server. If you consider the cost the FXS port it's cheaper than going through * anyway. Michael On Thu, 6 Jan 2005 19:46:48 -0800, Erik Espinoza wrote: Most digital devices such as modems, fax machines and tivo's can not be used without a lot of changes on VoIP. I've seen success with TiVo when you use a special code to kick it down to 14.4 kbps and use g711ulaw as the codec. I think your best bet is to try to eBay the custom nic for the TiVo series 1. Erik On Thu, 6 Jan 2005 20:39:45 -0500, David Ishmael [EMAIL PROTECTED] wrote: Hi everyone, I just got a Sipura SPA-1001 and have connected my Tivo Series 1 (yes its old). When I do a test call with Tivo, the call always fails (it seems to dial the number but never connects). I can pick up the phone line and hear the Tivo talking. I've tried looking around for anything special I need to do but its still not working. I can connect a phone to the SPA-1001 and can make outgoing calls just fine. I even tried calling the Tivo number and can hear the modem pick up. Has anyone done this? Any help would be greatly appreciated. Thanks, Dave ___ 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] NIC irq load balancing
2. NetGear Switch - I'm using FS-526T Switch, which has 24 10/100 ports and 2 Gb sorts. I want to know if this kind of general purpose switch is not suitable for voip. If so, could you recommand one? I've been doing network performance assessments for corporate clients in 40+ states since 1993, and we see an absolute ton of supposedly knowledgable engineers deploying switches from every major manufacturer. One item they just never address is making sure a server's interface matches the switch's interface settings. Over 90% of the time they tend to let the switch and serve auto-negotiate the speed and duplex settings. Most nic card vendors and most switch vendors get the negotiated speed correct (that's an easy one to do), but about 50% of the time the negotiated duplex setting is wrong. (Eg, the switch will negotiate half duplex while the server thinks he's in full duplex.) Under any reasonable load, the interface will cause damaged packets, dropped packets, etc. We've actually tested many of these and seen 100 meg interfaces maxing out at less then 1 meg throughput, for an absolute fact. Part of the negotiation problem is until recently there have been no industry standards as to how duplex settings should be negotiated. So, with every reboot and/or interface interruption, the negotiated duplex settings will be wrong about 50% of the time. Very few tech's actually have the skills/knowledge to see the mismatch. The only reasonable way to solve that issue is to lock both interfaces (the switch interface and the server nic) at full duplex. Since the FS-526T is a managed switch, if you lock the interface (and the server) to 100 meg full duplex it will work just fine. If you don't lock both interfaces, your actual throughput (and voip quality) is totally up for grabs. (Gig interfaces are always full duplex.) It also seems the majority of sys admins don't have a clue how to look at their systems to see what the nic interface has negotiated. For RH systems, take a look at the output from dmesg. Different distros will have different ways to look at (and set) the duplex setting. The duplex mismatch will have an increasingly negative impact with greater load/throughput. So, if your implementation is a home/soho system, duplex will seldom be an issue; however, if your implementation is within a larger corporate network, duplex will have a very serious impact. ___ 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] TE410P problem (Looping UP Span 1...)
Scott Stingel wrote: Sid- Try connecting one port to another. Note that one of the ports must be set up as cpe and the other as net in zapata.conf when you loop them together like this. A suitable crossover cable for testing can be constructed by cutting up a CAT 5 cable, and connecting: Pin 1 -- Pin 4 on the other end Pin 2 -- Pin 5 Pin 4 -- Pin 1 Pin 5 -- Pin 2 You should get green's on both the connected channels if your zaptel and zapata configurations are ok, and if you've run both modprobe and ztcfg as documented. Good luck Scott Stingel President EVT, Inc. www.evtmedia.com Sid wrote: Hi list, We have been trying to configure a Quad Span T1 card in a system running RH9. We have followed the instructions in the Wiki and searched the mailing lists, but so far havent got any success. Cable is connected to the first span, and module is loaded. Without loading the module the LED glows in red colour, but the moment we load module, it goes off. (No red or green) . We ran zttool and tried to run a loop test, but zttool simply hung with the message 'Looping UP Span 1...'. We had to terminate zttool with 'kill'. Here is the output of the lsmod command. Can someone shed some light on this? Thanks, -Sid Module Size Used byNot tainted wcusb 20128 0 (unused) wct4xxp54272 0 (unused) zaptel182432 0 [wcusb wct4xxp] tail -f /var/log/messages Jan 6 14:54:32 localhost kernel: TE410P: Launching card: 0 Jan 6 14:54:32 localhost kernel: TE410P: Setting up global serial parameters Jan 6 14:54:32 localhost kernel: Found a Wildcard: Wildcard TE410P-Xilinx Jan 6 14:54:32 localhost kernel: usb.c: registered new driver wcusb Jan 6 14:54:32 localhost kernel: Wildcard USB FXS Interface driver registered Jan 6 14:54:33 localhost kernel: Registered tone zone 0 (United States / North America) Jan 6 14:54:33 localhost kernel: TE410P: Span 1 configured for ESF/B8ZS Jan 6 14:54:33 localhost zaptel: Running ztcfg: succeeded Jan 6 14:55:07 localhost kernel: TE410P: Span 1 configured for ESF/B8ZS Jan 6 14:55:07 localhost kernel: Registered tone zone 0 (United States / North America) Do you Yahoo!? Yahoo! Mail http://us.rd.yahoo.com/mail_us/taglines/virus/*http://promotions.yahoo.com/new_mail/static/protection.html - Helps protect you from nasty viruses. ___ 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 Try this, Start the linux box unload zaptel driver by running modprobe -r wct4xxp and modprobe -r zaptel at this point you should see running red led (1 at the time) on all 4 ports Config your zaptel.conf with appropriate span ( say 2) the ex below is for EUROISDN span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 span=2,1,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 --- Now reload the driver by modprobe wct4xxp ( no need to run modprobe zaptel) and run ztcfg - and check channels numbering and status you should see red blinking light on span 1, and 2 now connect a cross cable like recommanded by Scott Stingle start asterisk with the appropriate zapata.conf you shouls see green llight if your config is correct Guck Jack ___ 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] Sipura SPA-1001 and Tivo Series 1
On Thu, Jan 06, 2005 at 11:50:40PM -0500, David Ishmael said: What about when users switch to 100% VoIP? I've been considering getting DirecTV with the HD PVR and I've heard it can't use broadband, in a case like that I would have to route a modem call through VoIP (or is there a better way I'm just not seeing). I've thought about this a little... It would be interesting to see if you could setup an spa2000 with a dialplan that calls another modem on the second port, and fake the PPP session. Maybe, just maybe, with the call being local to the device you can get it to work. Or some kind of T38 type solution... (BTW, your mail client doesn't quote properly. If running outlook, you can install quotefix to fix it.) ___ 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] x100p to X-lite works but x-lite to x-lite not (can not transmit audio)
Hello People, I am a newbie asterisk and happy user, i have configured a x100p card and everything works nice, i can forward incoming connections to a x-lite software client and works out of the box, However when i try to make a connection between two x-lite clients then no audio is transmited, i have followed the instructions on voip-info.org, the tutorials on onlamp and i have read some instructions on the net, and i still have not found the answer, in conclusion: I have two x-lite clients, that can call each other, connection is stablished but no audio is transmited, i follow the recomendations: 1. Install the iblc and spx registry patch (Windows 2K) 2. Work only with the alaw codec 3. Disable silence suppresion. but i still get: RFC3389 support incomplete. Turn off on client if possible RFC3389: 5 bytes, level 0... RFC3389: 5 bytes, level 0... The above message also is showing when the call is comming from a zap defice and the application Dial (Zap, SIP/313) is executed (without the RFC3389: 5 bytes, level 0...) but it works this way. I run asterisk from the command line as user asterisk like this: asterisk -vgcd This is my sip.conf: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=all ; Allow all codecs context = bogon-calls ; Send SIP callers that we don't know about here [312] type=friend username=312 secret=123456 host=dynamic disallow=all allow=alaw context=from-sip [313] type=friend username=313 secret=123456 host=dynamic disallow=all allow=alaw context=from-sip The extensions.conf: [from-sip] exten = 312,1,Dial(SIP/312,10) exten = 312,2,Voicemail(u312) exten = 312,102,Voicemail(b312) exten = 312,103,Hangup exten = 313,1,Dial(SIP/313,10) exten = 313,2,Voicemail(u313) exten = 313,102,Voicemail(b313) exten = 313,103,Hangup Voicemail works, but i can not leave a message from a sip phone: an 7 08:25:32 WARNING[393234]: app.c:615 ast_play_and_record: No audio available on SIP/313-47b0?? -- User hung up Urgent handler but i can do that from a zap device. I use asterisk debian's packages from testing. ii asterisk 1.0.2-2Open Source Private Branch Exchange (PBX) ii asterisk-doc 1.0.2-2Documentation for asterisk ii asterisk-sound 1.0.2-2Sound files for asterisk I like to have the x-lite clients working, any help will be apreciated. Thanks you very much for your time. -- Nestor A. Diaz LizarazoTel. +57.1.6005490 Ingeniero de Sistemas y Comp.Cel. 315 8190760 [EMAIL PROTECTED] http://soporte.tiendalinux.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] Confrences..kinda
Chris wrote: Hey all, Is there any software or something out there that anyone knows of that will allow me to have a conference in asterisk (or possibly not if you know another solution) where I can see who is talking at the time? Kinda like teamspeak or ventrillo. I'm not getting my hopes up, but any help would be much appreciated thanks everyone! -Chris MeetMe? http://www.voip-info.org/ -- see the wiki -- Andrew Thompson http://aktzero.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] How do I get version 1.x from theDigium CVS or elsewhere?
Anyone help me, I've looked at the Wiki and cant see anything ___ 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] fax e-mail spandsp
I'm trying to install spandsp But when I try to patch the Makefile it gives this error [EMAIL PROTECTED] apps]# patch apps_makefile.patch patching file Makefile Reversed (or previously applied) patch detected! Assume -R? [n] y Hunk #1 succeeded at 41 (offset -6 lines). Hunk #2 FAILED at 67. is it ok to go on ___ 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 do I get version 1.x from theDigium CVS orelsewhere?
