[asterisk-users] Outbound FXO call, getting You must first dial...
I am not sure what I might be set up wrong, but dialing out with my Zap/1
port seems to alwyas get the You must first dial a 1 when calling this
number message from what sounds like the actual PSTN. My zapatel.conf and
extensions.conf bits below. Any advice? (I do receive inbound calls, and
it does sound like I am getting the PSTN error. I do notice that when I
get an inbound call, I have 5 secs of sevear static before it suddenly
becomes clear.. could that be happening on the outboud as well munging the
first few digits?)
signalling=fxs_ks
language=us
context=inbound_qwest
sendcalleridafter=2
callerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
channel=1
exten = _9.,1,Dial(Zap/1/${EXTEN:1},60)
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
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Re: [asterisk-users] Outbound FXO call, getting You must first dial...
I did have a bit of trouble with searching (what to search on), though
looking for the w in the dial command did return quite a few hits as you
described. Thank you so much for taking the time to reanswer a covered
subject.
I played with the settings and 1 w and removing the :1 after EXTEN (not
stripping the leading digit?) makes it reliable. Not stripping the first
digit worked about 2 in 5 attempts. I stumbled onto that idea when I
missdialed a number 9215037 digit number and it worked! The 2 was a
fat finger mistake. So I tried 90xx and that worked.
As I have some success now, I can tune this so it works as the HowTo's
list. :)
Thank you again!
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Sat, 7 Oct 2006, Rich Adamson wrote:
Nick Ellson wrote:
I am not sure what I might be set up wrong, but dialing out with my Zap/1
port seems to alwyas get the You must first dial a 1 when calling this
number message from what sounds like the actual PSTN. My zapatel.conf and
extensions.conf bits below. Any advice? (I do receive inbound calls, and
it does sound like I am getting the PSTN error. I do notice that when I
get an inbound call, I have 5 secs of sevear static before it suddenly
becomes clear.. could that be happening on the outboud as well munging the
first few digits?)
signalling=fxs_ks
language=us
context=inbound_qwest
sendcalleridafter=2
callerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
channel=1
exten = _9.,1,Dial(Zap/1/${EXTEN:1},60)
You should probably do a little research before posting questions like this
as its been answered many many time.
The problem is that some pstn central offices are not ready to receive dtmf
digits as quickly as what asterisk sends them. So, an option w has been
added to the Dial command to instruct asterisk to wait about 200 milliseconds
before sending dtmf. Try something like this:
exten = _9.,1,Dial(Zap/1/w${EXTEN:1},60)
and notice that lower-case w in the string. If that doesn't fix the
problem, try two ww's in a row.
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Re: [asterisk-users] Caller ID on Zap not always working
Do you have sendcalleridafter=2 in your [channels] section of /etc/asterisk/zapata.conf? (I had to change it for mine to work) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Tue, 3 Oct 2006, [EMAIL PROTECTED] wrote: Hello, We are using asterisk with 6 POTS lines and Caller ID is not always read from the lines properly. Is there a way to make asterisk wait for the caller id before proceeding with the dial plan or is it possible a setting is wrong in a conf file somewhere? Any guidance would be helpful. Thanks, NB ___ --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] X100M location in circuit requirement?
I have just added a A1200P+FXO port to my home line for testing. In the interest of saving time, I wired it to the Phone port of my Fujitsu Speed Port DSL Modem. So in total, I have the line from the telco going to my Fujitsu, which goes to the FXO port. In parallel at the POP I have a DSL line filter in series with the rest of my house phones (2 phones, one modem, and the ADT alarm system). So I look at my console this morning as see all these events. Judging by the period, I am going to guess that my ADT alarm panel is calling home to check in on the parallel existsing phone system and Asterisk is seing that. Would that be correct? And.. When it comes to Asterisk, does it function fine as a second system to the same line as the house phones? (Also, can anyone point me to the list of configuration options for an X100M FXO module for the asterisk conf files?) Sep 27 21:50:16 NOTICE[17729]: chan_zap.c:6073 ss_thread: Got event 17 (Polarity Reversal)... Sep 27 21:50:23 WARNING[17729]: chan_zap.c:6149 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (inbound_qwest, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Sep 28 00:20:23 NOTICE[18249]: chan_zap.c:6073 ss_thread: Got event 17 (Polarity Reversal)... Sep 28 00:20:30 WARNING[18249]: chan_zap.c:6149 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (inbound_qwest, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Sep 28 01:50:23 NOTICE[18565]: chan_zap.c:6073 ss_thread: Got event 17 (Polarity Reversal)... Sep 28 01:50:31 WARNING[18565]: chan_zap.c:6149 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (inbound_qwest, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Sep 28 02:50:23 NOTICE[18874]: chan_zap.c:6073 ss_thread: Got event 17 (Polarity Reversal)... Sep 28 02:50:31 WARNING[18874]: chan_zap.c:6149 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (inbound_qwest, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Sep 28 04:20:25 NOTICE[20057]: chan_zap.c:6073 ss_thread: Got event 17 (Polarity Reversal)... Sep 28 04:20:32 WARNING[20057]: chan_zap.c:6149 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (inbound_qwest, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Sep 28 05:20:24 NOTICE[20387]: chan_zap.c:6073 ss_thread: Got event 17 (Polarity Reversal)... Sep 28 05:20:31 WARNING[20387]: chan_zap.c:6149 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (inbound_qwest, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Sep 28 05:50:25 NOTICE[20556]: chan_zap.c:6073 ss_thread: Got event 17 (Polarity Reversal)... Sep 28 05:50:33 WARNING[20556]: chan_zap.c:6149 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (inbound_qwest, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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 lockup and boot error after one day?
