RE: [Asterisk-Users] Most Reliable Proxy Server?
You could try these: voiptalk - www.voiptalk.org sipgate - www.sipgate.de Simon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McMillinSent: Thursday, 15 April 2004 15:29To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Most Reliable Proxy Server? Hi all, Do you know if there's any free public SIP proxy server that is more reliable that FWD and Iptel? Thanks Ron -This mail was content checked for malicious code and virusesby GFI MailSecurity.
Re: AW: [Asterisk-Users] PC based Switchboard application
Yes me to,how do I contact you On Tue, 2004-04-13 at 13:27, ePyron Felix Deierlein wrote: Hello Pertti, we would be interessted to, if you could send further informations... Thanks Regards Felix Deierlein [EMAIL PROTECTED] -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Pertti Pikkarainen Gesendet: Samstag, 10. April 2004 11:26 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] PC based Switchboard application We have switchboard application ( PC+browser+Java ) with quite a rich feature set. It talks to * via manager port. Works as a call center too. However, it is not open source. If you are interested in, please contact me directly. Best regards Pertti Keith D'Atrio wrote: Hello All I am looking for a PC based switchboard application. Cisco CallManager has a web attendant console that allows you to use the PC to transfer calls and the like and I was wondering if there was a similar program compatible with *. Thank you in advance Keith D'Atrio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: External access to voicemail
Hello. I have written a small patch to app_voicemail.c which provides the precise functionality Steve wants. I sent it to this list once, and got my subscription disabled for my trouble. so, if anyone's interested in it, it's about a 50 line diff file, which I'd be happy to mail anyone who writes and says they want it. If enough write, I'll post a URL on this list for it. If it's super popular, I'll figure out how to submit it as a feature request on the bug tracker. -Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: External access to voicemail
i'm interested in it. - Original Message - From: Brian Buhrow [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Thursday, April 15, 2004 2:54 AM Subject: [Asterisk-Users] Re: External access to voicemail Hello. I have written a small patch to app_voicemail.c which provides the precise functionality Steve wants. I sent it to this list once, and got my subscription disabled for my trouble. so, if anyone's interested in it, it's about a 50 line diff file, which I'd be happy to mail anyone who writes and says they want it. If enough write, I'll post a URL on this list for it. If it's super popular, I'll figure out how to submit it as a feature request on the bug tracker. -Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How many lines of IP phone can Asterisk support?
Hello, I am an Asterisk beginer and are using it. Now I have1 question, please ! How many lines of IP phone can Asterisk support, if we use only IP interface? Chunghwa Telecom BTA Tech. Lab.E-mail:[EMAIL PROTECTED]
Re[2]: [Asterisk-Users] dtmf for public telephony access
Grazie Matteo, I looked in wiki pages, but found nothing regarding dtmf tone regeneration, just the indication that inbound tones are not allowed over low bitrate codecs. Would you raccomend sip info or rfc2833 as tone handling method ? P.S. finalmente un compatriota :) MB * hint : did you searched the ml first? MB this has been discussed a lot, even little time ago... MB however... MB sure, just use oob dtmf like rfc2833 or sip info dtmf... MB so you can use a low bitrate codec and asterisk MB will generate them again when going to the pstn... MB matteo MB Il mer, 2004-04-14 alle 10:49, Alessio Focardi ha scritto: Hi, I would like to have some remote users with sip phones over adsl connections access our asterisk pbx and make out calls, currently we are using a zaptel pri interface for outdialing. What is the right way to manage dtmf over pstn lines and still retain low bandwith occupation ? In other words: if I use g729 (and sip info dtmf) for sip phones - asterisk communication will asterisk be able to regenerate real tones when going out to the pstn ? Tnx for any help ... currently I havent got g729 licenses so I cant test it out by myself. -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk List Digest down
For about a week the Asterisk list digest has not been sent. I have checked my subscription and all appears normal there. I sent messages to addresses indicated as the list supervisiors: [EMAIL PROTECTED] and [EMAIL PROTECTED] both were returned as cannot access mailbox /var/mail/clrhodes for user clrhodes. error writing message: File too large If someone at Digium could fix the problem all of us on the Digest would appreciate it. Steven Elliott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MeetMe - new e and E flags?
In article [EMAIL PROTECTED], Tilghman Lesher [EMAIL PROTECTED] wrote: If it's a pin-required conference, you will hear the conference number prior to being prompted to enter the associated pin. Obviously, in this case, any such conference would be static, so the pin would be pre-assigned in the config file. This might be useful if you ran a number of conferences, but did not want just anybody to be able to access them (i.e. in order to access the conferences, possibly dial-able from anywhere, you had to know the associated pin). You can also select an empty dynamic conference, with pin, by combining the flags 'eD', in which case you will be told the conference number prior to you specifying the pin. Or you could simply select an empty dynamic conference (no pin), with flags 'ed'. I'm trying hard to understand the usefulness of these features. It looks like, from what I've read here, if you dial an extension that routes to MeetMe(e), it will put you in an empty conference and tell you the number. Presumably for anyone else to join the same conference, you then have to tell them the number, e.g. by email, IM or another phone call, and they then have to dial a different extension which routes to MeetMe(without e). And if the empty conference also has a PIN, does the first user need a list of conference numbers to PINs so he can enter the correct PIN when told the conference number? This all seems rather cumbersome, and I haven't had the chance to experiment with this feature yet, so the above probably highlights both (a) my lack of understanding, and (b) the lack of documentation! Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Upgrade firmware on iaxy?
I've googled and grepped myself silly. I see the iaxy.bin file there in the contrib tree of the asterisk source, but nowhere have I been able to find out how to get it sent to the device. . . Anyone know? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calls to Cisco PSTN gateway
Hi all,
A Cisco ATA186 configured with g711ulaw, NAT=yes
and canreinvite=yes, made calls through Asterisk to a Cisco 5300 gateway
out to a PSTN line with errors as follows:
-- Executing
Dial("SIP/ata186-c1cf", "SIP/[EMAIL PROTECTED]:5060|30|r")
in new stack -- Called 29086988@110.100.231.2:5060Apr 15
16:11:22 WARNING[1116941120]: chan_sip.c:2049 process_sdp: Error in codec string
'ideo 0 '
Asterisk was configured with allow=ulaw. Any idea
for this problem??
Thanks.
Ben
Re: Re[2]: [Asterisk-Users] dtmf for public telephony access
depends on the device you're using, if are supported or not. i feel very confortable with INFO method, since is a sip message and can be easily debugged :) Il gio, 2004-04-15 alle 09:45, Alessio Focardi ha scritto: Grazie Matteo, I looked in wiki pages, but found nothing regarding dtmf tone regeneration, just the indication that inbound tones are not allowed over low bitrate codecs. Would you raccomend sip info or rfc2833 as tone handling method ? P.S. finalmente un compatriota :) MB * hint : did you searched the ml first? MB this has been discussed a lot, even little time ago... MB however... MB sure, just use oob dtmf like rfc2833 or sip info dtmf... MB so you can use a low bitrate codec and asterisk MB will generate them again when going to the pstn... MB matteo MB Il mer, 2004-04-14 alle 10:49, Alessio Focardi ha scritto: Hi, I would like to have some remote users with sip phones over adsl connections access our asterisk pbx and make out calls, currently we are using a zaptel pri interface for outdialing. What is the right way to manage dtmf over pstn lines and still retain low bandwith occupation ? In other words: if I use g729 (and sip info dtmf) for sip phones - asterisk communication will asterisk be able to regenerate real tones when going out to the pstn ? Tnx for any help ... currently I havent got g729 licenses so I cant test it out by myself. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] freebsd?
On Wed, Apr 14, 2004 at 04:56:46PM -0700, Randy Bush wrote: the freebsd port tree version is dead because of the openh323 issues. before i start hacking, i am hoping someone else has a freebsd version that will build on -current. and i do not care about h232. Just comment out the line with FORBIDDEN= in the port Makefile. (You will need to do it for pwlib and openh323 as well, if I recall correctly). You can then make install it in the usual fashion. If you are worried about the H323 security issues then I guess you will need to do some hacking or reconfiguring to get rid of it. Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP Phone Recommendations
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith AS for the WiSIP, STAY WELL AWAY!! I have 3 and they all started to exhibit the same problem after only 4 weeks out of the box. They become deaf and consistantly miss the call setup. The processor in the phone is very slow which can be demonstrated by the painfullnes endured when navigating the menu's or configuring its web interface. None of the features that Pulver claims work (hold, transfer, call waiting, second line etc) and he will not respond to any critisism of the product. You can forget trying to talk to their tech support too. She sucks. I second this, although I don't have any issues with speed -- Mind you I am not using any kind of encryption, which I hear really bogs it down. Hold works fine for me. No transfer, call waiting or second line though. I too have had no success in getting phone calls or email responded to. The volume is very low, even when cranked up. The standy time sucks ass, although I consistently get 3h of talk time out of it. Both the display contrast and the display backlight are substandard, IMO. I'm gonna dump this phone on ebay and try one of the other wireless SIP phones. I third this. Functionally it's a terrible implementation and not something I would ever give to users. Anyone want to buy it at a low, low price? I'll think twice before buying another Pulver product. Regards, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP Phone Recommendations
We are currently integration testing the wireless Zyxel Prestige 2000W, and if all goes well we'll have it for sale in 2 weeks. Has anyone any experience of this SIP device and asterisk? Tan telappliant.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adams, Gavin Sent: 15 April 2004 12:28 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP Phone Recommendations -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith AS for the WiSIP, STAY WELL AWAY!! I have 3 and they all started to exhibit the same problem after only 4 weeks out of the box. They become deaf and consistantly miss the call setup. The processor in the phone is very slow which can be demonstrated by the painfullnes endured when navigating the menu's or configuring its web interface. None of the features that Pulver claims work (hold, transfer, call waiting, second line etc) and he will not respond to any critisism of the product. You can forget trying to talk to their tech support too. She sucks. I second this, although I don't have any issues with speed -- Mind you I am not using any kind of encryption, which I hear really bogs it down. Hold works fine for me. No transfer, call waiting or second line though. I too have had no success in getting phone calls or email responded to. The volume is very low, even when cranked up. The standy time sucks ass, although I consistently get 3h of talk time out of it. Both the display contrast and the display backlight are substandard, IMO. I'm gonna dump this phone on ebay and try one of the other wireless SIP phones. I third this. Functionally it's a terrible implementation and not something I would ever give to users. Anyone want to buy it at a low, low price? I'll think twice before buying another Pulver product. Regards, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Booting error - Unable to specify channel 2: No such device
On Monday 12 April 2004 01:34 am, Dave Cotton wrote: On Mon, 2004-04-12 at 03:39, Anon wrote: On Sunday 11 April 2004 07:18 pm, Todd Lieberman wrote: do a cat /proc/interupts your should see your hardware showup. OK... cat /proc/interrupts CPU0 0: 494600 XT-PIC timer 1: 5588 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 9: 266296 XT-PIC ehci_hcd, es1371 10:4892925 XT-PIC usb-uhci, wctdm 11:4936093 XT-PIC ide1, usb-uhci, usb-uhci, wcfxo, eth0 12: 108609 XT-PIC PS/2 Mouse 14: 22056 XT-PIC ide2 15: 83 XT-PIC ide3 NMI: 0 LOC: 494575 ERR: 1278 MIS: 0 I don't see wcfxs. I did modprobe wcfxs and cat /proc/interrupts still shows the same output. What do you think is causing to not show on the list above? It is there it's hiding under the name of wctdm, but at the moment I'd say that's the least of your problems. Interrupts 10 and 11 both have other things sharing the interrupts with Digium cards, most people will tell you that that's not good. Interrupt 11 looks horrible. Having been through this hell recently I can only suggest repositioning cards in different slots. Interestingly where is ide0 and do you really need ide2/3? I thought about all those problems you mentioned, and it turns out I have an older version of the TDM400P that had some kind of defect, and needed to be sent back to Digium. While on the topic of Digium support, I would like to say that I am extremely impressed with the unusually high quality service at Digium. To my suprise and amazement, Digium shipped out a card before I sent mine back. Truely _first class_ service. BTW - I do use IDE 2/3. I do not know why the BIOS loads up the IRQ's the way it does. Yer, there does not seem to be any IRQ-related problems. Thanks for the help and ideas. :) Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Booting error - Unable to specify channel 2: No such device
