Re: [asterisk-users] Asterisk and TDM400P ?
Hi Noc, I had similar problem. Check If you have netjetpci module and try to delete it...this solved my problem. Giorgio Incantalupo Noc Phibee wrote: Hi i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect my asterisk to a french analog line. In my zaptel.conf, i have: loadzone=fr defaultzone=fr fxols=3 when i load the module, i have in logs: Nov 24 06:13:40 gw kernel: Freed a Wildcard Nov 24 06:13:40 gw kernel: Zapata Telephony Interface Unloaded Nov 24 06:13:40 gw zaptel: Removing zaptel module: succeeded Nov 24 06:13:42 gw kernel: Zapata Telephony Interface Registered on major 196 Nov 24 06:13:42 gw kernel: Zaptel Version: 1.2.9.1-1mdv2007.0 Echo Canceller: KB1 Nov 24 06:13:42 gw zaptel: Loading zaptel framework: succeeded Nov 24 06:13:42 gw kernel: ACPI: PCI Interrupt :02:0a.0[A] - GSI 21 (level, low) - IRQ 20 Nov 24 06:13:43 gw kernel: Freshmaker version: 73 Nov 24 06:13:43 gw kernel: Freshmaker passed register test Nov 24 06:13:43 gw kernel: Module 0: Not installed Nov 24 06:13:43 gw kernel: Module 1: Not installed Nov 24 06:13:43 gw kernel: Module 2: Not installed Nov 24 06:13:43 gw kernel: Module 3: Installed -- AUTO FXO (FCC mode) Nov 24 06:13:43 gw kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules) Nov 24 06:13:44 gw kernel: Registered tone zone 2 (France) Nov 24 06:13:44 gw zaptel: Running ztcfg: succeeded and my problems are whit all sample that i have, asterisk don't restart and put me: Nov 24 06:15:17 NOTICE[2943] cdr.c: CDR simple logging enabled. Nov 24 06:15:17 WARNING[2943] chan_zap.c: Unable to specify channel 3: No such device Nov 24 06:15:17 ERROR[2943] chan_zap.c: Unable to open channel 3: No such device here = 0, tmp-channel = 3, channel = 3 Nov 24 06:15:17 ERROR[2943] chan_zap.c: Unable to register channel '3' Nov 24 06:15:17 WARNING[2943] loader.c: chan_zap.so: load_module failed, returning -1 Nov 24 06:15:17 WARNING[2943] loader.c: Loading module chan_zap.so failed! for all channel (i have tested from 1 to 5) my zapata.conf: [trunkgroups] [channels] context=default signalling=fxo_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes musiconhold=default channel = 3 where is my errors ? Thanks for your help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
Thanks Giogio, but no i don't have this module bye Giorgio Incantalupo a écrit : Hi Noc, I had similar problem. Check If you have netjetpci module and try to delete it...this solved my problem. Giorgio Incantalupo Noc Phibee wrote: Hi i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect my asterisk to a french analog line. In my zaptel.conf, i have: loadzone=fr defaultzone=fr fxols=3 when i load the module, i have in logs: Nov 24 06:13:40 gw kernel: Freed a Wildcard Nov 24 06:13:40 gw kernel: Zapata Telephony Interface Unloaded Nov 24 06:13:40 gw zaptel: Removing zaptel module: succeeded Nov 24 06:13:42 gw kernel: Zapata Telephony Interface Registered on major 196 Nov 24 06:13:42 gw kernel: Zaptel Version: 1.2.9.1-1mdv2007.0 Echo Canceller: KB1 Nov 24 06:13:42 gw zaptel: Loading zaptel framework: succeeded Nov 24 06:13:42 gw kernel: ACPI: PCI Interrupt :02:0a.0[A] - GSI 21 (level, low) - IRQ 20 Nov 24 06:13:43 gw kernel: Freshmaker version: 73 Nov 24 06:13:43 gw kernel: Freshmaker passed register test Nov 24 06:13:43 gw kernel: Module 0: Not installed Nov 24 06:13:43 gw kernel: Module 1: Not installed Nov 24 06:13:43 gw kernel: Module 2: Not installed Nov 24 06:13:43 gw kernel: Module 3: Installed -- AUTO FXO (FCC mode) Nov 24 06:13:43 gw kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules) Nov 24 06:13:44 gw kernel: Registered tone zone 2 (France) Nov 24 06:13:44 gw zaptel: Running ztcfg: succeeded and my problems are whit all sample that i have, asterisk don't restart and put me: Nov 24 06:15:17 NOTICE[2943] cdr.c: CDR simple logging enabled. Nov 24 06:15:17 WARNING[2943] chan_zap.c: Unable to specify channel 3: No such device Nov 24 06:15:17 ERROR[2943] chan_zap.c: Unable to open channel 3: No such device here = 0, tmp-channel = 3, channel = 3 Nov 24 06:15:17 ERROR[2943] chan_zap.c: Unable to register channel '3' Nov 24 06:15:17 WARNING[2943] loader.c: chan_zap.so: load_module failed, returning -1 Nov 24 06:15:17 WARNING[2943] loader.c: Loading module chan_zap.so failed! for all channel (i have tested from 1 to 5) my zapata.conf: [trunkgroups] [channels] context=default signalling=fxo_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes musiconhold=default channel = 3 where is my errors ? Thanks for your help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing the b410p card, unable to install mISDN
Hi, I'm installing Asterisk on Ubuntu 6.10 When I first compiled the zaptel package I used: make clean make make install So far so good, but the following command failed: make b410p I did some digging on google and found a guide on how to install it manually, but the result was the same. I got these files ftp.digium.com/pub/telephony/zaptel/b410p/misdn-b410p.tar.gz ftp.digium.com/pub/telephony/zaptel/b410p/mISDNuser.tar.gz And did the following: make force make make install All the commands worked just fine. Remark: The guide I was using told me to cd into the mISDNuser directory, but didn't do anything with it. /etc/init.d/misdn-init scan gives me: [OK] found the following devices: card=1,0x4 So I ran /etc/init.d/misdn-init config The output: [OK] /etc/misdn-init.conf created. It's now safe to run /etc/init.d/misdn-init start [ii] make your ports (1-4) available in asterisk by editing /etc/asterisk/misdn.conf [ii] run /etc/init.d/misdn-init config to store this information to /etc/misdn-init.conf I then edited the setting in /etc/misdn-init.conf, guessing I'll be needing nt_ptmp=1,2,3,4 (although I have no idea) Now the problem, if I run /etc/init.d/misdn-init start I get the following: /etc/init.d/misdn-init: line 91: [: 5: unary operator expected - Loading module(s) for your misdn-cards: - modprobe --ignore-install hfcmulti type= protocol=,,, layermask=0x3,0x3,0x3,0x3 poll=128 debug=0xf De output of the lsmod | grep hfcmulti command is hfcmulti 74984 0 mISDN_core 85248 6 mISDN_dsp,mISDN_capi,l3udss1,mISDN_l2,mISDN_l1,hfcmulti The output of dmesg | grep Digium is [42949387.38] HFC-MULTI: Card 'HFC-4S Digium Card' found, but not given by module's options, ignoring... I recompile Asterisk and install it again, but I do not get the misdn command in asterisk. What am I missing ? Thnx. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Server Configuration for E1's
Dear Users, I am fairly new to Digium and Asterisk. I wanted to know that if I use the Digium product THREE Digium Wildcard TE412Ps (Quad E1 Card) how many calls can I handle simultaneously. I want to use the cards with the following Configurations: Intel® Xeon 3.00GHz/800MHz, 2M Processor 1GB PC2-3200 DDR2 SDRAM (2x512MB + 2X1GB) memory Integrated Dual Channel Ultra320 SCSI Adapter NC7781 Single Port PCI-X embedded NIC Hot plug drive cage - Ultra3 (6X1) High Speed IDE CD-ROM Drive 72.8GB Pluggable Ultra320 SCSI 15,000 rpm (1) Universal Hard Drive Asterisk Business Edition 3 X TE412P I have a requirement of handling 350 Calls using a single Server and please note the Server will used to transferring the call only. Other Servers will handle gateway Negotiation and Billing. This server will SIMPLY be a Gateway. Please let me know if this configuration too high or too low. If anybody has better solution please let me know that as well. Thank you, waiting eagerly for a response. Imran M Yousuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
Noc, it seems your zap*.conf are okhave you tried to do all the modules loading procedure step by step? ztcfg -s modprobe -r wctdm modprobe wctdm ztcfg -vv Giorgio Incantalupo Noc Phibee wrote: Thanks Giogio, but no i don't have this module bye Giorgio Incantalupo a écrit : Hi Noc, I had similar problem. Check If you have netjetpci module and try to delete it...this solved my problem. Giorgio Incantalupo Noc Phibee wrote: Hi i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect my asterisk to a french analog line. In my zaptel.conf, i have: loadzone=fr defaultzone=fr fxols=3 when i load the module, i have in logs: Nov 24 06:13:40 gw kernel: Freed a Wildcard Nov 24 06:13:40 gw kernel: Zapata Telephony Interface Unloaded Nov 24 06:13:40 gw zaptel: Removing zaptel module: succeeded Nov 24 06:13:42 gw kernel: Zapata Telephony Interface Registered on major 196 Nov 24 06:13:42 gw kernel: Zaptel Version: 1.2.9.1-1mdv2007.0 Echo Canceller: KB1 Nov 24 06:13:42 gw zaptel: Loading zaptel framework: succeeded Nov 24 06:13:42 gw kernel: ACPI: PCI Interrupt :02:0a.0[A] - GSI 21 (level, low) - IRQ 20 Nov 24 06:13:43 gw kernel: Freshmaker version: 73 Nov 24 06:13:43 gw kernel: Freshmaker passed register test Nov 24 06:13:43 gw kernel: Module 0: Not installed Nov 24 06:13:43 gw kernel: Module 1: Not installed Nov 24 06:13:43 gw kernel: Module 2: Not installed Nov 24 06:13:43 gw kernel: Module 3: Installed -- AUTO FXO (FCC mode) Nov 24 06:13:43 gw kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules) Nov 24 06:13:44 gw kernel: Registered tone zone 2 (France) Nov 24 06:13:44 gw zaptel: Running ztcfg: succeeded and my problems are whit all sample that i have, asterisk don't restart and put me: Nov 24 06:15:17 NOTICE[2943] cdr.c: CDR simple logging enabled. Nov 24 06:15:17 WARNING[2943] chan_zap.c: Unable to specify channel 3: No such device Nov 24 06:15:17 ERROR[2943] chan_zap.c: Unable to open channel 3: No such device here = 0, tmp-channel = 3, channel = 3 Nov 24 06:15:17 ERROR[2943] chan_zap.c: Unable to register channel '3' Nov 24 06:15:17 WARNING[2943] loader.c: chan_zap.so: load_module failed, returning -1 Nov 24 06:15:17 WARNING[2943] loader.c: Loading module chan_zap.so failed! for all channel (i have tested from 1 to 5) my zapata.conf: [trunkgroups] [channels] context=default signalling=fxo_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes musiconhold=default channel = 3 where is my errors ? Thanks for your help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
Noc Phibee wrote: Thanks Giogio, but no i don't have this module bye Check your zapata.conf. Your signalling and channel settings are wrong for FXO module. signalling=fxs_ls channel= 4 FXO module use fxs signalling, FXS module use fxo signalling. Leo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change IAX default port 4569 to some other port
On 23 Nov 2006, at 11:18, Zeeshan Zakaria wrote: Hi all, All of a sudden all my IAX DIDs have gone down. I couldn't find any reason other than that the ISP is blocking port 4569. DIDs register fine from my home server, but not from office server, which is not behind any NAT. SIP registers fine. I am trying to change IAX port but it apparantly IAX works only on 4569. Changing it in iax.conf doesn't do anything. Changing it is registration string also doesn't help. How can I make IAX work on some other port? I think that the bindport setting in iax.conf is ignored on reload, you have to stop and start asterisk for it to take effect. If that doesn't do the trick, you will have to write some tricksy iptables port mapping rules for 4569 Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Siemens Gigaset SL75
