Re: [Asterisk-Users] res_config
Matthew Boehm wrote: MailboxExists([EMAIL PROTECTED]): Conditionally branches to priority n+101 if the specified voice mailbox exists Bingo... MailboxExists([EMAIL PROTECTED]) is more like MailboxExists([EMAIL PROTECTED]). Omitting the context was my point of failure. When it's there everything works nicely (except CLI command to show voicemail users). I will document this in the wiki as soon as I have enough free time to figure out how :) Thanks for turning the lightbulb on for me! Regards, Trevor Peirce ___ 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] NewBie Question Modem Telephone -PSTN
Hello, I'm really new on Asterisk. Is it possible to use a telephone machine connected to a modem as an asterisk voice input output device? I do not need PSTN connection. The scheme i'm thinking about is; user - phone - modem - asterisk - ip - vice versa. If it is possible can a user dial another asterisk user via the phone? I've searched astersik lists but couldnot find any help on this issue. Sorry for bothering and thanks alot. -- ___ 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] Incoming SIP Address?
Hi, Is it possible to have an incoming SIP address like [EMAIL PROTECTED], where sip.mydomain.com points to a box running Asterisk? If so, please could someone give an example asterisk config snippet for this? If it is possible, I assume ports 5060 and 1-2 need to be opened in the firewall too. Thanks! --ian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 500, won't ring??
Jared Armstrong wrote: Hi, I have was testing some of the different ring types with my polycom 500, and the ALERT_INFO settings. Now when my phone receives a call it wont ring. I had the same thing happen to me - touched the files on the ftp server, rebooted the phone, it formatted/reinstalled itself and was fixed. probably not the right way, but definately quick n easy :) matt -- Matt Gibson VOIP Administrator NJ Tech Solutions 1.314.480.4550 ex. 6400 1.877.999.4678 ex. 6400 ___ 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] Gossiptel with Asterisk?
Hi, Has anyone got Gossiptel working with Asterisk? - I am having real problems getting it to register - i'm just getting timeout errors. Thanks --ian ___ 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] howto install
Hello, I am using Mandrake 10.1. Howto to install asterisk. I have downloaded tarball. I have not installed any hardware yet. Is it possible to install ? Thanks Varun ___ 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] Bluetooth with *
On Saturday 04 December 2004 04:43, Nate Carlson wrote: In other words, if it's something you really want, add more cash to the bounty, to help encourage the developer to spend more time on it *grin*: alright, alright - i'll work on it today :-) ~ Theo -- Theo P. Zourzouvillys [EMAIL PROTECTED] http://www.crazygreek.co.uk/ ___ 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] Snom 220 busy lamps [was: Receptionist phone...]
I am so far unable to get the busy lamps on a Snom 220 to work either with current cvs or asterisk 1.0. I am using the hint extension and the Snom 220 just as described in the mini-howto on: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg49781.html There are also a couple of wiki pages referencing this: http://www.voip-info.org/wiki-Asterisk+standard+extensions This one seems a bit out of date: http://www.voip-info.org/wiki-Asterisk+phone+SNOM I have created hint priorities in my dialplan: exten = l00,hint,SIP/100 exten = 100,1,Macro(stdexten,100,SIP/100) I have this set up for phones 100-110. Phone 110 is the receptionists phone and 100 is a normal users phone. I also have subscribecontext=default in all of my sip.conf entries which the mini-howto above completely neglects to mention. Is it necessary? I have gone to the configuration page on the phone and set the function keys to be type Destination and put in the numbers of extensions 100-110 which the phone then converts to a sip uri. The sip uri has user= at the end. It defaults to 'phone'. Does this matter? If I do a sip debug peer peername on the receptionists phone I can see it successfully subscribe to all of the appropriate channels. If I do a sip show subscriptions it shows me: *CLI sip show subscriptions Peer UserCall IDURI 10.1.2.199 110 3c26700c9cb2-p3yfx90d 10.1.2.199 110 3c26700c5f8f-edjolk9e 10.1.2.199 110 3c26700c3804-klsp2pey 10.1.2.199 110 3c26700c3069-tiq4v9lh 10.1.2.199 110 3c26700c2902-7mkiukbt 10.1.2.199 110 3c26700c1b20-j74r8659 10.1.2.199 110 3c26700c13b5-y76qc0og 10.1.2.199 110 3c26700c0c3f-w6r29j7l 10.1.2.199 110 3c26700c04b6-blfcx2lj 10.1.2.199 110 3c26700bf156-nt8ntdhc 10.1.2.199 110 3c26700be9f9-opq8ysgg 10.1.2.199 110 3c26700bdc79-7qctu66t 10.1.2.199 110 3c26700bcf0b-ixcy19rk 10.1.2.199 110 3c26700bc152-dz03km17 10.1.2.198 100 3c2670098a3f-ojydtir1 0 active SIP subscriptions(s) What does the User column represent? It is odd that all of them say 110 (the receptionists phone) except for one which says 100. Extension 100 is a Snom 200 but all of the rest of the phones are Cisco 7960's. But the lights do not work regardless of whether I make a call with a Cisco or the Snom phone. What should be in the URI column? Does the fact that it is empty mean anything? One odd thing is that if a phone is actually online and reporting a status the light on the keypad is illuminated. This seems opposite from what I would expect. I would expect the light to come on only if the phone is busy. If the phone is offline for some reason the light is dark. But the status of the light never changes even when I am making calls on one of the phones. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgpfs8hEeBP5R.pgp Description: PGP signature ___ 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] ZAP and IAX Trunks
HelloEveryone, I have recently come across these two terms. I am new at Asterisk and do appreciate your assistance in this. Some tools such as "astGUIclient" and "Asterisk Management Portal" require that the phone system be running Zap or IAX trunks as well as SIP devices. SIP devices are understadable but what about the other two? I am planning to use Cisco 7960/7940 IP phones. Thanks Walid ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Cisco IP Phones
Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid ___ 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] Snom 220 busy lamps [was: Receptionist phone...]
On Sat, 4 Dec 2004, Tracy R Reed wrote: I have created hint priorities in my dialplan: exten = l00,hint,SIP/100 exten = 100,1,Macro(stdexten,100,SIP/100) ^ I guess it may just be a typo during retyping, but you have 'l' (lower case L) in the hint line and a '1' (one) in the macro line. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_misdn and Dynalink IS64PH ISDN
Hi For use my isdn card in NT mode I have compiled chan_misdn. When I launch asterisk it stop with th e message : [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables the asterisk to use hardware which is supported by the new ) Locking Config Mutex UnLocking Config Mutex Init. Stack on port 1 TE Stack No Upper ID init_stack: File exists note : It work with chan_capi (but nnot nt mode ) Any ideas ? Thanks Jerome Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ Avec Yahoo! faites un don et soutenez le Téléthon en cliquant sur http://www.telethon.fr/030-Don/10-10_Don.asp ___ 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 with SMS
Nguyen Quang Hoa wrote: Hi all, I am trying to setup the SMS of Asterisk. I have a Siemens SMS enable fixed phone which connects to my Asterisk through PSTN. Currently, I can use my fixed phone to edit and send messages to the Asterisk. However, I cannot make my Asterisk to send messages to the fixed phone. The SMS command displays TX and RX records, hang for a while and then stops with non-zero exits. I read somewhere in the technical manual of the phone that the phone should be able to identify the caller id in order to receive messages. My telephone line for the fixed phone has the callerid feature, but I guess I should config the phone as well to identify the SMS calls from the Asterisk, but I don't know how. Have anyone tried Asterisk with SMS? Yes, I did. The phone number you need to have Asterisk dial to send SMS messages is NOT the phone number of the phone you want to receive the SMS message, but that of your local SMS service center. You can receive this number from your local telco or from browsing your PSTN phone menus. Hope this helps, Gilad ___ 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 crashes my router!?
On Thu, 2004-12-02 at 16:47, David Filion wrote: Hi, Does anybody else have problems like this. I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek Vigour 2600 ADSL router. My * box is configured with a public IP address which is presented on one of the switch ports on the rear of the router. When there is some SIP activity, incoming mainly, towards my * box, the router will lockup after a short period?! Maybe the router can't handle the traffic? If you have a modem before your router, try connecting * right to the modem and using rpppoe. The Vigor 2600 should just do fine, I have 2 Vigor 2600VGi's in use with Asterisk (IAX though) and Handytone's (SIP, not connected to the Asterisk) in use (router PPPoE though, connected to NetSource here in Ireland). Firmware is 2.5.3 (i think Draytek withdrew that again because of GUI problems), but 2.5.2 was fine, too. The router is SIP aware, so actually you shouldn't think much about NAT setup etc, it should work straight away. Have you talked with Draytek about that problem ? Maybe they have heard about it before and have a solution. Besides, is it the UK specific firmware you have loaded ? Slán leat, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) Yes, try a firmware upgrade. I actually saw a router one time that would lockup if a client behind it ran a trace route ___ 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] Incoming SIP Address?