I had installed a X100P clone just as a quick trial at real analog access and my system would not boot successfully (error below) but I figured it was just a bad card and at $20 I didn't care much. Now I have a 12 port card using the FXO/FXS modules, and installed it Tuesday. It booted great, got the drivers working and even made a few calls in each direction. Over last night the system locked up hard (IE: Dark screen, num-lock or caps lock won't toggle) After a hard reset, I see a lot of Dazed and Confused) messages and then it hangs, but after a second boot and all those errors again it makes it to run level 3 and I can shell in and grab dmesg before it locks up. Take the card out, it's all happy again. Ok, why did I ask Asterisk-Users? I have not seen this until I tried these two telecom based cards and I want to know if this is something that happens using cards or zaptel/zapata drivers, or should I be looking at a hardware specific forum? (IE: Any Asterisk users have this happen?) snip large section of normal boot process, no errors [17179587.012000] Zapata Telephony Interface Registered on major 196 [17179587.012000] Zaptel Version: 1.2.8 Echo Canceller: KB1 [17179587.052000] ACPI: PCI Interrupt :05:05.0[A] - GSI 20 (level, low) - IRQ 20 [17179587.052000] OpenVox A1200P version: 1.1 [17179587.052000] OpenVox A1200P passed register test [17179587.128000] evbug.c: Connected device: PC Speaker, isa0061/input0 [17179587.128000] evbug.c: Connected device: ImPS/2 Logitech Wheel Mouse, isa0060/serio1/input0 [17179587.128000] evbug.c: Connected device: AT Translated Set 2 keyboard, isa0060/serio0/input0 [17179587.952000] Module 0: Installed -- AUTO FXO (FCC mode) [17179589.104000] Module 1: Installed -- AUTO FXS/DPO [17179589.104000] Module 2: Not installed [17179589.104000] Module 3: Not installed [17179589.104000] Module 4: Not installed [17179589.104000] Module 5: Not installed [17179589.104000] Module 6: Not installed [17179589.104000] Module 7: Not installed [17179589.104000] Module 8: Not installed [17179589.104000] Module 9: Not installed [17179589.104000] Module 10: Not installed [17179589.104000] Module 11: Not installed [17179589.108000] Found a OpenVox A1200P: Version 1.1 (2 modules) [17179589.108000] buffer re-sync occur from 0 to 2 [17179589.268000] hw_random hardware driver 1.0.0 loaded [17179591.232000] ipmi message handler version 39.0 [17179591.24] IPMI Watchdog: driver initialized [17179591.248000] ipmi device interface [17179591.388000] IPMI SMB Interface driver [17179591.388000] ipmi_smb: DMI specifies SSIF @ 0x42, slave address 0x84 [17179594.024000] Uhhuh. NMI received for unknown reason 00 on CPU 0. [17179594.024000] Dazed and confused, but trying to continue [17179594.024000] Do you have a strange power saving mode enabled? [17179597.992000] ipmi: Found new BMC (man_id: 0x000322, prod_id: 0x4311, dev_id: 0x20) [17179598.012000] IPMI Watchdog: Unable to register misc device [17179598.112000] i2c /dev entries driver [17179598.132000] IPMI System Interface driver. [17179598.192000] ipmi_si: Unable to find any System Interface(s) [17179603.892000] Uhhuh. NMI received for unknown reason 00 on CPU 0. [17179603.892000] Dazed and confused, but trying to continue [17179603.892000] Do you have a strange power saving mode enabled? [17179605.724000] Uhhuh. NMI received for unknown reason 00 on CPU 0. [17179605.724000] Dazed and confused, but trying to continue [17179605.724000] Do you have a strange power saving mode enabled? [17179607.872000] Uhhuh. NMI received for unknown reason 00 on CPU 0. snip and this continues with either a lockup, or a login, then lock up Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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] Right way to prevent analog channel from answering the phone?