To get the current stable release, issue the following command: # cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds http://www.asterisk.org/index.php?menu=download -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of John Middleton Sent: Friday, January 07, 2005 7:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How do I get version 1.x from theDigium CVS orelsewhere? Anyone help me, I've looked at the Wiki and cant see anything ___ 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] fax e-mail spandsp
On Fri, 2005-01-07 at 16:07 +0200, Altus Snyman wrote: I'm trying to install spandsp But when I try to patch the Makefile it gives this error [EMAIL PROTECTED] apps]# patch apps_makefile.patch patching file Makefile Reversed (or previously applied) patch detected! Assume -R? [n] y Hunk #1 succeeded at 41 (offset -6 lines). Hunk #2 FAILED at 67. is it ok to go on The patch required is so trivial its better to do it manually. Look at the makefile.patch and edit the Makefile accordingly. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PolyCom IP3000, gnugk and * audio problems
Current setup: Polycom IP3000 - gnugk - asterisk - Cisco 7940 Asterisk and gnugk are on 10.20.98.6 IP3000 is H.323, using G.711 (10.20.98.2) 7940 is SIP, using g711ulaw (10.20.98.3) I've been asterisk for a while now, only using SIP devices. I'm happy with that side of things, but I've not used H.323 before this week, in trying to get the IP3000 to work. * is using the chan_h323 driver and I've got call routing working in both directions, so both phones can call each other. However I'm getting no voice data between them, just silence. I'm using the asterisk packages from Debian testing (1.0.2). gnugk is also from Debian testing, default config (2.2.0) Doing tcpdump on the asterisk server shows the 7940 sending a lot of UDP data to the server, but no other data. h323.conf: [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw gatekeeper=10.20.98.1 alias=asterisk context=staff [asterisk] type=h323 prefix=0,1,2,3,4,5,6,7,8,9 context=staff [777] type=user host=10.20.98.2 context=staff I don't really know enough about how h.323 works to go much further. I've enabled h.323 trace 9 and h.323 debug from the asterisk console but I get no output. gnugk -ttt gives me a lot of output while the call is being set up (until I hit Answer) but then shows nothing until hangup. I don't see any mention of codecs in the output (don't know if I should). Can someone please give me a pointer on where to look next, as I've exhausted all my ideas. One thing I've considered doing is installing the chan_oh323 driver, but I'd prefer to exhaust my options with chan_h323 first :) Thanks, Gareth 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
[Asterisk-Users] International area codes (incl. mobile)
Hello everybody, does anybody knows from where I can get an list of international area codes incl. the mobile numbers? Regards Bastian ___ 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] Monitoring
Hi, I have some trouble with the Monitor() application. I start and stop it via the management interface, giving no special parameters except the channel name. What happens is: - if I specify WAV as the format, the resulting files are exactly 44 bytes big and contain nothing at all - if I specify GSM as the format, the resulting files are of size 0. I did not request mixing of the files or anything else. Any ideas why the monitoring fails? Cheers Robert Spielmann - TAL.DE Klaus Internet Service GmbH [EMAIL PROTECTED] Robertstr. 6 * D-42107 Wuppertal, Germany Tel +49 (0) 202 495-364 * Fax +49 (0) 202 / 495-399 ___ 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] fax e-mail spandsp
I'm trying to install spandsp But when I try to patch the Makefile it gives this error [EMAIL PROTECTED] apps]# patch apps_makefile.patch patching file Makefile Reversed (or previously applied) patch detected! Assume -R? [n] y Hunk #1 succeeded at 41 (offset -6 lines). Hunk #2 FAILED at 67. is it ok to go on Since you did not mention which * release you're using (cvs head verses v1.0 stable), I'll assume cvs head. I'd have to guess that because there has been a substantial number code changes to cvs head, Steve probably needs to update the patch to match the cvs head code. If you look very close at the apps_makefile.patch, you can probably figure out where each of the patch items belong in the Makefile. ___ 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] Queue app following dialplan
Joe Dennick wrote: Yeah, set the queue timeout to be about 1 second less than the voicemail timeout (ya know, where you say Dial(SIP/, 15)). That way the queue times out the agent before the dialplan goes to voicemail. The more reasonable solution is to just put the agent's direct path (SIP/) into your queue's agent list, rather than a Local channel that dials out through their normal extension dialing path. If I add a line like this: member = SIP/3044, can I still get login/logoff functionality? We need agent login/logff functionality AND for calls to not goto voicemail. Example extensions.conf; 3044 is both an agent that logs in/off and receives regular calls: exten = 3044,1,Dial(SIP/3044,30) exten = 3044,2,Voicemail([EMAIL PROTECTED]) exten = 3044,102,Voicemail([EMAIL PROTECTED]) If 3044 is currently talking to anyone (be it queue call or a direct call), if anyone else calls his extension (be it queue call or a direct call) it will go directly to his voicemail (Pri 102). It needs to be so that if an outside call comes in, it follows the dialplan accordingly, but if a queue call comes in and his phone is truly busy, it needs to stop following the dialplan and go try another agent. -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] Problem with call pickup
Title: Message
I know that my
phone displays *8 because I dailed that.
But it's
definitly not what I would want, or most other people.
Any other
ordinary PBX would show the CID of the caller, but because this is a SIP-based
system we get this problem.
I was thinking
more in line of an alternate call-pickup procedure to realize this
option.
My idea would
be:
exten =
*8,1,SetVar(PICKEXT=${CALLERIDNUM})
exten =
*8,2,HangUp
exten =
*8,3, Here come the lines that will deflect the call to
$PICKEXT
Why
deflect?, well when a call is deflected CID information is also
transferred.
The
effectwould bethat when dialing *8the connection would be
closed, but immediatly after that your phone will ring showing you all the
information you need.. even before pickup.
You could
call this function remote deflecting???
This
function does not exist in * as far as I know, but perhaps there is some
work-around to this???
Anyone?!??!
I have configured
call pickup, and this works fine.
Although there are 2
problems, perhaps anyone would know a solution to this;
- When I pickup a
call from another set, the *8 code keeps being displayed in my screen (Snom
220).
I would like
it to show the phonenumber of the person calling me.
- When a caller that
I've answered through Call-Pickup disconnects, my phone does not close the
connection but acts like there is still someone on the otherside. (Logging shows
dat de Zap/channel has cleared, but not the SIP/channel)
I use Asterisk
1.0.2-BRIstuffed-0.2.0-RC2
Any help would be
greatly appreciated...
Ramon
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Re: [Asterisk-Users] Twin Cities Asterisk meeting still on for Saturday?
Yes. Quoting Roger Hanson [EMAIL PROTECTED]: Is the meeting still on for Saturday 1/8/05? 11:30am at 2375 University Av W STE120, Saint Paul. ___ 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] softphones
Hi, can someone tell be about some good and free softphones? Are they easy to use by non-tecnical users? Can someone share their experience about the implementation of VoIP softphones in a company? because usualy people dont want to make changes in the way they work I would like to know a way to convince peaple in my company to use them. Thanks Joao Pereira ___ 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] International area codes (incl. mobile)
Hello everybody, does anybody knows from where I can get an list of international area codes incl. the mobile numbers? Have you tried google ? http://www.google.com.au/search?hl=enq=international+dialing+codes ___ 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] Queue app following dialplan
Matthew Boehm wrote: If I add a line like this: member = SIP/3044, can I still get login/logoff functionality? We need agent login/logff functionality AND for calls to not goto voicemail. No, I was suggesting using SIP/3044 in agents.conf, not in queues.conf. If you put it into queues.conf, that channel will be dedicated to the queue app at all times. Word of warning, though: I don't use chan_agent, never have. All my queues are configured using dynamic members (AddQueueMember/RemoveQueueMember), so what I suggested above is based solely on the docs I've read. If you want a SetGroup/CheckGroup based solution, email me off-list; I have a patch for app_queue that causes it to assign groups to channels it creates and to check group counts before calling agents. This works well for us, it allows us to very easily control when the app will send a call to an agent and when it will consider them busy. I doubt this patch will ever go into CVS, though, unless I make it part of my new version of app_queue that'll be available in a few weeks. ___ 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: kind of Urgent (Fedora Core 3 Asterisk)
/SNIP/ On Thu, 2005-01-06 at 12:00 -0600, asterisk-users- [EMAIL PROTECTED] wrote: Andy Burns wrote: Shoval Tomer wrote: Can anyone comment why shouldn't we use FC 3 for an * production system? when I tried the X100P drivers on FC3 I had problems with udev, the workaround didn't work for me, maybe things have improved since ... /SNIP/ We are replacing all our Fedora Core2 Systems with Redhat Enterprise Linux. We found following problems, to list a few. 1)SMP Integration is poor. We could not use MOH as the drivers do not have compatibility for FC2 2)Oh323 compiles clean on RHEL and it hangs when compiling on FC2 3)Mouse and Monitor drivers are not stable on FC2. We had serious issues with LCD Panels and Trackballs 4)Slower than many other distros 5)Reboots are not clean 6)Updates are buggy and many a times fail. I do not recommend using Fedora for a production environment. Seshu Kanuri NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Test2
Robert Webb Posted: -Original Message- Sent: Thursday, January 06, 2005 3:53 PM Subject: [Asterisk-Users] Test2 Sorry for all the tests. Please excuse. /SNIP/ What are you trying to test? The list's patience? Seshu NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0.2 - Unable to allocate channel structure
Hi, This morning I had some failed calls. On the console (and in the log) I saw the error Unable to allocate channel structure. Before I restarted the process, I checked it's memory usage in ps and glanced at my free memory in top. Asterisk was using a normal ammount of memory, about 40M. I don't think this was a system limit. This was running Asterisk v1.0.2. Below is an excerpt of my messages log as well as the output of ps and top, if it helps. Has anyone seen this sort of error before? Any ideas what could be causing it? The changelog for 1.0.3 doesn't list anything related to memory or resource allocation.. Anyone know if there was any work done to ast_channel_alloc() or related functions? Thanks. - Eric Jan 7 07:24:50 WARNING[163850]: Unable to allocate channel structure Jan 7 07:24:50 WARNING[163850]: Unable to start PBX on channel 0/11, span 1 Jan 7 07:24:50 WARNING[163850]: Call specified, but not found? Jan 7 07:24:50 WARNING[163850]: Hangup on bad channel 0/11 on span 1 Jan 7 07:24:51 WARNING[180235]: Unable to allocate channel structure Jan 7 07:24:51 WARNING[180235]: Unable to start PBX on channel 0/1, span 2 Jan 7 07:24:51 WARNING[180235]: Call specified, but not found? Jan 7 07:24:51 WARNING[180235]: Hangup on bad channel 0/1 on span 2 Jan 7 07:24:54 WARNING[163850]: Call specified, but not found? Jan 7 07:24:54 WARNING[163850]: Hangup on bad channel 0/11 on span 1 Jan 7 07:24:55 WARNING[180235]: Call specified, but not found? Jan 7 07:24:55 WARNING[180235]: Hangup on bad channel 0/1 on span 2 Jan 7 08:20:24 WARNING[81925]: Unable to allocate channel structure Jan 7 08:20:24 NOTICE[81925]: Unable to create/find channel Jan 7 08:20:42 WARNING[81925]: Unable to allocate channel structure Jan 7 08:20:42 NOTICE[81925]: Unable to create/find channel Jan 7 08:21:03 WARNING[81925]: Unable to allocate channel structure Jan 7 08:21:03 NOTICE[81925]: Unable to create/find channel Jan 7 08:22:43 WARNING[81925]: Unable to allocate channel structure Jan 7 08:22:43 NOTICE[81925]: Unable to create/find channel Jan 7 08:23:01 WARNING[81925]: Unable to allocate channel structure Jan 7 08:23:01 NOTICE[81925]: Unable to create/find channel Jan 7 08:23:23 WARNING[81925]: Unable to allocate channel structure Jan 7 08:23:23 NOTICE[81925]: Unable to create/find channel Jan 7 08:26:09 WARNING[81925]: Unable to allocate channel structure Jan 7 08:26:09 NOTICE[81925]: Unable to create/find channel Jan 7 08:26:17 WARNING[81925]: Unable to allocate channel structure Jan 7 08:26:17 NOTICE[81925]: Unable to create/find channel Jan 7 08:28:23 WARNING[81925]: Unable to allocate channel structure Jan 7 08:28:23 NOTICE[81925]: Unable to create/find channel Jan 7 08:28:29 WARNING[81925]: Maximum retries exceeded on call 1636b9b523c778f [EMAIL PROTECTED] for seqno 102 (Non-critical Response) Jan 7 08:28:30 WARNING[163850]: Unable to allocate channel structure Jan 7 08:28:30 WARNING[163850]: Unable to start PBX on channel 0/12, span 1 Jan 7 08:28:31 WARNING[163850]: Call specified, but not found? Jan 7 08:28:31 WARNING[163850]: Hangup on bad channel 0/12 on span 1 Jan 7 08:28:31 WARNING[180235]: Unable to allocate channel structure Jan 7 08:28:31 WARNING[180235]: Unable to start PBX on channel 0/2, span 2 Jan 7 08:28:31 WARNING[180235]: Call specified, but not found? Jan 7 08:28:31 WARNING[180235]: Hangup on bad channel 0/2 on span 2 Jan 7 08:28:34 WARNING[163850]: Call specified, but not found? Jan 7 08:28:34 WARNING[163850]: Hangup on bad channel 0/12 on span 1 Jan 7 08:28:35 WARNING[180235]: Call specified, but not found? Jan 7 08:28:35 WARNING[180235]: Hangup on bad channel 0/2 on span 2 Jan 7 08:29:18 WARNING[81925]: Unable to allocate channel structure Jan 7 08:29:18 NOTICE[81925]: Unable to create/find channel Jan 7 08:29:30 WARNING[81925]: Unable to allocate channel structure Jan 7 08:29:30 NOTICE[81925]: Unable to create/find channel ([EMAIL PROTECTED]) ~ # ps aux USER PID %CPU %MEM VSZ RSS TTY STAT START TIME COMMAND root 1 0.0 0.1 1272 476 ?S 2004 0:06 init [3] root 2 0.0 0.0 00 ?SW2004 0:00 [keventd] root 3 0.0 0.0 00 ?SWN 2004 0:00 [ksoftirqd_CPU0] root 4 0.0 0.0 00 ?SW2004 0:00 [kswapd] root 5 0.0 0.0 00 ?SW2004 0:00 [bdflush] root 6 0.0 0.0 00 ?SW2004 0:00 [kupdated] root 7 0.0 0.0 00 ?SW2004 0:00 [kjournald] root20 0.0 0.0 00 ?SW2004 0:00 [kjournald] root21 0.0 0.0 00 ?SW2004 0:00 [kjournald] root22 0.0 0.0 00 ?SW2004 0:02 [kjournald] root23 0.0 0.0 00 ?SW2004 0:00 [kjournald] root24 0.0 0.0 00 ?SW2004 0:00 [kjournald] root37 0.0 0.2 1324 556 ?