Have you ever compared it to Linksys WIP 330 or Zyxel 2000 ? Those 2 seem to get average reviews from users (short range, autonomy, ...). On paper, it seems to me a decent WiFi phone is still lacking today. Maybe this Gigaset SL75 could fill the void. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
thanks for this information, but no change: Nov 24 10:32:42 WARNING[6346] chan_zap.c: Unable to specify channel 4: No such device or address Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to open channel 4: No such device or address here = 0, tmp-channel = 4, channel = 4 Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to register channel '4' Nov 24 10:32:42 WARNING[6346] loader.c: chan_zap.so: load_module failed, returning -1 Nov 24 10:32:42 WARNING[6346] loader.c: Loading module chan_zap.so failed! Leo Ann Boon a écrit : Noc Phibee wrote: Thanks Giogio, but no i don't have this module bye Check your zapata.conf. Your signalling and channel settings are wrong for FXO module. signalling=fxs_ls channel= 4 FXO module use fxs signalling, FXS module use fxo signalling. Leo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
Noc Phibee wrote: thanks for this information, but no change: Nov 24 10:32:42 WARNING[6346] chan_zap.c: Unable to specify channel 4: No such device or address Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to open channel 4: No such device or address here = 0, tmp-channel = 4, channel = 4 Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to register channel '4' Nov 24 10:32:42 WARNING[6346] loader.c: chan_zap.so: load_module failed, returning -1 Nov 24 10:32:42 WARNING[6346] loader.c: Loading module chan_zap.so failed! Can you check if your /dev/zap directory is created correctly? On my machine with a TDM400P with 2xFXS and 2xFXO. [EMAIL PROTECTED] ~]$ ls /dev/zap/ 1 2 3 4 channel ctl pseudo time If you don't see anything then you'll have to check if your security setting is prevent access to /dev/zap. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
Noc Phibee wrote: thanks for this information, but no change: Nov 24 10:32:42 WARNING[6346] chan_zap.c: Unable to specify channel 4: No such device or address Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to open channel 4: No such device or address here = 0, tmp-channel = 4, channel = 4 Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to register channel '4' Nov 24 10:32:42 WARNING[6346] loader.c: chan_zap.so: load_module failed, returning -1 Nov 24 10:32:42 WARNING[6346] loader.c: Loading module chan_zap.so failed! Hi, Check your /etc/zaptel.conf and ensure that it has the right kind of signalling set for the same channel number as that in you zapata.conf. do : cat /proc/zaptel/1 and it should show channels and the effective signalling settings for them. If signalling does not appear here,it means that, it is not configured properly, and loading chan_zap would fail. My first and fourth channels are configured as fxsks and the output i get is : #cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 1 WCTDM/0/0 FXSKS 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 FXSKS HTH :) Regards, Pranav -- Blessed are the pessimists, for they take backups !! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: How to change IAX default port 4569 to some other port :Debug Message Attached
On 23 Nov 2006, at 11:36, Zeeshan Zakaria wrote: iax2 debug is giving following messages repeatedly. Tx-Frame Retry[003] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 1ms SCall: 00010 DCall: 0 [xxx.xxx. 157.230:4569] USERNAME: XXX9072835 REFRESH : 60 Tx-Frame Retry[002] -- OSeqno: 002 ISeqno: 000 Type: IAX Subclass: PING Timestamp: 20001ms SCall: 6 DCall: 0 [xxx.xxx. 157.230:5070] Tx-Frame Retry[003] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ Timestamp: 1ms SCall: 5 DCall: 0 [xxx.xxx. 157.230:4569] I don't see any Rx-Frames - so I guess that something is wrong with your routing/firewalls/NAT/Ipaddresses - Asterisk isn't seeing any replies to the packets it is (re)sending. (Do you have a local IP tables config ?) T. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible, horrible firewall issues in * to * setup
On 22 Nov 2006, at 22:21, Lachek Butalek wrote: My mission is to get one * box to dial another * box' extensions. I have set this up previously without any issues by making a simple IAX trunk/extension pair on the two boxes and create a dial plan with a prefix like 9|XXX to select an extension on the other box. My problem is that I now have to do this with extremely restrictive firewalls thrown into the mix - firewalls I have no control over. Basically, the setup is: *1 --- FW1 --- (Internet) --- FW2 --- FW3 --- *2 I have control over firewall 1 and 3, but not 2. Using port forwarding (4569 UDP) on FW1, I have been able to make calls from *2 to *1. My problem lies with making calls the other way, as I have no way of port forwarding on FW2. If FW2 and FW3 permit outbound UDP and associated replies you won't need to. (even if they NAT them). Set up 4569 on FW1 to go to *1 Add *2 as a peer (and user) in iax.conf on *1 Do _nothing_ with FW3 Set up *2 to _register_ with *1 The repeated registration from 2 to 1 will keep the any NAT's and port maps open and tell 1 how to reach 2. (IAX is great) Tim. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Cisco 7970
Hello; Maybe this is a little off-topic, but I need help. I need to repair a cisco 7970, but in my country(spain) cisco is only selling, they don't repair if you're not client. Because I bought on ebay, I'm not client, so I have no chance. I tried to repair by myself, the problem is on the LCD screen, I need a replace, anyone know which part number is it (manufacturer and part number), and where I can get a replacement? Anyway, if someone knows a technical service in Spain, or Europe, where I can ask for the piece, it will help a lot. Thank you. David. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
Leo Ann Boon a écrit : Noc Phibee wrote: thanks for this information, but no change: Nov 24 10:32:42 WARNING[6346] chan_zap.c: Unable to specify channel 4: No such device or address Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to open channel 4: No such device or address here = 0, tmp-channel = 4, channel = 4 Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to register channel '4' Nov 24 10:32:42 WARNING[6346] loader.c: chan_zap.so: load_module failed, returning -1 Nov 24 10:32:42 WARNING[6346] loader.c: Loading module chan_zap.so failed! Can you check if your /dev/zap directory is created correctly? On my machine with a TDM400P with 2xFXS and 2xFXO. [EMAIL PROTECTED] ~]$ ls /dev/zap/ 1 2 3 4 channel ctl pseudo time If you don't see anything then you'll have to check if your security setting is prevent access to /dev/zap. Leo Yes i have ;=) [EMAIL PROTECTED] zap]# ll total 0 crw-rw 1 asterisk asterisk 196, 1 nov 24 06:29 1 crw-rw 1 asterisk asterisk 196, 2 nov 24 06:29 2 crw-rw 1 asterisk asterisk 196, 3 nov 24 06:29 3 crw-rw 1 asterisk asterisk 196, 4 nov 24 06:29 4 crw-rw 1 asterisk asterisk 196, 254 nov 24 06:29 channel crw-rw 1 asterisk asterisk 196, 0 nov 24 06:29 ctl crw-rw 1 asterisk asterisk 196, 255 nov 24 06:29 pseudo crw-rw 1 asterisk asterisk 196, 253 nov 24 06:29 timer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfers in SER + Asterisk architecture
Hi Ricardo, Could you post a specific example where your problem happens. That way would be easier for me to try to help you on this. Does asterisk is registred into SER , or you have trust based relationship between them? On 11/23/06, Ricardo Carvalho [EMAIL PROTECTED] wrote: Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem is that when the REFER message (that carries the Refer-To number to whom the call should be transferred) gets to Asterisk, it replies with a 404 Not Found message, and the Call Transfer isn't established! Any ideas on how can I solve this problem? Thanks in advance, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
Pranav Peshwe a écrit : Hi, Check your /etc/zaptel.conf and ensure that it has the right kind of signalling set for the same channel number as that in you zapata.conf. do : cat /proc/zaptel/1 and it should show channels and the effective signalling settings for them. If signalling does not appear here,it means that, it is not configured properly, and loading chan_zap would fail. My first and fourth channels are configured as fxsks and the output i get is : #cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 1 WCTDM/0/0 FXSKS 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 FXSKS HTH :) Regards, Pranav Thanks for your help, a cat: [EMAIL PROTECTED] zap]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 FXSKS 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 in my zaptel.conf, i have only: loadzone=fr defaultzone=fr fxsks=1 and zapata.conf: [trunkgroups] [channels] context=default signalling=fxs_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes musiconhold=default callerid=Filtrinov0477530573 channel = 5 if i understand, my error are channel ? In zaptel.conf, its fxsks=1 and in zapata.conf it's channel = 0 no ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] upgraded polycom to 2.0.1.0291 and...
Now i'm receiving this and my phone no longer can dial out... ERROR[4391]: chan_sip.c:11169 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete. I'm having to use the configs that came with the zip because apparently my previous configs no longer are valid and lock the phone from dialing with a url disabled message... anyway, these polycom phones are driving me crazy, expecially their configs! -- ~Shaun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Request for help with DISA (Not taking my input number correctly?)
Hi,
Thank you for response. I configured DISA and its working sometimes and not
working sometimes. Here I am sending the configuration and output on Asterisk
server console:
Extensions.conf file content:
[custom-CLID]
exten = s,1,Answer
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(10)
exten = s,4,Authenticate(1234)
exten = s,5,DISA(no-password|disa-ext)
[disa-ext]
exten = _X.,1,DIAL(IAX2/[EMAIL PROTECTED]/${EXTEN},30,tr)
Output on server console:
-- Playing 'custom/v1' (language 'en')
== CDR updated on Zap/1-1
-- Executing Goto(Zap/1-1, custom-CLID|s|1) in new stack
-- Goto (custom-CLID,s,1)
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing DigitTimeout(Zap/1-1, 5) in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout(Zap/1-1, 10) in new stack
-- Set Response Timeout to 10
-- Executing Authenticate(Zap/1-1, 1234) in new stack
-- Playing 'agent-pass' (language 'en')
-- Playing 'auth-thankyou' (language 'en')
-- Executing DISA(Zap/1-1, no-password|disa-ext) in new stack
-- Executing Dial(Zap/1-1, IAX2/[EMAIL PROTECTED]/187773456|30|tr) in
new stack
-- Called [EMAIL PROTECTED]/187773456
-- Hungup 'IAX2/teliax-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Hungup 'Zap/1-1'
What is happening is:
1) I called my zap number from my mobile
2) My IVR is responding
3) I entered a extension number to access DISA
4) Asterisk asked the secret (PIN) code to access DISA
5) I entered password of DISA
6) After validating the password, its giving Dial tone to dial a USA number
7) I entered 17187773456 (This is a toll free number) to test
8) Call is going sometimes and call is not going sometimes. If we observe on
server console, its not taking my input number properly and taking my input
phone number wrongly.