On Monday 12 April 2004 04:24 am, [EMAIL PROTECTED] wrote: Thanks. :) When my *NEW* TDM400P card gets here, I will use the setup you described and see what happens. I find your suggestion a bit curious, because the CVS version I am running (Asterisk CVS-04/10/04-21:44:51) automatically loads all the modules. The automatic loading suprised me a bit, as the Asterisk version I was used to (8 moths old now) did not automatically load the modules. Does your version of Asterisk automatically load the modules? What version are you using? Anon Hi .. This may seem odd, but this problem is reminescent of the troubel I had when first starting [*] on my test setup. Two suggestions. (a) on loading the modules make sure you explicitely load in the following order (forget modprobe) insmod zaptel; insmod wcfxo; insmod wcfxs; Do a ztcfg -vvv to confirm start [*]. (b) in my zapata.conf, I foumd it useful to put group=1 on top for the FXO channel and group=2 for FXS, i.e. see below: ; ; Zapata telephony interface ; Configuration file [channels] context=inbound-calls language=en musiconhold=default group=1 signalling=fxs_ks echocancel = 64 echocancelwhenbridged = no echotraining=yes rxgain = 20% txgain = -5% channel = 1 group=2 echocancel = no signalling=fxo_ks mailbox = 2100 channel = 2 -- Now it works every time. Hope this helps Willy - Original Message Follows - On Mon, 2004-04-12 at 03:39, Anon wrote: On Sunday 11 April 2004 07:18 pm, Todd Lieberman wrote: do a cat /proc/interupts your should see your hardware showup. OK... cat /proc/interrupts CPU0 0: 494600 XT-PIC timer 1: 5588 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 9: 266296 XT-PIC ehci_hcd, es1371 10:4892925 XT-PIC usb-uhci, wctdm 11:4936093 XT-PIC ide1, usb-uhci, usb-uhci, wcfxo, eth0 12: 108609 XT-PIC PS/2 Mouse 14: 22056 XT-PIC ide2 15: 83 XT-PIC ide3 NMI: 0 LOC: 494575 ERR: 1278 MIS: 0 I don't see wcfxs. I did modprobe wcfxs and cat /proc/interrupts still shows the same output. What do you think is causing to not show on the list above? It is there it's hiding under the name of wctdm, but at the moment I'd say that's the least of your problems. Interrupts 10 and 11 both have other things sharing the interrupts with Digium cards, most people will tell you that that's not good. Interrupt 11 looks horrible. Having been through this hell recently I can only suggest repositioning cards in different slots. Interestingly where is ide0 and do you really need ide2/3? -- Dave Cotton Directeur Linux Autrement 193 rue Marcel Cerdan 84270 Vedene 04 90 23 30 81 http://www.linuxautrement.com IAX 17004902330 FWD 42651 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP Phone Recommendations
It is the same product as listed below, with a different firmware The firmware does exhibit similar problems Jason At 12:42 15/04/2004 +0100, you wrote: We are currently integration testing the wireless Zyxel Prestige 2000W, and if all goes well we'll have it for sale in 2 weeks. Has anyone any experience of this SIP device and asterisk? Tan telappliant.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adams, Gavin Sent: 15 April 2004 12:28 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP Phone Recommendations -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith AS for the WiSIP, STAY WELL AWAY!! I have 3 and they all started to exhibit the same problem after only 4 weeks out of the box. They become deaf and consistantly miss the call setup. The processor in the phone is very slow which can be demonstrated by the painfullnes endured when navigating the menu's or configuring its web interface. None of the features that Pulver claims work (hold, transfer, call waiting, second line etc) and he will not respond to any critisism of the product. You can forget trying to talk to their tech support too. She sucks. I second this, although I don't have any issues with speed -- Mind you I am not using any kind of encryption, which I hear really bogs it down. Hold works fine for me. No transfer, call waiting or second line though. I too have had no success in getting phone calls or email responded to. The volume is very low, even when cranked up. The standy time sucks ass, although I consistently get 3h of talk time out of it. Both the display contrast and the display backlight are substandard, IMO. I'm gonna dump this phone on ebay and try one of the other wireless SIP phones. I third this. Functionally it's a terrible implementation and not something I would ever give to users. Anyone want to buy it at a low, low price? I'll think twice before buying another Pulver product. Regards, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk in pass-thru mode
Hi all, Below is what I did to run Asterisk in pass-thru mode: sip.conf: [general] disallow=all allow=ulaw canreinvite=yes For each channel, canreinvite=yes is enabled. No dial command has 't' option. However, it seems that Asterisk still stay in the media path and bridge the 2 end points.Am I missing something??? sip*CLI show channels Channel (Context Extension Pri ) State Appl. DataSIP/5001-c60b (company1 1 ) Up Bridged Call SIP/1234-faf1 SIP/1234-faf1 (company1 5001 1 ) Up Dial SIP/5001|20|r2 active channel(s) sip*CLI sip show channelsPeer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format192.168.1.101 5001 257684717aa 00104/0 0ms ms ULAW210.17.211.51234 003094c2-fd 00104/00102 0ms ms ULAW2 active SIP channel(s) Thanks. Ben
Re: [Asterisk-Users] Voicemail Question
On Monday 12 April 2004 06:49 am, Andrew Thompson wrote: Paul Tyreman wrote: du -sh /var/spool/asterisk/vm/* At the command line, do man du You will have to know a bit about the operating system, this is not point and click. John Chapman Yeah ok, I know I need to type it at the command prompt, I'm not stupid. I just wanted to know what it did before I actually typed it in to my system !! Allow me to reiterate the suggestion of: man du *Hint man is short for manual. If you use KDE, you can go to the location bar, clear it, and type #du[ENTER] You will get this: DU Section: User Commands (1) NAME du - estimate file space usage SYNOPSIS du [OPTION]... [FILE]... DESCRIPTION Summarize disk usage of each FILE, recursively for directories. Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Booting error - Unable to specify channel 2:
On Monday 12 April 2004 09:39 am, randulo wrote: Subject: Re: [Asterisk-Users] Booting error - Unable to specify channel 2: No such device I just went through all this as well. The best thing to do IMHO is to try to find a way to manually assign IRQ in the BIOS. Also, and this is what I didn't see at first, some slots SHARE IRQ. Avoid this! If you are not using USB at all, turn it off in BIOS if possible. Tell the BIOS NOT INSTALLED for any peripherals not installed. I was able to remove parallel and serial interfaces as well since nothing is connect to this box. Here's what I have now: 0: 27713309 XT-PIC timer 1:167 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 277073215 XT-PIC wctdm -- TDM410 4:1114724 XT-PIC eth0 7: 277082652 XT-PIC wcfxo -- X100P 8: 1 XT-PIC rtc 9: 277077218 XT-PIC wcfxo -- X100P 12: 0 XT-PIC PS/2 Mouse 14: 102780 XT-PIC ide0 15: 3 XT-PIC ide1 hth Thank you. Unfortunately, I do not have * running on a box dedicated to it (for now). So, I do need to use USP, serial, and the printer port. I wonder though, if my install of * works well while sharing IRQ's, is there really any compelling reason to get the Zap cards on their own IRQ's? Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk systems
On Monday 12 April 2004 10:19 am, James Moran wrote: Is anyone selling asterisks systems?? Just wanting to know if it's profitable to try and start selling them. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Go here: http://www.digium.com/index.php?menu=resellers Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk + Fritz!PCI + CAPI
Hello,
The ONLY issue I have is that I don't get ringing dialback so
calling out gives a silence until the other party picks up
Have you turned on early B3?
S,1,Dial,CAPI/12345678:b${EXTEN}|30 (always early B3)
(plus the recent changes to locks in * required a tweak to the
chan_capi source to match).
could you post this tweak here? I'm stuck with CVS-STABLE ;-(
Regards,
Andreas
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Re: [Asterisk-Users] Re: Booting error - Unable to specify channel 2:
I just went through all this as well. The best thing to do IMHO is to try to find a way to manually assign IRQ in the BIOS. Also, and this is what I didn't see at first, some slots SHARE IRQ. Avoid this! If you are not using USB at all, turn it off in BIOS if possible. Tell the BIOS NOT INSTALLED for any peripherals not installed. I was able to remove parallel and serial interfaces as well since nothing is connect to this box. Here's what I have now: 0: 27713309 XT-PIC timer 1:167 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 277073215 XT-PIC wctdm -- TDM410 4:1114724 XT-PIC eth0 7: 277082652 XT-PIC wcfxo -- X100P 8: 1 XT-PIC rtc 9: 277077218 XT-PIC wcfxo -- X100P 12: 0 XT-PIC PS/2 Mouse 14: 102780 XT-PIC ide0 15: 3 XT-PIC ide1 hth Thank you. Unfortunately, I do not have * running on a box dedicated to it (for now). So, I do need to use USP, serial, and the printer port. I wonder though, if my install of * works well while sharing IRQ's, is there really any compelling reason to get the Zap cards on their own IRQ's? No, there is no compelling reason at all. Some folks have had IRQ sharing problems, but the majority do not. Yet, the few keep repeating it as though its a major issue for everyone, which is not even close to the truth. Here's one example... 9: 871579742 XT-PIC ehci-hcd, eth0, wcfxo, Intel ICH4 10: 0 XT-PIC usb-uhci 11: 3631482514 XT-PIC usb-uhci, wcfxo where the same interrupt is shared by several devices with absolutely no problems whatsoever. (Notice the 100 meg nic card is sharing with a x100p card, etc.) The bottom line for sharing interrupts involves having some technical knowledge as to which devices actually use interrupts in the first place (even though an interrupt might be allocated to a card, doesn't mean its actually used for anything), the ability of the cards and drivers to handle sharing, motherboard, etc. Without that knowledge, it boils down to simply trying it. If the cards work at all, there is a high probability they will continue to work without messing around with that stuff. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls to Cisco PSTN gateway
Make sure you don't have videosupport=yes in sip.conf when using as5300. I found mine doesn't like that much got that codec error. Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Radius Sent: Thursday, April 15, 2004 2:37 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Calls to Cisco PSTN gateway Hi all, A Cisco ATA186 configured with g711ulaw, NAT=yes and canreinvite=yes, made calls through Asterisk to a Cisco 5300 gateway out to a PSTN line with errors as follows: -- Executing Dial(SIP/ata186-c1cf, SIP/[EMAIL PROTECTED]:5060|30|r) in new stack -- Called [EMAIL PROTECTED]:5060 mailto:[EMAIL PROTECTED]:5060 Apr 15 16:11:22 WARNING[1116941120]: chan_sip.c:2049 process_sdp: Error in codec string 'ideo 0 ' Asterisk was configured with allow=ulaw. Any idea for this problem?? Thanks. Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMailBox wav file format in EMAIL.
On Tuesday 13 April 2004 12:12 am, James Gardiner wrote: Hi all, I am not sure if tis is a bug but.. Was learning about VM etc to see how it all worked, and I noticed the following.. In the default install, the VM system leaves 3 different copies of the Voice message. Size filename 13kb Msg.gsm 13kb Msg.wav 122kb Msg.WAV - under UNIX we have case sensitive file names of course. I wanted to have a look at these files so loaded them into SOUND FORGE 6. This first thing I noticed was that the LARGER file is of much HIGHER volume. Like it had been normalised to 100% The smaller was file, when loaded into sound forge, did not load properly, only the first 2 seconds loads. Can anyone explain these issues and why they exist? All in all, I was wondering what would be the best format for best quality but with still great compression. I want to archive all calls for a period of time with self expire. (For example dedicate 5 gig disk space to the last number of calls that can fit in the 5gig.) I want to store the best quality possible but also make best use of disk space, so I can store for even longer periods. I was considering ogg but then is occurred to me that GSM or other codecs designed for audio with this frequency response may be better. (But the GSM file above is not as clear as the WAV ones produced.) I was also wondering if the VM system when emailing the audio can be setup to use something like ogg or MP3? Comments appreciated, James Gardiner That is a very interesting observation I had not seen yet. There are certainly a few ways to do this. Here is an idea: run a cron job that downsamples the WAV file using sox, then compresses it, and deletes the redundant files. Make sure to have the cron job run during the least-busy part of the day. Yes, it is a crude solution. ;) Yet it may get you by until you get a better solution. Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * Announcement * Astricon 2004 - call for speakers!