Ian Chilton wrote: I assume ports 5060 and 1-2 need to be opened in the firewall too. I don't know much about SIP and firewalls, but opening ten thousand ports doesn't sound good, you've just knocked 1/6 of your firewall down :-( ___ 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 sms voicemail notification
Hi Patric; I interested in your email on "Mon Oct 2004" with the subject "Howto get voicemail $VM_ vars into externnotify script?". Have you been able to set up such an application. If yes, plz help me to find out about that. Regards mohammad ___ 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] Incoming SIP Address?
Hi, Is it possible to have an incoming SIP address like [EMAIL PROTECTED], where sip.mydomain.com points to a box running Asterisk? If so, please could someone give an example asterisk config snippet for this? If it is possible, I assume ports 5060 and 1-2 need to be opened in the firewall too. Thanks! --ian Ian, you don't even have to create a subdomain for this. Include a 'SRV' entry in your DNS record and you can have [EMAIL PROTECTED] http://www.voip-info.org/wiki-DNS+SRV Cheers Shane ___ 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] Incoming SIP Address?
Hi, I assume ports 5060 and 1-2 need to be opened in the firewall too. I don't know much about SIP and firewalls, but opening ten thousand ports doesn't sound good, you've just knocked 1/6 of your firewall down That's what I thought but I was told it was the only way to get incoming SIP working when Asterisk was behind a firewall/NAT. I was told it was not a security risk to do this. Any thoughts anyone? --ian ___ 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] Incoming SIP Address?
Hi, Is it possible to have an incoming SIP address like [EMAIL PROTECTED], where sip.mydomain.com points to a box running Asterisk? If so, please could someone give an example asterisk config snippet for this? snip --ian Ian, you don't even have to create a subdomain for this. Include a 'SRV' entry in your DNS record and you can have [EMAIL PROTECTED] http://www.voip-info.org/wiki-DNS+SRV Cheers Shane Another good link Ian with working examples... http://slacker.com/~nugget/asterisk7.php -Shane ___ 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 crashes my router!?
Hi Martin, my router is a vanilla 2600, not the V model, as far as I know it has no special SIP features, other than SIP seeming to crash it when a SIP call is made from the internet to the * box here! :( I mentioned the problem on the draytek forum but I;ve not contacted Draytek themselves per se. One big difference is you are using PPPoE and I'm using PPPoA, unfortunately! I've tried several different firmware, all UK specific, still the same. thanks. Mike On Fri, 03 Dec 2004 23:39:50 +, Martin List-Petersen [EMAIL PROTECTED] wrote: The Vigor 2600 should just do fine, I have 2 Vigor 2600VGi's in use with Asterisk (IAX though) and Handytone's (SIP, not connected to the Asterisk) in use (router PPPoE though, connected to NetSource here in Ireland). Firmware is 2.5.3 (i think Draytek withdrew that again because of GUI problems), but 2.5.2 was fine, too. The router is SIP aware, so actually you shouldn't think much about NAT setup etc, it should work straight away. Have you talked with Draytek about that problem ? Maybe they have heard about it before and have a solution. Besides, is it the UK specific firmware you have loaded ? Slán leat, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom 500, won't ring??
Title: Message You might want to check your phone directory file. In there you can specify a ring type for a identified incoming caller - perhaps you have specified ring type 0 which is by default silent. Peter -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared ArmstrongSent: Saturday, 4 December 2004 8:31 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Polycom 500, won't ring?? Hi, I have was testing some of the different ring types with my polycom 500, and the ALERT_INFO settings. Now when my phone receives a call it wont ring. All the other phones ring fine, and my phone wasnt the only one I rebooted with the changed sip.conf and impd.conf. I have reverted back to a standard sip.conf and impd.conf and I still can not get my phone to ring for any incoming calls. Does anyone have any suggestions to look for? Jared Armstrong ___ 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] Incoming SIP Address?
Hi Shane, http://www.voip-info.org/wiki-DNS+SRV http://slacker.com/~nugget/asterisk7.php The SRV page was useful - i've done that in my domain now. But, the other page is talking more about dialing sip addresses through Asterisk rather than incoming sip addresses. However, after adding the SRV record into DNS and the following into Asterisk in extensions.conf, it seems to work: [default] exten = ian,1,Dial(SIP/spa3k_line1,10) exten = ian,2,Voicemail(u4) exten = ian,3,Hangup Is this the right/best way to do it? Is there any way to get such calls coming into a dedicated context, rather than default? Thanks! --ian ___ 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] Bluetooth with *
Thanks! :) On Sat, 4 Dec 2004 10:19:59 +, Theo P. Zourzouvillys [EMAIL PROTECTED] wrote: On Saturday 04 December 2004 04:43, Nate Carlson wrote: In other words, if it's something you really want, add more cash to the bounty, to help encourage the developer to spend more time on it *grin*: alright, alright - i'll work on it today :-) ~ Theo -- Theo P. Zourzouvillys [EMAIL PROTECTED] http://www.crazygreek.co.uk/ ___ 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] Udev setup question for zaptel
Trying to setup asterisk and zaptel on a Fedora Core 3. Its all working after reading up on udev but I still get errors. [EMAIL PROTECTED] ~]# ztcfg -v Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 2 channels configured. Notice: Configuration file is /etc/zaptel.conf line 4: Unable to open master device '/dev/zap/ctl' I added the suggested lines to /etc/udev/rules.d/50-udev.rules that were in the zaptel README.udev, as I understood them? # Section for zaptel device KERNEL=zapctl, NAME=zap/ctl KERNEL=zaptimer, NAME=zap/timer KERNEL=zapchannel, NAME=zap/channel KERNEL=zappseudo, NAME=zap/pseudo KERNEL=zap[0-9]*, NAME=zap/%n When I load the zaptel modules, they work the errors are just distracting. Any suggestions would be great. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI debug output - still not working :(
Hi all, I'm debugging a PRI problem, i can see the calling number but i get a busy all the time. From the output below, I guess asterisk hangs up immediately. Can anyone point out what the problem is? Thanks in advance. *CLI Protocol Discriminator: Q.931 (8) len=32 Call Ref: len= 2 (reference 4865/0x1301) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 9f] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 31 ] [1e 02 81 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 06 01 80 33 39 31 30] Calling Number (len= 8) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3910' ] [70 02 81 35] Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5' ] [a1] Sending Complete (len= 1) -- Making new call for cr 4865 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) -- Processing IE 161 (cs0, Sending Complete) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 37633/0x9301) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI debug - weird behaviour
Hi all, another thing i noticed, when i start asterisk and type pri show span 1, i get the following: Primary D-channel: 16 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 As soon as i type pri debug span 1, i get lots of messages that is appended at the end of the email and now pri show span 1 shows only this: *CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Up, Active Is this a normal behavior? [messages that appear after i enable debugging] *CLI -- Making new call for cr 32768 Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 32768/0x8000) (Terminator) Message type: RESTART ACKNOWLEDGE (78) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 121 (cs0, Restart Indicator) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 32768/0x8000) (Terminator) Message type: RESTART ACKNOWLEDGE (78) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 121 (cs0, Restart Indicator) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 83] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 3 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 32768/0x8000) (Terminator) Message type: RESTART ACKNOWLEDGE (78) [18 03 a9 83 83] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 3 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 121 (cs0, Restart Indicator) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 84] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 4 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 32768/0x8000) (Terminator) Message type: RESTART ACKNOWLEDGE (78) [18 __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP Address?