Re: [Asterisk-Users] Queue app following dialplan
On Fri, 2005-01-07 at 08:08 -0700, Kevin P. Fleming wrote: Matthew Boehm wrote: If I add a line like this: member = SIP/3044, can I still get login/logoff functionality? We need agent login/logff functionality AND for calls to not goto voicemail. No, I was suggesting using SIP/3044 in agents.conf, not in queues.conf. If you put it into queues.conf, that channel will be dedicated to the queue app at all times. Word of warning, though: I don't use chan_agent, never have. All my queues are configured using dynamic members (AddQueueMember/RemoveQueueMember), so what I suggested above is based solely on the docs I've read. If you want a SetGroup/CheckGroup based solution, email me off-list; I have a patch for app_queue that causes it to assign groups to channels it creates and to check group counts before calling agents. This works well for us, it allows us to very easily control when the app will send a call to an agent and when it will consider them busy. I doubt this patch will ever go into CVS, though, unless I make it part of my new version of app_queue that'll be available in a few weeks. I would like to know more about your solution. What do you mean about having the agents in queue.conf or agents.conf? I thought they have to be both places? ___ 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 -- 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
[Asterisk-Users] oh323 driver installation - It works now
Joao, Thanks for sending the Installation tips as pasted below. It works. Seshu -- Get oh323 fromhttp://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gzGet pwlib fromhttp://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gzGet asterisk-oh323 fromhttp://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.6.5.tar.gzUntar the files#tar zxvf openh323-Janus_patch4-src-tar.gz#tar zxvf pwlib-Janus_patch4-src-tar.gz#tar zxvf asterisk-oh323-0.6.5.tar.gz#tar zxvf asterisk-1.0.3.tar.gzInstall Pwlib#cd pwlib#./configure make clean make opt make install ldconfigPatch and Install OpenH323#cd openh323#patch -p1 ../asterisk-oh323-0.6.5/openh323_1.13.5-make.patch#./configure make clean make opt make install ldconfigAsterisk#cd asterisk-1.0.3#make make install make samplesAsterisk-oh323#cd asterisk-oh323-0.6.5Edit the Makefile#make make install ldconfig From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of João AmaroSent: Thursday, January 06, 2005 7:37 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] asterisk - oh323 driver Hi allI've managed to get chan-oh323-0.6.5 working with asterisk-1.0.3I've downloaded all the files from www.inaccessnetworks.com pwlib + pwlib-janus patch openh323 + openh323-janus patch chan-oh323 0.6.5Don't forget to apply the chan-oh323 patch to openh323 before compiling.Hope it helpsRegardsJoão Amaro NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] International area codes (incl. mobile)
I can send a list, mobile is not complete but it has a lot of numbers... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de PHP Mechanic Enviado el: Viernes, 07 de Enero de 2005 11:57 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] International area codes (incl. mobile) Hello everybody, does anybody knows from where I can get an list of international area codes incl. the mobile numbers? Have you tried google ? http://www.google.com.au/search?hl=enq=international+dialing+codes ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 2005-01-06 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 2005-01-06 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500
That's what I'm about to try, I keep getting pulled off of this project to go do other things. Thanks for the input. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Thursday, January 06, 2005 5:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim Jackson wrote: Copied your sip.conf and changed the settings and I'm getting the exact same error. I'm also running 1.3.4 of the SIP app for the IP500. Someone has already pointed out that you might have ran into a network problem. What's the network setup between phone and the server? Asterisk CVS-v1-0-01/06/05-00:11:36 built by [EMAIL PROTECTED] on a i686 I was unable to use Asterisk from latest CVS, I am using version from 12/02 CVS. I was getting authorization failed in CLI, and phone could not make calls with CVS-latest Asterisk. Might be something similar in your setup? Just copy /usr/src/asterisk from old server and try make install.. Please, someone, comment on latest changes in CVS for SIP configurations? Might enforced md5 passwords etc? Or anything like that? context=noawnser A typo, right? Andrei ___ 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] Queue app following dialplan
-Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Friday, January 07, 2005 3:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Queue app following dialplan The more reasonable solution is to just put the agent's direct path (SIP/) into your queue's agent list, rather than a Local channel that dials out through their normal extension dialing path. Another possible scenario is to specify the context to call the agent when using AgentCallBackLogin. This way you can have one set of behaviors for reaching an agent at an extension and another set for simply reaching the extension outside of an ACD context. This is how we have it setup and it seems to work pretty well. Hope this helps, Robert Jackson ___ 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] Setting up Polycom IP 500 with *
I am in the process of setting up an * system using Polycom IP 500's. I don't want to spend time setting a ftp server for application and configuration files at the moment, just want to use the web interface to the Polycoms. DCHP works OK and IP is obtained correctly. Polycom fails to load .cfg file and holts. I have read the 143 page admin user guide a couple of times...and I missing somthing? Adrian Walker [EMAIL PROTECTED] === This email has been scanned for Virus infection by MessageLabs For more information please contact [EMAIL PROTECTED] === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.2 - Unable to allocate channelstructure
Holy cow! Why are there so many asterisk instances running? There should only be 1. kill them all and start just 1 asterisk -Matthew - Original Message - From: Eric [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 9:35 AM Subject: [Asterisk-Users] Asterisk 1.0.2 - Unable to allocate channelstructure Hi, This morning I had some failed calls. On the console (and in the log) I saw the error Unable to allocate channel structure. Before I restarted the process, I checked it's memory usage in ps and glanced at my free memory in top. Asterisk was using a normal ammount of memory, about 40M. I don't think this was a system limit. This was running Asterisk v1.0.2. Below is an excerpt of my messages log as well as the output of ps and top, if it helps. Has anyone seen this sort of error before? Any ideas what could be causing it? The changelog for 1.0.3 doesn't list anything related to memory or resource allocation.. Anyone know if there was any work done to ast_channel_alloc() or related functions? Thanks. - Eric Jan 7 07:24:50 WARNING[163850]: Unable to allocate channel structure Jan 7 07:24:50 WARNING[163850]: Unable to start PBX on channel 0/11, span 1 Jan 7 07:24:50 WARNING[163850]: Call specified, but not found? Jan 7 07:24:50 WARNING[163850]: Hangup on bad channel 0/11 on span 1 Jan 7 07:24:51 WARNING[180235]: Unable to allocate channel structure Jan 7 07:24:51 WARNING[180235]: Unable to start PBX on channel 0/1, span 2 Jan 7 07:24:51 WARNING[180235]: Call specified, but not found? Jan 7 07:24:51 WARNING[180235]: Hangup on bad channel 0/1 on span 2 Jan 7 07:24:54 WARNING[163850]: Call specified, but not found? Jan 7 07:24:54 WARNING[163850]: Hangup on bad channel 0/11 on span 1 Jan 7 07:24:55 WARNING[180235]: Call specified, but not found? Jan 7 07:24:55 WARNING[180235]: Hangup on bad channel 0/1 on span 2 Jan 7 08:20:24 WARNING[81925]: Unable to allocate channel structure Jan 7 08:20:24 NOTICE[81925]: Unable to create/find channel Jan 7 08:20:42 WARNING[81925]: Unable to allocate channel structure Jan 7 08:20:42 NOTICE[81925]: Unable to create/find channel Jan 7 08:21:03 WARNING[81925]: Unable to allocate channel structure Jan 7 08:21:03 NOTICE[81925]: Unable to create/find channel Jan 7 08:22:43 WARNING[81925]: Unable to allocate channel structure Jan 7 08:22:43 NOTICE[81925]: Unable to create/find channel Jan 7 08:23:01 WARNING[81925]: Unable to allocate channel structure Jan 7 08:23:01 NOTICE[81925]: Unable to create/find channel Jan 7 08:23:23 WARNING[81925]: Unable to allocate channel structure Jan 7 08:23:23 NOTICE[81925]: Unable to create/find channel Jan 7 08:26:09 WARNING[81925]: Unable to allocate channel structure Jan 7 08:26:09 NOTICE[81925]: Unable to create/find channel Jan 7 08:26:17 WARNING[81925]: Unable to allocate channel structure Jan 7 08:26:17 NOTICE[81925]: Unable to create/find channel Jan 7 08:28:23 WARNING[81925]: Unable to allocate channel structure Jan 7 08:28:23 NOTICE[81925]: Unable to create/find channel Jan 7 08:28:29 WARNING[81925]: Maximum retries exceeded on call 1636b9b523c778f [EMAIL PROTECTED] for seqno 102 (Non-critical Response) Jan 7 08:28:30 WARNING[163850]: Unable to allocate channel structure Jan 7 08:28:30 WARNING[163850]: Unable to start PBX on channel 0/12, span 1 Jan 7 08:28:31 WARNING[163850]: Call specified, but not found? Jan 7 08:28:31 WARNING[163850]: Hangup on bad channel 0/12 on span 1 Jan 7 08:28:31 WARNING[180235]: Unable to allocate channel structure Jan 7 08:28:31 WARNING[180235]: Unable to start PBX on channel 0/2, span 2 Jan 7 08:28:31 WARNING[180235]: Call specified, but not found? Jan 7 08:28:31 WARNING[180235]: Hangup on bad channel 0/2 on span 2 Jan 7 08:28:34 WARNING[163850]: Call specified, but not found? Jan 7 08:28:34 WARNING[163850]: Hangup on bad channel 0/12 on span 1 Jan 7 08:28:35 WARNING[180235]: Call specified, but not found? Jan 7 08:28:35 WARNING[180235]: Hangup on bad channel 0/2 on span 2 Jan 7 08:29:18 WARNING[81925]: Unable to allocate channel structure Jan 7 08:29:18 NOTICE[81925]: Unable to create/find channel Jan 7 08:29:30 WARNING[81925]: Unable to allocate channel structure Jan 7 08:29:30 NOTICE[81925]: Unable to create/find channel ([EMAIL PROTECTED]) ~ # ps aux USER PID %CPU %MEM VSZ RSS TTY STAT START TIME COMMAND root 1 0.0 0.1 1272 476 ?S 2004 0:06 init [3] root 2 0.0 0.0 00 ?SW2004 0:00 [keventd] root 3 0.0 0.0 00 ?SWN 2004 0:00 [ksoftirqd_CPU0] root 4 0.0 0.0 00 ?SW2004 0:00 [kswapd] root 5 0.0 0.0 00 ?SW2004 0:00 [bdflush] root 6 0.0 0.0 00 ?SW2004 0:00 [kupdated] root 7 0.0 0.0 00 ?SW2004 0:00
[Asterisk-Users] Moderator on vacation?