We're proud to announce Astricon 2004 - the first Asterisk user's and developer's conference! * Where? Atlanta, USA * When? September 22-24, 2004 The conference is arranged in partnership with Digium.inc and the keynote speaker is Mark Spencer, lead developer of Asterisk - the Open Source PBX. Among the speakers already signed on are Ed Guy of Pulver.com, John Todd, Jeremy McNamara (NuFone) and collegues from the SIP Foundry Open Source project. Main topics: * Integrating the PBX with the IT infrastructure: Asterisk for the Enterprise * VOIP migration in-a-box: Asterisk for Service providers * Lower cost, more flexibility: Asterisk for Call Centers * Your VoIP Swiss Army Knife: Asterisk for developers * Managing your Asterisk PBX: from the CLI to the GUI Agenda in brief: * Wednesday: Tutorials - in depth sessions held by VoIP and Asterisk gurus Tutorials will be arranged both for newbies and pro's * Thursday: Conference and exhibition * Friday: Asterisk developer's meeting Early bird registration will start soon at discounted rates on the web site, http://www.astricon.net We're now in the process of setting up the agenda and are looking for speakers and sponsors. Send a tutorial or speaker's proposal to [EMAIL PROTECTED] including * A subject * A brief description (five-six lines) * Target group (if tutorial) * Name and contact information * A digital picture of yourself (for the conference web) We need proposals no later than april 30, 2004. You may of course also propose other speakers than yourself :-) If you're working for a company that sells Asterisk-related products and services, there's an oppurtunity to show your products and sponsor the event. Contact us at [EMAIL PROTECTED] for more information. Looking forward to meeting you all in Atlanta! Steven SokolOlle E. Johansson [EMAIL PROTECTED] [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [semi-OT] Channelbanks for european market / Alternatives
Hello list, does anyone know if there are channelbanks homologated for the eruopean/german market, i.e. labeled with the socalled CE certificate? So far, I know that the products from Adtran (TA750 and TSU600) and Carrieraccess (Adit 600) are not CE-labeled, but I have no alternatives when it comes to terminating 12 128 FXS ports. [It has been already discussed somewhere on the list: what number of TDM40Bs in a box would you consider safe regarding the limitations of the PCI bus / interrupts and the power supply? I have something like maximum three cards in mind...] Or am I deadlocked in the idea that this has to be done via multiple E1/T1 trunks terminated to FXS by a channelbank? Are there other approaches imaginable? TIA, Tobias F. Leucht ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [semi-OT] Channelbanks for european market / Alternatives
Tobias F. Leucht wrote: does anyone know if there are channelbanks homologated for the eruopean/german market, i.e. labeled with the socalled CE certificate? So far, I know that the products from Adtran (TA750 and TSU600) and Carrieraccess (Adit 600) are not CE-labeled, but I have no alternatives when it comes to terminating 12 128 FXS ports. Or am I deadlocked in the idea that this has to be done via multiple E1/T1 trunks terminated to FXS by a channelbank? Are there other approaches imaginable? If you are not connecting the channel bank to the telco, then you should not need the Euro certification for telco stuff, just the certification for EMI/RFI. i.e. if you are connecting an E-1 into Asterisk on one port of a 4 port Digium card, then you should be able to connect pretty much any E-1 OR T-1 channel bank to another port of the card for your FXS ports. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Warning message
Does anyone know what this means Warning [65542]: chan_sip:c:501 retrans_plct: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno102 (Non-critical Request. 172.16.0.52 is the Asterisk Server I'm guessing that I have something miss configured just not sure what it is. If you need more info just ask. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] onhold bug?
I'm running the latest version of cvs (not stable), I'm not sure what the other end is running and if this has been fixed or not yet, however I was playing round with onhold earlier, the call went to onhold, and came back from it, then 2 seconds later was hung up unexpectedly, below is what was on console... -- Started music on hold, class 'default', on IAX2[x.x.x.x:4569]/1 -- Stopped music on hold on IAX2[x.x.x.x:4569]/1 Apr 15 23:46:33 WARNING[458768]: chan_iax2.c:2723 iax2_send: timestamp is 0? -- Hungup 'IAX2[x.x.x.x:4569]/1' -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Registering Asterisk to Lucent's MVAM Gatekeeper
Hi there, I am trying to register asterisk to a Lucent's MVAM GK. It is not registering to the GK with both h323 channels(chan_h323 and oh323). The problem is that if I set the GK(in asterisk) through the GK's IP, the GK answers with GCF without it's GK ID, and after that it does not answer to the RRQ, because the RRQ message has no GK ID. And when I set the GK through the GK ID, asterisk does not respond to the GK's GCF message because there is no GK ID in this message. What I would need to do is to set GK's ID and IP at the same time in asterisk. Is this possible? I've tried setting both: id and IP, for the oh323 and it did not work. Asterisk would assume that I had only set the GK's id. And for the chan_h323, it isn't possible to set GK ID inside the h323.conf file. I am really in a hurry here. Could somebody help me? thanks in advance, Pablo Salinas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] freebsd?
On Wed, Apr 14, 2004 at 04:56:46PM -0700, Randy Bush wrote: the freebsd port tree version is dead because of the openh323 issues. before i start hacking, i am hoping someone else has a freebsd version that will build on -current. and i do not care about h232. dare i hope? randy make install -DNO_IGNORE I'm also working on a freebsd port that uses the cvs version of asterisk, let me know if you're interested in taking a look. http://something.inethouston.net/~dwcjr/asterisk.patch -- David W. Chapman Jr. [EMAIL PROTECTED] Raintree Network Services, Inc. www.inethouston.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Announcement * Astricon 2004 - call for speakers!
eh, very good idea... but how about for alaw people? Any plans to make another conference in EU world? Matteo. P.S. unfortunately I cannot join... too much money for me. Il gio, 2004-04-15 alle 15:16, Olle E. Johansson ha scritto: We're proud to announce Astricon 2004 - the first Asterisk user's and developer's conference! * Where? Atlanta, USA * When? September 22-24, 2004 The conference is arranged in partnership with Digium.inc and the keynote speaker is Mark Spencer, lead developer of Asterisk - the Open Source PBX. Among the speakers already signed on are Ed Guy of Pulver.com, John Todd, Jeremy McNamara (NuFone) and collegues from the SIP Foundry Open Source project. Main topics: * Integrating the PBX with the IT infrastructure: Asterisk for the Enterprise * VOIP migration in-a-box: Asterisk for Service providers * Lower cost, more flexibility: Asterisk for Call Centers * Your VoIP Swiss Army Knife: Asterisk for developers * Managing your Asterisk PBX: from the CLI to the GUI Agenda in brief: * Wednesday: Tutorials - in depth sessions held by VoIP and Asterisk gurus Tutorials will be arranged both for newbies and pro's * Thursday: Conference and exhibition * Friday: Asterisk developer's meeting Early bird registration will start soon at discounted rates on the web site, http://www.astricon.net We're now in the process of setting up the agenda and are looking for speakers and sponsors. Send a tutorial or speaker's proposal to [EMAIL PROTECTED] including * A subject * A brief description (five-six lines) * Target group (if tutorial) * Name and contact information * A digital picture of yourself (for the conference web) We need proposals no later than april 30, 2004. You may of course also propose other speakers than yourself :-) If you're working for a company that sells Asterisk-related products and services, there's an oppurtunity to show your products and sponsor the event. Contact us at [EMAIL PROTECTED] for more information. Looking forward to meeting you all in Atlanta! Steven SokolOlle E. Johansson [EMAIL PROTECTED] [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] external voicemail access - solved (mostly)
thanks to those who replied. I have managed to get the functionality I
was looking for working with a series of Macros. However, it doesn't
work as simply as I would like. There are two issues I've run into:
(1)Goto provides no way to pass variables between one context and
another.
(2)I can't find any way to Goto a specific point within a Macro when
calling it.
Mostly this is a result of the background command listening for
extensions in the current context. If background is run from within a
Macro, then it will terminate the macro and return to the current
context to execute whatever user input was just captured. In order to
get the behavior I'm looking for (user calls into voicemail, presses *
to be prompted for a password and check messages, press # to skip the
greeting and leave a message) I had to have 3 macros:
(1)vm - leave voicemail
(2)vm-nogreet - simply provide a beep
(3)checkmessage
Ideally this would be one larger macro, where the starting point could
be specified as well as passing the arguments along.
[macro-vm]
exten = s,1,Answer
exten = s,2,Background(${VMAILPATH}/${ARG2}/${ARG1}/unavail)
exten = s,3,VoiceMail2(s${ARG1})
exten = s,4,Hangup
[macro-vm-nogreet]
exten = s,1,Answer
exten = s,2,VoiceMail2(s${ARG1})
exten = s,3,Hangup
[macro-checkmessage]
exten = s,1,VoiceMailMain2(${ARG1})
exten = s,2,Hangup
In the inbound context I do the following (xxx is the rest of the
phone number):
[line-in]
exten = xx1638,1,Dial(${P1}${P2}${P3},25,Tr)
exten = xx1638,2,Macro(vm,${P1_VM},${P1_VM_CONTEXT})
exten = xx1638,3,Hangup
exten = *,1,Macro(checkmessage,${P1_VM})
exten = i,1,Macro(vm-nogreet,${P1_VM},${P1_VM_CONTEXT})
This does everything in a fairly general way. However, if anyone knows
how to address my points above this could be done much cleaner in a
single macro (and if there is a way to keep background isolated within
the macro it would be easier still). One note - this will actually
interrupt the greeting and send to the beep for voicemail regardless
of what the user presses, as long as it isn't the * key. Since
background will always bounce to here I thought it would be better to
force someone to leave voicemail than get a fast busy with an
inadvertent button press.
-Steve
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[Asterisk-Users] T1 Line install.. (UK Muppet)
Hi all, Muppet from the UK asking for help We are just about to have a T1 line installed in our office in Dallas and Advantex the supplier has sent a questionnaire asking a number of questions. I have put the question area at the bottom of the email, we will be using Digium's hardware. could anybody help :-) In the UK when I asked for a E1, number of trunks required and the number of DID numbers, then they stick the socket on the wall! No mass question... I feel I'm a little out of my depth. Thanks in Advance for any help. David Stubbs = Questionnaire Follows = 1. DS1 Facilities Required: - Superframe or Extended (ESF/SF)? - Line Coding (B8ZS,AMI,. Etc.)? - Number of DS1 facilities required? - Channel numbers used for this trunk group? 2. Circuit Signaling (DS0 or DS1 Signal): DSO Analog Line Trunk Interface 2 wire, 4 wire, (or na, not applicable)? - Supervision Loop Start, Ground Start, Reverse Battery, E M DNIS - Dialed Number Identification service Not applicable with DS0 Analog Line trunks. DS1 Digital Signal : Digital Interface - Supervision Loop Start, Ground Start, E M - In-Band or Robbed Bit (EM) signaling? (Y/N) - CO Supplied Dial tone (Y/N) DS1 Digital Signal: ISDN PRI Only - Primary Rate Interface (ISDN PRI) - If multiple DS1s, Does the customer want Non-Facility Associated Signaling? (NFAS) (Y/N) If No, Does the customer want One D-Channel per DS1, or 1 D-Channels for all DS1s. How many D-Channels? If NFAS=Yes, Number of D-Channels Required TWO - (PRI ISDN Protocol)= example National PRI (NIPRI) NIPRI, NTNAPRI, U449PRI, U459PRI, N449PRI 3. Trunk Group Billing Telephone Number (10 digit): 4. Trunk Group Originating Area Code for PBX 5. Calling Scope (Basic, Basic+, Extended, Metro, Metro+)? 6. Primary Interexchange Carrier (PIC) 7. NPA/NXX Call Blocking ( 700, 900, 976) 8. Trunk Group Directional call flow and Member numbering: (If sub-grouped, select subgroup rollover) - 2-Way, 1-Way Into PBX, or 1-Way Out of PBX - Member Numbering (i.e. 0001 thru 0024) 9. Trunk Member selection - Selection for calls originating at the PBX toward the Central Office (CO) Asending (ASEQ), Desending (DSEQ), Most Idle (MIDL), Least Idle (LIDL) - Selection for calls terminating to the PBX from the Central Office (CO) Asending (ASEQ), Desending (DSEQ), Most Idle (MIDL), Least Idle (LIDL) 10. Direct Inward Dial to PBX - DID (Y/N): - Number or Number Blocks routed toward PBX (i.e. 972-668-20xx) - Porting Coordination Required (Y/N)? - Number of digits to outpulse (None, 4, 7, 10 digits) to PBX? 11. Inband Ringing required from the CO when terminating calls to PBX (Y/N)? 12. NON-PRI DS1Switching - Start Signaling and Digit Pulsing for In-Band Signaling for NON-PRI DS1: - CO Incoming Start Signaling for calls originating at the PBX (ISTARTSG) Start Signal expected by the PBX, sent by the CO telling the PBX to begin sending digits to the CO for collection. - Wink Start, Delay Dial, Immediate Dial, Dialtone, Loop Start, Grd Start - CO Incoming Digit Pulsing for calls originating at the PBX (IPULSTYP) Type of digits sent from the PBX to CO after the PBX receives the start signal (ISTARTSG) from the CO - Dial Pulse, DT/ DTMF, MF, NP/No Pulsing - CO Outgoing Start Signaling for calls terminating to the PBX (OSTARTSG) Start Signal expected by the CO, sent by the PBX telling the CO to begin sending digits to the PBX for collection. - Wink Start, Immediate Dial, Delay Dial, Loop Start, Grd Start - CO Outgoing Digit Pulsing for calls terminating to the PBX (OPULSTYP) Type of digits sent from the CO to the PBX after the CO receives the start signal (OSTARTSG) fom the PBX - Dial Pulse, DT/ DTMF, MF, NP/No Pulsing 13. Manufacturer Model of PBX. *** I know this One :o) *** END OF Questionnaire == ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] t1 won't dial outbound
Title: Message
I've posted this problem a couple of times before with
little or no response. Basically I have a T100P in my * box.