I assume ports 5060 and 1-2 need to be opened in the firewall too. I don't know much about SIP and firewalls, but opening ten thousand ports doesn't sound good, you've just knocked 1/6 of your firewall down That's what I thought but I was told it was the only way to get incoming SIP working when Asterisk was behind a firewall/NAT. I was told it was not a security risk to do this. Any thoughts anyone? If your configuration and firewall actually require you to open a group of ports to *, then take a look at limiting the rtp ports that are actually used. Examples: - in /etc/asterisk/rtp.conf, look at changing rtpstart and rtpend - for cisco 7960's, look in SIPDefault.cnf for start_media_port and end_media_port - other sip phones often times use other rtp ports, some of which are configurable (and some phones not). Each sip phone vendor use a different range of rtp ports. To reduce the security exposures, one can also use firewall filters to allow only certain external IP addresses (if your firewall supports that function), and/or sip.conf definitions that include something like: deny=0.0.0.0/0.0.0.0 permit=47.136.1.129/255.255.255.0 If you really need to do this, you will almost always need a packet sniffer to see what is actually happening on the inside edge of your firewall and on the outside edge. Without such packet traces changing parameters is nothing more then a guessing game. ___ 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] non blind call transfers
On Friday 29 October 2004 21:17, lenz wrote: Hello list, I was looking for a way to implement non-blind call transfers with *, i.e. the usual behaviour of most PBXs when pressing the flash button: - A and B are talking - A pushes flash - A is free to compose a new number - B hears music on hold - A's call is answered by C - A hangs up - B and C are in conversation As much as I can understand, * only supports blind transfers, where if C does not answer the phone there is no way for A to get back to B. Is there a way to have a standard flash behaviour? The above is exactly what happens with my system - I've not done anything special (ie patches) to make this happen. I can do attended transfers by simply doing 'flash' while in a call, dial the new number and talk, press flash again and hang up. It works perfectly for me :) pressing # while in a call allows blind transfers. Jon ___ 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] TDM400P FXO channel remains Offhook after outoing or incoming call / line is parallel with other telephone equipment
Inline... I am preparing to roll out Asterisk setup with TDM400P, 4 FXO modules in a small office. Asterisk will replace legacy system (4 telco lines, 8 extensions PBX), but before the new system and ip phones would be installed, the legacy system is still in use. The four telco lines are now connected in parallel both to legacy PBX and the 4 FXO modules in TDM400P. Asterisk is configured not to pick up any incoming calls. snip I am dialing out via Asterisk and it works fine untill the following situation: - one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that line) - both via old and new PBX. - zap show channel n would show that line as 'Offhook', though no telephone is off hook. If physical line would be unplugged from TDM card, the line would become normal again. Others have posted similar type issues relative to the TDM400P card. Not sure if that issue has been addressed in code or not. I'm not seeing that problem with current cvs head. Sorry, if it is a well known problem, but I did not find any specific information yet.. Please answer two questions: - is it really bad to have parallel connection on TDM400P FXO lines to an additional telephone equipment, does it prevent TDM400P to detect Offhook/Onhook correctly? - will the problems go away when parallel lines would be disconnected (legacy PBX shut down)? In very general terms, no its not a problem. It certainly can be a problem when, as an example only, the legacy pbx is actually using a line and * attempts to use the same line. (eg, * is not going to check to see if dialtone or voice is present before dialing.) Same in reverse; if * has a line in use, will the legacy pbx detect that before dialing? The 3050 chip on the TDM fxo module has the capability of sensing whether another analog device on the pstn is off hook (ie, much lower line voltage), but I'm 80% sure that flag is not currently handled by the tdm drivers, etc. As you may understand the office personnel has anxiety that this may be a bad Asterisk setup / bad TDM card etc (which I am sure so far that it is not). Sounds like you really need to go through a pre-cutover test plan without the legacy pbx attached to validate your config, etc. The tdm card does have some unusual issues that appear to be driver oriented, but there are lots of folks using the card in production. ___ 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] DIALSTATUS weirdness (CHANUNAVAIL instead of BUSY, NOANSWER instead of CHANUNAVAIL)
On December 3, 2004 03:36 pm, Andrew Kohlsmith wrote:
exten = 1234,1,Dial(Zap/g1/5551234,,g)
exten = 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
${DIALSTATUS})
Why, if 5551234 is busy, is DIALSTATUS set to CHANUNAVAIL? Should it not
be BUSY?
Brian West pointed me at chan_zap.c where there is a configuration parameter
called priindication which can be used to set the pri indication to inband
or out of band, defaulting to out of band.
I have set priindication=outofband in zapata.conf, now I will test this later
but it looks like it will work.
Posting a new thread now in -dev as to why the blue f*ck there is a pri
inband
configuration option that is
a) undocumented and
b) defaults to inband
The mind boggles -- PRI is *always* out of band.
Looks like the command is documented in the current config samples.
I'm not knowledgable/experienced (as yet) on where it is actually used,
but just reading the comments in the config sample led me question the
writers use of the terms inband and outofband relative to a pri.
Since the comments use words like doesn't work with all telcos,
could this have something to do with detecting busy when a call
reaches a destination lurking behind an analog system? (eg, pri
call placed to a DID number on an analog pbx where the d channel
isn't aware of the destination's status?)
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Re: [Asterisk-Users] TDM400P FXO channel remains Offhook after outoing or incoming call / line is parallel with other telephone equipment
Andrei : In zapata.confyou must activate the following lines busydetect=yes busycount=4 regards Rodrigo - Original Message - From: Andrei (MPI) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, December 04, 2004 4:57 AM Subject: [Asterisk-Users] TDM400P FXO channel remains Offhook after outoing or incoming call / line is parallel with other telephone equipment Hi, I am preparing to roll out Asterisk setup with TDM400P, 4 FXO modules in a small office. Asterisk will replace legacy system (4 telco lines, 8 extensions PBX), but before the new system and ip phones would be installed, the legacy system is still in use. The four telco lines are now connected in parallel both to legacy PBX and the 4 FXO modules in TDM400P. Asterisk is configured not to pick up any incoming calls. zapata.conf: signalling=fxs_ks musiconhold=default languages=en context=inbound-analog group = 1 channel = 1-4 zaptel.conf: loadzone = us defaultzone=us fxsks=1-4 I am dialing out via Asterisk and it works fine untill the following situation: - one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that line) - both via old and new PBX. - zap show channel n would show that line as 'Offhook', though no telephone is off hook. If physical line would be unplugged from TDM card, the line would become normal again. Sorry, if it is a well known problem, but I did not find any specific information yet.. Please answer two questions: - is it really bad to have parallel connection on TDM400P FXO lines to an additional telephone equipment, does it prevent TDM400P to detect Offhook/Onhook correctly? - will the problems go away when parallel lines would be disconnected (legacy PBX shut down)? As you may understand the office personnel has anxiety that this may be a bad Asterisk setup / bad TDM card etc (which I am sure so far that it is not). Please help. Sincerely, Andrei ___ 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] Ouch, part reset, quickly
On December 4, 2004 12:59 am, Dinesh Nair wrote: i've debugged the driver well enough and know that the Ouch message happens when register 0x08 of the module returns 0, which indicates in most times that digital loopback is enabled on the card. this register is set to /disable/ digital loopback upon an init. the power alarm happens when the line feed (hookstate) of the module is not in sync with a driver variable which tracks hookstate. the resetting bit you see is just informational to let you know that the driver is setting the on-module registers back to what the internal variable says it should be. Very interesting; thank you for sharing this. In my experience the card starts to act funny when I get ONE of these -- perhaps in some situations there are more than one register that is going awry and the resetting code doesn't reset them all? I should hack in some debug code that dumps the registers whenever it detects this power alarm. I should also grab the datasheets and any erratta for the SLIC chipset and see if anything interesting turns up. Thanks for giving me a direction to start in. i can explain what the driver does when these things happen, however, i'm thinking that it's more of a hardware issue than anything else. based on my (admittedly limited) reading of the Tiger320 ProSLIC datasheet, the registers mentioned shouldnt go awry, yet they do. I thought the TJ320 was a PCI bridge that provided an 8-bit parallel interface, timer and a serial interface or two, and that there was a separate SLIC chipset which did the actual interfacing to the phone line. -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] howto install
[EMAIL PROTECTED] wrote: Hello, I am using Mandrake 10.1. Howto to install asterisk. I have downloaded tarball. I have not installed any hardware yet. Is it possible to install ? Yes, http://www.voip-info.org/wiki-Asterisk Doug ___ 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] drive space for voice mail
snip Use a good card like the 3ware 7500 series (parallel IDE ATA) and there are no problems using IDE ATA drives. 3ware uses hardware raid unlike the garbage promise chips that Claim hardware raid, but are not in reality. IED Raidsets on 3ware show up as scsi drives to the system. 3ware is one of those rare companies that have Great linux support. You get what you pay for. The controller card may cost as much or more than the drives. Linux SATA support is still a little weak, but the performance can be much better for the higher-end SATA drives. Use of a good raid card like 3ware makes Linux compatability a non-issue. I agree that software raid should be avoided. /snip Thanks for the tip on the 3Ware cards. Looks like I can pick up the 8000 (SATA RAID 0,1) 2-port for around $150.00 US. Thanks again, -Ron ___ 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
RES: [Asterisk-Users] howto install
Yes it is possible, you will running IAX and SIP phones only. You can get more details about the installation www.voip-info.org This are some good links to start. http://www.voip-info.org/wiki-Asterisk+introduction http://www.voip-info.org/wiki-Asterisk+installation+tips Bye Gerald -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de [EMAIL PROTECTED] Enviada em: sábado, 4 de dezembro de 2004 08:18 Para: [EMAIL PROTECTED] Assunto: [Asterisk-Users] howto install Hello, I am using Mandrake 10.1. Howto to install asterisk. I have downloaded tarball. I have not installed any hardware yet. Is it possible to install ? Thanks Varun ___ 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] DIALSTATUS weirdness (CHANUNAVAIL instead of BUSY, NOANSWER instead of CHANUNAVAIL)
On Sat, 4 Dec 2004, Rich Adamson wrote: The mind boggles -- PRI is *always* out of band. Looks like the command is documented in the current config samples. I'm not knowledgable/experienced (as yet) on where it is actually used, but just reading the comments in the config sample led me question the writers use of the terms inband and outofband relative to a pri. Since the comments use words like doesn't work with all telcos, could this have something to do with detecting busy when a call reaches a destination lurking behind an analog system? (eg, pri call placed to a DID number on an analog pbx where the d channel isn't aware of the destination's status?) From what I can see the only thing it changes is that the Busy and Congestion applications / indications from other sources send audio signals using the normally opened reverse path from the B subscriber to the A subscriber before the channel is answered. It may be used by Dial as well, I have not checked. With the priindication = outofband those situations will send an isdn release with the specified code. This can also be achieved by setting the PRI_CAUSE variable prior to calling Hangup(). Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP Address?