OK, I'm trying to send an email to the list the contiune a thread which describes a problem I'm having. This particualy email I wish to send contains an ls -l describing my problem (too many open files) and is apparently too large to be considered a normal post, so I get a message that it's being held until a moderator can view it. Fine. So now I get an autoresponder from the moderator telling me he's on vacation until someone near the end of the month. Seriously, what gives. Can we make some changes here? I'd like to post my findings and get some help. - Eric ___ 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] softphones
Hi I tried Xten, its very good, because it can stay in the taskbar (next to the clock) and start when windows starts, and is allways ready to receive calls. Maybe it s the best way to introduce VoIP to my company workers But theres a feature that s missing (or I couldnt find), there s no way to connect this softphone with the adress book. I think this feature is very important, because everybody has allready a big adressbook with the friends emails, and we dont want to have this adressbook replicated (windows adressbook and Xlite phonebook). Thanks Joao - Original Message - From: Walter Carter [EMAIL PROTECTED] To: 'Joao Pereira' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Friday, January 07, 2005 3:17 PM Subject: RE: [Serusers] softphones Try Xten: http://www.xten.com/index.php?menu=productssmenu=xlite Regards, WSC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira Sent: Friday, January 07, 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: [Serusers] softphones Hi, can someone tell be about some good and free softphones? Are they easy to use by non-tecnical users? Can someone share their experience about the implementation of VoIP softphones in a company? because usualy people dont want to make changes in the way they work I would like to know a way to convince peaple in my company to use them. Thanks Joao Pereira ___ 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] spandsp and app_rxfax (alternative topic: t38modem)
H. Did I just ask in the wrong forum, or has _nobody_ experienced image corruption using app_rxfax that was NOT due to using the wrong version of libtiff? If that's the case, then my secondary approach is going to have to be: PSTN - Asterisk + chan_h323 - t38modem + Hylafax Is there anybody that could help me with either of these solutions? A thousand thank yous in advance, Ryan VanMiddlesworth On Thursday, January 6th, I wrote: I've been pulling my hair out trying to get Asterisk to receive and decode a fax using spandsp and app_rxfax. It seems like it should be working. The fax machine on the other end connects and Asterisk reports a fax coming in. But when it's done all I have is a 2 or 3 KB TIF (see attachment). The console activity looks completely normal: -- Starting simple switch on 'Zap/3-1' -- Executing SetVar(Zap/3-1, FAXFILE=/var/spool/asterisk/fax/1105043880.0.tif) in new stack -- Executing RxFAX(Zap/3-1, /var/spool/asterisk/fax/1105043880.0.tif) in new stack -- Hungup 'Zap/3-1' And there are no errors in the log file. Here's my config: Wildcard TDM40B hardware (Zaptel) asterisk-1.0.2 spandsp-0.0.2pre6 libtiff-3.6.1 (with the fax fix patches) (also tried libtiff-3.6.0 and libtiff-3.5.7) I've tried multiple sending fax machines and get the same effect. Any tips on getting this setup working? I've run out of ideas. Alternately, I'd also be willing to offload the DSP processing to a HylaFAX machine using some sort of software fax driver. I tinkered with t38modem and chan_h323, but couldn't get it to do anything once the HylaFAX machine answered. So if anybody has any experience with that, I'd be interested. Thanks in advance, Ryan VanMiddlesworth ___ 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] Setting up Polycom IP 500 with *
The FTP server option works very well so you should do it when get time. The phone has an option where you tell it to load via FTP, believe it is the server config. To get to it, reboot the phone and enter setup on the phone, not the web. Remove the settings if you want no configs from network and your settings via browser should work if correct. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Walker Sent: Friday, January 07, 2005 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Setting up Polycom IP 500 with * I am in the process of setting up an * system using Polycom IP 500's. I don't want to spend time setting a ftp server for application and configuration files at the moment, just want to use the web interface to the Polycoms. DCHP works OK and IP is obtained correctly. Polycom fails to load .cfg file and holts. I have read the 143 page admin user guide a couple of times...and I missing somthing? Adrian Walker [EMAIL PROTECTED] === This email has been scanned for Virus infection by MessageLabs For more information please contact [EMAIL PROTECTED] === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with MySQL
Hello I am getting this error message, when i try to authenticate my users through database. Jan 7 20:28:08 WARNING[26487]: res_config_odbc.c:69 realtime_odbc: SQL Alloc Handle failed! Jan 7 20:28:08 NOTICE[26487]: chan_sip.c:7974 handle_request: Registration from 'rizwan sip:[EMAIL PROTECTED]' failed for '192.168.0.149' My conf files are: ;res_odbc.conf [test] dsn = test username = root password = pre-connect = yes ;extensions.conf [test] switch = Realtime/@realtime_ext ;extconfig.conf sipfriends = odbc,test,sip_buddies realtime_ext = odbc,test,extensions_table Can you please help me, what to do here? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Moderator on vacation?
Eric wrote: Seriously, what gives. Can we make some changes here? I'd like to post my findings and get some help. I can't get google to show me any, but there are sites that allow you to drop off large files and give you a url for retreiving them. Perhaps someone can come up with the name of one. Find a site, upload it there, post your message with info and point us at the link. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Moderator on vacation?
On January 7, 2005 11:08 am, Eric wrote: I'm trying to send an email to the list the contiune a thread which describes a problem I'm having. This particualy email I wish to send contains an ls -l describing my problem (too many open files) and is apparently too large to be considered a normal post, so I get a message that it's being held until a moderator can view it. If you got that message it means you posted to the list from an address that is not subscribed. It's a little misleading -- I've *never* had a moderator post or deny a message I've posted from a nonsubscriber address, on vacation or not. Seriously, what gives. Can we make some changes here? I'd like to post my findings and get some help. Post to the list from an address that is subscribed, like you just did here. No human intervention required. :-) -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] spandsp and app_rxfax (alternative topic: t38modem)
On January 7, 2005 11:13 am, Ryan wrote: H. Did I just ask in the wrong forum, or has _nobody_ experienced image corruption using app_rxfax that was NOT due to using the wrong version of libtiff? Seems to be correct, or at least image corruption from a really crappy fax reception. I know I've been receiving between 30-50 faxes a day with app_rxfax without issue. I also note that you posted your initial message at 4:14pm, and now, less than 24 hours later you are expecting the entire asterisk community to have received your message, parsed it in the sea of other messages to the list, had it apply to them and responded. That's just a little pretentious, don't you think? -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] Moderator on vacation?
On January 7, 2005 11:22 am, Andrew Thompson wrote: I can't get google to show me any, but there are sites that allow you to drop off large files and give you a url for retreiving them. Perhaps someone can come up with the name of one. http://pastebin.ca is what is used on the IRC channel almost exlcusively. Also its big brother, http://pastebin.com, although it is frequenty slow. :-) -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] spandsp and app_rxfax (alternative topic: t38modem)
On Fri, 7 Jan 2005, Ryan wrote: H. Did I just ask in the wrong forum, or has _nobody_ experienced image corruption using app_rxfax that was NOT due to using the wrong version of libtiff? Hello Ryan, I have. There was a discussion on this list a short while ago on howto debug frameslips. I think you could find useful information in that thread. I had the same problems using hfc cards with bristuff. (with patched zaptel drivers). When I applied Florian Zumbiehls patch the problem went away. (The link to the patch can be found in the wiki: asterisk zaphfc) Is it a possibility that there is a problem with interrupt handling in the zaptel driver for other cards as well? /Nils If that's the case, then my secondary approach is going to have to be: PSTN - Asterisk + chan_h323 - t38modem + Hylafax Is there anybody that could help me with either of these solutions? A thousand thank yous in advance, Ryan VanMiddlesworth On Thursday, January 6th, I wrote: I've been pulling my hair out trying to get Asterisk to receive and decode a fax using spandsp and app_rxfax. It seems like it should be working. The fax machine on the other end connects and Asterisk reports a fax coming in. But when it's done all I have is a 2 or 3 KB TIF (see attachment). The console activity looks completely normal: -- Starting simple switch on 'Zap/3-1' -- Executing SetVar(Zap/3-1, FAXFILE=/var/spool/asterisk/fax/1105043880.0.tif) in new stack -- Executing RxFAX(Zap/3-1, /var/spool/asterisk/fax/1105043880.0.tif) in new stack -- Hungup 'Zap/3-1' And there are no errors in the log file. Here's my config: Wildcard TDM40B hardware (Zaptel) asterisk-1.0.2 spandsp-0.0.2pre6 libtiff-3.6.1 (with the fax fix patches) (also tried libtiff-3.6.0 and libtiff-3.5.7) I've tried multiple sending fax machines and get the same effect. Any tips on getting this setup working? I've run out of ideas. Alternately, I'd also be willing to offload the DSP processing to a HylaFAX machine using some sort of software fax driver. I tinkered with t38modem and chan_h323, but couldn't get it to do anything once the HylaFAX machine answered. So if anybody has any experience with that, I'd be interested. Thanks in advance, Ryan VanMiddlesworth ___ 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 Nils Segerdahl --- Upsala Systemkonsult, UPSYS AB Telefon:(+46) (0)18 56 80 41 Glunten, 751 83 UppsalaMobil: (+46) (0)703 55 65 03 http://www.upsys.seFax: (+46) (0)18 56 80 49 --- Jan 8 Battle of New Orleans Jan 9 Fellowship reaches Lorien (LOTR) Jan 9 Plough Monday Jan 10 First meeting of United Nations General Assembly in London, 1946 Jan 10 Thomas Paine's Common Sense published, 1776 Jan 8 American Telephone and Telegraph loses antitrust case, 1982 Jan 8 Herman Hollerith patents first data processing computer, 1889 Jan 8 Justice Dept. drops IBM suit, 1982 Jan 10 First CDC 1604 delivered to Navy, 1960 --- ___ 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] Monitoring
What version of sox do you use? Lamine - Original Message - From: Robert Spielmann [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 2:40 PM Subject: [Asterisk-Users] Monitoring Hi, I have some trouble with the Monitor() application. I start and stop it via the management interface, giving no special parameters except the channel name. What happens is: - if I specify WAV as the format, the resulting files are exactly 44 bytes big and contain nothing at all - if I specify GSM as the format, the resulting files are of size 0. I did not request mixing of the files or anything else. Any ideas why the monitoring fails? Cheers Robert Spielmann - TAL.DE Klaus Internet Service GmbH [EMAIL PROTECTED] Robertstr. 6 * D-42107 Wuppertal, Germany Tel +49 (0) 202 495-364 * Fax +49 (0) 202 / 495-399 ___ 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] Multiple lines on Cisco 7960
I did set these to the correct poxy serveras well in the SIPDefault.cnf file. This is very frustrating problem, I have ready dozens of posts that refer to how to set this up and I see mto have followed all the suggestions. I had not looked at the phones settings yet, thanks for the suggestion. The setting indicate that there is no configuration on the second line it is listed as UNPROVISIONED Scott Nathan Alberti wrote: Do you have: # Proxy Server proxy1_address: x.x.x.x proxy2_address: x.x.x.x Unsure if this is required, does your phone list the correct server ? (settings | 4 | 2 | 6) Nathan. Scott Henderson wrote: I have been trying to get multiple lines on the 7960 to work for several days. i have read all the posts I can find and have run multiple sip debug and have gotten no place on this. Here are the relevant section of the config files: sip.conf [scott] type=friend host=dynamic username=scott secret=scott context=default mailbox=6101 callerid=Scott Henderson [scott1] type=friend host=dynamic username=scott1 secret=scott1 context=default mailbox=6101 callerid=Scott Henderson 1 macaddress.cnf # Line 1 line1_name: Scott line1_authname: scottline1_password: scott # Line 2 line2_name: Scott1 line2_authname: scott1 line2_password: scott1 sip debug output from resetting the phone: Sip read: REGISTER sip:192.168.17.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: CSCO/7 Contact: sip:[EMAIL PROTECTED]:5060 Content-Length: 0 Expires: 3600 10 headers, 0 lines Using latest request as basis request Sending to 192.168.17.114 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.17.114:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK61a1a63a From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=0045611f Content-Length: 0 to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms argon*CLI Sip read: REGISTER sip:192.168.17.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: CSCO/7 Contact: sip:[EMAIL PROTECTED]:5060 Authorization: Digest username=scott,realm=asterisk,uri=sip:192.168.17.13,response=7b9f392d15161ef76ae35f283e876497,nonce=0045611f,algorithm=md5 Content-Length: 0 Expires: 3600 11 headers, 0 lines Using latest request as basis request Sending to 192.168.17.114 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.17.114:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.17.114:5060;branch=z9hG4bK49f2aa87 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as00424045 Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:[EMAIL PROTECTED]:5060;expires=3600 Date: Fri, 07 Jan 2005 02:56:25 GMT Content-Length: 0 to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK29419b41 From: asterisk sip:[EMAIL PROTECTED];tag=as42c5efcf To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 (no NAT) to 192.168.17.114:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms argon*CLI Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK29419b41 From: asterisk sip:[EMAIL PROTECTED];tag=as42c5efcf To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] Date: Fri, 07 Jan 2005 02:56:26 GMT CSeq: 102 NOTIFY Content-Length: 0 8 headers, 0 lines Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' argon*CLI The result of this configuration is that I always get the first line line_1 but never the second
Re: [Asterisk-Users] Asterisk 1.0.2 - Unable to allocate channelstructure
On Friday 07 January 2005 16:04, Matthew Boehm wrote: Holy cow! Why are there so many asterisk instances running? There should only be 1. kill them all and start just 1 asterisk Do not top post, learn to trim. There is 1 process and many threads. ___ 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] Moderator on vacation?