Incoming calls are working great. However outgoing calls are not working
at all. I've copied a previous post into this message which should have
all the necessary info. Any ideas or suggestions would be greatly
appreciated. Thanks.
Mark
#
OK...I've got an *
box with a T100P in it. For the most part incoming calls are going through
just fine. Outgoing calls, however, I'm having some more trouble
with. Whenever I make an outgoing call, the call begins, however after the
dialing process all I hear is dead air. Here's the output from my *
console:
-- Executing
Dial("SIP/mark-2d08", "Zap/g3/2550559") in new stack --
Called g3/2550559 -- Hungup 'Zap/6-1' == Spawn
extension (uti-mainst, 2550559, 1) exited non-zero on
'SIP/mark-2d08'
I've checked with
the switch guy...and whatever channel I'm trying to dial out on is coming up as
"blocked" on his switch. We've compared as many settings as we can think
of and they all seem to be set the same. I'll post the entries from my
zaptel.conf and my zapata.conf in here...if you have any ideas please send them
my way...
zaptel.conf
span=1,1,0,d4,amiem=1-24fxsks=25loadzone=usdefaultzone=us
zapata.conf
context=conferencesignalling=emswitchtype=5essgroup=3callgroup=3pickupgroup=3channel
= 6
busydetect=yescallerid=asreceivedcallprogress=yescallreturn=yescallwaiting=yescallwaitingcallerid=yescancallforward=yesechocancel=yesechocancelwhenbridged=yesimmediate=nolanguage=usmusiconhold=defaultthreewaycalling=yestransfer=yesusecallerid=yes##
Re: [Asterisk-Users] Dropped calls
I see this very same effect rather often in the following setup: SIP (GS101) -- * -- IAX2 -- * -- MGCP (ip10) In fact I think I've seen it also with SIP instead of MGCP at the end. The first client is behind NAT, by the way. That must be it. I have seen this happening with sip -- * -- IAX as well. I take it you don't know a cure? Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VOIP Spam
Hi, Some people have suggested maintaining black lists and white lists to avoid spammers and allow legitimate callers into the network. However, the problem with this method is that the spammer's IP address might change due to DHCP. Today a spammer might get aaa.bbb.ccc.ddd and lets say that I put this address in my blacklist. To my annoyance, tomorrow a legitimate caller might get aaa.bbb.ccc.ddd and the spammer might get a different IP address. In the end, I end up blocking the legitimate caller also. Any ideas or thoughts to on this problem is appreciated. Thanks, Tom __ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] freebsd?
make install -DNO_IGNORE h, scary considering i don't need h323. or am i misunderstanding something? I'm also working on a freebsd port that uses the cvs version of asterisk, let me know if you're interested in taking a look. o! but i am about to go back on the road. so i don't know if i will have time this week. randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP Spam
Tom Green wrote: Hi, Some people have suggested maintaining black lists and white lists to avoid spammers and allow legitimate callers into the network. However, the problem with this method is that the spammer's IP address might change due to DHCP. Today a spammer might get aaa.bbb.ccc.ddd and lets say that I put this address in my blacklist. To my annoyance, tomorrow a legitimate caller might get aaa.bbb.ccc.ddd and the spammer might get a different IP address. In the end, I end up blocking the legitimate caller also. Any ideas or thoughts to on this problem is appreciated. Thanks, Tom __ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yeah, for a relatively modern protocol SIP has some surprisingly glaring omissions, such as: - certificate based authentication - encryption - NAT-awareness -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Whats the best audio compresion format for the following?
All in all, I was more hoping to get some words of wisdom from the more worldly Audio Compression experienced people in regard of the question below about what is the best way to store audio recorded with asterisk. Ie, to keep the BEST possible quality asterisk can record but still getting great compression. And not having to use any real time compression formats. (mp3, ogg, etc) Thanks, James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anon Sent: Thursday, 15 April 2004 11:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoiceMailBox wav file format in EMAIL. On Tuesday 13 April 2004 12:12 am, James Gardiner wrote: Hi all, I am not sure if tis is a bug but.. Was learning about VM etc to see how it all worked, and I noticed the following.. In the default install, the VM system leaves 3 different copies of the Voice message. Sizefilename 13kbMsg.gsm 13kbMsg.wav 122kb Msg.WAV - under UNIX we have case sensitive file names of course. I wanted to have a look at these files so loaded them into SOUND FORGE 6. This first thing I noticed was that the LARGER file is of much HIGHER volume. Like it had been normalised to 100% The smaller was file, when loaded into sound forge, did not load properly, only the first 2 seconds loads. Can anyone explain these issues and why they exist? All in all, I was wondering what would be the best format for best quality but with still great compression. I want to archive all calls for a period of time with self expire. (For example dedicate 5 gig disk space to the last number of calls that can fit in the 5gig.) I want to store the best quality possible but also make best use of disk space, so I can store for even longer periods. I was considering ogg but then is occurred to me that GSM or other codecs designed for audio with this frequency response may be better. (But the GSM file above is not as clear as the WAV ones produced.) I was also wondering if the VM system when emailing the audio can be setup to use something like ogg or MP3? Comments appreciated, James Gardiner That is a very interesting observation I had not seen yet. There are certainly a few ways to do this. Here is an idea: run a cron job that downsamples the WAV file using sox, then compresses it, and deletes the redundant files. Make sure to have the cron job run during the least-busy part of the day. Yes, it is a crude solution. ;) Yet it may get you by until you get a better solution. Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How many lines of IP phone can Asterisk support?
On Thursday 15 April 2004 01:44 am, PTCHEN wrote: Hello, I am an Asterisk beginer and are using it. Now I have 1 question, please ! How many lines of IP phone can Asterisk support, if we use only IP interface? Chunghwa Telecom BTA Tech. Lab. E-mail:[EMAIL PROTECTED] That will depend on many factors: - How fast is your Asterisk box? - How much memory your asterisk box has - How fast your ethernet card is - How fast your Internet connection is (bandwidth) - Which codec(s) you decide to use - What other programs/services you are running on the Asterisk box I think it is safe to say that if you have a decent, relatively new computer to run Asterisk on, you will find it able to support a large number of VoIP connections. Based on bandwidth alone, if you have a 640K Internet connection and your codec uses 8K, you could have at least 70 simultaneous connections. (If I am poking my head up my butt on this one, please correct me). Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP Spam
Brian Cuthie wrote: Yeah, for a relatively modern protocol SIP has some surprisingly glaring omissions, such as: - certificate based authentication - encryption - NAT-awareness I'd love nothing more to see some decent crypto in the IAX2 protocol, it already covers the third item on your list... Especially with government agencies so eager to get their mits into voip tapping, why make it any easier on them... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Booting error - Unable to specify channel 2:
On Thursday 15 April 2004 07:45 am, Rich Adamson wrote: I just went through all this as well. The best thing to do IMHO is to try to find a way to manually assign IRQ in the BIOS. Also, and this is what I didn't see at first, some slots SHARE IRQ. Avoid this! If you are not using USB at all, turn it off in BIOS if possible. Tell the BIOS NOT INSTALLED for any peripherals not installed. I was able to remove parallel and serial interfaces as well since nothing is connect to this box. Here's what I have now: 0: 27713309 XT-PIC timer 1:167 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 277073215 XT-PIC wctdm -- TDM410 4:1114724 XT-PIC eth0 7: 277082652 XT-PIC wcfxo -- X100P 8: 1 XT-PIC rtc 9: 277077218 XT-PIC wcfxo -- X100P 12: 0 XT-PIC PS/2 Mouse 14: 102780 XT-PIC ide0 15: 3 XT-PIC ide1 hth Thank you. Unfortunately, I do not have * running on a box dedicated to it (for now). So, I do need to use USP, serial, and the printer port. I wonder though, if my install of * works well while sharing IRQ's, is there really any compelling reason to get the Zap cards on their own IRQ's? No, there is no compelling reason at all. Some folks have had IRQ sharing problems, but the majority do not. Yet, the few keep repeating it as though its a major issue for everyone, which is not even close to the truth. Here's one example... 9: 871579742 XT-PIC ehci-hcd, eth0, wcfxo, Intel ICH4 10: 0 XT-PIC usb-uhci 11: 3631482514 XT-PIC usb-uhci, wcfxo where the same interrupt is shared by several devices with absolutely no problems whatsoever. (Notice the 100 meg nic card is sharing with a x100p card, etc.) The bottom line for sharing interrupts involves having some technical knowledge as to which devices actually use interrupts in the first place (even though an interrupt might be allocated to a card, doesn't mean its actually used for anything), the ability of the cards and drivers to handle sharing, motherboard, etc. Without that knowledge, it boils down to simply trying it. If the cards work at all, there is a high probability they will continue to work without messing around with that stuff. Rich Thank you very much for that useful perspective. :) Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] freebsd?