Ian Chilton wrote: That's what I thought but I was told it was the only way to get incoming SIP working when Asterisk was behind a firewall/NAT. I was told it was not a security risk to do this. If you *know* that only asterisk is listening on the relevant ports it's less of a risk, but it's such a wide range and (in theory at least) leaves plenty of scope for a trojan to listen on one of those ports. Perhaps SElinux can help here, does it allpw you to say that only a cerain process has access to the those ports? Arrghh, I hate the way to:, from: and reply-to: addresses get mangled by lists! ___ 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] IAXy and ADPCM codec problem
Carlos Clemares wrote: Hi everyone, I'm using the IAXy boxes and i'm having some trouble when I use it with the ADPCM codec. The IAXy only does ULAW ___ 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] Blank Machine Again.
Alan Ingleby wrote: I also wanted to set up this machine to be our network firewall/nat Our existing firewall runs linux on a p90, and runs fine, but I figured it's time to upgrade.. Will this cause any problems for *? You might want to look into fli4l (http://www.fli4l.de). It is a router/whatever plus there is a module add-on with asterisk. Might be worth a try. hth rgds pos Is there a good site to check this out that is in English? ___ 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] ASTCC configuration problem
Same thing here. It used to work perfectly until I re-installed. Hi I need some advice in this issue, I installed astcc again and creates database from configure menu but I am still getting errors messages: in Brands menu: Something is wrong with the brands database in Cards menu: Please define at least one brand before creating cards in Trunks menu: Something is wrong with the trunks database in Routes menu: Please define at least one trunk before creating routes in Users_Configure: Not Configured! Also I have noted that created database exists but its tables are empty the same as the file astcc-config.conf although the apache user has rights on it... any idea thanks Rafael On Wed, 1 Dec 2004 16:23:25 -0500, Rafael J. Risco G.V. [EMAIL PROTECTED] wrote: hi Today I´ve installed, apache 2.0.52, mysql-4.1.7, asterisk-perl-0.08 and ASTCC prepaid card aplication from CVS, so now I have access to the astcc-admin.cgi from web server http://asterisk/cgi-bin/astcc-admin/astcc-admin.cgi and I´ve been able to create the database from Configure menu but I have some doubts to continue: - Do I have to reinstall asterisk with mysql support? - How asterisk or astcc knows what db should use, where the configuration files are? - when I go to Cards I ge this: Please define at least one brand before creating cards and when I go to Brands, I have this error: Something is wrong with the brands database ...in Users_Configure menu I have this error: Not Configured! please send me some advise to continue, thanks Rafael Lima-Peru -- rrgv -- rrgv ___ 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] Codec translator problem (g723.1,ilbc = alaw)
Hi, I cannot get SIP channel working with folowing codec configuration: [sip] disallow=all allow=g723.1 ;I need this codec between sip phones (BT100) allow=ilbc ;Use this codec to others Calling between BT100 SIP phones is OK - asterisk makes native bridge (with g723.1) between them. When I'm calling from SIP to other channel (iax,zap,...), asterisk is not able to chose right codec and is trying transalate g723.1 to alaw, instead of choose ilbc and translate to alaw. Thanks in advance Petr Michalek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_misdn and Dynalink IS64PH ISDN
Hi, TE Stack No Upper ID init_stack: File exists You need to set the layermask when loading the card driver. For a TE port, use 15 (layer 0-3) and for an NT port, use 3 (layer 0-1). Simon ___ 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] Blank Machine Again.
Steve Totaro schrieb: You might want to look into fli4l (http://www.fli4l.de). It is a router/whatever plus there is a module add-on with asterisk. Might be worth a try. Is there a good site to check this out that is in English? For fli4l itself, yes. For the opt_modul, no. After reading the documentation of fli4l in english the module will should be fairly easy to understand. You can find the location of the OPT_Module using http://www.voip-info.org (search for fli4l) If you have any specific parts of the documentation you do not understand, feel free to mail me. rgds pos ___ 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] XML to monitor queues on Cisco display ?
Hi everybody, I'd like to know if anybody tried to write a xml doc to monitor the number of calls in Q, when working with an ACD it's convenient to see how many calls are waiting so the agent can speed up the conversation when it gets too busy :-) I was wondering if it was poss to display this info on a display of a cisco 7940 / 60 ? any idea ? jl ___ 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] XML to monitor queues on Cisco display ?
Hi everybody, I'd like to know if anybody tried to write a xml doc to monitor the number of calls in Q, when working with an ACD it's convenient to see how many calls are waiting so the agent can speed up the conversation when it gets too busy :-) I was wondering if it was poss to display this info on a display of a cisco 7940 / 60 ? any idea ? jl ___ 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] (no subject)
test ___ 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] XML to monitor queues on Cisco display ?
Jean-Louis curty to Asterisk More options 4:38pm (7 minutes ago) Hi everybody, I'd like to know if anybody tried to write a xml doc to monitor the number of calls in Q, when working with an ACD it's convenient to see how many calls are waiting so the agent can speed up the conversation when it gets too busy :-) I was wondering if it was poss to display this info on a display of a cisco 7940 / 60 ? any idea ? ___ 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] iaxy to iaxy call drops out of show channels
I place a call from an IAXY to an IAXY device. INitially the calls show in the output of show channels. Then after a few seconds the show channels command shows 0 active channels even though I am still talking on the channels. Any ideas on this? THanks, Jerry ___ 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] XML to monitor queues on Cisco display ?
I attempted this but I got stuck on one issue. Cisco phones pull data so I couldn't get them to autoupdate. In other words push data to them. I am working on an app to run on a windows desktop that will show the queues, the amount of calls in each queue, the longest wait time and the average wait time. I am also planning on creating the app with alarm thresholds. When the app is minimized it will go to the task bar and if the queue gets too full it will popup the window on the desktop and/or make the icon in the taskbar turn red. Henry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Louis curty Sent: Saturday, December 04, 2004 9:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] XML to monitor queues on Cisco display ? Hi everybody, I'd like to know if anybody tried to write a xml doc to monitor the number of calls in Q, when working with an ACD it's convenient to see how many calls are waiting so the agent can speed up the conversation when it gets too busy :-) I was wondering if it was poss to display this info on a display of a cisco 7940 / 60 ? any idea ? jl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with SMS
My intention is to setup Asterisk to be a message center to receive from and send SMS to fixed phones. Can it be possible? My fixed phone can dial to Asterisk and send SMS to Asterisk, but I cannot setup the other way: make Asterisk dial to fixed phone and send SMS to fixed phone. On Sat, 04 Dec 2004 13:09:27 +0200, Gilad Ben-Yossef [EMAIL PROTECTED] wrote: Nguyen Quang Hoa wrote: Hi all, I am trying to setup the SMS of Asterisk. I have a Siemens SMS enable fixed phone which connects to my Asterisk through PSTN. Currently, I can use my fixed phone to edit and send messages to the Asterisk. However, I cannot make my Asterisk to send messages to the fixed phone. The SMS command displays TX and RX records, hang for a while and then stops with non-zero exits. I read somewhere in the technical manual of the phone that the phone should be able to identify the caller id in order to receive messages. My telephone line for the fixed phone has the callerid feature, but I guess I should config the phone as well to identify the SMS calls from the Asterisk, but I don't know how. Have anyone tried Asterisk with SMS? Yes, I did. The phone number you need to have Asterisk dial to send SMS messages is NOT the phone number of the phone you want to receive the SMS message, but that of your local SMS service center. You can receive this number from your local telco or from browsing your PSTN phone menus. Hope this helps, Gilad ___ 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] Why, why, why???