Andrew Thompson wrote: Find a site, upload it there, post your message with info and point us at the link. And then everyone who is not involved in the thread about the OP's problem will be very thankful! To the OP: There is an obvious reason why the list does not allow posting larger than a pre-defined limit, and even if the moderator was not on vacation it still wouldn't have been let through. There are thousand(s) of subscribers to this list, many of whom have poor and/or low-speed access to their mailboxes. Forcing all of them to download your large attachment would be very disrespectful of their limitations. Andrew's response is right on target: find some other place to host your file, and send the list a link to it. We'll all be much happier :-) ___ 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] spandsp and app_rxfax (alternative topic: t38modem)
On 2005.01.07 08:13 Ryan wrote: H. Did I just ask in the wrong forum, or has _nobody_ experienced image corruption using app_rxfax that was NOT due to using the wrong version of libtiff? Oh, you can get image corruption on any non-ECM fax, and that doesn't have anything to do with anything other than data corruption over-the-wire. Since spandsp doesn't support ECM, you can get immage corruption and not have it be anything's fault except for the nature of non-ECM faxes. During fax Phase C (the part where the TIFF image data is communicated) if some data gets garbled due to whatever reason, then the image data will be messed up. If that's the case, then my secondary approach is going to have to be: PSTN - Asterisk + chan_h323 - t38modem + Hylafax Well, that would be a lovely thought... if only you could get Asterisk to talk T.38 through chan_h323, and Asterisk does not support T.38 in any way, shape, or form right now. Is there anybody that could help me with either of these solutions? As far as I'm aware, the only way to get HylaFAX working behind Asterisk is to connect a HylaFAX-controlled hardmodem either into an FXS port or a passthru span. For example: PSTN - X100P - Asterisk - SPA-2000 - analog modem - HylaFAX or T1 - TE405P - Asterisk - TE405P - T1 modem - HylaFAX The first configuration, which will generally work tolerably well with modern HylaFAX and most fax machines, is subject to a fair risk in data corruption due to the combined analog-to-digital (digital === ulaw) and digital-to-analog conversions. A mere 20ms delay in faxing can make a huge impact. The only reason that this configuration works tolerably well is due to the ECM support in modern HylaFAX and most fax machines. The ECM protocol is able to recover the corrupted data through retransmission attempts. Now, if the sender doesn't support ECM, then you're generally stuck with whatever corruption occurs (maybe none, but probably some). The second configuration seems to be quite a flawless way to do this as Asterisk is merely forwarding the already-digital signal. The downside is, of course, that it's probably not really an option unless you have a T1, a TE405P, and a T1 modem (either an Eicon Diva Server or a Patton DataFire 2977). Lee. ___ 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] Moderator on vacation?
On Friday 07 January 2005 16:08, Eric wrote: OK, I'm trying to send an email to the list the contiune a thread which describes a problem I'm having. This particualy email I wish to send contains an ls -l describing my problem (too many open files) and is apparently too large to be considered a normal post, so I get a message that it's being held until a moderator can view it. Do you really think we need to see the entire output from an ls command? If there are duplicates either at the file or directory level, then use the normal [...] syntax to show similar line have been deleted. ___ 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] Message light on 7960 or in this case no message light
I think the issue is the context specification. In this application I had two contexts in voicemail.conf that were not default. I have modified the sip.conf as suggested. Scott Nathan Alberti wrote: Ensure you have mailbox= in sip.conf, you must also make sure in voicemail.conf the mailbox declarations are under the [default] context. If this is not the case you need to specify the context. http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20mailbox i.e. # voicemail.conf # [admin] 4060 = 4060,fred,[EMAIL PROTECTED] [sales] 4061 = 4061,Sales Team,[EMAIL PROTECTED],,delete=yes # sip.conf # [4060] .. [EMAIL PROTECTED] [4061] .. [EMAIL PROTECTED] Scott Henderson wrote: I have just finished setting up a new asterisk system which is basically the same as our first system. We are using 7960 phones and I used the phone config files the first installation with appropriate changes. The problem is that on the new system I get no message lights, I can't figure this out. One thing I do notice is that when I monitor the sip debug on the second system the sip chatter is almost none existent and the sip chatter on the first system that works is quite regular. There is a version difference as follows: The system that is working is: Asterisk 1.0.1 built by [EMAIL PROTECTED] on a i686 running Linux The system that isn't working is: Asterisk 1.0.2 built by [EMAIL PROTECTED] on a i686 running Linux I have reviewed everything I can think of but now message lights and the chatter that seems to have the Message information doesn't seem to be occurring on the system that isn't work like it is on the working system. ___ 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 -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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] can the dialtone be changed after pressing 9?
extensions.conf has
ignorepat = 9
exten = _9X.,1,Dial(Zap/G2/${EXTEN:1})
The first user to try it asked if instead of keeping the same dialtone
after pressing 9, if I could play a different dialtone. Can this be
done? I'm running asterisk 1.0.0 in case that matters.
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Re: [Asterisk-Users] Asterisk 1.0.2 - Unable to allocate channelstructure
Um, that's about normal here. It runs like 16 threads on a fresh startup. Maybe you don't have threading enabled on your box? - Eric On Fri, 7 Jan 2005 10:04:59 -0600 Matthew Boehm [EMAIL PROTECTED] wrote: Holy cow! Why are there so many asterisk instances running? There should only be 1. kill them all and start just 1 asterisk -Matthew - Original Message - From: Eric [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 9:35 AM Subject: [Asterisk-Users] Asterisk 1.0.2 - Unable to allocate channelstructure Hi, This morning I had some failed calls. On the console (and in the log) I saw the error Unable to allocate channel structure. Before I restarted the process, I checked it's memory usage in ps and glanced at my free memory in top. Asterisk was using a normal ammount of memory, about 40M. I don't think this was a system limit. This was running Asterisk v1.0.2. Below is an excerpt of my messages log as well as the output of ps and top, if it helps. Has anyone seen this sort of error before? Any ideas what could be causing it? The changelog for 1.0.3 doesn't list anything related to memory or resource allocation.. Anyone know if there was any work done to ast_channel_alloc() or related functions? Thanks. - Eric Jan 7 07:24:50 WARNING[163850]: Unable to allocate channel structure Jan 7 07:24:50 WARNING[163850]: Unable to start PBX on channel 0/11, span 1 Jan 7 07:24:50 WARNING[163850]: Call specified, but not found? Jan 7 07:24:50 WARNING[163850]: Hangup on bad channel 0/11 on span 1 Jan 7 07:24:51 WARNING[180235]: Unable to allocate channel structure Jan 7 07:24:51 WARNING[180235]: Unable to start PBX on channel 0/1, span 2 Jan 7 07:24:51 WARNING[180235]: Call specified, but not found? Jan 7 07:24:51 WARNING[180235]: Hangup on bad channel 0/1 on span 2 Jan 7 07:24:54 WARNING[163850]: Call specified, but not found? Jan 7 07:24:54 WARNING[163850]: Hangup on bad channel 0/11 on span 1 Jan 7 07:24:55 WARNING[180235]: Call specified, but not found? Jan 7 07:24:55 WARNING[180235]: Hangup on bad channel 0/1 on span 2 Jan 7 08:20:24 WARNING[81925]: Unable to allocate channel structure Jan 7 08:20:24 NOTICE[81925]: Unable to create/find channel Jan 7 08:20:42 WARNING[81925]: Unable to allocate channel structure Jan 7 08:20:42 NOTICE[81925]: Unable to create/find channel Jan 7 08:21:03 WARNING[81925]: Unable to allocate channel structure Jan 7 08:21:03 NOTICE[81925]: Unable to create/find channel Jan 7 08:22:43 WARNING[81925]: Unable to allocate channel structure Jan 7 08:22:43 NOTICE[81925]: Unable to create/find channel Jan 7 08:23:01 WARNING[81925]: Unable to allocate channel structure Jan 7 08:23:01 NOTICE[81925]: Unable to create/find channel Jan 7 08:23:23 WARNING[81925]: Unable to allocate channel structure Jan 7 08:23:23 NOTICE[81925]: Unable to create/find channel Jan 7 08:26:09 WARNING[81925]: Unable to allocate channel structure Jan 7 08:26:09 NOTICE[81925]: Unable to create/find channel Jan 7 08:26:17 WARNING[81925]: Unable to allocate channel structure Jan 7 08:26:17 NOTICE[81925]: Unable to create/find channel Jan 7 08:28:23 WARNING[81925]: Unable to allocate channel structure Jan 7 08:28:23 NOTICE[81925]: Unable to create/find channel Jan 7 08:28:29 WARNING[81925]: Maximum retries exceeded on call 1636b9b523c778f [EMAIL PROTECTED] for seqno 102 (Non-critical Response) Jan 7 08:28:30 WARNING[163850]: Unable to allocate channel structure Jan 7 08:28:30 WARNING[163850]: Unable to start PBX on channel 0/12, span 1 Jan 7 08:28:31 WARNING[163850]: Call specified, but not found? Jan 7 08:28:31 WARNING[163850]: Hangup on bad channel 0/12 on span 1 Jan 7 08:28:31 WARNING[180235]: Unable to allocate channel structure Jan 7 08:28:31 WARNING[180235]: Unable to start PBX on channel 0/2, span 2 Jan 7 08:28:31 WARNING[180235]: Call specified, but not found? Jan 7 08:28:31 WARNING[180235]: Hangup on bad channel 0/2 on span 2 Jan 7 08:28:34 WARNING[163850]: Call specified, but not found? Jan 7 08:28:34 WARNING[163850]: Hangup on bad channel 0/12 on span 1 Jan 7 08:28:35 WARNING[180235]: Call specified, but not found? Jan 7 08:28:35 WARNING[180235]: Hangup on bad channel 0/2 on span 2 Jan 7 08:29:18 WARNING[81925]: Unable to allocate channel structure Jan 7 08:29:18 NOTICE[81925]: Unable to create/find channel Jan 7 08:29:30 WARNING[81925]: Unable to allocate channel structure Jan 7 08:29:30 NOTICE[81925]: Unable to create/find channel ([EMAIL PROTECTED]) ~ # ps aux USER PID %CPU %MEM VSZ RSS TTY STAT START TIME COMMAND root 1 0.0 0.1 1272 476 ?S 2004 0:06 init [3] root 2 0.0 0.0 00 ?SW2004 0:00 [keventd] root 3 0.0 0.0 00 ?SWN 2004 0:00
Re: [Asterisk-Users] Moderator on vacation?