9) I tested from other mobiles also. But, its not taking my input number as i
entered sometimes.
9) What is the wrong?
Please tell me. Looking forward to your response. Thank you.
Regards,
Chandra.
-
Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates.___
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[asterisk-users] DB9 e1 to RJ45 pinout
Hi all, anyone known how can i create an adapter from a DB9 port of a E1 to an RJ45 plug? My telco left active the db9 port, but on my te407p card i have rj45 connection. Anyone can help me pls ? Thanks in advance -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.14/548 - Release Date: 23/11/2006 15.22 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
Hi Noc, why channel 5 in zapata? It must be channel 1 the same as in zaptel.conf. Giorgio Incantalupo Noc Phibee wrote: Pranav Peshwe a écrit : Hi, Check your /etc/zaptel.conf and ensure that it has the right kind of signalling set for the same channel number as that in you zapata.conf. do : cat /proc/zaptel/1 and it should show channels and the effective signalling settings for them. If signalling does not appear here,it means that, it is not configured properly, and loading chan_zap would fail. My first and fourth channels are configured as fxsks and the output i get is : #cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 1 WCTDM/0/0 FXSKS 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 FXSKS HTH :) Regards, Pranav Thanks for your help, a cat: [EMAIL PROTECTED] zap]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 FXSKS 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 in my zaptel.conf, i have only: loadzone=fr defaultzone=fr fxsks=1 and zapata.conf: [trunkgroups] [channels] context=default signalling=fxs_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes musiconhold=default callerid=Filtrinov0477530573 channel = 5 if i understand, my error are channel ? In zaptel.conf, its fxsks=1 and in zapata.conf it's channel = 0 no ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
Noc Phibee wrote: Thanks for your help, a cat: [EMAIL PROTECTED] zap]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 FXSKS 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 in my zaptel.conf, i have only: loadzone=fr defaultzone=fr fxsks=1 and zapata.conf: [trunkgroups] [channels] context=default signalling=fxs_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes musiconhold=default callerid=Filtrinov0477530573 channel = 5 if i understand, my error are channel ? In zaptel.conf, its fxsks=1 and in zapata.conf it's channel = 0 no ? Hi, The cat output suggests that channel number 1 is configured(as fxsks) by the zaptel kernel module as instructed in the zaptel.conf file. So, you should have : channel=1 instead of channel=5 and signalling=fxs_ks instead of signalling=fxs_ls (mark the 'k' and 'l') in zapata.conf This will tell the chan_zap module to use the channel that is actually initialised by the driver in the kernel i.e channel no 1.Currently chan_zap must be looking for a configured channel number 5 which is not present on the system.That is why it fails. Hopefully, doing the above will fix it :) Best regards, Pranav We don't see things as they are, we see them as we are. - Anais Nin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller Id not propagated to the analog line
Dear all, a newbie question... I have two external lines (PSTN SIP) and two internal lines. When receiving an incoming call, I correctly get the CID, but it's not propagated to the internal lines. My analog phones shows External call instead of the CID. My analog device is a TDM400P (2 FXO + 2 FXS) with two analog phones attached to it (Siemens Gigaset and a wired Logicom). I searched the web and the lists, found several pages like this one: http://www.voip-info.org/wiki/view/CID+Issues+with+some+Siemens+DECT+phones+in+France (I'm in France), but nothing helps. Any advice is welcome. Thanks! AF. -- extensions.conf: exten = s,1,Dial(Zap/2-1,,otw) zapata.conf: usecallerid=yes usecallingpres=yes callwaitingcallerid=yes callerid=asreceived hidecallerid=no callwaiting=yes treewaycalling=yes transfer=yes echocancel=yes echotraining=yes immediate=no context=home ;cidsignalling=v23 ;cadence=250,1500,1500,3000,1500,3000 signalling=fxo_ks channel = 1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More than one asterisk process
On Thu, Nov 23, 2006 at 05:31:03PM -0300, Ard wrote: This is the output. [EMAIL PROTECTED] ~]# ps auxw | grep asterisk root 4392 0.0 0.6 50604 13968 ? Ssl 11:02 0:00 asterisk root 5050 0.0 0.4 38416 9268 ?S11:07 0:00 asterisk root 5242 0.0 0.4 38528 9420 ?S11:09 0:00 asterisk root 5495 0.0 0.4 38448 9500 ?S11:10 0:00 asterisk root 5499 0.0 0.4 38472 9504 ?S11:10 0:00 asterisk root 5548 0.0 0.4 38404 9488 ?S11:10 0:00 asterisk root 5551 0.0 0.4 38408 9488 ?S11:10 0:00 asterisk root 5566 0.0 0.4 38360 9520 ?S11:10 0:00 asterisk root 5594 0.0 0.4 38420 9592 ?S11:10 0:00 asterisk root 5626 0.0 0.4 38512 9776 ?S11:10 0:00 asterisk root 5629 0.0 0.4 38524 9776 ?S11:10 0:00 asterisk root 5740 0.0 0.4 39528 9848 ?S11:10 0:00 asterisk root 5741 0.0 0.4 39532 9848 ?S11:10 0:00 asterisk root 5743 0.0 0.4 39540 9852 ?S11:10 0:00 asterisk root 5892 0.0 0.4 39352 9732 ?S11:10 0:00 asterisk root 5912 0.0 0.4 39332 9716 ?S11:10 0:00 asterisk root 5914 0.0 0.4 39336 9716 ?S11:10 0:00 asterisk root 7011 0.0 0.4 39828 10272 ? S11:11 0:00 asterisk Different processes, indeed. What do you see in /var/run/asterisk.pid or /var/eun/asterisk/asterisk.pid ? Strange. This is the second report I see of such a situation in this list recently. Asterisk should fail to daemonize if it finds a different Asterisk through the PID file. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
A summary of some of the mistakes in this thread, as it was full of them. On Fri, Nov 24, 2006 at 06:32:14AM +0100, Noc Phibee wrote: Hi i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect my asterisk to a french analog line. In my zaptel.conf, i have: loadzone=fr defaultzone=fr fxols=3 An FXO module needs FXS signalling: fxsls=3 And probably fxsks=3: I'm not sure whether or not KS is used in the French telco . Didn't you know this obvious fact? And what is the channel number of this module? when i load the module, i have in logs: Nov 24 06:13:40 gw kernel: Freed a Wildcard Nov 24 06:13:40 gw kernel: Zapata Telephony Interface Unloaded Nov 24 06:13:40 gw zaptel: Removing zaptel module: succeeded Nov 24 06:13:42 gw kernel: Zapata Telephony Interface Registered on major 196 Nov 24 06:13:42 gw kernel: Zaptel Version: 1.2.9.1-1mdv2007.0 Echo Canceller: KB1 Nov 24 06:13:42 gw zaptel: Loading zaptel framework: succeeded Nov 24 06:13:42 gw kernel: ACPI: PCI Interrupt :02:0a.0[A] - GSI 21 (level, low) - IRQ 20 Nov 24 06:13:43 gw kernel: Freshmaker version: 73 Nov 24 06:13:43 gw kernel: Freshmaker passed register test Nov 24 06:13:43 gw kernel: Module 0: Not installed Nov 24 06:13:43 gw kernel: Module 1: Not installed Nov 24 06:13:43 gw kernel: Module 2: Not installed Nov 24 06:13:43 gw kernel: Module 3: Installed -- AUTO FXO (FCC mode) The card reports that first three slots are unused, and only slot 4 is used. You should have known that this is a zero-based counting, whereas the counting of Zaptel channels is 1-based. Luckily for you you have just one card, and didn't need any further offsets calculations. Those two tiny this are obvious to us who havve been messing long enough with Zaptel. However they are far from being intuitive. My suggestions: either: * Use genzaptelconf from xpp/utils/genzaptelconf to save you from this guesswork. Or: * Apply http://bugs.digium.com/view.php?id=7613 (preferably with some optimizations from the wctdm driver itself, but the latter is clearly not my department). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing the b410p card, unable to install mISDN
Slightly off-topic: On Fri, Nov 24, 2006 at 09:30:57AM +0100, Timothy Parez wrote: Hi, I'm installing Asterisk on Ubuntu 6.10 When I first compiled the zaptel package I used: Zaptel has nothing to do with misdn. make clean make make install Sadly. This was unnecessary for you. 'm-a a-i zaptel' next time. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
Tzafrir Cohen wrote: And probably fxsks=3: I'm not sure whether or not KS is used in the French telco . signalling=fxs_ks works Ok in France with both: France Telecom and Freebox. AF. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Installing Ztdummy on Fedora Core 5
On Thu, Nov 16, 2006 at 07:57:12AM +0300, Rogers Ochieng wrote: Am trying to make zaptel with ztdummy uncommented in FC5 but am getting make error. Has anyone gotten this to work? You should post the errors :) But there are ready-to-use packages at http://ATrpms.net/name/zaptel/, so you don't have to build it yourself at all. -- Axel.Thimm at ATrpms.net pgpJqWwzd5fvt.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
Tzafrir Cohen a écrit : * Use genzaptelconf from xpp/utils/genzaptelconf to save you from this guesswork. Hi, thanks ;=) with genzaptelconf, now that's works ... correct channel are put into zaptel.conf and zapata.conf small question if you know the TDM400P: if the fxo module are at the slot 4, the RJ11 connector are the number 4 ? a show channels done: gw*CLI zap show channels Chan Extension Context Language MusicOnHold pseudointerne 4interne gw*CLI gw*CLI zap show status Description Alarms IRQ bpviol CRC4 Wildcard TDM400P REV I Board 1 OK 0 0 0 now, i can add to my extension ZAP/4 ;=) for see if the card answer, what is the process ? very very thanks at all for this result ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Encrypted password for voicemail
Hello all, I was wondering whether the only way to store the passwords for the voicemail is in extensions.conf. Is it perhaps possible to store it (encrypted) in a DB or store the hash of the password (as is standard in unix) in the extensions.conf file? Finally, it does not seem to me that the password the user enters to obtain his voicemail is encrypted between the user and Asterisk. Is this correct? Thanks, jez Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
In my zaptel.conf, i have: loadzone=fr defaultzone=fr fxols=3 loadzone=fr defaultzone=fr fxsls=4 my zapata.conf: [trunkgroups] [channels] context=default signalling=fxo_ls signalling=fxs_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes musiconhold=default channel = 3 channel=4 run ztcfg -vv a restart asterisk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom 360 / firmware 6.5.1 / registration problems with Asterisk
Hi, we just upgraded from 1.2.10 to 1.2.13 and now encounter strange problems with our snom phones (FW 6.2.3 to 6.5.1). Upon phone boot everything works fine. Phone registers and asterisk is happy. Soon afterwards the registration is lost however. Sometimes after a few minutes the phone reregisters, sometimes not. This only seems to happen on the first configured line. Switching back to 1.2.10 solved the problem. What changed between those to versions? Maybe a new setting on the snoms we have to take care of? Funny thing: I set defaultexpiry=60 and told the phone to use 1min as well. After the phone registered I watched the expiry counter with sip show peer. It counted backwards from 60 to about 40. Then it jumped to 70, counted to 0 and the phone was gone. This is somewhat reproducable. And it simply does not look right... Any ideas? Kind regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970
Hello; Maybe this is a little off-topic, but I need help. I need to repair a cisco 7970, but in my country(spain) cisco is only selling, they don't repair if you're not client. Because I bought on ebay, I'm not client, so I have no chance. I tried to repair by myself, the problem is on the LCD screen, I need a replace, anyone know which part number is it (manufacturer and part number), and where I can get a replacement? Anyway, if someone knows a technical service in Spain, or Europe, where I can ask for the piece, it will help a lot. Thank you. David. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: How to change IAX default port 4569 to some other port :Debug Message Attached
Its a hosted server with public IP. I also noted that it doesn't get replies from the service provide. (But my home computer does get replies with no problem from the same service provide). SIP registers fine. I can login using SSH. Apache server on port 80 works ok. Only IAX is giving trouble. It all started all of a sudden. ISP days nothing was changed on their side. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DB9 e1 to RJ45 pinout
On 24 Nov 2006, at 11:34, Giordano Grandis wrote: Hi all, anyone known how can i create an adapter from a DB9 port of a E1 to an RJ45 plug? My telco left active the db9 port, but on my te407p card i have rj45 connection. Probably not. Call the telco and get them to fit the correct termination (120ohms I think). Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfers in SER + Asterisk architecture
Hi Marco, Ser has IP of Asterisk server in his trusted table, Asterisk isn't registered in Ser. When Ser needs to use Asterisk, it simply rewrites the IP destination with Asterisk's IP, and routes them to him. For example, here's one failed attempt in transferring a call PSTN - VoIP - VoIP: PSTN Asterisk Ser phone_A phone_B |INVITE| | | | | -- | | | | | 100 Trying | | | | | --- | | | | | | INVITE| | | | | -- |INVITE | | | | | --- | | | | |100 trying | | | 100 trying | --- | | | 100 trying | --- | 180 Ringing | | | -- | 180 Ringing | --- | | | 180 Ringing | -- | | | | -- | | | | | ACK | | | | | --- | ACK | | | | | --- | ACK | | | | | --- | | | | RTP | | | | == | | | | | | | | | | REFER | | | | REFER| --- | | | | -- | | | | | 404 Not Found | | | | | --- | 404 Not Found | | | | | -- | | | | | | | In this example, phone_A answers the PSTN originated call, and wants to transfer the call to phone_B. A REFER message is than routed backwards to Asterisk, and he replies with those 404 Not Found messages. Phone_B never gets called! Should Asterisk be registered in Ser so this can work properly? How can that be done? Thanks, Ricardo. Marco Mouta wrote: Hi Ricardo, Could you post a specific example where your problem happens. That way would be easier for me to try to help you on this. Does asterisk is registred into SER , or you have trust based relationship between them? On 11/23/06, *Ricardo Carvalho* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem is that when the REFER message (that carries the Refer-To number to whom the call should be transferred) gets to Asterisk, it replies with a 404 Not Found message, and the Call Transfer isn't established! Any ideas on how can I solve this problem? Thanks in advance, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options
Re: [asterisk-users] Asterisk and TDM400P ?