We have Grandstream SIP phones with the latest firmware versions and have also have this problem. It appears to be something to do with RTP, I believe. I don't know exactly what (simply because I don't know much about RTP as yet), but the packets don't seem to reach the Grandstream from the other phone. The phones appear to work correctly when located on the same LAN segment. But, when one is placed behind a NAT router, the dynamic changes and one-way audio seems to happen frequently. I've Are you forwarding ports? What ports have you set asterisk to? IIRC the GS phones come with 8000 by default and asterisk comes with 1. ___ 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: calling an iaxy
Thomas Niesel wrote: Hallo rich allen I get this same error. Very strange. Dialing out from the IAXy works fine, message: Accepted AUTHENTICATED TBD call from 192.168.2.111 Accepted DIAL from 192.168.2.111, formats = 0x4 I also turned on Qualify and IAX debugging, and it reported my IAXy was alive and well. Yet dialing into the IAXy produces the error below. Portions of my config files are at the tail of the email. i have an IAXy which i can make calls from but am unable to call. when i dial the extension assigned, i get the following from the console; -- Executing Dial(SIP/5801-b665, IAX2/5899 at 192.168.0.5) in new stack -- Called 5899 at 192.168.0.5 -- Call accepted by 192.168.0.5 (format ULAW) Nov 1 12:28:33 NOTICE[163850]: chan_iax2.c:5546 socket_read: Rejected call to 192.168.0.5, format 0x4 incompatible with our capability 0xff03. Hm, I'm not an expert on iaxY but it looks like that the codec is the prob. If both sides do not find a common codec the call will be rejected. Try to call with alaw or gsm and see if it helps. The IAXy doesn't do anything but ulaw and adpcm. I would stick with ulaw for testing. I would start looking in the iax.conf entry for the iaxy for the culpit. B. provision file: ip: 192.168.2..111 netmask: 255.255.255.0 codec: adpcm ; also tried adpcm server: 192.168.2.110 user: myuser pass: mypass iax.conf: [myuser] type=friend accountcode=iaxy host=192.168.2.111 secret=mypass context=iaxycontext disallow=all allow=ulaw ; also tried adpcm callerid=My IAXy trunk=no extensions.conf exten = s,2,Dial(IAX2/myuser/s) ___ 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] iaxy to iaxy call drops out of show channels
Sure, the IAXy's do a reinvite and * drops out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jerry Geis Sent: Saturday, December 04, 2004 10:50 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] iaxy to iaxy call drops out of show channels I place a call from an IAXY to an IAXY device. INitially the calls show in the output of show channels. Then after a few seconds the show channels command shows 0 active channels even though I am still talking on the channels. Any ideas on this? THanks, Jerry ___ 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] iaxy to iaxy call drops out of show channels
On Sat, 04 Dec 2004 10:49:42 -0500, Jerry Geis wrote: I place a call from an IAXY to an IAXY device. INitially the calls show in the output of show channels. Then after a few seconds the show channels command shows 0 active channels even though I am still talking on the channels. Any ideas on this? This could happen if the wo devices negotiate a direct connection. IAX has a mechanism to do this just like REINVITE in SIP. Once the call is setup * connects the two end points directly and takes itself out of the call path. This happens as long as * is not required to transcode between codecs or protocols. Also, there is a parameter for IAX.CONF called NOTRANSFER= that can user enable/disable such things on a per peer basis. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list [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] compiling asterisk-addons for Mysql-cdr
Hi ALL; I got the latest Asterisk-addons for Mysql-Cdr, but I have problem compiling that.It says: # make . res_config_mysql.c: In function `realtime_mysql':res_config_mysql.c:143: warning: passing arg 1 of `ast_strlen_zero' makes pointer from integer without a castres_config_mysql.c: In function `realtime_multi_mysql':res_config_mysql.c:242: warning: passing arg 1 of `ast_strlen_zero' makes pointer from integer without a castres_config_mysql.c: In function `load_module':res_config_mysql.c:467: structure has no member named `static_func'res_config_mysql.c:468: structure has no member named `realtime_func'res_config_mysql.c:469: structure has no member named `update_func'res_config_mysql.c:470: structure has no member named `realtime_multi_func'make: *** [res_config_mysql.o] Error 1rm app_saycountpl.o Appreciate any help mohammad ___ 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: calling an iaxy
Well, if these are the latest versio,ns of your files... provision file: codec: adpcm iax.conf: disallow=all allow=ulaw ___ 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] Codec translator problem (g723.1,ilbc = alaw)
asterisk wrote: Hi, I cannot get SIP channel working with folowing codec configuration: [sip] disallow=all allow=g723.1 ;I need this codec between sip phones (BT100) allow=ilbc ;Use this codec to others Calling between BT100 SIP phones is OK - asterisk makes native bridge (with g723.1) between them. When I'm calling from SIP to other channel (iax,zap,...), asterisk is not able to chose right codec and is trying transalate g723.1 to alaw, instead of choose ilbc and translate to alaw. Thanks in advance Petr Michalek Petr, Asterisk only has passthrough for G.723.1. It cannot transcode it at all. You will have to use ilbc, GSM, G726, etc... -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: calling an iaxy
-- Called 5899 at 192.168.0.5 -- Call accepted by 192.168.0.5 (format ULAW) Nov 1 12:28:33 NOTICE[163850]: chan_iax2.c:5546 socket_read: Rejected call to 192.168.0.5, format 0x4 incompatible with our capability 0xff03. Hm, I'm not an expert on iaxY but it looks like that the codec is the prob. If both sides do not find a common codec the call will be rejected. Try to call with alaw or gsm and see if it helps. The IAXy doesn't do anything but ulaw and adpcm. I would stick with ulaw for testing. That's according to the docs It only supports ULAW as far as I can tell in trying to provision mine to anything else. Maybe digium's marketing is driving development? ;) ___ 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 create channel of type 'Zap' (cause 0)
Leonardo J. Tramontina wrote: No, I don't have anything connected on the TE110P. After the Unable to create channel of type 'Zap' (cause 0) message, I also get the CHANUNAVAIL... Is not possible test a channel from the card without connections on it?? Leonardo *snipped no, the only way is if you have another t1/e1 device that can 'act' like net/cpe. so if you had a 4 port digium t1/e1 card, you could have made a t1 crossover cable, and connected say port 1 to port 4 and setup a dialplan that would have the one port group for sending out, and one group for receiving the 'fakeout' traffic. ___ 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] Ouch, part reset, quickly
i've debugged the driver well enough and know that the Ouch message happens when register 0x08 of the module returns 0, which indicates in most times that digital loopback is enabled on the card. this register is set to /disable/ digital loopback upon an init. the power alarm happens when the line feed (hookstate) of the module is not in sync with a driver variable which tracks hookstate. the resetting bit you see is just informational to let you know that the driver is setting the on-module registers back to what the internal variable says it should be. Very interesting; thank you for sharing this. In my experience the card starts to act funny when I get ONE of these -- perhaps in some situations there are more than one register that is going awry and the resetting code doesn't reset them all? I should hack in some debug code that dumps the registers whenever it detects this power alarm. I should also grab the datasheets and any erratta for the SLIC chipset and see if anything interesting turns up. Thanks for giving me a direction to start in. i can explain what the driver does when these things happen, however, i'm thinking that it's more of a hardware issue than anything else. based on my (admittedly limited) reading of the Tiger320 ProSLIC datasheet, the registers mentioned shouldnt go awry, yet they do. I thought the TJ320 was a PCI bridge that provided an 8-bit parallel interface, timer and a serial interface or two, and that there was a separate SLIC chipset which did the actual interfacing to the phone line. FYI, the tdm fxo chip set from Silicon Labs ( www.silabs.com ) uses the 3019 for handling pstn line-side interface (and electrical isolation) and the 3050 for PCM encode/decode, impedance, hybrid, near-end echo cancellation, interrupts, etc. The *.pdf's are rather hard to find on their site but very detailed (3050 has 110 pages). A quick check of SI's revision history tends to suggest very few anomalies since released in 2003. The very first tdm fxo modules sold by digium used the rev-C and rev-D chips. The Tiger320 handles, as you mentioned, the pci v2.2 bus interfacing. Given the sophistication of the SI chip set, it would appear that at least some functionality exists that has not been taken advantage of within the wctdm/zaptel drivers, etc. Part of that history is probably related to attempted reuse of code that was written for the x100p (in multiple * modules and drivers). ___ 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] ZAP and IAX Trunks
Hi Walid, Welcome to the list. Zap are the connections from ordinary pstn (or telco lines) to your digium hardware. IAX is an Asterisk protocol for incoming lines via IP from another asterisk PABX. Hope this helps. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walid Azab Sent: Saturday, December 04, 2004 5:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ZAP and IAX Trunks HelloEveryone, I have recently come across these two terms. I am new at Asterisk and do appreciate your assistance in this. Some tools such as astGUIclient and Asterisk Management Portal require that the phone system be running Zap or IAX trunks as well as SIP devices. SIP devices are understadable but what about the other two? I am planning to use Cisco 7960/7940 IP phones. Thanks Walid ___ 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 with SMS