On Thu, Apr 15, 2004 at 08:38:32AM -0700, Randy Bush wrote: make install -DNO_IGNORE h, scary considering i don't need h323. or am i misunderstanding something? NO_IGNORE is going to bypass all of the forbidden lines for all of the dependencies -- David W. Chapman Jr. [EMAIL PROTECTED] Raintree Network Services, Inc. www.inethouston.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 Line install.. (UK Muppet)
In a way Europe does make this easy, at the expense of choice. These questionaires cover the gamut of standard analog lines to hi-cap service, and ISDN hi-cap. An E1 is almost (maybe always) ISDN, where a T1 is not, and a PRI is. Consider it a cultural quirk. Both are 24 channels, delivered on exactly the same telco gear. A T1 suggest inband signalling for each channel, and no advanced features (like caller id is advanced). A PRI is functionally identical to an E1, with a D channel for call control. So if you want a similar service to what you get with an E1, you want to order a ciruit whit these features: B8ZS ESF 24 channels PRI NFAS is no for a single span 1 D-Channel NIPRI (National signalling is common. Many other choices exist) 2-Way trunk (otherwise it is inbound only) DID (Number range provided by you carrier) Billing # (Number range provided by you carrier) PIC is your long distance carrier These are the key elements, and should get you going. Dan -Original Message- From: David Stubbs [mailto:[EMAIL PROTECTED] Sent: Thursday, April 15, 2004 7:37 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] T1 Line install.. (UK Muppet) Hi all, Muppet from the UK asking for help We are just about to have a T1 line installed in our office in Dallas and Advantex the supplier has sent a questionnaire asking a number of questions. I have put the question area at the bottom of the email, we will be using Digium's hardware. could anybody help :-) In the UK when I asked for a E1, number of trunks required and the number of DID numbers, then they stick the socket on the wall! No mass question... I feel I'm a little out of my depth. Thanks in Advance for any help. David Stubbs = Questionnaire Follows = 1. DS1 Facilities Required: - Superframe or Extended (ESF/SF)? - Line Coding (B8ZS,AMI,. Etc.)? - Number of DS1 facilities required? - Channel numbers used for this trunk group? 2. Circuit Signaling (DS0 or DS1 Signal): DSO Analog Line Trunk Interface 2 wire, 4 wire, (or na, not applicable)? - Supervision Loop Start, Ground Start, Reverse Battery, E M DNIS - Dialed Number Identification service Not applicable with DS0 Analog Line trunks. DS1 Digital Signal : Digital Interface - Supervision Loop Start, Ground Start, E M - In-Band or Robbed Bit (EM) signaling? (Y/N) - CO Supplied Dial tone (Y/N) DS1 Digital Signal: ISDN PRI Only - Primary Rate Interface (ISDN PRI) - If multiple DS1s, Does the customer want Non-Facility Associated Signaling? (NFAS) (Y/N) If No, Does the customer want One D-Channel per DS1, or 1 D-Channels for all DS1s. How many D-Channels? If NFAS=Yes, Number of D-Channels Required TWO - (PRI ISDN Protocol)= example National PRI (NIPRI) NIPRI, NTNAPRI, U449PRI, U459PRI, N449PRI 3. Trunk Group Billing Telephone Number (10 digit): 4. Trunk Group Originating Area Code for PBX 5. Calling Scope (Basic, Basic+, Extended, Metro, Metro+)? 6. Primary Interexchange Carrier (PIC) 7. NPA/NXX Call Blocking ( 700, 900, 976) 8. Trunk Group Directional call flow and Member numbering: (If sub-grouped, select subgroup rollover) - 2-Way, 1-Way Into PBX, or 1-Way Out of PBX - Member Numbering (i.e. 0001 thru 0024) 9. Trunk Member selection - Selection for calls originating at the PBX toward the Central Office (CO) Asending (ASEQ), Desending (DSEQ), Most Idle (MIDL), Least Idle (LIDL) - Selection for calls terminating to the PBX from the Central Office (CO) Asending (ASEQ), Desending (DSEQ), Most Idle (MIDL), Least Idle (LIDL) 10. Direct Inward Dial to PBX - DID (Y/N): - Number or Number Blocks routed toward PBX (i.e. 972-668-20xx) - Porting Coordination Required (Y/N)? - Number of digits to outpulse (None, 4, 7, 10 digits) to PBX? 11. Inband Ringing required from the CO when terminating calls to PBX (Y/N)? 12. NON-PRI DS1Switching - Start Signaling and Digit Pulsing for In-Band Signaling for NON-PRI DS1: - CO Incoming Start Signaling for calls originating at the PBX (ISTARTSG) Start Signal expected by the PBX, sent by the CO telling the PBX to begin sending digits to the CO for collection. - Wink Start, Delay Dial, Immediate Dial, Dialtone, Loop Start, Grd Start - CO Incoming Digit Pulsing for calls originating at the PBX (IPULSTYP) Type of digits sent from the PBX to CO after the PBX receives the start signal (ISTARTSG) from the CO - Dial Pulse, DT/ DTMF, MF, NP/No Pulsing - CO Outgoing Start Signaling for calls terminating to the PBX (OSTARTSG) Start Signal expected by the CO, sent by the PBX telling the CO to begin sending digits to the PBX for collection. - Wink Start, Immediate Dial, Delay Dial, Loop Start, Grd Start - CO Outgoing Digit Pulsing for calls terminating to the PBX (OPULSTYP) Type of digits sent
[Asterisk-Users] Call transfer with sipura
I can transfer a call from my sipura using flash, *98 and number, the problem is If I hangup before the destination extension picks-up, the transfer is lost. Is there a way to transfer and hangup without having to wait for the destination extension to pickup? Regards, Victor Perez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Severe hum in recording
Hello, We've made thousands of recording on our Asterisk system, and we had one last night that ended up having a constant loud hum throughout the entire recording. None of the other recordings from last night have the hum and the caller said the conversation had no hum in it. Anyone have any ideas how the recording could've had such a loud hum when the calling parties didn't hear it? The call was a SIP - T1 channel call over a T400P Digium quad T1 card. Thanks, MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Announcement * Astricon 2004 - call for speakers!
On Thu, 2004-04-15 at 08:16, Olle E. Johansson wrote: We're proud to announce Astricon 2004 - the first Asterisk user's and developer's conference! * Where? Atlanta, USA * When? September 22-24, 2004 We're now in the process of setting up the agenda and are looking for speakers and sponsors. Do we need a presentation on how to behave on the list to avoid getting flamed by me, or should I just show up with an appropriate LART device to fix problem people during the normal presentations? Too many conferences I want to attend, but this one is a must attend. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail
Voicemail,from wiki pages: Returns -1 on error or mailbox not found, or if the user hangs up. Otherwise, it returns 0. Question is ... how can I trap the error code to divert to a message like This user is offline and has no mailbox, goodbye!. Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP Spam
Brian, Encrypted SIP messages can be sent using TLS. However, I don't think it is realistic to expect everyone calling you to have a public/private key pair. Cryptographic solutions have been suggested for email spams also but they have been found to be ineffective because of scalability problems. I looking for a spam control solution that avoids cryptography (I prefer non-cryptographic solutions but I am open to new ideas). Thanks, Tom. --- Brian Cuthie [EMAIL PROTECTED] wrote: Tom Green wrote: Hi, Some people have suggested maintaining black lists and white lists to avoid spammers and allow legitimate callers into the network. However, the problem with this method is that the spammer's IP address might change due to DHCP. Today a spammer might get aaa.bbb.ccc.ddd and lets say that I put this address in my blacklist. To my annoyance, tomorrow a legitimate caller might get aaa.bbb.ccc.ddd and the spammer might get a different IP address. In the end, I end up blocking the legitimate caller also. Any ideas or thoughts to on this problem is appreciated. Thanks, Tom __ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yeah, for a relatively modern protocol SIP has some surprisingly glaring omissions, such as: - certificate based authentication - encryption - NAT-awareness -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PC based Switchboard application
Im interested can you send information? Kyle [EMAIL PROTECTED] - Original Message - From: Pertti Pikkarainen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, April 10, 2004 2:26 AM Subject: Re: [Asterisk-Users] PC based Switchboard application We have switchboard application ( PC+browser+Java ) with quite a rich feature set. It talks to * via manager port. Works as a call center too. However, it is not open source. If you are interested in, please contact me directly. Best regards Pertti Keith D'Atrio wrote: Hello All I am looking for a PC based switchboard application. Cisco CallManager has a web attendant console that allows you to use the PC to transfer calls and the like and I was wondering if there was a similar program compatible with *. Thank you in advance Keith D'Atrio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] too many arguments to function `ast_queue_hangup' compiling asterisk-oh323
when trying to build asterisk-oh323 I get the following: make[1]: Entering directory `/usr/src/asterisk-oh323-0.5.10/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declara tions -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1128: error: too many arguments to function `ast_queue_hangup' chan_oh323.c:1142: error: too many arguments to function `ast_queue_hangup' chan_oh323.c: In function `oh323_hangup': chan_oh323.c:1182: error: too many arguments to function `ast_queue_hangup' chan_oh323.c: In function `oh323_read': chan_oh323.c:1581: error: too many arguments to function `ast_dsp_process' chan_oh323.c: In function `ast_oh323_new': chan_oh323.c:2030: warning: assignment from incompatible pointer type chan_oh323.c: In function `cleanup_h323_connection': chan_oh323.c:2835: error: too many arguments to function `ast_queue_hangup' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.5.10/asterisk-driver' make: *** [subdirs_all] Error 1 I checked chan_oh323.c and indeed it only takes one parameter now so I am wondering what was that old parameter for and when did they take it off so I may try pulling that version of asterisk to try with. Regards, Victor Perez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropped calls
Hi! I see this very same effect rather often in the following setup: SIP (GS101) -- *1 -- IAX2 -- *2 -- MGCP (ip10) In fact I think I've seen it also with SIP instead of MGCP at the end. The first client is behind NAT, by the way. That must be it. I have seen this happening with sip -- * -- IAX as well. I take it you don't know a cure? Unfortunately not, no. By the way I am not on latest CVS as that would disable my MGCP phones. And so far I didn't even get a chance to debug this since it happens approx 1 out of 10 calls only. By the way, I can now conirm that it can be both MGCP or SIP at the end, it doesn't matter. So to me it looks like IAX2 is involved as well, not just SIP. *1: CVS-02/10/04-16:49:37 *2: CVS-03/05/04-00:50:56 Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Announcement * Astricon 2004 - call for speakers!
Matteo Brancaleoni wrote: eh, very good idea... but how about for alaw people? And E1's and EuroISDN :-) Any plans to make another conference in EU world? We'll start with one conference for everyone. As I'm also based in Europe, having a European followup is an idea that is within our plans. (Also, Steven needs to visit Alaw-land :-) Right now, we need to focus and gather the community in one spot for the first time. For those of you visiting Von Europe in june, maybe you can set up an Asterisk community meeting - like the one we had in Santa Clara. Best regards, /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP Spam
Tom Green wrote: Brian, Encrypted SIP messages can be sent using TLS. However, I don't think it is realistic to expect everyone calling you to have a public/private key pair. SMTP servers that support SMTP-TLS and have valid certs + config do exactly that already... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Line install.. (UK Muppet)
On Thu, 2004-04-15 at 09:36, David Stubbs wrote: Hi all, Muppet from the UK asking for help There is 2 sections here. One for channelized T1 and one for PRI. I'll answer all questions, you choose which one you needed. Channelized section. = Questionnaire Follows = 1. DS1 Facilities Required: - Superframe or Extended (ESF/SF)? ESF - Line Coding (B8ZS,AMI,. Etc.)? B8ZS - Number of DS1 facilities required? This is your number of T1's. A DS0 is actual lines or channels. - Channel numbers used for this trunk group? This is the DS0 question. 2. Circuit Signaling (DS0 or DS1 Signal): DSO Analog Line Trunk Interface 2 wire, 4 wire, (or na, not applicable)? If you are putting this into a T1 interface on a single or quad span card, this is na. - Supervision Loop Start, Ground Start, Reverse Battery, E M DNIS - Dialed Number Identification service Not applicable with DS0 Analog Line trunks. You will probably want EM as you get good answer detection and hangup detection and the easy ability to add DID info to it. DS1 Digital Signal : Digital Interface - Supervision Loop Start, Ground Start, E M Odd question, would assume it is the same as above. - In-Band or Robbed Bit (EM) signaling? (Y/N) Robbed bit. - CO Supplied Dial tone (Y/N) If you want to call anywhere this would be a yes. PRI section. DS1 Digital Signal: ISDN PRI Only - Primary Rate Interface (ISDN PRI) - If multiple DS1s, Does the customer want Non-Facility Associated Signaling? (NFAS) (Y/N) I think this is a No, but I am not sure. If No, Does the customer want One D-Channel per DS1, or 1 D-Channels for all DS1s. How many D-Channels? Currently you need one D channel per DS1 for digium hardware/software. If NFAS=Yes, Number of D-Channels Required TWO - (PRI ISDN Protocol)= example National PRI (NIPRI) NIPRI, NTNAPRI, U449PRI, U459PRI, N449PRI National is good. These are pretty easy and I'll skip as they would be specific to your location and policy. 3. Trunk Group Billing Telephone Number (10 digit): 4. Trunk Group Originating Area Code for PBX 5. Calling Scope (Basic, Basic+, Extended, Metro, Metro+)? 6. Primary Interexchange Carrier (PIC) 7. NPA/NXX Call Blocking ( 700, 900, 976) 8. Trunk Group Directional call flow and Member numbering: (If sub-grouped, select subgroup rollover) - 2-Way, 1-Way Into PBX, or 1-Way Out of PBX - Member Numbering (i.e. 0001 thru 0024) 9. Trunk Member selection - Selection for calls originating at the PBX toward the Central Office (CO) Asending (ASEQ), Desending (DSEQ), Most Idle (MIDL), Least Idle (LIDL) - Selection for calls terminating to the PBX from the Central Office (CO) Asending (ASEQ), Desending (DSEQ), Most Idle (MIDL), Least Idle (LIDL) These 2 questions aren't really important. While we where on MCI we used least idle, now on Telcove I think it is asending. I think asending is normal for most analog groups. 10. Direct Inward Dial to PBX - DID (Y/N): - Number or Number Blocks routed toward PBX (i.e. 972-668-20xx) - Porting Coordination Required (Y/N)? - Number of digits to outpulse (None, 4, 7, 10 digits) to PBX? This will probably be a yes, and the number of blocks is per what you need. Whether you need 4,7, or 10 digits depends on how many digits you will need to unique. On a PRI a 10 digit outpulse is fine, but you wouldn't want it if it was on a channelized T1 because of the length of time to connect. 11. Inband Ringing required from the CO when terminating calls to PBX (Y/N)? In the US where you will probably get all your incoming calls free, and the majority of callers won't notice the extra few seconds of billable time, you can say no here and just answer the call right away. If you where porting it somewhere else and didn't want to actually answer the line to a human was ready to talk, then say yes. Back to channelized T1. 12. NON-PRI DS1Switching - Start Signaling and Digit Pulsing for In-Band Signaling for NON-PRI DS1: - CO Incoming Start Signaling for calls originating at the PBX (ISTARTSG) Start Signal expected by the PBX, sent by the CO telling the PBX to begin sending digits to the CO for collection. - Wink Start, Delay Dial, Immediate Dial, Dialtone, Loop Start, Grd Start Wink if you are doing DID lines. Otherwise it depends on the supervision you can get on each of the signaling support. - CO Incoming Digit Pulsing for calls originating at the PBX (IPULSTYP) Type of digits sent from the PBX to CO after the PBX receives the start signal (ISTARTSG) from the CO - Dial Pulse, DT/ DTMF, MF, NP/No Pulsing DTMF - CO Outgoing Start Signaling for calls terminating to the PBX (OSTARTSG) Start Signal expected by the CO, sent by the PBX telling the CO to begin