Noc Phibee wrote: small question if you know the TDM400P: if the fxo module are at the slot 4, the RJ11 connector are the number 4 ? accordingly to the documentation I have, yes, it's the connector #4. btw, why don't you put your module to the first slot? In this case you would avoid this mess with the numbers... AF. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfers in SER + Asterisk architecture
do you have created Asterisk views to SER database? Are you using sip realtime on asterisk? please post your extensions.conf. By the way, I'm Portuguese:) Qualquer coisa manda mail pode ser q consiga ajudar. On 11/24/06, Ricardo Carvalho [EMAIL PROTECTED] wrote: Hi Marco, Ser has IP of Asterisk server in his trusted table, Asterisk isn't registered in Ser. When Ser needs to use Asterisk, it simply rewrites the IP destination with Asterisk's IP, and routes them to him. For example, here's one failed attempt in transferring a call PSTN - VoIP - VoIP: PSTN Asterisk Ser phone_A phone_B |INVITE| | | | | -- | | | | | 100 Trying | | | | | --- | | | | | | INVITE| | | | | -- |INVITE | | | | | --- | | | | |100 trying | | | 100 trying | --- | | | 100 trying | --- | 180 Ringing | | | -- | 180 Ringing | --- | | | 180 Ringing | -- | | | | -- | | | | | ACK | | | | | --- | ACK | | | | | --- | ACK | | | | | --- | | | | RTP | | | | == | | | | | | | | | | REFER | | | | REFER| --- | | | | -- | | | | | 404 Not Found | | | | | --- | 404 Not Found | | | | | -- | | | | | | | In this example, phone_A answers the PSTN originated call, and wants to transfer the call to phone_B. A REFER message is than routed backwards to Asterisk, and he replies with those 404 Not Found messages. Phone_B never gets called! Should Asterisk be registered in Ser so this can work properly? How can that be done? Thanks, Ricardo. Marco Mouta wrote: Hi Ricardo, Could you post a specific example where your problem happens. That way would be easier for me to try to help you on this. Does asterisk is registred into SER , or you have trust based relationship between them? On 11/23/06, *Ricardo Carvalho* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem is that when the REFER message (that carries the Refer-To number to whom the call should be transferred) gets to Asterisk, it replies with a 404 Not Found message, and the Call Transfer isn't established! Any ideas on how can I solve this problem? Thanks in advance, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DB9 e1 to RJ45 pinout
from my aging memory. Standard E1 pinouts for RJ45 jack are 1-2 (TX) and 4-5(RX). So, trim DB9 and make standard RJ45 jack. Also, http://www.pccables.com/01910.htm http://www.zytrax.com/tech/layer_1/cables/tech_rs232.htm#t1 On 11/24/06, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all, anyone known how can i create an adapter from a DB9 port of a E1 to an RJ45 plug? My telco left active the db9 port, but on my te407p card i have rj45 connection. Anyone can help me pls ? Thanks in advance -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.14/548 - Release Date: 23/11/2006 15.22 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DB9 e1 to RJ45 pinout
I forgot to mention that in the db9 part I guess pins 2-3(tx) and 6-8(rx) are used. I'm sorry I dont recall ground. On 11/24/06, Erick Perez [EMAIL PROTECTED] wrote: from my aging memory. Standard E1 pinouts for RJ45 jack are 1-2 (TX) and 4-5(RX). So, trim DB9 and make standard RJ45 jack. Also, http://www.pccables.com/01910.htm http://www.zytrax.com/tech/layer_1/cables/tech_rs232.htm#t1 On 11/24/06, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all, anyone known how can i create an adapter from a DB9 port of a E1 to an RJ45 plug? My telco left active the db9 port, but on my te407p card i have rj45 connection. Anyone can help me pls ? Thanks in advance -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.14/548 - Release Date: 23/11/2006 15.22 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Junk faxes
Hey everybody, I wanted to know what other may be doing to stem the flood of inbound junk faxes? We currently using Asterisk/iaxmodem/Hylafax for fax services and get a number of junk faxes daily. Most (If not all) of them have caller-id blocked and have a TSI of . I was hoping that, since we are using a PRI, there would be other information coming across that I could use to identify these spammers. Any suggestion would be appreciated. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Junk faxes
Unfortunately a lot of people don't bother to set TSI and have blocked Caller ID on their fax line so you would get false positives if you filtered out those faxes. I just did a HylaFAX install last week where the enduser was extremely pleased about the fax-to-email - when a junk fax came in (about 30% of their faxes!) she just deleted it from Outlook. She felt totally empowered. -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Friday, November 24, 2006 8:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Junk faxes Hey everybody, I wanted to know what other may be doing to stem the flood of inbound junk faxes? We currently using Asterisk/iaxmodem/Hylafax for fax services and get a number of junk faxes daily. Most (If not all) of them have caller-id blocked and have a TSI of . I was hoping that, since we are using a PRI, there would be other information coming across that I could use to identify these spammers. Any suggestion would be appreciated. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Doubling up; redunancy with DUNDi
Hi :) We currently have a single * box with 4-port E1 card terminating
60 channels:
[PSTN]
| | 2 x E1
[Asterisk]
| | 2 x E1
[Legacy PBX]
What I'd like to have is this:
[PSTN]
| \__
||
[*1]- - - -[*2] - DUNDi peering between 2 * boxes
||
[Legacy PBX]
Whereby a call in either direction would be routed either 'straight
through' to/from the PSTN from/to the Legacy PBX, or in the case where
all channels were in use (we max out at about 40-45 channels usage), it
would connect via IAX to a free channel on the other * box.
This is slightly different from the DUNDi tutorials I have found on the
web (including markster's excellent one on voip-magazine.com) whereby
DUNDi is only consulted if 'is this extension local?' fails. In my
case, both sets of extensions are local to both boxes, but I want to
utilise any spare capacity on the other machine. Also, I am not using
any SIP devices here, only Zaptel ones.
Should this be as straightforward as replacing our current 'Dial'
entries in extensions.conf :
exten = _31.,1,Dial(Zap/G2/${EXTEN})
with a 'DUNDi-aware' version like:
exten = _31.,1,Dial(Zap/G2/${EXTEN})
switch = DUNDi/priv
How does this deal gracefully with the scenario of 'no free channels' ?
Is there much variance in DUNDi between 1.2.X and 1.4.X ?
Cheers,
Gavin.
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Re: [asterisk-users] Junk faxes
Hey everybody, I wanted to know what other may be doing to stem the flood of inbound junk faxes? We currently using Asterisk/iaxmodem/Hylafax for fax services and get a number of junk faxes daily. Most (If not all) of them have caller-id blocked and have a TSI of . I was hoping that, since we are using a PRI, there would be other information coming across that I could use to identify these spammers. Any suggestion would be appreciated. Take a good look at the resources on the Internet for dealing with junk fax. The TCPA of 1991 made them essentially illegal. The 2005 update broadened the scope a bit, but there are still a bunch of rules you need to follow or you're subject to penalties. You can also do a much better job of getting caller-id by subscribing to an 800# service that puts ANI information in the caller-id field before delivering it to you; this assumes you're willing to pick up the tab for incoming calls. See http://www.fcc.gov/cgb/consumerfacts/unwantedfaxes.html for more info. The biggest reason we still have junk faxes is that so few people make use of the available remedies. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Junk faxes
Joe Greco wrote: You can also do a much better job of getting caller-id by subscribing to an 800# service that puts ANI information in the caller-id field We have our own 1800 lines, hm I'll have to look into this. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
for see if the card answer, what is the process ? since your port is configured to be in the interne context, just add this to this context exten = s,1,Answer exten = s,2,Playback(tt-monkeys) exten = s,3,Hangup watch the console and dial-in. if you get monkeys screaming at you, it worked ! hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
Hi, i receive a call on my analog line but my asterisk don't answer ;=) do you know if they hae a solution for know if the card see the call ? for see if it's not my cable don't work .. thanks bye ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DB9 e1 to RJ45 pinout
I had some troubles once with db9 and RJ45... It was due to an impedance pb. As I remember, you have to change the impedance in the adaptor (with adding some R-C)... I haven't done it, and my line was half-working (sometimes yes, sometimes not...). So, I suggest you to check if the impedance are the same... Le Vendredi 24 Novembre 2006 16:21, Erick Perez a écrit : I forgot to mention that in the db9 part I guess pins 2-3(tx) and 6-8(rx) are used. I'm sorry I dont recall ground. On 11/24/06, Erick Perez [EMAIL PROTECTED] wrote: from my aging memory. Standard E1 pinouts for RJ45 jack are 1-2 (TX) and 4-5(RX). So, trim DB9 and make standard RJ45 jack. Also, http://www.pccables.com/01910.htm http://www.zytrax.com/tech/layer_1/cables/tech_rs232.htm#t1 On 11/24/06, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all, anyone known how can i create an adapter from a DB9 port of a E1 to an RJ45 plug? My telco left active the db9 port, but on my te407p card i have rj45 connection. Anyone can help me pls ? Thanks in advance -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.14/548 - Release Date: 23/11/2006 15.22 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and TDM400P ?