I am in the process of learning my A1200P, and i would like an elegant way to prevent it from answering the phone, but still make outbound calls. I tried zap destroy channel 1 (which worked, but pissed off Asterisk ;) Is there a more elegant way to tell it to answer/not answer on command? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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] Right way to prevent analog channel from answering the phone?
Well, that just makes too much sense.. starting to feel a tad embarrased here ;) Ok, I will simply remove the Dial(IAX2/4005) and have it not do anything, that will error on the console, but that's ok and let the parallel land line have the call (AKA: The wife) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Wed, 27 Sep 2006, Rich Adamson wrote: Nick Ellson wrote: I am in the process of learning my A1200P, and i would like an elegant way to prevent it from answering the phone, but still make outbound calls. I tried zap destroy channel 1 (which worked, but pissed off Asterisk ;) Is there a more elegant way to tell it to answer/not answer on command? I don't have an A1200P, but most zap channel interfaces are built to not answer an incoming call unless you specifically configure asterisk to do it. There are only two basic conditions under which an incoming call will be answered: 1. by including the answer statement, like: exten = 3556,1,Answer exten = 3556,2,Wait,1 exten = 3556,3,Authenticate(3017) exten = 3556,4,Meetme(3556|pM) 2. a SIP phone (or other phone) user picks up the handset. So, in zapata.conf you have definitions for each of the A1200P ports, and one of the items in those definitions is context=something. If that context statement points to some non-existent context name (like context=xyz), there is nothing that would answer the incoming call. If the context=something points to a real context (in extensions.conf), then review that context to ensure there is nothing there to answer the incoming call. (Note: some asterisk applications will automatically answer incoming calls.) You could also define that context and include statements like: [no-answer] exten = _X.,1,Hangup ___ --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] Right way to prevent analog channel from answering the phone?
Erm.. nothing that I know of, other than I do not yet know what that means? :) -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Wed, 27 Sep 2006, Eric ManxPower Wieling wrote: What is wrong with using the WaitForRing app? Rich Adamson wrote: Nick Ellson wrote: I am in the process of learning my A1200P, and i would like an elegant way to prevent it from answering the phone, but still make outbound calls. I tried zap destroy channel 1 (which worked, but pissed off Asterisk ;) Is there a more elegant way to tell it to answer/not answer on command? I don't have an A1200P, but most zap channel interfaces are built to not answer an incoming call unless you specifically configure asterisk to do it. There are only two basic conditions under which an incoming call will be answered: 1. by including the answer statement, like: exten = 3556,1,Answer exten = 3556,2,Wait,1 exten = 3556,3,Authenticate(3017) exten = 3556,4,Meetme(3556|pM) 2. a SIP phone (or other phone) user picks up the handset. So, in zapata.conf you have definitions for each of the A1200P ports, and one of the items in those definitions is context=something. If that context statement points to some non-existent context name (like context=xyz), there is nothing that would answer the incoming call. If the context=something points to a real context (in extensions.conf), then review that context to ensure there is nothing there to answer the incoming call. (Note: some asterisk applications will automatically answer incoming calls.) You could also define that context and include statements like: [no-answer] exten = _X.,1,Hangup ___ --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] Anybody have the opvx1200.c driver?
The link is not working at OpenVox. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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
(GOT IT) Re: [asterisk-users] Anybody have the opvx1200.c driver?