[Asterisk-Users] Missing vm feature - turn off voicemail
Listening to the options on the voicemail system it seems to be missing a feature for users to turn voicemail off completely. This seems a rather glaring omission. Does the feature of turning off message recording via the phone exist - or does it need a patch? Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] t1 won't dial outbound
It looks like your channel and group statements in the zapata.conf are the problem. Notice that when it tries to dial out it does so on Zap/6-1. You have the T-1 defined as 'Span 1,' but you are trying to send the calls to span 6. It ain't gonna work! I don't see anywhere where you've assigned the rest of the channels on that T-1, either. I would recommend either grouping them all together (that's the easiest), or at least making sure you've got all of the channels assigned to groups. My zapata.conf is much simpler: signalling=pri_net group=1 channel = 1-23 When it dials, then you will see the calls going out on Zap/1-1 or Zap/1-2, etc. Good luck; and have fun! Joe Mark Messmore, Technical Support, University Telcom Inc. [EMAIL PROTECTED] wrote: I've posted this problem a couple of times before with little or no response. Basically I have a T100P in my * box. Incoming calls are working great. However outgoing calls are not working at all. I've copied a previous post into this message which should have all the necessary info. Any ideas or suggestions would be greatly appreciated. Thanks. Mark # OK...I've got an * box with a T100P in it. For the most part incoming calls are going through just fine. Outgoing calls, however, I'm having some more trouble with. Whenever I make an outgoing call, the call begins, however after the dialing process all I hear is dead air. Here's the output from my * console: -- Executing Dial(SIP/mark-2d08, Zap/g3/2550559) in new stack -- Called g3/2550559 -- Hungup 'Zap/6-1' == Spawn extension (uti-mainst, 2550559, 1) exited non-zero on 'SIP/mark-2d08' I've checked with the switch guy...and whatever channel I'm trying to dial out on is coming up as blocked on his switch. We've compared as many settings as we can think of and they all seem to be set the same. I'll post the entries from my zaptel.conf and my zapata.conf in here...if you have any ideas please send them my way... zaptel.conf span=1,1,0,d4,ami em=1-24 fxsks=25 loadzone=us defaultzone=us zapata.conf context=conference signalling=em switchtype=5ess group=3 callgroup=3 pickupgroup=3 channel = 6 busydetect=yes callerid=asreceived callprogress=yes callreturn=yes callwaiting=yes callwaitingcallerid=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes immediate=no language=us musiconhold=default threewaycalling=yes transfer=yes usecallerid=yes ## _ This mail sent using Secure Message Center Put SAI Address and NASD blah blah stuff here. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Line install.. (UK Muppet)
- Supervision Loop Start, Ground Start, Reverse Battery, E M DNIS - Dialed Number Identification service Not applicable with DS0 Analog Line trunks. You will probably want EM as you get good answer detection and hangup detection and the easy ability to add DID info to it. I always wondered what difference this made on DS1 interfaces -- you're either getting the signalling from the robbed bits or from the D channel, depending on whether you're using ct1 or PRI. How are these supervision options changing anything at all in the DS1 bitstream? -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Line install.. (UK Muppet)
PRI section. DS1 Digital Signal: ISDN PRI Only - Primary Rate Interface (ISDN PRI) - If multiple DS1s, Does the customer want Non-Facility Associated Signaling? (NFAS) (Y/N) I think this is a No, but I am not sure. I don't think that Zaptel PRI software can currently handle NFAS. If NFAS=Yes, Number of D-Channels Required TWO Hmm so they enforce the use of a backup D channel. I never found much use for this when all the DS1s came in on the same trunk. That, and it wastes another B channel. :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMailBox wav file format in EMAIL.
On Tue, 2004-04-13 at 01:12, James Gardiner wrote: Hi all, I am not sure if tis is a bug but.. Was learning about VM etc to see how it all worked, and I noticed the following.. In the default install, the VM system leaves 3 different copies of the Voice message. Size filename 13kb Msg.gsm This is a raw gsm frame dump of the audio. It contains no headers for a sound editing app to understand it. 13kb Msg.wav This is the previous GSM frames wrapped in a RIFF wav header and the appropriate bit shifting to make windows happy. 122kb Msg.WAV - under UNIX we have case sensitive file names of course. This is raw PCM. That is why it is larger. I wanted to have a look at these files so loaded them into SOUND FORGE 6. This first thing I noticed was that the LARGER file is of much HIGHER volume. Like it had been normalised to 100% The smaller was file, when loaded into sound forge, did not load properly, only the first 2 seconds loads. Can anyone explain these issues and why they exist? For some reason the PCM files are bit shifted up. This gives the effect of doubling the volume. If I understand it though, it is also bit shifted down when played back via asterisk. So you loose the volume if asterisk replays the audio file. All in all, I was wondering what would be the best format for best quality but with still great compression. GSM is fine enough for the prompts. I want to archive all calls for a period of time with self expire. (For example dedicate 5 gig disk space to the last number of calls that can fit in the 5gig.) I want to store the best quality possible but also make best use of disk space, so I can store for even longer periods. I was considering ogg but then is occurred to me that GSM or other codecs designed for audio with this frequency response may be better. (But the GSM file above is not as clear as the WAV ones produced.) GSM is good. 33 bytes per 20ms will get you a long ways. From what I see, here is your capacity. (5*1024*1024*1024)/(50 blocks a second * 33 bytes per block)/60 seconds per minutes /60 minutes per hour and you get 903 hours. then remove a certain amount for disk block alignment, formatting, and the lost fragments at the end of a 4k cluster and you are now down to a realistic 800 hours of record time or 33 and a 3rd hours of a constant T1 of audio calls. I was also wondering if the VM system when emailing the audio can be setup to use something like ogg or MP3? ogg and mp3 are not good choices for telephone quality. They don't get that great of compression unless you start sacrificing a lot of quality. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropped calls
On Thu, 2004-04-15 at 18:21, Philipp von Klitzing wrote: So to me it looks like IAX2 is involved as well, not just SIP. Are you sure? I did some analysis of my traffic. Here is what I found so far: Only Grandstream phones appear to be affected. All phones affected have been behind a coned NAT, running firmware 1.0.4.39 with STUN enabled. The hangup only occurs in dialogs with CSeq set to '0'. I will test whether another firmware will solve this issue. Let's hoep the best. Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Line install.. (UK Muppet)
On Thu, 2004-04-15 at 12:15, Andrew Kohlsmith wrote: - Supervision Loop Start, Ground Start, Reverse Battery, E M DNIS - Dialed Number Identification service Not applicable with DS0 Analog Line trunks. You will probably want EM as you get good answer detection and hangup detection and the easy ability to add DID info to it. I always wondered what difference this made on DS1 interfaces -- you're either getting the signalling from the robbed bits or from the D channel, depending on whether you're using ct1 or PRI. How are these supervision options changing anything at all in the DS1 bitstream? When we had a MCI ct1, they couldn't send us proper supervised hangup on a loopstart encoded DS0. They claimed it to be a problem with the software on their switch. Their solution was to switch to groundstart. Our end solution was to drop them and switch to Telcove(formerly Adelphia) and get a PRI where we controlled the signaling with asterisk. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Line install.. (UK Muppet)
When we had a MCI ct1, they couldn't send us proper supervised hangup on a loopstart encoded DS0. They claimed it to be a problem with the software on their switch. Their solution was to switch to groundstart. Our end solution was to drop them and switch to Telcove(formerly Adelphia) and get a PRI where we controlled the signaling with asterisk. Again, how the hell is switching the supervision type changing anything in the bitstream? You have 193 bits being sent 8000 times a second. There is no grounding or reversing of a battery on any kind of T1 -- that is strictly FXO-side signalling. You have 2 (SF) or 4 (ESF) bits with a CT1 to determine channel state. And with ESF those extra bits are usually just duplicates of the original 2. You have onhook, offhook, ringing and something I can't remember at the present moment. I am positive I'm showing my T1 newbieness here but again... how does the switch changing their FXO signalling/supervision change the bitstream and fix CT1-related issues? I completely understand EM, LS, GS and all of that on the actual analog side -- but how does it change a damn thing on the CT1 side? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] too many arguments to function `ast_queue_hangup' compiling asterisk-oh323
Victor Perez wrote: when trying to build asterisk-oh323 I get the following: make[1]: Entering directory `/usr/src/asterisk-oh323-0.5.10/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declara tions -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1128: error: too many arguments to function `ast_queue_hangup' chan_oh323.c:1142: error: too many arguments to function `ast_queue_hangup' chan_oh323.c: In function `oh323_hangup': chan_oh323.c:1182: error: too many arguments to function `ast_queue_hangup' chan_oh323.c: In function `oh323_read': chan_oh323.c:1581: error: too many arguments to function `ast_dsp_process' chan_oh323.c: In function `ast_oh323_new': chan_oh323.c:2030: warning: assignment from incompatible pointer type chan_oh323.c: In function `cleanup_h323_connection': chan_oh323.c:2835: error: too many arguments to function `ast_queue_hangup' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.5.10/asterisk-driver' make: *** [subdirs_all] Error 1 I checked chan_oh323.c and indeed it only takes one parameter now so I am wondering what was that old parameter for and when did they take it off so I may try pulling that version of asterisk to try with. Some functions of Asterisk have recently changed. asterisk-oh323 has not been updated yet to work with them, so stay with an older version of Asterisk (date to use with cvs 20040407 or older). Regards, Victor Perez Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Missing vm feature - turn off voicemail
directly into voicemail I don't think that's possibile. but you can fake this function, simply using in the right way dbput / dbget and if conditions... Matteo. Il gio, 2004-04-15 alle 18:45, Iain Stevenson ha scritto: Listening to the options on the voicemail system it seems to be missing a feature for users to turn voicemail off completely. This seems a rather glaring omission. Does the feature of turning off message recording via the phone exist - or does it need a patch? Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] t1 won't dial outbound
Thanks for the reply. I didn't include my entire zapata.conf...just the portion that applied to this call (i.e. group #3) Please correct me if I have misunderstood how this all works together. When I see: -- Executing Dial(SIP/mark-2d08, Zap/g3/2550559) in new stack -- Called g3/2550559 -- Hungup 'Zap/6-1' I'm interpreting that this is dialing out on Zap group 3 (which happens to begin on channel 6). Please correct me if I'm wrong here... I'm attaching my entire zapata.conf just to defer any confusion...and to see if you can see anything. Also, I'm going to take your suggestion and create another zapata.conf which will be simplified just to see if there is a conflict somewhere in there. Thanks for your help! Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick Sent: Thursday, April 15, 2004 1:46 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] t1 won't dial outbound It looks like your channel and group statements in the zapata.conf are the problem. Notice that when it tries to dial out it does so on Zap/6-1. You have the T-1 defined as 'Span 1,' but you are trying to send the calls to span 6. It ain't gonna work! I don't see anywhere where you've assigned the rest of the channels on that T-1, either. I would recommend either grouping them all together (that's the easiest), or at least making sure you've got all of the channels assigned to groups. My zapata.conf is much simpler: signalling=pri_net group=1 channel = 1-23 When it dials, then you will see the calls going out on Zap/1-1 or Zap/1-2, etc. Good luck; and have fun! Joe Mark Messmore, Technical Support, University Telcom Inc. [EMAIL PROTECTED] wrote: I've posted this problem a couple of times before with little or no response. Basically I have a T100P in my * box. Incoming calls are working great. However outgoing calls are not working at all. I've copied a previous post into this message which should have all the necessary info. Any ideas or suggestions would be greatly appreciated. Thanks. Mark # OK...I've got an * box with a T100P in it. For the most part incoming calls are going through just fine. Outgoing calls, however, I'm having some more trouble with. Whenever I make an outgoing call, the call begins, however after the dialing process all I hear is dead air. Here's the output from my * console: -- Executing Dial(SIP/mark-2d08, Zap/g3/2550559) in new stack -- Called g3/2550559 -- Hungup 'Zap/6-1' == Spawn extension (uti-mainst, 2550559, 1) exited non-zero on 'SIP/mark-2d08' I've checked with the switch guy...and whatever channel I'm trying to dial out on is coming up as blocked on his switch. We've compared as many settings as we can think of and they all seem to be set the same. I'll post the entries from my zaptel.conf and my zapata.conf in here...if you have any ideas please send them my way... zaptel.conf span=1,1,0,d4,ami em=1-24 fxsks=25 loadzone=us defaultzone=us zapata.conf context=conference signalling=em switchtype=5ess group=3 callgroup=3 pickupgroup=3 channel = 6 busydetect=yes callerid=asreceived callprogress=yes callreturn=yes callwaiting=yes callwaitingcallerid=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes immediate=no language=us musiconhold=default threewaycalling=yes transfer=yes usecallerid=yes ## _ This mail sent using Secure Message Center Put SAI Address and NASD blah blah stuff here. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users zapata.conf Description: Binary data
[Asterisk-Users] What's in a number? say.c internationalization!