On Fri, Nov 24, 2006 at 02:25:38PM +0100, Anton Frolov wrote: Tzafrir Cohen wrote: And probably fxsks=3: I'm not sure whether or not KS is used in the French telco . signalling=fxs_ks Nitpicking: 'signalling=fxs_ks' is the line to add to /etc/asterisk/zapata.conf . 'fxsks=4' is the line to put in /etc/zaptel.conf works Ok in France with both: France Telecom and Freebox. Thanks -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitoring awareness
Hello, I'm discovering asterisk, it seem to be a great soft. I have seen a fonction to record calls that's a great fontion but there is something disturbing me. When the record start, except if the recorder prevent the other part, he is not aware of the recording... I dont find a way from the feature.conf how to play a sound when a monitor start to record :/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Encrypted password for voicemail
On Fri, Nov 24, 2006 at 06:15:12AM -0800, je . wrote: Hello all, I was wondering whether the only way to store the passwords for the voicemail is in extensions.conf. voicemail.conf ? Is it perhaps possible to store it (encrypted) in a DB or store the hash of the password (as is standard in unix) in the extensions.conf file? Storing an encrypted DB? Encrypted in what way? What exactly do you want to defend against? Finally, it does not seem to me that the password the user enters to obtain his voicemail is encrypted between the user and Asterisk. Is this correct? Basically, not. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto recording calls?
On 21 Nov 2006, at 14:34, Jay Moore wrote:
Tim Panton wrote:
On 20 Nov 2006, at 21:46, Jay Moore wrote:
Doug wrote:
Hmmm. I think this may work for WinAmp and
incidently for Windows Media Player:
http://www.mlkj.net/gsm/
No luck with WMP. Anyone else have any suggestions on
playing .gsm files in Windows Media Player?
Jay, would you be interested in a java applet that played gsm files ?
I think I have the bones of one kicking around that I could dust off
and polish up.
This only really works if you are providing your customers access
to the gsm files
via http and can easily wrap a page around them...
Well, ideally I'd like for my customers to be able to download the
file and play it on their computer, but a Java applet that plays
them on our website would be a cool idea, too. So, yeah, I'd be
interested.
Here's my bare bones implementation GSM player for voicemail etc
http://www.westhawk.co.uk/software/playGSM/PlayGSM.html
As the web page says, you can use it in 2 ways:
1) as an applet - arrange for the web app (php?) to set the 'url' param
and it will play download and play the selected file
2) or if you download the jar file (http://www.westhawk.co.uk/
software/playGSM/PlayGSM.jar)
you can also run it as an application
java -jar playGSM.jar {url of gsm}
It is GPL - so enjoy and fix bugs
Tim.
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Re: [asterisk-users] Server Configuration for E1's
Imran M Yousuf wrote: Dear Users, *I am fairly new to Digium and Asterisk. I wanted to know that if I use the Digium product THREE Digium Wildcard TE412Ps’ (Quad E1 Card) how many calls can I handle simultaneously.* I want to use the cards with the following Configurations: /Intel® Xeon™ 3.00GHz/800MHz, 2M Processor/ /1GB PC2-3200 DDR2 SDRAM (2x512MB + 2X1GB) memory/ /Integrated Dual Channel Ultra320 SCSI Adapter/ /NC7781 Single Port PCI-X embedded NIC/ /Hot plug drive cage - Ultra3 (6X1)/ /High Speed IDE CD-ROM Drive/ / / /72.8GB Pluggable Ultra320 SCSI 15,000 rpm (1) Universal Hard Drive/ *Asterisk Business Edition* * * *3 X TE412P* I have a requirement of handling 350 Calls using a single Server and please note the Server will used to transferring the call only. Other Servers will handle gateway Negotiation and Billing. This server will SIMPLY be a Gateway. Please let me know if this configuration too high or too low. If anybody has better solution please let me know that as well. Thank you, waiting eagerly for a response. Imran M Yousuf I have always read never to go beyond two cards per box due to interrupts (Digium, not sure about Sangoma) but have never tried it. I have found a perfect fit for me. I do exactly as you describe, the box just converts TDM to SIP to handoff except I use T1 instead of E1. 1 HPDL320 + 1 Sangoma 4 port T1 card + 2x 80 gig SATA RAID 1 + 1 gig memory + 3Ghz Pentium 4 Core Solo Running minimal services and modules, with NFAS I get 95 B channels. When all 95 are lit and handing off calls with no codec translation (Ulaw), I hit exactly 50% CPU utilization which I feel very comfortable with and would not want to go over. These boxes are the most stable piece of my pie so far. Comparable IBM systems work just as well. Failover is provided by the RAID 1 and the fact that I have rollover on seven trunk groups from the telco. Dual power supplies would be nice. I see very little HD usage since all processing is mostly done from the wire (PRI) to solid state parts and put on the wire (Ethernet) with no swap. If the server dies, up to 95 calls could drop but if the caller calls back, they will be routed to one of the other remaining trunk groups. Originally, I was going to go with big beefy servers and put two cards in each but the prospect of dropping 190 calls as opposed to 95 along with the fact that the smaller server can be readily purchased at ~$1,500 made the choice pretty clear. To answer your question. I have no idea because of the Digium interrupts, you never mentioned codec conversion, hopefully straight Alaw since you are talking E1. I would think not, plus, do you really want to put all your eggs in one basket? Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Request for help with DISA (Not taking my input number correctly?)
Crazy Boy wrote:
Hi,
Thank you for response. I configured DISA and its working sometimes
and not working sometimes. Here I am sending the configuration and
output on Asterisk server console:
Extensions.conf file content:
[custom-CLID]
exten = s,1,Answer
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(10)
exten = s,4,Authenticate(1234)
exten = s,5,DISA(no-password|disa-ext)
[disa-ext]
exten = _X.,1,DIAL(IAX2/[EMAIL PROTECTED]/${EXTEN},30,tr)
Output on server console:
-- Playing 'custom/v1' (language 'en')
== CDR updated on Zap/1-1
-- Executing Goto(Zap/1-1, custom-CLID|s|1) in new stack
-- Goto (custom-CLID,s,1)
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing DigitTimeout(Zap/1-1, 5) in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout(Zap/1-1, 10) in new stack
-- Set Response Timeout to 10
-- Executing Authenticate(Zap/1-1, 1234) in new stack
-- Playing 'agent-pass' (language 'en')
-- Playing 'auth-thankyou' (language 'en')
-- Executing DISA(Zap/1-1, no-password|disa-ext) in new stack
-- Executing Dial(Zap/1-1,
IAX2/[EMAIL PROTECTED]/187773456|30|tr) in new stack
-- Called [EMAIL PROTECTED]/187773456
-- Hungup 'IAX2/teliax-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Hungup 'Zap/1-1'
What is happening is:
1) I called my zap number from my mobile
2) My IVR is responding
3) I entered a extension number to access DISA
4) Asterisk asked the secret (PIN) code to access DISA
5) I entered password of DISA
6) After validating the password, its giving Dial tone to dial a USA
number
7) I entered 17187773456 (This is a toll free number) to test
8) Call is going sometimes and call is not going sometimes. If we
observe on server console, its not taking my input number properly and
taking my input phone number wrongly.
9) I tested from other mobiles also. But, its not taking my input
number as i entered sometimes.
9) What is the wrong?
Please tell me. Looking forward to your response. Thank you.
Regards,
Chandra.
Do you have relaxdtmf set in your zap conf file? Try from a landline
phone and see if you have the same issue. If you have relaxdtmf=yes try
no and test again, or do the opposite. It is obvious that asterisk is
getting dtmf but it is mixing it up.
Also try dialing other companies IVRs and navigating the menus, maybe
your cell phone is just screwed up?
Is the call going through any other boxes that may have DTMF settings
misconfigured? I have seen DTMF come out in doubles (ie you press 911
and asterisk sees 99111).
Thanks,
Steve Totaro
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Re: [asterisk-users] Junk faxes
Doug Lytle wrote: Hey everybody, I wanted to know what other may be doing to stem the flood of inbound junk faxes? We currently using Asterisk/iaxmodem/Hylafax for fax services and get a number of junk faxes daily. Most (If not all) of them have caller-id blocked and have a TSI of . I was hoping that, since we are using a PRI, there would be other information coming across that I could use to identify these spammers. Any suggestion would be appreciated. Doug Is this an 800 number? If so, and maybe if your telco is very nice, you can get ANI. Toll free lines get ANI for billing purposes but it is much more useful than just billing. Sometimes you just have to request it. ANI cannot be blocked like callerID. If the spammers are sophisticated enough, they can still get around the ANI but I doubt they are. BTW OT from another thread, I upgraded IAXmodem and used the example Hylafax modem config files instead of using Hylafax's addmodem and things seem to be much better. Keeping my fingers crossed and will load first thing Monday. Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring awareness
[EMAIL PROTECTED] wrote: Hello, I'm discovering asterisk, it seem to be a great soft. I have seen a fonction to record calls that's a great fontion but there is something disturbing me. When the record start, except if the recorder prevent the other part, he is not aware of the recording... I dont find a way from the feature.conf how to play a sound when a monitor start to record :/ Either play a file with a beep or a verbal message that this call may be recorded for such and such reason. This can be done easily in the dialplan by calling playback or background prior to monitor. Depending on local laws, you may be OK if just one party on the call knows it is being recorded. Other states have different laws. I have no idea how the law works when one caller is in one state with one set of laws and the other caller is in a different state with different laws. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mfcr/R2
Hello! I'm tryuing to bring up an R2 connection but eventhough I've followed the guidelines in: http://zarzamora.com.mx/asterisk/17 something seems to be missing. When an incomming call is generated I get: Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 - 0001 [1/ 1/Idle /Idle] Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 Detected Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 Making a new call with CRN 32771 Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 1101 - [2/ 2/Idle /Idle] Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:2672 handle_uc_event: Unicall/24 event Detected and that's it, afterwards just a busy tone and the telco guy says the channel turns sealed. When I try an outbound call I get: -- Attempting call on Unicall/g2/12345678 for [EMAIL PROTECTED]:1 (Retry 1) Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 Call control(1) Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 Make call Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 Making a new call with CRN 32769 Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 0001 - [1/ 1/Idle /Idle] Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:2672 handle_uc_event: Unicall/24 event Dialing Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 seize_ack_wait_expired Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 R2 prot. err. [1/ 40/Seize /Idle] cause 32776 - Seize ack timed out Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 1001 - [1/ 1/Idle /Idle] Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:2672 handle_uc_event: Unicall/24 event Protocol failure -- Unicall/24 protocol error. Cause 32776 Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 Channel echo cancel Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 Channel gains Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 Channel switching -- Hungup 'UniCall/24-1' Nov 24 05:53:54 NOTICE[-286483536]: pbx_spool.c:234 attempt_thread: Call failed to go through, reason 1 Something else funny is happening: as soon as the telco makes a reset of the trunk they say they start receving data as if the PBX would be generating calls, then all channels go sealed except for 1. Anyone having an idea on how to solve this? Alyed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Junk faxes
Doug Lytle wrote: Joe Greco wrote: You can also do a much better job of getting caller-id by subscribing to an 800# service that puts ANI information in the caller-id field We have our own 1800 lines, hm I'll have to look into this. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. Try dialing from a phone that lets you block callerid *67 + number with pri debug span x turned on. You will see all of the info that is being passed in the IEs. You may already have ANI enabled and not know it. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cisco 7961 , asterisk and busy lamp
How can i programming a Cisco 7961 to be used as busy lamp field? my configs : sccp.conf : [devices] type= 7961 tzoffset= 0 autologin = 601 speeddial = *31, Hanna -- other SIP telefon extensions.conf : exten = *31,hint,SIP/hanna exten = *34,hint,SCCP/601 on SIP Telefon ( SNOM 360 ) everything functions good and i have busy lamp when cisco telefon Offhook, but differently does not function any idea ? Any input is greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FS: Sangoma 10 port FXO card
Hi all, I have a surplus Sangoma 10 port FXO card for sale. This model could be upgraded to 12 ports or even changed to FXS or a combo of FXO/FXS by changing the grand-daughter cards (each card supports 2 lines). You could also downgrade the card by removing any or all of the daughter cards. I'm asking US$450 plus shipping to the lower 48. Paypal or Master/Visa only. Thanks Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FS: Sangoma 10 port FXO card
Please don't cross post FS items to *-users - that's what *-biz is for. CP On Nov 24, 2006, at 10:45 AM, Mark Phillips wrote: Hi all, I have a surplus Sangoma 10 port FXO card for sale. This model could be upgraded to 12 ports or even changed to FXS or a combo of FXO/FXS by changing the grand-daughter cards (each card supports 2 lines). You could also downgrade the card by removing any or all of the daughter cards. I'm asking US$450 plus shipping to the lower 48. Paypal or Master/Visa only. Thanks Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Junk faxes
Steve Totaro wrote:
Try dialing from a phone that lets you block callerid *67 + number
with pri debug span x turned on. You will see all of the info that is
being passed in the IEs. You may already have ANI enabled and not
know it.