Andrew Kohlsmith wrote: If you got that message it means you posted to the list from an address that is not subscribed. It's a little misleading -- I've *never* had a moderator post or deny a message I've posted from a nonsubscriber address, on vacation or not. That may not be the only reason for the awaiting moderator approval, but it is the one I often get when I forget to hit the dropdown and change the From address to asteriskuser (list email). Post to the list from an address that is subscribed, like you just did here. No human intervention required. :-) He said he was posting a large file, it may have been larger than what the list allows. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ringing an extension on multiple phones
I am using Cisco 7960 phones and have had a request to have the receptionist phone ring on multiple phones just in case she is not around. Call pickup is the theory here but the issue is that not all the people that need to hear the ring would here the receptionist phone ring so I think I need to have a second line appearance on the phones in question so that line will ring. Can this be done or is there a better way. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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] mantis password reset link
Greetings, Does someone have the link to reset your password on bugs.digium.com? I can't seem to find one. Thanks. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Setting up Polycom IP 500 with *
On Fri, 2005-01-07 at 09:18 -0700, Wiley Siler wrote: The FTP server option works very well so you should do it when get time. The phone has an option where you tell it to load via FTP, believe it is the server config. To get to it, reboot the phone and enter setup on the phone, not the web. Remove the settings if you want no configs from network and your settings via browser should work if correct. Wiley What user and pass does the polycom use to connect to the ftp server? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Walker Sent: Friday, January 07, 2005 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Setting up Polycom IP 500 with * I am in the process of setting up an * system using Polycom IP 500's. I don't want to spend time setting a ftp server for application and configuration files at the moment, just want to use the web interface to the Polycoms. DCHP works OK and IP is obtained correctly. Polycom fails to load .cfg file and holts. I have read the 143 page admin user guide a couple of times...and I missing somthing? Adrian Walker [EMAIL PROTECTED] === This email has been scanned for Virus infection by MessageLabs For more information please contact [EMAIL PROTECTED] === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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] Multiple lines on Cisco 7960
I had not looked at the phones settings yet, thanks for the suggestion. The setting indicate that there is no configuration on the second line it is listed as UNPROVISIONED Go into the phone and program Line 2 Settings directly, without using the SIPMAC.cnf file. If that works, then your .cnf file is wrong. -- Nabeel Jafferali tel: 416.491.9136 (toronto) 646.225.7426 (new york) fwd: 46990 email/msn : nabeelatjafferali.net ___ 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] Linksys RT31P2
Hello, Is there any way to unlock the Linksys router? -- Richard Cook [EMAIL PROTECTED] Tel: 705-497-9320 ext 2010 Blank Bkgrd.gif___ 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] New York?
Hey I noticed this posting, is anyone in New York interested in catching up? I'd be happy to host it at my place on 72nd/york if it wasn't too big a group, or we can always head out and grab some lunch or something somewhere. Email me your interest and we'll see what the numbers are. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Hanson Sent: Thursday, January 06, 2005 8:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Twin Cities Asterisk meeting still on for Saturday? Is the meeting still on for Saturday 1/8/05? 11:30am at 2375 University Av W STE120, Saint Paul. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with MySQL
post your /etc/odbc.ini and /etc/odbcinst.ini -matthew - Original Message - From: rizwan [EMAIL PROTECTED] To: Asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 10:19 AM Subject: [Asterisk-Users] Asterisk with MySQL Hello I am getting this error message, when i try to authenticate my users through database. Jan 7 20:28:08 WARNING[26487]: res_config_odbc.c:69 realtime_odbc: SQL Alloc Handle failed! Jan 7 20:28:08 NOTICE[26487]: chan_sip.c:7974 handle_request: Registration from 'rizwan sip:[EMAIL PROTECTED]' failed for '192.168.0.149' My conf files are: ;res_odbc.conf [test] dsn = test username = root password = pre-connect = yes ;extensions.conf [test] switch = Realtime/@realtime_ext ;extconfig.conf sipfriends = odbc,test,sip_buddies realtime_ext = odbc,test,extensions_table Can you please help me, what to do here? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR question
I use the CDR CVS file for logging my home phone system. Can I force write data to a CDR Field from an extensions macro? Say if the callerid was empty and I dumped the call to put data in the CDR to let me know that is what happened. Thanks --John ___ 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] spandsp and app_rxfax (alternative topic: t38modem)
On Friday 07 January 2005 11:24 am, Andrew Kohlsmith wrote: I also note that you posted your initial message at 4:14pm, and now, less than 24 hours later you are expecting the entire asterisk community to have received your message, parsed it in the sea of other messages to the list, had it apply to them and responded. You are correct - my timing was inappropriate. It's just that this is a project of escalating priority for my employer. My apologies and my gratitude, Ryan VanMiddlesworth ___ 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] spandsp and app_rxfax (alternative topic: t38modem)
Nils Segerdahl wrote: On Fri, 7 Jan 2005, Ryan wrote: I had the same problems using hfc cards with bristuff. (with patched zaptel drivers). Which zaptel patches did you use? Thanks -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.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] can the dialtone be changed after pressing 9?
Title: Re: [Asterisk-Users] can the dialtone be changed after pressing 9?
Yes you can but it only works for zap devices. IP based would be a function of the hardware.
-Original Message-
From: [EMAIL PROTECTED] [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Sent: Fri Jan 07 11:42:41 2005
Subject: [Asterisk-Users] can the dialtone be changed after pressing 9?
extensions.conf has
ignorepat = 9
exten = _9X.,1,Dial(Zap/G2/${EXTEN:1})
The first user to try it asked if instead of keeping the same dialtone
after pressing 9, if I could play a different dialtone. Can this be
done? I'm running asterisk 1.0.0 in case that matters.
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Re: [Asterisk-Users] spandsp and app_rxfax (alternative topic:t38modem)
I have the same problem and thought I would wait for someone else to post... (just kidding Ryan) I have used an analog trunk (FXO) AND a station (FXS) both on the same card. I thought that it might be related to the hardware so I hooked up an old Brother Intellifax 9000 on the station port. Both of these attempts had the same problem. It is my speculation that the 'cutoff' problem was related to some type of 'line noise' and that others successfully using the spandsp code _might_ be using T1/E1 rather than analog lines (1FL) but when I started testing using an old Fax machine plugged into a station port with a six foot RJ11 on either end, I realized that this setup really shouldn't be introducing much noise (if any) so I am lost. It happens approximately half way through EVERY fax I attempt regardless of sending machine (I tried Dialogic and some modems) or port (FXO or FXS) so I just gave up on it. (It should be noted that Asterisk 1.0.3 runs on FC3 with libtiff-3.6.1-8.fc3/kernel-2.6.9-1.724_FC3 otherwise) There IS a link (search spandsp cutoff fax on Google) to a similar problem that was apparently fixed with version 0.0.1(h). I assumed that the fix was already applied to the 0.0.2pre6 version. I thought I would wait until another version of either * OR spandsp was posted but if anyone else has any suggestions (or can corroborate the Digital vs. Analog theory) I would love to hear from them otherwise I will test this on another PC when I get the chance. Just thought I would chime in ;-) Jeff - Original Message - From: Ryan [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 11:13 AM Subject: Re: [Asterisk-Users] spandsp and app_rxfax (alternative topic:t38modem) H. Did I just ask in the wrong forum, or has _nobody_ experienced image corruption using app_rxfax that was NOT due to using the wrong version of libtiff? snip ___ 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] Ringing an extension on multiple phones
Title: Re: [Asterisk-Users] Ringing an extension on multiple phones There are several options here. You can set up a queue and have the phones ring un the order you like. Setup an additional extension on every phone. Set up an AGI script that allows them to login to the receptionist calls. That way they can turn it on and off when they want. -Original Message- From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Sent: Fri Jan 07 11:45:37 2005 Subject: [Asterisk-Users] Ringing an extension on multiple phones I am using Cisco 7960 phones and have had a request to have the receptionist phone ring on multiple phones just in case she is not around. Call pickup is the theory here but the issue is that not all the people that need to hear the ring would here the receptionist phone ring so I think I need to have a second line appearance on the phones in question so that line will ring. Can this be done or is there a better way. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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] spandsp and app_rxfax (alternative topic:t38modem)
Seems to be correct, or at least image corruption from a really crappy fax reception. I know I've been receiving between 30-50 faxes a day with app_rxfax without issue. What versions of everything are you using? Using PRI? libtiff? spandsp? asterisk? diagram? I can't get any faxes via rxfax. -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] spandsp and app_rxfax (alternative topic:t38modem)