http://bugs.digium.com/bug_view_page.php?bug_id=0001429 We need to architect a general structure for saying numbers in voices. Say.c is broken as it is now and it needs to be changed. If we work quickly, this can be sorted out and fixed to 1.1, if not before that. There's a number of separate patches in bugs that fixes one language per patch, but no general structure. I would be perfectly satisfied if Swedish worked, but I guess that there's other users out there that need Danish, Polish, Portuguese, French and possibly other languages like Chinese, Mandarin and Turkish. This is the time to add your input :-) We need to * Design a general structure * Create one unified patch that fixes the general solution * Add patches that works with these for each syntax needed This is easier to fix than it sounds, so roll up your sleeves and join me in the bug tracker. Please go to the bug tracker and add your comments, thought or solutions. Coders welcome :-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: MeetMe - new e and E flags?
On Thursday 15 April 2004 03:01, Tony Mountifield wrote: In article [EMAIL PROTECTED], Tilghman Lesher [EMAIL PROTECTED] wrote: If it's a pin-required conference, you will hear the conference number prior to being prompted to enter the associated pin. Obviously, in this case, any such conference would be static, so the pin would be pre-assigned in the config file. This might be useful if you ran a number of conferences, but did not want just anybody to be able to access them (i.e. in order to access the conferences, possibly dial-able from anywhere, you had to know the associated pin). You can also select an empty dynamic conference, with pin, by combining the flags 'eD', in which case you will be told the conference number prior to you specifying the pin. Or you could simply select an empty dynamic conference (no pin), with flags 'ed'. I'm trying hard to understand the usefulness of these features. It looks like, from what I've read here, if you dial an extension that routes to MeetMe(e), it will put you in an empty conference and tell you the number. Presumably for anyone else to join the same conference, you then have to tell them the number, e.g. by email, IM or another phone call, and they then have to dial a different extension which routes to MeetMe(without e). And if the empty conference also has a PIN, does the first user need a list of conference numbers to PINs so he can enter the correct PIN when told the conference number? That's an administrative matter, not a detail of implementation. You could, of course, have the same PIN for multiple conferences. This all seems rather cumbersome, and I haven't had the chance to experiment with this feature yet, so the above probably highlights both (a) my lack of understanding, and (b) the lack of documentation! If the feature doesn't make any sense to you, then don't use it. For a customer of ours, though, it was necessary to have this feature. I would suggest actually trying out the feature a couple times, if your goal is to learn how to use it. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA 188 and fax
Hi, Does anybody have ATA 188 working with any kind of fax machine? I've tried many different configuration following the Cisco Online Manual and I couldn't get this working with Asterisk. I were trying do change the ATA Connect Mode and Audio Mode reading the (http://www.cisco.com/en/US/products/hw/gatecont/ps514/ products_configuration_example09186a00800d698e.shtml) and allowing all codecs on Asterisk and did not work either. best regards Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] * Announcement * Astricon 2004 - call for speakers!
Can someone direct me to the site for Von Europe? Thanks. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ===8==Original message text=== Matteo Brancaleoni wrote: eh, very good idea... but how about for alaw people? And E1's and EuroISDN :-) Any plans to make another conference in EU world? We'll start with one conference for everyone. As I'm also based in Europe, having a European followup is an idea that is within our plans. (Also, Steven needs to visit Alaw-land :-) Right now, we need to focus and gather the community in one spot for the first time. For those of you visiting Von Europe in june, maybe you can set up an Asterisk community meeting - like the one we had in Santa Clara. Best regards, /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ===8===End of original message text=== ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to process inband DTMF
Hi All, Since I updated my * (CVS 2004-03-24), daily, I am getting a strange message just before a segmentation fault: Unable to process inband DTMF on 2 frames. What could it be? Should it cause seg.faults? Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA 188 and fax
Osvaldo Mundim wrote: Hi, Does anybody have ATA 188 working with any kind of fax machine? I've tried many different configuration following the Cisco Online Manual and I couldn't get this working with Asterisk. I don't know what the difference is between the 186 and 188 other then the extra nic port. But we gave up on the 186 for doing any fax or data calls. We switched to Sipura-2000 and using Ulaw faxing works. Data calls well we can get them working but only at 28,800 bps. Good luck I were trying do change the ATA Connect Mode and Audio Mode reading the (http://www.cisco.com/en/US/products/hw/gatecont/ps514/ products_configuration_example09186a00800d698e.shtml) and allowing all codecs on Asterisk and did not work either. best regards Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail notification - LED solution
Walker Haddock wrote: I just hit google and got this. However, there are 35 hits so take a look. http://lists.digium.com/pipermail/asterisk-users/2003-October/022796.html or, more recently: http://lists.digium.com/pipermail/asterisk-users/2003-February/007855.html Bottom line, you have to put the mailbox= record in the sip.conf file in the stanza for the device. Thanks I put mailbox=extention@context ie [EMAIL PROTECTED] The extension came from the voicemail.conf file. The handset LED comes on solid and the line where I'm connected to has a envelope that flashes, although not very prominatly. Thanks a bunch! I now know that I'm missing calls! ;-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dropped calls
We also are having randomly dropped calls with our IAX2 connections, we have tried IAX2 with and without trunking enabled. the phones are snom 200's with SIP and there is an asterisk box at each site so no sip nat problems. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Philipp von Klitzing Sent: Thursday, April 15, 2004 11:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dropped calls Hi! I see this very same effect rather often in the following setup: SIP (GS101) -- *1 -- IAX2 -- *2 -- MGCP (ip10) In fact I think I've seen it also with SIP instead of MGCP at the end. The first client is behind NAT, by the way. That must be it. I have seen this happening with sip -- * -- IAX as well. I take it you don't know a cure? Unfortunately not, no. By the way I am not on latest CVS as that would disable my MGCP phones. And so far I didn't even get a chance to debug this since it happens approx 1 out of 10 calls only. By the way, I can now conirm that it can be both MGCP or SIP at the end, it doesn't matter. So to me it looks like IAX2 is involved as well, not just SIP. *1: CVS-02/10/04-16:49:37 *2: CVS-03/05/04-00:50:56 Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VON Europe (was * Announcement)
Here's the link. It's 7-10 June, in London. http://pulver.com/europe2004/register.html Warning: Unlike Santa Clara, they've for some reason decided not to have an exhibits-only pass available. You have to pay a lot for a full conference pass to attend. Maybe some feedback would be in order to the organiser. Please let the list know if they change their mind! Cheers Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Karrington Sent: Thursday, April 15, 2004 11:40 AM To: Olle E. Johansson Subject: Re[2]: [Asterisk-Users] * Announcement * Astricon 2004 - call for speakers! Can someone direct me to the site for Von Europe? Thanks. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ===8==Original message text=== Matteo Brancaleoni wrote: eh, very good idea... but how about for alaw people? And E1's and EuroISDN :-) Any plans to make another conference in EU world? We'll start with one conference for everyone. As I'm also based in Europe, having a European followup is an idea that is within our plans. (Also, Steven needs to visit Alaw-land :-) Right now, we need to focus and gather the community in one spot for the first time. For those of you visiting Von Europe in june, maybe you can set up an Asterisk community meeting - like the one we had in Santa Clara. Best regards, /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ===8===End of original message text=== ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MeetMe - new e and E flags?
In article [EMAIL PROTECTED], Tilghman Lesher [EMAIL PROTECTED] wrote: On Thursday 15 April 2004 03:01, Tony Mountifield wrote: This all seems rather cumbersome, and I haven't had the chance to experiment with this feature yet, so the above probably highlights both (a) my lack of understanding, and (b) the lack of documentation! If the feature doesn't make any sense to you, then don't use it. I'd much rather be educated so that it does make sense to me. Then I too might find it useful. For a customer of ours, though, it was necessary to have this feature. Could you explain why? I'm sure that knowing why it was important to them would help a lot in understanding (a) its purpose, and (b) the correct way to use it. I didn't know until you said so that it was you who implemented this feature - please could I ask you to add something to the wiki about it? I would suggest actually trying out the feature a couple times, if your goal is to learn how to use it. That's what I've been doing this evening, with not much success. I have the following in extensions.conf: exten = 4003,1,Answer exten = 4003,2,Wait(1) exten = 4003,3,MeetMe(|eMp) exten = 4003,4,Hangup exten = 4004,1,Answer exten = 4004,2,Wait(1) exten = 4004,3,MeetMe(|EMp) exten = 4004,4,Hangup And the following in meetme.conf: conf = conf = 1234 conf = 2345,9938 conf = 3131 conf = 4242 conf = 5353 conf = 6464 conf = 7575 conf = 8686 conf = 9797 I observed exactly the same behaviour whether I dialled 4003 or 4004: The first phone to dial was told I was in conference , and then received Music On Hold. The second and subsequent phones to dial 4003 or 4004 just received silence, not a new empty conference. No announcement, no MoH. sip show channels showed a new channel that did not disappear on hangup. I suspect this is a bug of some sort, but it's possible I have just done something wrong. Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] music on hold problems
i've been searching the archives but can't find anything substantive on this. most of the music on hold documentation discusses integrating with zap hardware, but i am trying to send it across a sip channel. I have the following in extensions.conf: exten = 2100,1,Answer exten = 2100,2,MusicOnHold(default) and have uncommented the default line in musiconhold.conf: [classes] default = quietmp3:/var/lib/asterisk/mohmp3 I also installed mpg123, and created a symlink to /usr/bin (which is where it seems asterisk looks for it). Does anyone have any idea as to what I'm doing wrong here? Regards, -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] All mates in Australia: Check this
http://bugs.digium.com/bug_view_page.php?bug_id=0001396 Indications for australia. Please confirm if this works for you so we know if this is something to include in CVS or not. Thanks, mate :-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA 188 and fax
On a Grandstream ATA and CVS HEAD from last night, and with echo off, I'm able to receive faxes. With echo on, no go. HTH, Ryan On Apr 15, 2004, at 1:58 PM, Ariel Batista wrote: Osvaldo Mundim wrote: Hi, Does anybody have ATA 188 working with any kind of fax machine? I've tried many different configuration following the Cisco Online Manual and I couldn't get this working with Asterisk. I don't know what the difference is between the 186 and 188 other then the extra nic port. But we gave up on the 186 for doing any fax or data calls. We switched to Sipura-2000 and using Ulaw faxing works. Data calls well we can get them working but only at 28,800 bps. Good luck I were trying do change the ATA Connect Mode and Audio Mode reading the (http://www.cisco.com/en/US/products/hw/gatecont/ps514/ products_configuration_example09186a00800d698e.shtml) and allowing all codecs on Asterisk and did not work either. best regards Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to process inband DTMF
Daniel Bichara wrote: Hi All, Since I updated my * (CVS 2004-03-24), daily, I am getting a strange message just before a segmentation fault: Unable to process inband DTMF on 2 frames. That message is usually caused by using inband DTMF and using a compressed codec. All codecs except ulaw and alaw are compressed. Don't use a compressed codec or use out of band DTMF on the PHONE and on Asterisk (RFC2833 or INFO) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Announcement * Astricon 2004 - call for speakers!