I'll give that a try when I have access to that feature.
As for ANI, I tried the NoOP(${CALLIERID(ANI)}) and it only showed my
cell phone caller-id. I wonder if that could be a feature of a PRI.
I'll have to talk to the phone admin.
Doug
-- Ben Franklin quote: Those who would give up Essential Liberty to
purchase a little Temporary Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] Junk faxes
Steve Totaro wrote: Is this an 800 number? If so No, they do have 1 800 number though, but it goes to the main inbound number of the facility. BTW OT from another thread, I upgraded IAXmodem and used the example Hylafax modem config files instead of using Hylafax's addmodem and things seem to be much better. Keeping my fingers crossed and will load first thing Monday. Great! I love the package. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FS: Sangoma 10 port FXO card
Why not. Users by stuff too! On Fri, 2006-11-24 at 11:40 -0800, Anthony Rodgers wrote: Please don't cross post FS items to *-users - that's what *-biz is for. CP On Nov 24, 2006, at 10:45 AM, Mark Phillips wrote: Hi all, I have a surplus Sangoma 10 port FXO card for sale. This model could be upgraded to 12 ports or even changed to FXS or a combo of FXO/FXS by changing the grand-daughter cards (each card supports 2 lines). You could also downgrade the card by removing any or all of the daughter cards. I'm asking US$450 plus shipping to the lower 48. Paypal or Master/Visa only. Thanks Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Rewriting caller ID from database?
At 22:07 22/11/2006 -0700, Marco Mouta wrote: You can do it using AstDB, just load the database with callerid names and numbers and then include a lookup on database in all incoming calls, so you can override whatever you wanted:) Thanks everyone. Indeed, it seems like using the embedded BerkeleyDB-based AstDB is good enough for what I'm trying to do. However, asterisk barfs on the following script that I used to import data: #/bin/bash asterisk -rx database put cidname 1234567 'Me - cellular' asterisk -rx database put cidname 1234567 'Me - home' etc. Any idea why? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Rewriting caller ID from database?
Am Freitag, den 24.11.2006, 22:22 +0100 schrieb Vincent Delporte: At 22:07 22/11/2006 -0700, Marco Mouta wrote: You can do it using AstDB, just load the database with callerid names and numbers and then include a lookup on database in all incoming calls, so you can override whatever you wanted:) Thanks everyone. Indeed, it seems like using the embedded BerkeleyDB-based AstDB is good enough for what I'm trying to do. However, asterisk barfs on the following script that I used to import data: #/bin/bash asterisk -rx database put cidname 1234567 'Me - cellular' asterisk -rx database put cidname 1234567 'Me - home' etc. Do try asterisk -rx database put cidname 12345676 \Me - cellular\ or asterisk -rx 'database put cidname 3871263 Me - home' These quotations seem to work. Hth Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
At 03:14 PM 11/22/2006, you wrote: The missing piece of the puzzle: I'm extension 203. I want any call I park to get parked at extension 2203. I want a call my boss parks to park at 2205, since he's ext. 205. In other words, I want calls parked FROM extension XYZ to be parked AT extension (XYZ+2000). I don't see a way to force parked calls to a specific extension. I'm probably just missing the answer, but I've googled for it and I can't find it. That doesn't seem to be the way parking was designed. It's a first available distribution of a series of numbers you choose. The problem with your plan is that it can't handle a second call on an extension. Coming up in V1.4 is something called SLA or shared line appearance which might do what you want depending upon how it's implemented. For the moment you just need to tell people extension to pick up to retrieve a parked call. Here it's always 701 as we've never yet parked 2 calls. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible, horrible firewall issues in * to * setup
Okay, I *think* I got it, but I must be missing something. Here is
what the files say on the various boxen:
On *1:
[401]
type=friend
secret=password
qualify=yes
port=4569
notransfer=yes
host=dynamic
dial=IAX/401
context=from-internal
[601]
type=friend
secret=password
qualify=yes
port=4569
notransfer=no
host=dynamic
dial=IAX/601
context=from-internal
On *2:
iax.conf:
[601]
type=friend
disallow=all
context=from-internal
canreinvite=yes
allow=ulaw
[asterisk-1]
username=601
type=peer
secret=777
qualify=yes
host=asterisk-1.someplace.net
disallow=all
context=from-internal
canreinvite=yes
allow=ulaw
register=601:[EMAIL PROTECTED]
extensions.conf:
[outrt-003-CallA1]
exten = _4XXX,1,Macro(dialout-trunk,1,${EXTEN:1},,)
exten = _4XXX,n,Macro(outisbusy,)
So now, of course, I can call from *2 to extension 401 on *1 (by
dialing 4401) without a problem, but I still cannot seem to call from
*1 to extensions on *2. It's complaining about there not being a route
to the given extension, which makes sense I guess. I don't know how to
create a proper outbound route on *1 to *2 since I don't have a trunk
to direct it to, just a registration. I'm sure I'm lacking something
fundamental here - any help would be greatly appreciated.
Thanks!
On 11/24/06, Tim Panton [EMAIL PROTECTED] wrote:
On 22 Nov 2006, at 22:21, Lachek Butalek wrote:
My mission is to get one * box to dial another * box' extensions. I
have set this up previously without any issues by making a simple IAX
trunk/extension pair on the two boxes and create a dial plan with a
prefix like 9|XXX to select an extension on the other box.
My problem is that I now have to do this with extremely restrictive
firewalls thrown into the mix - firewalls I have no control over.
Basically, the setup is:
*1 --- FW1 --- (Internet) --- FW2 --- FW3 --- *2
I have control over firewall 1 and 3, but not 2. Using port forwarding
(4569 UDP) on FW1, I have been able to make calls from *2 to *1. My
problem lies with making calls the other way, as I have no way of port
forwarding on FW2.
If FW2 and FW3 permit outbound UDP and associated replies you won't
need to.
(even if they NAT them).
Set up 4569 on FW1 to go to *1
Add *2 as a peer (and user) in iax.conf on *1
Do _nothing_ with FW3
Set up *2 to _register_ with *1
The repeated registration from 2 to 1 will keep the any
NAT's and port maps open and tell 1 how to reach 2.
(IAX is great)
Tim.
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Re: [asterisk-users] Terrible, horrible firewall issues in * to * setup
Correction: those 'secret' lines are of course all supposed to say '777'. :)
On 11/24/06, Lachek Butalek [EMAIL PROTECTED] wrote:
Okay, I *think* I got it, but I must be missing something. Here is
what the files say on the various boxen:
On *1:
[401]
type=friend
secret=password
qualify=yes
port=4569
notransfer=yes
host=dynamic
dial=IAX/401
context=from-internal
[601]
type=friend
secret=password
qualify=yes
port=4569
notransfer=no
host=dynamic
dial=IAX/601
context=from-internal
On *2:
iax.conf:
[601]
type=friend
disallow=all
context=from-internal
canreinvite=yes
allow=ulaw
[asterisk-1]
username=601
type=peer
secret=777
qualify=yes
host=asterisk-1.someplace.net
disallow=all
context=from-internal
canreinvite=yes
allow=ulaw
register=601:[EMAIL PROTECTED]
extensions.conf:
[outrt-003-CallA1]
exten = _4XXX,1,Macro(dialout-trunk,1,${EXTEN:1},,)
exten = _4XXX,n,Macro(outisbusy,)
So now, of course, I can call from *2 to extension 401 on *1 (by
dialing 4401) without a problem, but I still cannot seem to call from
*1 to extensions on *2. It's complaining about there not being a route
to the given extension, which makes sense I guess. I don't know how to
create a proper outbound route on *1 to *2 since I don't have a trunk
to direct it to, just a registration. I'm sure I'm lacking something
fundamental here - any help would be greatly appreciated.
Thanks!
On 11/24/06, Tim Panton [EMAIL PROTECTED] wrote:
On 22 Nov 2006, at 22:21, Lachek Butalek wrote:
My mission is to get one * box to dial another * box' extensions. I
have set this up previously without any issues by making a simple IAX
trunk/extension pair on the two boxes and create a dial plan with a
prefix like 9|XXX to select an extension on the other box.
My problem is that I now have to do this with extremely restrictive
firewalls thrown into the mix - firewalls I have no control over.
Basically, the setup is:
*1 --- FW1 --- (Internet) --- FW2 --- FW3 --- *2
I have control over firewall 1 and 3, but not 2. Using port forwarding
(4569 UDP) on FW1, I have been able to make calls from *2 to *1. My
problem lies with making calls the other way, as I have no way of port
forwarding on FW2.
If FW2 and FW3 permit outbound UDP and associated replies you won't
need to.
(even if they NAT them).
Set up 4569 on FW1 to go to *1
Add *2 as a peer (and user) in iax.conf on *1
Do _nothing_ with FW3
Set up *2 to _register_ with *1
The repeated registration from 2 to 1 will keep the any
NAT's and port maps open and tell 1 how to reach 2.