B G wrote: My intention is to setup Asterisk to be a message center to receive from and send SMS to fixed phones. Can it be possible? My fixed phone can dial to Asterisk and send SMS to Asterisk, but I cannot setup the other way: make Asterisk dial to fixed phone and send SMS to fixed phone. Ah, I see. In that case your phone must have set in it's menu some caller ID (number) that Asterisk should be made to set the caller ID to when it is trying to send the SMS. That's the only way for the phone to figure out that an SMS is being sent to it when Asterisk rings. Gilad ___ 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] more than 3 msns with chan_capi
Hi, sorry for newbie Fritz question. I always thought that AVM Fritz has 2 devices for 2 MSNs. So does this mean, that Fritz can handle more ISDN lines ? Does this mean you can have more than 2 calls at once ? What is MAX number of parallel calls ? Thanks in advance, Regards, Robert. - Original Message - From: Martin List-Petersen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, December 04, 2004 12:57 AM Subject: RE: [Asterisk-Users] more than 3 msns with chan_capi On Thu, 2004-12-02 at 22:18, Derek Conniffe wrote: I notice that you've put the msns in as the msn field and have the incomingmsn as a * character. I have lots of msns too and they all work just fine (SuSE 9.1 AVM Fritz chan_capi) *BUT* I have the msn field as my OUTGOING MSN for others to see exactly, and you would want to define more than one MSN there, if you want to show different MSNs, as Jens did define it. However for Denmark he should specify the whole 8 digits for every number. For Ireland you need to specify the whole number without prefix. (7 digits for Dublin) and my incomingmsn as the list of comma separated MSNs to accept incoming calls on - maybe you should try this out?. Nope * works without problems in Ireland, too. It'll happily take any MSN then, that comes in. Convienient, since the two MSN lines take a max of 5 or 6 MSNs, but i as a example got Eircom to provision 12 MSNs on my ISDN line (in PtMP mode !!) Here in Ireland I have to supply the last 4 digits of the msn Yep, Eircom is only sending you the last 4 digits, but when you want to show your MSN you need to send all 7 digits. Very odd setup, but it's the way Eircom seems to do things. Slán leat, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) ___ 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] TDM400P FXO channel remains Offhook after outoing or incoming call / line is parallel with other telephone equipment
Rich Adamson wrote: Inline... snip Rich, Thank you for your answer. Now I've figured that one of the FXO modules on the card may be defective. Whenever I plug in telco line in it - that line will be like shortened (if you pick up parallel telephone, the dial tone will be heard weaker than usually). So the FXO module is always in Offhook state, unable to dial out, unable to detect rings. Reboot and Power off/Power on did not help. Any suggestions? Might be just my luck.. just my luck. Sincerely, Andrei ___ 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: calling an iaxy
Typing trouble on my part. Should have said: provision file: codec: ulaw iax.conf: disallow=all allow=ulaw Elided email follows at end. - Original Message - From: Wilson Pickett [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, December 04, 2004 10:33 AM Subject: Re: [Asterisk-Users] Re: calling an iaxy Well, if these are the latest versio,ns of your files... provision file: codec: adpcm iax.conf: disallow=all allow=ulaw ___ 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 provision file: ip: 192.168.2..111 netmask: 255.255.255.0 codec: adpcm ; also tried adpcm server: 192.168.2.110 user: myuser pass: mypass iax.conf: [myuser] type=friend accountcode=iaxy host=192.168.2.111 secret=mypass context=iaxycontext disallow=all allow=ulaw ; also tried adpcm callerid=My IAXy trunk=no extensions.conf exten = s,2,Dial(IAX2/myuser/s) -- Executing Dial(SIP/5801-b665, IAX2/5899 at 192.168.0.5) in new stack -- Called 5899 at 192.168.0.5 -- Call accepted by 192.168.0.5 (format ULAW) Nov 1 12:28:33 NOTICE[163850]: chan_iax2.c:5546 socket_read: Rejected call to 192.168.0.5, format 0x4 incompatible with our capability 0xff03. ___ 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] IAX Native Transfer
Hello, I'm having an issue with native transfer not happening. I have a * machine speaking ILBC in the middle of two * machines - everybody on ILBC, but for some reason they will not transfer. All machines have public IP addresses and can communicate directly with one another. One thing I notice is that one of the endponts registers itself on port 1025 as opposed to 4569 for some strange reason. I don't see any errors about binding to port 4569, so I'm wondering what's the deal. Any ideas? Thanks very much in advance! Thomas Hutton ___ 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] TDM400P FXO channel remains Offhook after outoing or incoming call / line is parallel with other telephone equipment
Rich Adamson wrote: The tdm card does have some unusual issues that appear to be driver oriented, but there are lots of folks using the card in production. Usually in situations where the client knows how to and tolerates having to reload drivers and/or reboot his PBX periodically... Regards, 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] XML to monitor queues on Cisco display ?
Henry Devito wrote: I attempted this but I got stuck on one issue. Cisco phones pull data so I couldn't get them to autoupdate. In other words push data to them. I am working on an app to run on a windows desktop that will show the queues, the amount of calls in each queue, the longest wait time and the average wait time. I am also planning on creating the app with alarm thresholds. When the app is minimized it will go to the task bar and if the queue gets too full it will popup the window on the desktop and/or make the icon in the taskbar turn red. Henry Henry, that is a very useful app indeed! Do you plan to share that, sell it, ?? Love to get more info or help.. Cheers, Wayne ___ 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] XML to monitor queues on Cisco display ?
Quoting Henry Devito [EMAIL PROTECTED]: I attempted this but I got stuck on one issue. Cisco phones pull data so I couldn't get them to autoupdate. In other words push data to them. You can use an http Refresh to keep the screen updating once you've accessed your XML application. It's not the best solution, but it is a step closer. --Shane ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN kernel 2.6 problems chapi isdn4lin
Hello! I've encauntered some serious problems with asterisk. I have to install it on system: 1. Mandrake 10.1 2. kernel 2.8.1 3. four ISDN cards. And I am in big trouble, isdn4linux is no longer supported for kernels 2.6 (on this system there are not any /dev/ttyI0 and similar devices)/ msidn - is unstable and for brave people chapi - I can't compile (lot of errors and I don't know why) i tired to patch it but it didn't help :(. I don't know what to do and I need solution very fast. Thanks for any help. BR, Corvin ___ 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] Incoming SIP Address?
Hi Rick, If your configuration and firewall actually require you to open a group of ports to *, then take a look at limiting the rtp ports that are actually used. How many do I need (or how do I find out?) and why does Asterisk specify so many by default? Thanks --ian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco IP Phones
No you dont have to use SIP. You can also use the SCCP channel on * with Cisco phones. Message: 16 Date: Sat, 4 Dec 2004 12:45:53 +0200 From: "Walid Azab" [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco IP Phones To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset="us-ascii" Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid ___ 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] XML to monitor queues on Cisco display ?
I, too would be very interested in this application. I have a small call center with Cisco Phones, and one of our biggest problems is alerting the Agents that a) there are calls in the queue; and b) they have been logged out with the auto-logout feature. Most of our agents converted from a Nortel system where the display always indicated their status as well as the queue status. As a corporate user, we would be willing to pay for this application; but as an Open Source Proponent, I would love to see it hit the Open Source community. Thank you! Joe Dennick Director, IS Operations Securities America Financial Corporation Omaha, Nebraska -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shane Young Sent: Saturday, December 04, 2004 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Henry Devito Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] XML to monitor queues on Cisco display ? Quoting Henry Devito [EMAIL PROTECTED]: I attempted this but I got stuck on one issue. Cisco phones pull data so I couldn't get them to autoupdate. In other words push data to them. You can use an http Refresh to keep the screen updating once you've accessed your XML application. It's not the best solution, but it is a step closer. --Shane ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.799 / Virus Database: 543 - Release Date: 11/19/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.799 / Virus Database: 543 - Release Date: 11/19/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN kernel 2.6 problems chapi isdn4lin
Corvin wrote: Hello! I've encauntered some serious problems with asterisk. I have to install it on system: 1. Mandrake 10.1 2. kernel 2.8.1 3. four ISDN cards. And I am in big trouble, isdn4linux is no longer supported for kernels 2.6 (on this system there are not any /dev/ttyI0 and similar devices)/ msidn - is unstable and for brave people chapi - I can't compile (lot of errors and I don't know why) i tired to patch it but it didn't help :(. I don't know what to do and I need solution very fast. I have exactly the same problem. Tried compiling chan_capi on Mandrake 10.1 and SuSE 9.1, but it failed with lots of weird errors... I posted it to the group a couple of days ago, got no reply :( Tomek ___ 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] Remote-Party-ID + CallerID + VoicemailMain
Hey All,
Quick Question. We just started using Remote-Party-ID
on our IAD endpoints and now when one of our customers has caller-ID blocked
(Privacy=full in the remote-party-id SIP header) and they call voicemail via
asterisks and get VoiceMailMain then they get a prompt for "Comedian
Mail, Mailbox?" instead of just the password.