im using libtiff-3-7 and im getting corruption constatnly. I posted to Steve's bug site but I've not heard from him in over a month. i guess he's still on vacation. -Matthew - Original Message - From: Ryan [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 10:13 AM Subject: Re: [Asterisk-Users] spandsp and app_rxfax (alternative topic:t38modem) H. Did I just ask in the wrong forum, or has _nobody_ experienced image corruption using app_rxfax that was NOT due to using the wrong version of libtiff? If that's the case, then my secondary approach is going to have to be: PSTN - Asterisk + chan_h323 - t38modem + Hylafax Is there anybody that could help me with either of these solutions? A thousand thank yous in advance, Ryan VanMiddlesworth On Thursday, January 6th, I wrote: I've been pulling my hair out trying to get Asterisk to receive and decode a fax using spandsp and app_rxfax. It seems like it should be working. The fax machine on the other end connects and Asterisk reports a fax coming in. But when it's done all I have is a 2 or 3 KB TIF (see attachment). The console activity looks completely normal: -- Starting simple switch on 'Zap/3-1' -- Executing SetVar(Zap/3-1, FAXFILE=/var/spool/asterisk/fax/1105043880.0.tif) in new stack -- Executing RxFAX(Zap/3-1, /var/spool/asterisk/fax/1105043880.0.tif) in new stack -- Hungup 'Zap/3-1' And there are no errors in the log file. Here's my config: Wildcard TDM40B hardware (Zaptel) asterisk-1.0.2 spandsp-0.0.2pre6 libtiff-3.6.1 (with the fax fix patches) (also tried libtiff-3.6.0 and libtiff-3.5.7) I've tried multiple sending fax machines and get the same effect. Any tips on getting this setup working? I've run out of ideas. Alternately, I'd also be willing to offload the DSP processing to a HylaFAX machine using some sort of software fax driver. I tinkered with t38modem and chan_h323, but couldn't get it to do anything once the HylaFAX machine answered. So if anybody has any experience with that, I'd be interested. Thanks in advance, Ryan VanMiddlesworth ___ 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] Multiple lines on Cisco 7960
I set this up manually on the phone and it works just fine so config files ... I attached the complete config files so maybe someone can see what I am missing. argon:/tftpboot# cat SIPDefault.cnf # SIP Default Generic Configuration File # Image Version image_version: P0S3-07-3-00 ; # Proxy Server proxy1_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy2_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy3_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy4_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy5_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy6_address: "192.168.17.13" ; Can be dotted IP or FQDN # Proxy Server Port (default - 5060) proxy1_port: 5060 proxy2_port: 5060 proxy3_port: 5060 proxy4_port: 5060 proxy5_port: 5060 proxy6_port: 5060 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 1 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: none # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; Default 6 timer_invite_expires: 180 ; Default 180 sec ### New Parameters added in Release 2.0 ### # Dialplan template (.xml format file relative to the TFTP root directory) dial_template: dialplan # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: "" ; Example: ./sip_phone/ # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: "192.168.17.11" ; SNTP Server IP Address sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default) time_zone: YST ; Time Zone Phone is in dst_offset: 1 ; Offset from Phone's time when DST is in effect dst_start_month: April ; Month in which DST starts dst_start_day: "" ; Day of month in which DST starts dst_start_day_of_week: Sun ; Day of week in which DST starts dst_start_week_of_month: 1 ; Week of month in which DST starts dst_start_time: 02 ; Time of day in which DST starts dst_stop_month: Oct ; Month in which DST stops dst_stop_day: "" ; Day of month in which DST stops dst_stop_day_of_week: Sunday ; Day of week in which DST stops dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last week of month dst_stop_time: 2 ; Time of day in which DST stops dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment time_format_24hr: 0 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr) # Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control) dnd_control: 0 ; Default 0 (Do Not Disturb feature is off) # Caller ID Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control) callerid_blocking: 0 ; Default 0 (Disable sending all calls as anonymous) # Anonymous Call Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls) # DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127) dtmf_avt_payload: 101 ; Default 101 # Sync value of the phone used for remote reset sync: 1 ; Default 1 ### New Parameters added in Release 2.1 ### # Backup Proxy Support proxy_backup: "" ; Dotted IP of Backup Proxy proxy_backup_port: 5060 ; Backup Proxy port (default is 5060) # Emergency Proxy Support proxy_emergency: "" ; Dotted IP of Emergency Proxy proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060) # Configurable VAD option enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable ### New Parameters added in Release 2.2 ## # NAT/Firewall Traversal nat_enable: 0 ; 0-Disabled (default), 1-Enabled nat_address: "" ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled # Outbound Proxy Support outbound_proxy: "" ; restricted to dotted IP or DNS A record only outbound_proxy_port: 5060 ; default is 5060 ### New Parameter added in Release 3.0 ### # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default) ### New Parameters added in Release 3.1 ### # Allow Transfer to be completed while target phone is still ringing semi_attended_transfer: 1 ; 0-Disabled, 1-Enabled (default) #
Re: [Asterisk-Users] Asterisk with MySQL
Please find the attached files, Thanks On Friday 07 January 2005 22:24, you wrote: post your /etc/odbc.ini and /etc/odbcinst.ini -matthew - Original Message - From: rizwan [EMAIL PROTECTED] To: Asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 10:19 AM Subject: [Asterisk-Users] Asterisk with MySQL Hello I am getting this error message, when i try to authenticate my users through database. Jan 7 20:28:08 WARNING[26487]: res_config_odbc.c:69 realtime_odbc: SQL Alloc Handle failed! Jan 7 20:28:08 NOTICE[26487]: chan_sip.c:7974 handle_request: Registration from 'rizwan sip:[EMAIL PROTECTED]' failed for '192.168.0.149' My conf files are: ;res_odbc.conf [test] dsn = test username = root password = pre-connect = yes ;extensions.conf [test] switch = Realtime/@realtime_ext ;extconfig.conf sipfriends = odbc,test,sip_buddies realtime_ext = odbc,test,extensions_table Can you please help me, what to do here? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users [FB_SAMPLE] Driver = FIREBIRD Description = Firebird driver Database= server:employee.gdb User= sysdba Password= masterkey With_Schema = 0 Dialect = 3 Charset = Role= Nowait = 0 OldMetaData = 0 ExecProc= 0 Dquote = 0 WithDefault = 1 TxnMode = 1 Flusfcommit = 0 Padvarchar = 0 Nullschema = 0 Fixprecision= 0 Simpleunicode = 0 wchardefault= 0 [demo] Driver = OOB Description = Easysoft ODBC-ODBC Bridge demo data source SERVER = demo.easysoft.com PORT= TRANSPORT = tcpip TARGETDSN = pubs LOGONUSER = demo LOGONAUTH = easysoft TargetUser = demo TargetAuth = easysoft [PostgreSQL] Description = ODBC for PostgreSQL Driver = /usr/lib/libodbcpsql.so Setup = /usr/lib/libodbcpsqlS.so FileUsage = 1 [FIREBIRD] Description = Easysoft Firebird ODBC Driver Driver = /usr/local/easysoft/fb/libfbodbc.so Setup = /usr/local/easysoft/fb/libfbodbcS.so FileUsage = 1 DontDLClose = 1 [OOB] Description = Easysoft ODBC-ODBC Bridge Driver = /usr/local/easysoft/oob/client/libesoobclient.so Setup = /usr/local/easysoft/oob/client/libesoobsetup.so FileUsage = 2 [MySQL ODBC 3.51 Driver] DRIVER = /usr/lib/libmyodbc3.so SETUP = /usr/lib/libmyodbc3S.so FileUsage = 1 ___ 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] Linksys RT31P2 {Scanned}
Check this out. http://voip.weblogsinc.com/entry/0142584371536804/ David On Fri, 2005-01-07 at 09:15, Richard Cook wrote: Hello, Is there any way to unlock the Linksys router? -- Richard Cook [EMAIL PROTECTED] Tel: 705-497-9320 ext 2010 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. Plase contact [EMAIL PROTECTED] if you have questions about this email. __ ___ 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] Multiple lines on Cisco 7960
Someone on the list spotted the problem, there is a typo in my line definitions. Thanks all Scott Henderson wrote: I set this up manually on the phone and it works just fine so config files ... I attached the complete config files so maybe someone can see what I am missing. argon:/tftpboot# cat SIPDefault.cnf # SIP Default Generic Configuration File # Image Version image_version: P0S3-07-3-00 ; # Proxy Server proxy1_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy2_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy3_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy4_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy5_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy6_address: "192.168.17.13" ; Can be dotted IP or FQDN # Proxy Server Port (default - 5060) proxy1_port: 5060 proxy2_port: 5060 proxy3_port: 5060 proxy4_port: 5060 proxy5_port: 5060 proxy6_port: 5060 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 1 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: none # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; Default 6 timer_invite_expires: 180 ; Default 180 sec ### New Parameters added in Release 2.0 ### # Dialplan template (.xml format file relative to the TFTP root directory) dial_template: dialplan # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: "" ; Example: ./sip_phone/ # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: "192.168.17.11" ; SNTP Server IP Address sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default) time_zone: YST ; Time Zone Phone is in dst_offset: 1 ; Offset from Phone's time when DST is in effect dst_start_month: April ; Month in which DST starts dst_start_day: "" ; Day of month in which DST starts dst_start_day_of_week: Sun ; Day of week in which DST starts dst_start_week_of_month: 1 ; Week of month in which DST starts dst_start_time: 02 ; Time of day in which DST starts dst_stop_month: Oct ; Month in which DST stops dst_stop_day: "" ; Day of month in which DST stops dst_stop_day_of_week: Sunday ; Day of week in which DST stops dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last week of month dst_stop_time: 2 ; Time of day in which DST stops dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment time_format_24hr: 0 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr) # Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control) dnd_control: 0 ; Default 0 (Do Not Disturb feature is off) # Caller ID Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control) callerid_blocking: 0 ; Default 0 (Disable sending all calls as anonymous) # Anonymous Call Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls) # DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127) dtmf_avt_payload: 101 ; Default 101 # Sync value of the phone used for remote reset sync: 1 ; Default 1 ### New Parameters added in Release 2.1 ### # Backup Proxy Support proxy_backup: "" ; Dotted IP of Backup Proxy proxy_backup_port: 5060 ; Backup Proxy port (default is 5060) # Emergency Proxy Support proxy_emergency: "" ; Dotted IP of Emergency Proxy proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060) # Configurable VAD option enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable ### New Parameters added in Release 2.2 ## # NAT/Firewall Traversal nat_enable: 0 ; 0-Disabled (default), 1-Enabled nat_address: "" ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled # Outbound Proxy Support outbound_proxy: "" ; restricted to dotted IP or DNS A record only outbound_proxy_port: 5060 ; default is 5060 ### New Parameter added in Release 3.0 ### # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default) ###
RE: [Asterisk-Users] New York?
-Original Message- Hey I noticed this posting, is anyone in New York interested in catching up? I'd be happy to host it at my place on 72nd/york if it wasn't too big a group, or we can always head out and grab some lunch or something somewhere. Email me your interest and we'll see what the numbers are. Cheers, Dean / SNIP/ Add me to the RSVP list. I am at 633 Broadway, between 50th and 49th. Ph: 212-537-2849 Seshu Kanuri NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Moderator on vacation?