(IAX is great)
Tim.
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Re: [asterisk-users] Terrible, horrible firewall issues in * to * setup
Two apologies to be made: #1: Sorry for all the spamming. #2: Sorry for all the top-posting. Gmail doesn't lend itself well to email lists since it neatly tucks all the quoted text away (out-of-sight, out-of-mind for the poster), leaving a mess for those whose email clients do not. Again, my apologies. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does voicemail authentication take place?
res |= ast_register_application(app4, vmauthenticate, synopsis_vmauthenticate, descrip_vmauthenticate); You need to look more closely at the code. This snippet registers the dial plan application VMAuthenticate so vmauthenticate is called wherever you use that function in your dial plan. static char *app4 = VMAuthenticate; static char *synopsis_vmauthenticate = Authenticate with Voicemail passwords; static char *descrip_vmauthenticate = VMAuthenticate([EMAIL PROTECTED]|options]): This application behaves the\n same way as the Authenticate application, but the passwords are taken from\n voicemail.conf.\n If the mailbox is specified, only that mailbox's password will be considered\n valid. If the mailbox is not specified, the channel variable AUTH_MAILBOX will\n be set with the authenticated mailbox.\n\n Options:\n s - Skip playing the initial prompts.\n; --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and UK ISDN 30
Anyone know if Asterisk will work with ISDN 30 and what sort of device I'll need to connect it? neil safeharbour IT Ltd Your IT Department ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and UK ISDN 30
On Fri, Nov 24, 2006 at 11:51:41PM -, Neil Tancock wrote: Anyone know if Asterisk will work with ISDN 30 and what sort of device I'll need to connect it? It will work with UK ISDN, but ensure it's EuroISDN and NOT UK ISDN (it's set in the telco switch and can generally be changed). UK ISDN is v85 and EuroISDN v110. ISDN2e (as in basic rate) is the Euro variety. Modern PRI lines should be Euro, but some telcos still provision the older UK variant. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Card don't hangup but Asterisk hangup
Hi , I have a problem with a X100, i do a external call to the asterisk server . The dialplan its simple answer and hangup.. when it's done , the telephone which i did the call , is in line but asterisk server is finish. I'll apreciate all your suggestion. Greetings, txus. The asterisk output: -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' zapata.conf [channels] language=es context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes ;===Añadido= busydetect=yes answeronpolarityswitch=yes hanguponpolarityswitch=yes callprogress=no progzone=es ; callerid=asreceived ; asi los telf saben kien llama.??? rxgain=8.2 txgain=1.0 ; echocancelwhenbridged=yes ; echotraining=yes ;=== usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no group=0 callgroup=1 pickupgroup=1 immediate=no ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re:Call Transfers in SER + Asterisk
Can u show SER asterisk configuration or logs? -- Regards -- M Emran Managing Director InSpiration Software Ltd. E-mail: [EMAIL PROTECTED] [EMAIL PROTECTED] Web: www.inspiresoftbd.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Card don't hangup but Asterisk hangup
Jesus Jimenez wrote: Hi , I have a problem with a X100, i do a external call to the asterisk server . The dialplan its simple answer and hangup.. when it's done , the telephone which i did the call , is in line but asterisk server is finish. I'll apreciate all your suggestion. Greetings, txus. The asterisk output: -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' zapata.conf [channels] language=es context=from-pstn signalling=fxs_ks Is your PSTN line really kwelstart? If it is loopstart, please use fxs_ls and busydetect. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Rewriting caller ID from database?
Steven wrote: There are two I can think of. Hoodahek and asterdex (or asteridex) We used hoodahek at first, but now use asterdex(sp?) It has a web interface to enter the new names into. We use it to fixup, corp. cell phones and used to use it for our leagcy PBX extensions. I use some custom scripts to do database lookups and rewrite CallerID information. Everything works fine with regard to the CID name, however my Cisco 7960 and Linksys SPA-942 phones do not display the calling number. Instead, they display the called number. This makes the phone's call return feature not work. The calling number and name are both properly displayed on all of the softphone clients that I've tried. Here's the format I'm using to set the CallerID. SET CALLERID JONES DARYL A6508701826 Can anyone help? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Correct syntax to access a shell variable?
I would like to access my shell environment variable MYIP from within
sip.conf to put in externip.
I've tried some variations of syntax after reading The Future of
Telephony but it's not working yet.
Should it be
externip=${ENV{$MYIP}}
or some other syntax??
Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Re: [asterisk-users] How to park calls on a specific extension
You need the valet parking module. It's a third pary add-on app. This is a direct link to the current code http://www.pbxfreeware.org/app_valetparking.c Here is a link to the original README http://www.loligo.com/asterisk/misc/apps/app_valetparking.README On 11/22/06, Steve Sobol [EMAIL PROTECTED] wrote: Currently at our office, if I want someone else to pick up a call, I have to transfer the call to them. So I'm looking into call parking, which is ALMOST perfect. The missing piece of the puzzle: I'm extension 203. I want any call I park to get parked at extension 2203. I want a call my boss parks to park at 2205, since he's ext. 205. In other words, I want calls parked FROM extension XYZ to be parked AT extension (XYZ+2000). I don't see a way to force parked calls to a specific extension. I'm probably just missing the answer, but I've googled for it and I can't find it. TIA for any help you can offer. -- Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows Victorville, California PGP:0xE3AE35ED It's all fun and games until someone starts a bonfire in the living room. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re:Call Transfers in SER + Asterisk
HI,
thanks for your reply. Here is my ser.cfg and other config files please
guide me.
ser.cfg
--
debug=5
fork=no
log_stderror=yes
listen=2xx.xxx.xxx.xxx # INSERT YOUR IP ADDRESS HERE
port=5060
children=4
dns=no
rev_dns=no
fifo=/tmp/ser_fifo
fifo_db_url=mysql://ser:[EMAIL PROTECTED]/ser
loadmodule /usr/lib/ser/modules/mysql.so
loadmodule /usr/lib/ser/modules/sl.so
loadmodule /usr/lib/ser/modules/tm.so
loadmodule /usr/lib/ser/modules/rr.so
loadmodule /usr/lib/ser/modules/maxfwd.so
loadmodule /usr/lib/ser/modules/usrloc.so
loadmodule /usr/lib/ser/modules/registrar.so
loadmodule /usr/lib/ser/modules/auth.so
loadmodule /usr/lib/ser/modules/auth_db.so
loadmodule /usr/lib/ser/modules/uri.so
loadmodule /usr/lib/ser/modules/uri_db.so
loadmodule /usr/lib/ser/modules/domain.so
loadmodule /usr/lib/ser/modules/mediaproxy.so
loadmodule /usr/lib/ser/modules/nathelper.so
loadmodule /usr/lib/ser/modules/textops.so
loadmodule /usr/lib/ser/modules/avpops.so
loadmodule /usr/lib/ser/modules/permissions.so
modparam(auth_db|permissions|uri_db|usrloc|domain, db_url,
mysql://ser:[EMAIL PROTECTED]/ser)
modparam(auth_db, calculate_ha1, 1)
modparam(auth_db, password_column, password)
modparam(nathelper, rtpproxy_disable, 1)
modparam(nathelper, natping_interval, 0)
modparam(mediaproxy,natping_interval, 30)
modparam(mediaproxy,mediaproxy_socket, /var/run/mediaproxy.sock)
modparam(mediaproxy,sip_asymmetrics,/etc/ser/sip-clients)
modparam(mediaproxy,rtp_asymmetrics,/etc/ser/rtp-clients)
modparam(usrloc, db_mode, 2)
modparam(registrar, nat_flag, 6)
modparam(rr, enable_full_lr, 1)
modparam(tm, fr_inv_timer, 27)
modparam(tm, fr_inv_timer_avp, inv_timeout)
modparam(permissions, db_mode, 1)
modparam(permissions, trusted_table, trusted)
# - request routing logic ---
# main routing logic
route {
# -
# Sanity Check Section
# -
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483, Too Many Hops);
break;
};
if (msg:len max_len) {
sl_send_reply(513, Message Overflow);
break;
};
# -
# Record Route Section
# -
if (method==INVITE client_nat_test(3)) {
# INSERT PROXY IP ADDRESS HERE
record_route_preset( 2xx.xxx.xxx.xxx:5060;nat=yes);
} else if (method!=REGISTER) {
record_route();
};
# -
# Call Tear Down Section
# -
if (method==BYE || method==CANCEL) {
end_media_session();
};
# -
# Loose Route Section
# -
if (loose_route()) {
if ((method==INVITE || method==REFER) !has_totag())
{
sl_send_reply(403, Use From=ID);
break;
};
if (method==INVITE)
{
if (!allow_trusted())
{
if (!proxy_authorize(,subscriber))
{
proxy_challenge(,0);
break;
} else if (!check_from()) {
sl_send_reply(403, user From=ID);
break;
};
consume_credentials();
};
if (client_nat_test(3) || search(^Route:.*;nat=yes)){
setflag(6);
use_media_proxy();
};
};
route(1);
break;
};
# -
# Call Type Processing Section
# -
if (!is_uri_host_local()) {
if (is_from_local() || allow_trusted()) {
route(4);
route(1);
} else { sl_send_reply(403, Forbidden-two);
};
break;
};
if (method==ACK) {
route(1);
break;
} if (method==CANCEL) {
route(1);
break;
} else if (method==INVITE) {
route(3);
break;
}
Re: [asterisk-users] How to change IAX default port 4569 to some other port
Something like this should work in your iptables: iptables -A PREROUTING -t nat -p tcp --dport 1234 -i eth0 -j DNAT --to-destination 127.0.0.1:4569 iptables -I FORWARD 1 -d 127.0.0.1 -p tcp --dport 4569 -j ACCEPT This would forward port 1234 to port 4569. bp On 11/23/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Hi all, All of a sudden all my IAX DIDs have gone down. I couldn't find any reason other than that the ISP is blocking port 4569. DIDs register fine from my home server, but not from office server, which is not behind any NAT. SIP registers fine. I am trying to change IAX port but it apparantly IAX works only on 4569. Changing it in iax.conf doesn't do anything. Changing it is registration string also doesn't help. How can I make IAX work on some other port? -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to kill a meet me room at midnight
Not quite what I'm looking for. I ant to hang up all channels (zap or sip) in meetme room 5 On 11/23/06, Michiel van Baak [EMAIL PROTECTED] wrote: On 19:18, Thu 23 Nov 06, Eric Bishop wrote: Other than rebooting the server or restarting Asterisk from cron does anyone know how to kill a meetme room at midnight. Or perhaps other creative ways people deal with callers who don't hang up. You can use soft hangup chan -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to kill a meet me room at midnight