We are calling VoiceMailMain as: VoiceMailMain(${CALLERIDNUM})
So, after some investigation it seems that the reason is
because the CALLERIDNUM and CALLERID variables now always contain a value of "Anonymous"
when the "Privacy" flag is set in the Remote-Party-ID SIP header.
Is there a way to disable Remote-Party-ID in Asterisk?
So that asterisk always looks at the SIP From: header instead of
Remote-Party-ID?
Or, is there a variable that I am unaware of that contains
the calling-number other than caller-id? I just need the calling number
available somewhere ... I can easily use an AGI script to parse it out of a
string and pass it to VoiceMailMain .. I just need access to it from an AGI
script in order to do that.
Any ideas?
Thanks for any help you can provide!
Darren Nay
Ionosphere, Inc.
VoIP Network Development
[EMAIL PROTECTED]
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RE: [Asterisk-Users] Asterisk and Cisco IP Phones
Pfft ya right if you want half assed supported channel drivers. Use SIP. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Keith O'Brien Sent: Saturday, December 04, 2004 12:57 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones No you don't have to use SIP. You can also use the SCCP channel on * with Cisco phones. Message: 16 Date: Sat, 4 Dec 2004 12:45:53 +0200 From: Walid Azab [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco IP Phones To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid ___ 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] SJPhone SIP Tab
Hi, I'm following, http://www.voip-info.org/wiki-Asterisk+phone+sjphone. However, I cannot find the SIP tab. Would someone please give me a few pointers? The screen capture can be seen at URL below http://www.dslreports.com/forum/remark,12022987~mode=flat Regards, Norman Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco IP Phones
Let me CLARIFY for those that might not get what I ment.. DO NOT RECOMMEND SCCP unless you have actually installed and used it. Its crap... SIP is what you want if you use a cisco phone with asterisk. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Saturday, December 04, 2004 1:33 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones Pfft ya right if you want half assed supported channel drivers. Use SIP. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Keith O'Brien Sent: Saturday, December 04, 2004 12:57 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones No you don't have to use SIP. You can also use the SCCP channel on * with Cisco phones. Message: 16 Date: Sat, 4 Dec 2004 12:45:53 +0200 From: Walid Azab [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco IP Phones To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid ___ 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] DIALSTATUS weirdness (CHANUNAVAIL instead of BUSY, NOANSWER instead of CHANUNAVAIL)
On December 4, 2004 08:43 am, Rich Adamson wrote: Looks like the command is documented in the current config samples. Yeah I see that now. :-) Since the comments use words like doesn't work with all telcos, could this have something to do with detecting busy when a call reaches a destination lurking behind an analog system? (eg, pri call placed to a DID number on an analog pbx where the d channel isn't aware of the destination's status?) Yeah but from the config: ; PRI Out of band indications. ; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to work ; with all telcos. ; ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT ; inband: Signal Busy/Congestion using in-band tones ; ; priindication = outofband This seems to be for * notifying the PRI, not the other way around. i.e. if someone calls me and I'm busy, not me calling out to a busy POTS line. -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] Why, why, why???
Noah, Thanks for the reply. I will try your instructions on Monday. I appreciate it very much Ferg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Friday, December 03, 2004 6:16 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Why, why, why??? Hi Michael - Thanks very much. See below. I do not have a zaptel.conf I made the assumption you were using Digium hardware, sorry. What device are you using for your incoming lines? For the fast busy: [incoming] exten = 321XXX,1,Goto(incoming,s,1) exten = s,1,Answer exten = s,2,DigitTimeout(10) exten = s,3,ResponseTimeout(20) exten = s,4,Background(swelcome) exten = t,1,Hangup include =extensions Are you dialing in on one of the 321XXX lines, or another number? For the one way audio on the grandstream: [5001] type=friend ; either friend (peer+user), peer or user host=dynamic username=5001 context=toll-access canreinvite=no quality=300 callerid=5001 disallow=all allow=ulaw allow=alaw [EMAIL PROTECTED] nat=no dtmfmode=rfc2833 It looks like it should work, but I don't use grandstream phones. Has anybody else had this problem? Have you tried the latest version of the Grandstream firmware - I know older versions had a number of problems. Thanks, Noah ___ 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] Why, why, why???
The * server is behind a Watchguard Firewall and I do have ports forwarded. I will chyeck them on Monday. Thanks to all. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Saturday, December 04, 2004 10:54 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Why, why, why??? We have Grandstream SIP phones with the latest firmware versions and have also have this problem. It appears to be something to do with RTP, I believe. I don't know exactly what (simply because I don't know much about RTP as yet), but the packets don't seem to reach the Grandstream from the other phone. The phones appear to work correctly when located on the same LAN segment. But, when one is placed behind a NAT router, the dynamic changes and one-way audio seems to happen frequently. I've Are you forwarding ports? What ports have you set asterisk to? IIRC the GS phones come with 8000 by default and asterisk comes with 1. ___ 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] Why, why, why???
I do not have the Digium card on this box. I have it on another box that I will eventually from it from. All incoming calls are through IP and not any POTS line -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Friday, December 03, 2004 6:16 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Why, why, why??? Hi Michael - Thanks very much. See below. I do not have a zaptel.conf I made the assumption you were using Digium hardware, sorry. What device are you using for your incoming lines? For the fast busy: [incoming] exten = 321XXX,1,Goto(incoming,s,1) exten = s,1,Answer exten = s,2,DigitTimeout(10) exten = s,3,ResponseTimeout(20) exten = s,4,Background(swelcome) exten = t,1,Hangup include =extensions Are you dialing in on one of the 321XXX lines, or another number? For the one way audio on the grandstream: [5001] type=friend ; either friend (peer+user), peer or user host=dynamic username=5001 context=toll-access canreinvite=no quality=300 callerid=5001 disallow=all allow=ulaw allow=alaw [EMAIL PROTECTED] nat=no dtmfmode=rfc2833 It looks like it should work, but I don't use grandstream phones. Has anybody else had this problem? Have you tried the latest version of the Grandstream firmware - I know older versions had a number of problems. Thanks, Noah ___ 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] Voicemail for Current Extension?
Hi, Is it possible to create an extension (say *1) that will give access to the voicemail for the current extension without entering the mailbox or password? (or if this is not possible, at least not have to enter the mailbox - only the password?) Thanks! --ian ___ 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 for Current Extension?
Hello Ian,
VoiceMailMain(${CALLERIDNUM})
should do the trick (unless you have the blocked number problem a
previous poster had)
-yair
On Sat, 4 Dec 2004 20:01:58 +, Ian Chilton
[EMAIL PROTECTED] wrote:
Hi,
Is it possible to create an extension (say *1) that will give access to
the voicemail for the current extension without entering the mailbox or
password?
(or if this is not possible, at least not have to enter the mailbox -
only the password?)
Thanks!
--ian
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RE: [Asterisk-Users] Voicemail for Current Extension?
Forgot the s
VoiceMailMain(s${CALLERIDNUM})
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Yair Hakak
Sent: Saturday, December 04, 2004 2:08 PM
To: Ian Chilton; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail for Current Extension?
Hello Ian,
VoiceMailMain(${CALLERIDNUM})
should do the trick (unless you have the blocked number problem a
previous poster had)
-yair
On Sat, 4 Dec 2004 20:01:58 +, Ian Chilton
[EMAIL PROTECTED] wrote:
Hi,
Is it possible to create an extension (say *1) that will give access to
the voicemail for the current extension without entering the mailbox or
password?
(or if this is not possible, at least not have to enter the mailbox -
only the password?)
Thanks!
--ian
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Re: [Asterisk-Users] Voicemail for Current Extension?
true enough, forgot the s...the s skips the password
my bad
-yair
On Sat, 4 Dec 2004 14:14:06 -0600, Brian West [EMAIL PROTECTED] wrote:
Forgot the s
VoiceMailMain(s${CALLERIDNUM})
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Yair Hakak
Sent: Saturday, December 04, 2004 2:08 PM
To: Ian Chilton; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail for Current Extension?