Steven Critchfield wrote: Do we need a presentation on how to behave on the list to avoid getting flamed by me, or should I just show up with an appropriate LART device to fix problem people during the normal presentations? Maybe we should we bring the cattle prod tradition from that other CON :) Too many conferences I want to attend, but this one is a must attend. Most certainly! Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] too many arguments to function `ast_queue_hangup' compiling asterisk-oh323
Victor Perez wrote: I checked chan_oh323.c and indeed it only takes one parameter now so I am wondering what was that old parameter for and when did they take it off so I may try pulling that version of asterisk to try with. The author of that software needs to update because the asterisk API has changed. It is a simple fix, just look at the new function prototype(s) and remove the offending parameter or wait until asterisk-oh323 gets updated. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP response 404 Not Found AND circuit-busy ??
I have a dlink dvg-1120s voip-router. I can make calls out from the router, but when calling the router I got -- Executing Dial(SIP/2010-b437, SIP/2021|30|r) in new stack -- Called 2021 -- Got SIP response 404 Not Found back from 62.79.78.74 -- SIP/2021-473b is circuit-busy What does this meen ? Or what can I do ? The router is behind nat, but if I put the router on the same network as asterisk it work ok /Hans-Henrik Andresen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: music on hold problems
In article [EMAIL PROTECTED], Steven Kokinos [EMAIL PROTECTED] wrote: i've been searching the archives but can't find anything substantive on this. most of the music on hold documentation discusses integrating with zap hardware, but i am trying to send it across a sip channel. Music on hold requires a zaptel timing source. If you do not have any zaptel cards in your system, you will need to install either ztdummy (only if you have a uhci type of USB) or zaprtc. See http://www.voip-info.org/tiki-index.php?page=Asterisk+timer Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: music on hold problems
I wrote: In article [EMAIL PROTECTED], Steven Kokinos [EMAIL PROTECTED] wrote: i've been searching the archives but can't find anything substantive on this. most of the music on hold documentation discusses integrating with zap hardware, but i am trying to send it across a sip channel. Music on hold requires a zaptel timing source. If you do not have any zaptel cards in your system, you will need to install either ztdummy (only if you have a uhci type of USB) or zaprtc. See http://www.voip-info.org/tiki-index.php?page=Asterisk+timer Having just read that article again, I see that the part which said a zaptel timer was necessary for MoH was deleted on 3 Apr. Is it correct that MoH doesn't need a zaptel timer? If so, was this always the case, or did it change at some point? Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropped calls
Hi! Only Grandstream phones appear to be affected. All phones affected have been behind a coned NAT, running firmware 1.0.4.39 with STUN enabled. The hangup only occurs in dialogs with CSeq set to '0'. Ok, I'll watch for that as well since I upgraded my desk's Grandstream to 1.0.4.54 an hour ago (previously I had either 4.26 or 4.17). Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with greek leters in CLI
I have been having a major problem with after some installations of Asterisk about every 3rd one the CLI will come up in some strange looking greek letters. This problem does not happen all the time but once it happens I was not able to clear it up. Well with the help of a unix/linux expert we have found a fix for it. If this happens to your system here in the US you need to change the following lines (This mainly has been happening on RH 8/9 and Fedora core 1). edit file /etc/sysconfig/i18n and make the following changes. LANG=en_US SUPPORTED=en_US:en SYSFONT=lat0-16 Save it and you then need to reboot the system. - \ \\_ Ariel Batista // / Avionica, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange T1 Problem
When people call into my * box over the T1 interface, they get no ring tone. It rings the SIP phone and when the SIP user picks up, both parties can hear each other ok, its just the PSTN user calling in hears no ring. What could be causing this? I tried setting immediate to yes in zapata.conf, but that causes my DNIS and CallerID to stop being available. T100P with E M Wink start signaling, all 24 channels are inbound channels (no channel bank or anything like that) to SIP ATAs. The ATA is sending a 180 Ringing reply to the invite, but still no ring. Same symptoms with different vendor ATA devices. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange T1 Problem
Fixed in CVS STABLE around 2pm CDT today. It's been fixed in CVS HEAD for a while. Mike Machado wrote: When people call into my * box over the T1 interface, they get no ring tone. It rings the SIP phone and when the SIP user picks up, both parties can hear each other ok, its just the PSTN user calling in hears no ring. What could be causing this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy problems (was music on hold problems)
Actually, after rebooting my machine music on hold started working properly. Not sure what the issue was. As for ztdummy, I am having a more substantive issue with that, which is keeping me from getting meetme working. while ztdummy compiles cleanly, i can't actually get it to load properly. [EMAIL PROTECTED] root]# modprobe ztdummy /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: insmod /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o failed /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: insmod ztdummy failed however, usb-uhci.o does in-fact exist. Does anyone have any thoughts? Regards, -Steve On Apr 15, 2004, at 5:27 PM, Tony Mountifield wrote: In article [EMAIL PROTECTED], Steven Kokinos [EMAIL PROTECTED] wrote: i've been searching the archives but can't find anything substantive on this. most of the music on hold documentation discusses integrating with zap hardware, but i am trying to send it across a sip channel. Music on hold requires a zaptel timing source. If you do not have any zaptel cards in your system, you will need to install either ztdummy (only if you have a uhci type of USB) or zaprtc. See http://www.voip-info.org/tiki-index.php?page=Asterisk+timer Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange T1 Problem
On Thu, 2004-04-15 at 15:26, Steven Critchfield wrote:
Explicitly answer the line. If that doesn't handle inband audio, there
is a r flag to dial. This was discussed very recently.
This must be a different problem, because neither of those solutions
worked.
zapata.conf sends call to fixup context:
[fixup]
; Receive call as *calling*called
exten = _.,1,Answer
exten = _.,2,Cut(CALLING=EXTEN,*,2)
exten = _.,3,SetCIDNum(${CALLING})
exten = _.,4,Cut(CALLED=EXTEN,*,3)
exten = _.,5,Goto(default|${CALLED}|1)
[default]
exten = 1234567890,1,Answer
exten = 1234567890,2,Dial(SIP/user1|r)
user1's phone rings, but no ring from PSTN caller. user1 picks up, both
can talk ok.
I have been using cvs stable branch. I will try HEAD and see if that
fixes it as suggested by Eric.
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Re: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
cvs HEAD did infact fix the ringing problem. Thanks Eric! I have another question for all you T1 buffs out there. The T1 I am working with goes into our local phone switch (Excel switch). Currently we are using E M Wink signaling. The problem is we cannot set callerid on the outbound side. My minimal understanding is that if we had a PRI, I could set the callerID. Unfortunately PRI is one signaling type they cannot do (not have expensive PRI card in switch). So, my question is what other signaling types CAN I set the callerID outbound? My local switch techs cannot seem to answer that question. They just always use E M for everything. But if I can ask them to specifically try a certain signaling type (such as Feature Group D) or one of the others in the t100p supported list, I could probably get them to change the signaling type on my trunk. Do any signaling types other than PRI support passing outbound callerID? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Most Reliable Proxy Server?
Thank you.Simon Brown [EMAIL PROTECTED] wrote: You could try these: voiptalk - www.voiptalk.org sipgate - www.sipgate.de Simon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McMillinSent: Thursday, 15 April 2004 15:29To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Most Reliable Proxy Server? Hi all, Do you know if there's any free public SIP proxy server that is more reliable that FWD and Iptel? Thanks Ron -This mail was content checked for malicious code and virusesby GFI MailSecurity.
Re: [Asterisk-Users] T1 Line install.. (UK Muppet)
Andrew Kohlsmith wrote: When we had a MCI ct1, they couldn't send us proper supervised hangup on a loopstart encoded DS0. They claimed it to be a problem with the software on their switch. Their solution was to switch to groundstart. Our end solution was to drop them and switch to Telcove(formerly Adelphia) and get a PRI where we controlled the signaling with asterisk. Again, how the hell is switching the supervision type changing anything in the bitstream? You have 193 bits being sent 8000 times a second. There is no grounding or reversing of a battery on any kind of T1 -- that is strictly FXO-side signalling. You have 2 (SF) or 4 (ESF) bits with a CT1 to determine channel state. And with ESF those extra bits are usually just duplicates of the original 2. You have onhook, offhook, ringing and something I can't remember at the present moment. I am positive I'm showing my T1 newbieness here but again... how does the switch changing their FXO signalling/supervision change the bitstream and fix CT1-related issues? I completely understand EM, LS, GS and all of that on the actual analog side -- but how does it change a damn thing on the CT1 side? The robbed bit T1 has 2 signalling bits, but the usually do the same thing, so its really like having just one signalling bit. However, the timing of changes to that signalling bit can follow one of several patterns. You will see the terms immediate start and delayed dial used, as well as the terms LS, GS, etc. These are more descriptive, as they describe what will actually happen to the signalling bit. Immediate start and I want to dial: I raise my signalling bit, wait a moment, then send my DTMF. When the far end answers the exchange's signalling bit is raised. When the far end drops the exchange's signalling bit drops. Delayed dial and I want to dial: I raise my signalling bit, I wait for the far end to give a little pulse (called a wink) on its signalling bit, I then know it is ready to receive DTMF. I send my DTMF. When the far end answers the exchange's signalling bit is raised. When the far end drops the exchange's signalling bit drops. Other variants are that only signalling bit A or only bit B changes; the signalling bits are inverted; the line only works during a full moon; etc. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
Mike Machado wrote: cvs HEAD did infact fix the ringing problem. Thanks Eric! I have another question for all you T1 buffs out there. The T1 I am working with goes into our local phone switch (Excel switch). Currently we are using E M Wink signaling. The problem is we cannot set callerid on the outbound side. My minimal understanding is that if we had a PRI, I could set the callerID. Unfortunately PRI is one signaling type they cannot do (not have expensive PRI card in switch). So, my question is what other signaling types CAN I set the callerID outbound? My local switch techs cannot seem to answer that question. They just always use E M for everything. But if I can ask them to specifically try a certain signaling type (such as Feature Group D) or one of the others in the t100p supported list, I could probably get them to change the signaling type on my trunk. Do any signaling types other than PRI support passing outbound callerID? You can usually get CLI on an EM robbed bit T1 by configuring it right. Instead of just sending you the DNIS as a string of DTMF they usually send *cli*dnis*. The DNIS and CLI may be swapped, and there may be less than 3 *s in the string - wonderful consistency, eh? :-\ Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
Mike Machado wrote: cvs HEAD did infact fix the ringing problem. Thanks Eric! As I said, CVS STABLE also has the fix as of this afternoon. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA 186 SIP behind XP Dynamic IP Firewall to Static Public Asterisk
Is it possible to set up the following? public IP Asterisk Server to ata 186 behind a XP server firewall. I think my biggest problem is that I don't know how to make XP forward the RTP port to the private ata address. I would put up some configs, but was hoping that someone one who has this working can share the working configurations incase mine are all messed up (which is likely.) Thank for any help you can offer Centauri _ MSN Toolbar provides one-click access to Hotmail from any Web page FREE download! http://toolbar.msn.com/go/onm00200413ave/direct/01/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip videosupport
Hi all I was tryed to connect to mysip.ch scs_client by siemens that isn't works well. Does anyones knows to work H/W or S/W applictations in asterisk SIP videosupport? Regards mack_jpn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