You could write an extension which executes meetme kick, for all the channels, but I am not sure how to execute such a thing at a given time. on Saturday 11/25/2006 Eric Bishop([EMAIL PROTECTED]) wrote Not quite what I'm looking for. I ant to hang up all channels (zap or sip) in meetme room 5 On 11/23/06, Michiel van Baak [EMAIL PROTECTED] wrote: On 19:18, Thu 23 Nov 06, Eric Bishop wrote: Other than rebooting the server or restarting Asterisk from cron does anyone know how to kill a meetme room at midnight. Or perhaps other creative ways people deal with callers who don't hang up. You can use soft hangup chan -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Not quite what I'm looking for. I ant to hang up all channels (zap or sip) in meetme room 5brbrdivspan class=gmail_quoteOn 11/23/06, b class=gmail_sendernameMichiel van Baak/b lt;a href=mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]/agt; wrote:/spanblockquote class=gmail_quote style=border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;On 19:18, Thu 23 Nov 06, Eric Bishop wrote:brgt; Other than rebooting the server or restarting Asterisk from cron does anyone brgt; know how to kill a meetme room at midnight. Or perhaps other creative waysbrgt; people deal with callers who don't hang up.brbrYou can use soft hangup lt;changt;brbr--brMichiel van Baakbra href=mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]/abra href=http://michiel.vanbaak.eu;http://michiel.vanbaak.eu/abrGnuPG key: a href=http://pgp.mit.edu:11371/pks/lookup?op=getamp;search=0x71C946BD;http://pgp.mit.edu:11371/pks/lookup?op=getamp;search=0x71C946BD /abrbrquot;Why is it drug addicts and computer afficionados are both called users?quot;brbr___br--Bandwidth and Colocation provided by a href=http://Easynews.com; Easynews.com/a --brbrasterisk-users mailing listbrTo UNSUBSCRIBE or update options visit:brnbsp;nbsp; a href=http://lists.digium.com/mailman/listinfo/asterisk-users;http://lists.digium.com/mailman/listinfo/asterisk-users /abr/blockquote/divbr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help needed - Can anyone please explain to me what is causing this - TDM2400P
Hello all. I have two identically configured asterisk servers each with a TDM2422P (with two S400M FXS modules and two X400M FXO modules). *1 works perfectly and the sound quality is great. However, I am having audio quality problem with *2 when making or receiving calls over the PSTN - the sound gets cut off from the earpiece when I start to speak in the microphone and then when I stop talking, I can hear the other party!! I do not have any problems with SIP calls. Also, if I connect a POTs line to Channel 5 - (WCTDM/0/4 FXSKS) and dial the number, it just keeps ring and the ZAP channel does not answer the call, nor does it even show up in the console. If I call out to the PSTN through ZAP 5, All other channels are working ok. I have tried to recompile zaptel but that did not solve the problem. I even went as far as swapping the TDM2400P cards in the two asterisk servers and the problem still persisted, confirming that the digium card is not faulty. I did not notice any conflicting IRQs either. I would really appreciate help in solving this problem. Asterisk1: centOS 4.3 asterisk 1.2.8 zaptel 1.2.6 Asterisk2: CentOS 4.4 Asterisk 1.2.12.1 zaptel 1.2.9.1 Thanks. Naija Man ** [EMAIL PROTECTED] asterisk]# cat /proc/interrupts CPU0 CPU1 0: 225479540 225426528IO-APIC-edge timer 1: 31 34IO-APIC-edge i8042 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 12: 75 3IO-APIC-edge i8042 14:20273802027327IO-APIC-edge ide0 169: 0 0 IO-APIC-level uhci_hcd 185: 0 0 IO-APIC-level ehci_hcd, uhci_hcd 193: 0 0 IO-APIC-level uhci_hcd 201: 0 0 IO-APIC-level uhci_hcd 209: 162967 162431 IO-APIC-level 3ware Storage Controller 217: 225465762 225426370 IO-APIC-level wctdm24xxp 233:9805871 0 PCI-MSI eth0 NMI: 0 0 LOC: 450936535 450936534 ERR: 0 MIS: 0 ** Below is the output of cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM2400P Board 1 IRQ misses: 1 1 WCTDM/0/0 FXSKS (In use) 2 WCTDM/0/1 FXSKS (In use) 3 WCTDM/0/2 FXSKS (In use) 4 WCTDM/0/3 FXSKS (In use) 5 WCTDM/0/4 FXSKS (In use) 6 WCTDM/0/5 FXSKS (In use) 7 WCTDM/0/6 FXSKS (In use) 8 WCTDM/0/7 FXSKS (In use) 9 WCTDM/0/8 FXOKS (In use) 10 WCTDM/0/9 FXOKS (In use) 11 WCTDM/0/10 FXOKS (In use) 12 WCTDM/0/11 FXOKS (In use) 13 WCTDM/0/12 FXOKS (In use) 14 WCTDM/0/13 FXOKS (In use) 15 WCTDM/0/14 FXOKS (In use) 16 WCTDM/0/15 FXOKS (In use) 17 WCTDM/0/16 18 WCTDM/0/17 19 WCTDM/0/18 20 WCTDM/0/19 21 WCTDM/0/20 22 WCTDM/0/21 23 WCTDM/0/22 24 WCTDM/0/23 *** [EMAIL PROTECTED] asterisk]# lsmod Module Size Used by wcusb 23840 0 wctdm 41280 0 wcfxo 16928 0 wcte11xp 30496 0 wct1xxp20640 0 wct4xxp 251328 0 tor2 93600 0 wctdm24xxp 65344 15 zaptel196740 40 wcusb,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2,wctdm24xxp crc_ccitt 6209 1 zaptel ** [EMAIL PROTECTED] asterisk]# cat zapata.conf ; ; Zapata telephony interface ; ; Configuration file [channels] ; usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no callerid=asreceived callwaiting=no usedistinctiveringdetection=no callwaitingcallerid=no threewaycalling=no transfer=no canpark=no cancallforward=no callreturn=no ;callreturn=yes faxdetect=no echocancel=yes echocancelwhenbridged=no callprogress=no busydetect=no ;busydetect=yes musiconhold=default useincomingcalleridonzaptransfer=yes ;busycount=4 ;group=3 context=from-desks signalling=fxo_ks callerid=CORDLESS 1132 channel = 9 group=1 usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no ;usecallingpres=yes useincomingcalleridonzaptransfer=yes rxgain=8.0 txgain=2.0 context=from-pstn signalling=fxs_ks channel = 1-6 ** tel2*CLI zap show status Description Alarms IRQbpviol CRC4 Wildcard TDM2400P Board 1OK 1 0 0 tel2*CLI zap show status * tel2*CLI zap show channels Chan Extension Context Language MusicOnHold pseudofrom-pstn default 1from-pstn default
Re: [asterisk-users] FOP is not displaying all my SIP extensions neither all E1 channels , why?
Hi, I must say that i'm not very used with customization of FOP. I've a box runing Flash Op.Panel, and i notice that the screen is full of buttons from my sip users, as well as Zapata channels. The problem is that i have more Zapata channels as well as SIP users, is there any way to get a scroll on this to display everything? do i need to resize the buttons? For sure someone now how to solve this basic question:) You can reduce the button size, update to the latest snapshot that includes 'slow' horizontal scrolling, or both. Best regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
On Wed, Nov 22, 2006 at 04:51:26PM -0800, Ira wrote: At 03:14 PM 11/22/2006, you wrote: The missing piece of the puzzle: I'm extension 203. I want any call I park to get parked at extension 2203. I want a call my boss parks to park at 2205, since he's ext. 205. In other words, I want calls parked FROM extension XYZ to be parked AT extension (XYZ+2000). I don't see a way to force parked calls to a specific extension. I'm probably just missing the answer, but I've googled for it and I can't find it. That doesn't seem to be the way parking was designed. It's a first available distribution of a series of numbers you choose. The problem with your plan is that it can't handle a second call on an extension. Coming up in V1.4 is something called SLA or shared line appearance which might do what you want depending upon how it's implemented. For the moment you just need to tell people extension to pick up to retrieve a parked call. Here it's always 701 as we've never yet As I was noting in an earlier message, the parking lot concept is to my view not a thrilling interface at best, and I can't see many times one would want it in a SOHO environment.It seems best for a large PBX where people are moving to random places to pick up calls, and many calls may be parked at any given time. For many people, a far simpler interface is to just put the call on hold -- by pressing just one hold button, and then go pick it up as easily as possible somewhere else.Shared line systems help to do that but from a different direction. The parking lot approach has you remember a somewhat random number told to you, and then to go dial it.People can remember their own extension much more easily, so one good interface in that case is a way to dial a number prefixNNN to pick up a call held on a specific extension (in my pickup group). Or more simply, to dial the pickup number, and if there is only one call on hold, it gives it to you, and if there is more than one, it lets you dial the extension that put it on hold and reads the extensions that have calls on hold to remind you. This is a better interface in an environment were the small security risk here is minimal, such as a home or small office. The nice thing about this interface is that a phone speed-dial function button can be programmed to the pickup number. This means that parking and getting a call can amount to pressing one button to put the call on hold, moving to another phone and pushing another button to get the call, which is about the simplest interface and the one found on key systems and some pbx. Where security is a concern (and the current call parking lot does not actually provide a great deal) you can have a call transferred to a valet, but not require the user to remember a parking lot number if they know the number of the extension that put the call on hold. The valet system gets us partway from what I read, but it still uses the arbitrary number slots. It still requires the user know to transfer a call to the valet. Of course if you know what phone you are going to, you can just do unattended xfer to it, as long as there is not too short a voicemail timeout. But again that's a way more complex interface than push hold and push pickup. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Snom 360 Multiple calls on hold help
Upgraded the client this morning, lets hope this works for good :) Can you tell us what the best way to suggest bug features or additions is? For example, Arizona is not a option under time zones, we dont have DST and never change time, would be nice if that was added! Also, a digital clock, alot of americans are lazy and I guess still cant figure out how to read a analog clock :( Thanks again for this new beta release, I couldnt of asked for a quicker response time, my hat is truly off to Snom for actually caring about the customer! Thanks again! On 11/23/06, Sven Fischer [EMAIL PROTECTED] wrote: Hi, try our latest beta version 6.5.2 which can be found here: http://www.snom.com/wiki/index.php/Snom360/Firmware/Beta_Versions http://www.snom.com/wiki/index.php/Snom320/Firmware/Beta_Versions http://www.snom.com/wiki/index.php/Snom300/Firmware/Beta_Versions Release Notes: http://www.snom.com/wiki/index.php/Snom360/Firmware/Release_Notes#6.5.2_beta http://www.snom.com/wiki/index.php/Snom320/Firmware/Release_Notes#6.5.2_beta http://www.snom.com/wiki/index.php/Snom300/Firmware/Release_Notes#6.5.2_beta Regards, Sven On Wednesday 22 November 2006 17:56, Ron McCarthy wrote: Yeah, doing more testing shows that the speed keys are broken, but dialing it works!!! Ugg!!! can you let me know if you get a new firmware? Im going to try and downgrade... Thanks! On 11/22/06, Alban [EMAIL PROTECTED] wrote: Yes, already. Waiting now for a new firmware... Regards, Alban Le Mercredi 22 Novembre 2006 13:50, Steve Davies a écrit: On 11/22/06, Alban [EMAIL PROTECTED] wrote: I'm having the same problem, pressing a speed dial/extension when 2 calls are on the phone connect the 2 calls together. Typing the number instead of using speed dial works. With older firmware, 6.2.1 or 6.3, it was working... But then other problem with pickup, deadlocking the phone (or slowing it down). Certainly due to the dp bug (fixed in 6.5.1). Regards, Alban. Has this been reported to snom by anyone? They are generally pretty good at fixing this type of issue and providing beta firmware. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