Hello Ian,
VoiceMailMain(${CALLERIDNUM})
should do the trick (unless you have the blocked number problem a
previous poster had)
-yair
On Sat, 4 Dec 2004 20:01:58 +, Ian Chilton
[EMAIL PROTECTED] wrote:
Hi,
Is it possible to create an extension (say *1) that will give access to
the voicemail for the current extension without entering the mailbox or
password?
(or if this is not possible, at least not have to enter the mailbox -
only the password?)
Thanks!
--ian
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[Asterisk-Users] chan_zap.c:6181 mkintf: Unable to get parameters
Hi! I want to install a X100P. I think I did everything according to the manuals I found in the net. I loaded the modules and I edited the config files according to http://www.digium.com/downloads/hw_article. I start ztcfg that tells me everything is alright. But when I start asterisk it tells me the error as posted in the subject. Why? I'm using kernel 2.6 could that be a problem? Bye! 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] ISDN kernel 2.6 problems chapi isdn4lin
check it: http://rpm.pbone.net/index.php3/stat/4/idpl/1516256/com/asterisk-chan_capi-0. 3.5-2mdk.i586.rpm.html but I don't know if it resolve all problems . Corvin --- ___ 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] Bluetooth with *
This is spam, but WOOT! I wonder if the bluez guys have put any further work into fixing SCO. On Sat, 4 Dec 2004 11:42:33 +, Mike Dent [EMAIL PROTECTED] wrote: Thanks! :) On Sat, 4 Dec 2004 10:19:59 +, Theo P. Zourzouvillys [EMAIL PROTECTED] wrote: On Saturday 04 December 2004 04:43, Nate Carlson wrote: In other words, if it's something you really want, add more cash to the bounty, to help encourage the developer to spend more time on it *grin*: alright, alright - i'll work on it today :-) ~ Theo -- Theo P. Zourzouvillys [EMAIL PROTECTED] http://www.crazygreek.co.uk/ ___ 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 -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_misdn and Dynalink IS64PH ISDN
--- Simon Richter [EMAIL PROTECTED] a écrit : Hi, TE Stack No Upper ID init_stack: File exists You need to set the layermask when loading the card driver. For a TE port, use 15 (layer 0-3) and for an NT port, use 3 (layer 0-1). Simon Thanks, I add layermask in my modprobe script : /sbin/modprobe --ignore-install w6692pci protocol=2 layermask=3 Now I have another error : Init. Stack on port 1 TE Stack No lower Id init_stack: File exists In syslog : kernel: MISDN free_device: entitylist not empty What can I do to resolv that ? thanks Jerome Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ Avec Yahoo! faites un don et soutenez le Téléthon en cliquant sur http://www.telethon.fr/030-Don/10-10_Don.asp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PolyCom MWI Chirp issue
James Milne wrote: Is there any workaround as of yet? Or is this something that polycom will have to update in firmware? It will have to be fixed in firmware, unless the problem is actually in Asterisk; I do not know the actual cause of the problem. Unfortunately since Polycom is not interested in supporting Asterisk, we can't get any help from them to debug it. ___ 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] Using Pocket PC over cell phone connection?
Anyone tried using a pocket pc with sjphone or x-ten over a cell phone connection? I'd like to be able to connect using my cell phone data connection, but so far I've come across bandwidth constraints - The closest to success I've found so far is to use the GSM codec, but even then the bandwidth seems to much for it. Anyone had any luck? Paul ___ 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 Gossiptel - 1 way audio???
Hi, I have Asterisk setup and registered with Gossiptel but i'm only getting 1 way audio. If I call 160 (echo test) or 123 (talking clock), it makes the call but I just get silence. If I call my Gossiptel number from a pstn line, I get gossiptel - pstn audio but not pstn - gossiptel audio. I've got ports 5060 and the rtp ports forwarded in on the firewall and I have 3 other sip providers setup and working on the same Asterisk box - it's only Gossiptel i'm having problems with. Any ideas? Anyone had a similar problem? Anyone got it working? Here is my config snippet from sip.conf: register = USERID:[EMAIL PROTECTED]/USERID [gossiptel] type=friend username=USERID fromuser=USERID authuser=USERID secret=PASSWORD host=sip.gossiptel.com nat=yes insecure=very dtmfmode=inband canreinvite=no fromdomain=sip.gossiptel.com disallow=all allow=ilbc allow=gsm allow=ulaw allow=all Thanks --ian ___ 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] Budgetone 100 Caller ID
Hi, Is there an * configuration that will allow the BT100 to display the numeric callerid instead of the broken text? Regards Greg ___ 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] more DIALSTATUS/HANGUPSTATUS woes with IAX2
Phone - TDM430P - home* - IAX2 - office* - PRI - Telco I dial a busy number from the Phone. Home* shows this in the CLI: -- Executing Macro(Zap/1-1, dial-wu|2922004) in new stack -- Executing Dial(Zap/1-1, IAX2/[EMAIL PROTECTED]/2922004||g) in new stack -- Called [EMAIL PROTECTED]/2922004 -- Call accepted by wu-ast (format gsm) -- Format for call is gsm -- IAX2/wu-ast/1 is busy -- Hungup 'IAX2/wu-ast/1' == Everyone is busy/congested at this time -- Executing NoOp(Zap/1-1, HANGUPCAUSE is 0 and DIALSTATUS is CHANUNAVAIL) in new stack **WHY** is DIALSTATUS set to CHANUNAVAIL instead of BUSY? Is the BUSY DIALSTATUS ever used?? How can I indicate that the line is busy if I can't detect when it's busy?! Frustrated, -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is this possible?
hi; i have a conference room setup on the asterisk server. And say that one of the sip peers (say A) wants to dial outbound to a PSTN destination (say B). Can i have A join the conference room and some how at the same time ask B to join the conference room? I think this feature is very important, because if we have few people in a conference room, it would be very nice to invite people to join the conference without the need of putting everybody on hold. Any feedback is appreciated. thanks moe smadi ___ 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 dabbling...
Newbee here I would like to play around with Asterisk a little. First, I need to prepare a server with FreeBSD. It's a PII 433mHz/256mb box. Good enough? Then install Asterisk. I have a broadband (cable) internet presence. Could I do anything with this connection and Asterisk? Thanks, Rayasterisk Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard.___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco IP Phones
If you get a Cisco phone, chance are it won't have SIP support right off the bat, but you can upgrade the firmware to a SIP version. They have the downloads available at their website but you have to buy some sort of license/account with them. I've heard it's pretty cheap, but I don't know first hand because my company bought the one I've use at work. Oh yeah, with the 7940/60's you might have to upgrade to SIP firmware version 5.0 first and then from there upgrade to 7.3. -Chris On Sat, 4 Dec 2004 13:36:09 -0600, Brian West [EMAIL PROTECTED] wrote: Let me CLARIFY for those that might not get what I ment.. DO NOT RECOMMEND SCCP unless you have actually installed and used it. Its crap... SIP is what you want if you use a cisco phone with asterisk. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Saturday, December 04, 2004 1:33 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones Pfft ya right if you want half assed supported channel drivers. Use SIP. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Keith O'Brien Sent: Saturday, December 04, 2004 12:57 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones No you don't have to use SIP. You can also use the SCCP channel on * with Cisco phones. Message: 16 Date: Sat, 4 Dec 2004 12:45:53 +0200 From: Walid Azab [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco IP Phones To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid ___ 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] Broadvoice outbound 404 error
Is anyone else experiencing 404 errors on outbound dial with Broadvoice? I've followed the instructions posted by Broadvoice to configure sip.conf, and inbound calling works fine. Every time I try to dial out, I get a 404 Not Found error. Here are the relevant sections from my configs. sip.conf: context=broadvoice-in register = [EMAIL PROTECTED]:xxpasswordxx:[EMAIL PROTECTED] [bv-home] type=peer host=proxy.dca.broadvoice.com fromdomain=sip.broadvoice.com fromuser=3215551212 context=inbound canreinvite=no qualify=yes disallow=all allow=ilbc allow=gsm allow=ulaw dtmfmode=inband secret=xxpasswordxx insecure=very Thank you, Reid Forrest, CISSP Max-IS, Inc. [EMAIL PROTECTED] ofc: 407.786.9600 x1200 cell: 321.439.8903 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_zap.c:6181 mkintf: Unable to get parameters
Michael Vogel schrieb: But when I start asterisk it tells me the error as posted in the subject. The problem is solved. I had a version mismatch between zaptel and asterisk. No I have different problems. But I will first try to find an answer in the wiki before posting them. Bye! 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