streamload.com dropload.com - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 5:25 PM Subject: Re: [Asterisk-Users] Moderator on vacation? On January 7, 2005 11:22 am, Andrew Thompson wrote: I can't get google to show me any, but there are sites that allow you to drop off large files and give you a url for retreiving them. Perhaps someone can come up with the name of one. http://pastebin.ca is what is used on the IRC channel almost exlcusively. Also its big brother, http://pastebin.com, although it is frequenty slow. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp and app_rxfax (alternative topic:t38modem)
On 2005.01.07 09:42 Jeff wrote: It is my speculation that the 'cutoff' problem was related to some type of 'line noise' and that others successfully using the spandsp code _might_ be using T1/E1 rather than analog lines (1FL) but when I started testing using an old Fax machine plugged into a station port with a six foot RJ11 on either end, I realized that this setup really shouldn't be introducing much noise (if any) so I am lost. It happens approximately half way through EVERY fax I attempt regardless of sending machine (I tried Dialogic and some modems) or port (FXO or FXS) so I just gave up on it. Since spandsp doesn't use ECM, what I'm about to say doesn't apply to spandsp. If you receive truncated (versus corrupted) fax images from spandsp, then I'm not sure what the problem would be. What I'm about to say only applies to ECM-enabled fax sessions such as usually will happen with most modern fax machines and modern HylaFAX. Truncated fax images usually only occur in an ECM-enabled fax session when the total image is larger than 64KB. With images larger than 64KB it is required that the image data be broken up into 64KB blocks and each block is transmitted separately. In the fax protocol this essentially works out to the same thing as sending a multipage fax except that the in-between-blocks signals indicate a multiple-block scenario rather than a multiple-page one. The timing sensitivities between pages and between blocks are crucial. A 20 ms lag at this point will most certainly terminate the fax session. Most pauses between signal exchanges during faxing are 75 ms +/- 20 ms. This means that most senders will wait pause for what it believes to be exactly 75 ms, with the buffer to compensate for any lags incurred by the telco or other timing issues. Consequently, if Asterisk (or the VoIP configuration) introduces a 20 ms lag at this point, then the timing tolerances will be exceeded, and a fax machine following the specifications will terminate the fax session after a few attempts to recover from this. So, you end up with a page of image data missing one or more blocks, and this produces a truncated (not corrupted) fax image. Now, depending on other factors the very end of that image data could, in theory, also look corrupted. But, most ECM sessions are going to use MMR compression, meaning that any data corruption (only possible in that last block received) would also likely truncate the image at that point (since any data after the point of corruption becomes meaningless). As far as I've been able to determine, there's nothing that can be done about this working with analog fax equipment behind Asterisk. In order for things to work correctly here, either Asterisk needs to support T.38 (FoIP specification), or Asterisk needs to produce pseudo-modem interfaces for fax packages like HylaFAX. I think the spandsp author is working on both of these over time. Lee. ___ 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] Ringing an extension on multiple phones
You can Dial() extension SIP/line1SIP/line2 even more you can and that will call both extensions only after a 5 seconds timeout exten = xxx,1,Dial(SIP/line1,5) exten = xxx,2,Dial(SIP/line1SIP/line2,10) etc... that's if I understood what ou needed... bye, M. - Original Message - From: Scott Henderson [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 1:45 PM Subject: [Asterisk-Users] Ringing an extension on multiple phones I am using Cisco 7960 phones and have had a request to have the receptionist phone ring on multiple phones just in case she is not around. Call pickup is the theory here but the issue is that not all the people that need to hear the ring would here the receptionist phone ring so I think I need to have a second line appearance on the phones in question so that line will ring. Can this be done or is there a better way. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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] Multiple lines on Cisco 7960
Theres your problem right there; All of them say line2_X Nathan. # Line 2 line2_name: Scott1 line2_authname: scott1 line2_password: scott1 # Line 3 line2_name: Line 2 line2_authname: UNPROVISIONED line2_password: UNPROVISIONED # Line 4 line2_name: Line 4 line2_authname: UNPROVISIONED line2_password: UNPROVISIONED # Line 5 line2_name: Line 5 line2_authname: UNPROVISIONED line2_password: UNPROVISIONED # Line 6 line2_name: Line 6 line2_authname: UNPROVISIONED line2_password: UNPROVISIONED Scott Henderson wrote: I set this up manually on the phone and it works just fine so config files ... I attached the complete config files so maybe someone can see what I am missing. argon:/tftpboot# cat SIPDefault.cnf # SIP Default Generic Configuration File # Image Version image_version: P0S3-07-3-00 ; # Proxy Server proxy1_address: 192.168.17.13 ; Can be dotted IP or FQDN proxy2_address: 192.168.17.13 ; Can be dotted IP or FQDN proxy3_address: 192.168.17.13 ; Can be dotted IP or FQDN proxy4_address: 192.168.17.13 ; Can be dotted IP or FQDN proxy5_address: 192.168.17.13 ; Can be dotted IP or FQDN proxy6_address: 192.168.17.13 ; Can be dotted IP or FQDN # Proxy Server Port (default - 5060) proxy1_port: 5060 proxy2_port: 5060 proxy3_port: 5060 proxy4_port: 5060 proxy5_port: 5060 proxy6_port: 5060 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 1 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: none # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10; Default 10 sip_invite_retx: 6 ; Default 6 timer_invite_expires: 180 ; Default 180 sec ### New Parameters added in Release 2.0 ### # Dialplan template (.xml format file relative to the TFTP root directory) dial_template: dialplan # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: ; Example: ./sip_phone/ # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: 192.168.17.11; SNTP Server IP Address sntp_mode: directedbroadcast; unicast, multicast, anycast, or directedbroadcast (default) time_zone: YST ; Time Zone Phone is in dst_offset: 1 ; Offset from Phone's time when DST is in effect dst_start_month: April ; Month in which DST starts dst_start_day:; Day of month in which DST starts dst_start_day_of_week: Sun ; Day of week in which DST starts dst_start_week_of_month: 1 ; Week of month in which DST starts dst_start_time: 02 ; Time of day in which DST starts dst_stop_month: Oct ; Month in which DST stops dst_stop_day: ; Day of month in which DST stops dst_stop_day_of_week: Sunday; Day of week in which DST stops dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last week of month dst_stop_time: 2; Time of day in which DST stops dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment time_format_24hr: 0 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr) # Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control) dnd_control: 0 ; Default 0 (Do Not Disturb feature is off) # Caller ID Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control) callerid_blocking: 0; Default 0 (Disable sending all calls as anonymous) # Anonymous Call Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls) # DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127) dtmf_avt_payload: 101 ; Default 101 # Sync value of the phone used for remote reset sync: 1 ; Default 1 ### New Parameters added in Release 2.1 ### # Backup Proxy Support proxy_backup: ; Dotted IP of Backup Proxy proxy_backup_port: 5060 ; Backup Proxy port (default is 5060) # Emergency Proxy Support proxy_emergency: ; Dotted IP of Emergency Proxy proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060) # Configurable VAD option enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable ### New Parameters added in Release 2.2 ## # NAT/Firewall
[Asterisk-Users] Question to authenficate client automaticlly
Hi, i have setting up asterisk for mysql. i using the template-database: sipfriends. i have a vpn in the office. i like to setup asterisk: when a client make authentification request: username and password stores automaticlly in the sql database. any users in the vpn can setup the own name and password. how can setup mysql when come user: test and secret: test will be stored? thank you for your help. --- tel : 089 2500 7676 homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc ___ 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] spandsp and app_rxfax (alternative topic:t38modem)
On January 7, 2005 12:26 pm, Matthew Boehm wrote: Seems to be correct, or at least image corruption from a really crappy fax reception. I know I've been receiving between 30-50 faxes a day with app_rxfax without issue. What versions of everything are you using? Using PRI? libtiff? spandsp? asterisk? diagram? I can't get any faxes via rxfax. Slackware 10.0 system libtiff NOT FROM slackware install, compiled manually 3.5.7 libpri and asterisk from CVS HEAD (~20041216) spandsp 0.0.2 You will note that the standard slackware install has libtiff in the aaa_elflibs package. You must go in there and manually obliterate anything TIFF (IIRC it's only /usr/lib/libtiff.so.3.6.1 and the symlinks) Diagram is deceptive: PRI - TE405P - Asterisk1 - IAX2 - Asterisk2 IAX2 link is a 1-hop SDSL link over Pairgain Megabit Modem 300S devices. Ethernet cards are Intel gigE on Asterisk1 and a Realtek RTL8139. Asterisk1 is a Supermicro server - Xeon processor, SCSI hard disks, ECC RAM. Asterisk2 is a simple plain-jane P3/800. Asterisk2 also has a TDM430P in it which I send faxes from (Canon IR3300 and an ancient Epson fax) -- I cannot RECEIVE faxes to either of these reliably through the TDM card (they worked fine when I had a T100P+Adit600 channel bank), which is why I set up app_rxfax. I wanted to see if it was the TDM card botching up or faxing over the IAX2 link; it's the TDM430P. It's strange, I can send through the TDM430 just fine, but neither fax can receive worth a shit through it. And both machines support ECM and so on. My fax rx rate hovers around 50% though the TDM430. It's at 100% with app_rxfax. -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] Setting up Polycom IP 500 with *
Default for IP 500 (prolly the other too, but not sure) username: PlcmSpIp password: PlcmSpIp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Friday, January 07, 2005 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Setting up Polycom IP 500 with * On Fri, 2005-01-07 at 09:18 -0700, Wiley Siler wrote: The FTP server option works very well so you should do it when get time. The phone has an option where you tell it to load via FTP, believe it is the server config. To get to it, reboot the phone and enter setup on the phone, not the web. Remove the settings if you want no configs from network and your settings via browser should work if correct. Wiley What user and pass does the polycom use to connect to the ftp server? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Walker Sent: Friday, January 07, 2005 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Setting up Polycom IP 500 with * I am in the process of setting up an * system using Polycom IP 500's. I don't want to spend time setting a ftp server for application and configuration files at the moment, just want to use the web interface to the Polycoms. DCHP works OK and IP is obtained correctly. Polycom fails to load .cfg file and holts. I have read the 143 page admin user guide a couple of times...and I missing somthing? Adrian Walker [EMAIL PROTECTED] === This email has been scanned for Virus infection by MessageLabs For more information please contact [EMAIL PROTECTED] === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 1/6/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 1/6/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with MySQL
ok. you have in your res_odbc: dsn= test but you don't have a dsn called test in any of your odbc config stuff. -Matthew - Original Message - From: Muhammad Rizwan Khan [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 11:51 AM Subject: Re: [Asterisk-Users] Asterisk with MySQL Please find the attached files, Thanks On Friday 07 January 2005 22:24, you wrote: post your /etc/odbc.ini and /etc/odbcinst.ini -matthew - Original Message - From: rizwan [EMAIL PROTECTED] To: Asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 10:19 AM Subject: [Asterisk-Users] Asterisk with MySQL Hello I am getting this error message, when i try to authenticate my users through database. Jan 7 20:28:08 WARNING[26487]: res_config_odbc.c:69 realtime_odbc: SQL Alloc Handle failed! Jan 7 20:28:08 NOTICE[26487]: chan_sip.c:7974 handle_request: Registration from 'rizwan sip:[EMAIL PROTECTED]' failed for '192.168.0.149' My conf files are: ;res_odbc.conf [test] dsn = test username = root password = pre-connect = yes ;extensions.conf [test] switch = Realtime/@realtime_ext ;extconfig.conf sipfriends = odbc,test,sip_buddies realtime_ext = odbc,test,extensions_table Can you please help me, what to do here? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] xmitting CallerID
Attempted to get this info from Digium but my efforts have failed... I am thinking of getting a TE410P from digium. My local Telco uses B8ZSESF and does support PBX customizing ANIs on a per call basis. What I need to know is, can I use the SetCallerID command in extensions.conf to transmit the DID# of the extension making the call with the TE410P or is there a different one that does support, customizing your ANI. -Mark 707-735-1038 ___ 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] Ringing an extension on multiple phones
You can Dial() extension SIP/line1SIP/line2 Yes, and if the multiple extensions that ring are members of the same group then any one of the phones can pickup the call. So the next question is: how does the receptionist put the system into group ring mode. The answer is to have the receptionist call a nominated number such as **221 (enable group ringing) and **222 (to disable group ringing). When the receptionist calls **221 a global variable (or an entry in the registry is created) is made to contain a value that indicates group ringing is in effect. When **222 is called, calls ring on the operator extension. We use a similar approach to have support calls forwarded to mobile phones out of office hours. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Listas Sent: January 07, 2005 6:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Ringing an extension on multiple phones You can Dial() extension SIP/line1SIP/line2 even more you can and that will call both extensions only after a 5 seconds timeout exten = xxx,1,Dial(SIP/line1,5) exten = xxx,2,Dial(SIP/line1SIP/line2,10) etc... that's if I understood what ou needed... bye, M. - Original Message - From: Scott Henderson [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 1:45 PM Subject: [Asterisk-Users] Ringing an extension on multiple phones I am using Cisco 7960 phones and have had a request to have the receptionist phone ring on multiple phones just in case she is not around. Call pickup is the theory here but the issue is that not all the people that need to hear the ring would here the receptionist phone ring so I think I need to have a second line appearance on the phones in question so that line will ring. Can this be done or is there a better way. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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