Thanks all, I have it now :) -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Tue, 26 Sep 2006, Nick Ellson wrote: The link is not working at OpenVox. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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] Dual core
My home Asterisk server is running dual proc dual core zeon 3ghz, seems happy, no crashes that I didn't bring about myself. ;) mpg123 does occasionally hang a pid at 100% now and then, but it does that on single proc/single core systems too. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Sat, 23 Sep 2006, Matt Florell wrote: Asterisk is very happy on dual core. It greatly reduces load. We just put a Pentium-D in poduction last week and it is working verry well. We have a Core 2 Duo on order that we should be putting in production next week. MATT--- On 9/22/06, Tomislav Par?ina [EMAIL PROTECTED] wrote: Hi list. I have one quick question. Does Asterisk work with dual core processors in version 1.2? Will it work with dual core processors in 1.4? I'm planning to buy new machine for one installation and I have to decide will I buy single or dual core processor. -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Forwarding
How might you identify a mobile #? (assuming you refer to cellular phones) Now that phone companies are allowing you to transfer your land line to a mobile, it's no longer practical to use prefix blocking. Where I worked, they just gave up and just restricted forwarding to long distant numbers except by exclusion (for those at the top of the food chain, so to speak) If there is a way to identify, from the number dialed, that the destination is a mobile phone, I'd be interested as well. And curious, why such a preference? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 22 Sep 2006, Paul Hales wrote: I am trying to find a way to stop phones from being forwarded to mobiles - the clients are allowed to forward phones in general, but we want to stop them forwarding calls to mobiles. Is there a SIP header I can check for in the dialplan? I have searched around, but I probably don't quite know what keyword to use in my search... PaulH ___ --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] A1200+fxo, anyone using this?
Thanks for the feedback Peter, I am going to try one with an FXO and then one of the $30 fixed port single FXO PCI cards from pbxeq.com as well. See if there is a real difference there. Anybody try the A-100PCI card? When I do, I'll post what I find. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Mon, 18 Sep 2006, Peter Lindquist wrote: Nick, I use one and it works just fine for me with 2 FXO and 2 FXS at the moment. I would say it is a great board to have and experiment with and as you say not too big or too small. Peter Nick Ellson wrote: I know it's not a digium product, but the 12 port A1200P card with a single FXO module at pbxeq.com at first glance would seem to be the way to get started for me with an in-system controller card. 4 ports seems too small for expansion, the huge 24 port card a tad too big (and spendy). So has anyone used this card with Asterisk? I googled for reviews and have not found anything, and I am tryingto find a way to search the archives without looking at each month one at a time. Nick ___ --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] A1200+fxo, anyone using this?
I know it's not a digium product, but the 12 port A1200P card with a single FXO module at pbxeq.com at first glance would seem to be the way to get started for me with an in-system controller card. 4 ports seems too small for expansion, the huge 24 port card a tad too big (and spendy). So has anyone used this card with Asterisk? I googled for reviews and have not found anything, and I am tryingto find a way to search the archives without looking at each month one at a time. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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] saved.gsm - Voicemail greeting ??
I seem to have stumped myself on this one. I had my son rattle off some really great sound bytes for his own extension (busy, after hours, etc) and that was easy to set up with the dial plan. Now I have his actual VM greeting in a .gsm and no idea how to get it into his VM Greeting, I am guessing that these are not stored where the other sounds are, maybe in the database? I looked through the .pdf book, not many helpful hits on google. Help? :) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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] saved.gsm - Voicemail greeting ??
Ok, I did finally find the file structure (/var/spool/asterisk/voicemail/default/exten/*) , but everything is stored in 3 formats with set names, so I bet there is a process that creates all that, if I have just the 8000hz mono .gsm, is there an entry point or program I can feed this file to? The sample I did by recording my own using the Voicemail system: ls /var/spool/asterisk/voicemail/default/4003 INBOX busy.WAV busy.gsm busy.wav greet.WAV greet.gsm greet.wav tmp unavail.WAV unavail.gsm unavail.wav Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 15 Sep 2006, Nick Ellson wrote: I seem to have stumped myself on this one. I had my son rattle off some really great sound bytes for his own extension (busy, after hours, etc) and that was easy to set up with the dial plan. Now I have his actual VM greeting in a .gsm and no idea how to get it into his VM Greeting, I am guessing that these are not stored where the other sounds are, maybe in the database? I looked through the .pdf book, not many helpful hits on google. Help? :) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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] saved.gsm - Voicemail greeting ??
Hi John, Yes, I followed an example that put all my family sound files in /var/lib/asterisk/sounds/local, which is also where this file is. Now I am trying to figure out how to get the unavailable|name|Busy .gsm's I made loaded into a mailbox without playing my sounds back into a phone ;) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 15 Sep 2006, John covici wrote: Check in /var/spool/asterisk/voicemail/default/extension number for a particular extension, don't know how you want to differentiate after hours, etc. Also, you can put files in /var/lib/asterisk/sounds/custom and do with them what you want. on Friday 09/15/2006 Nick Ellson([EMAIL PROTECTED]) wrote I seem to have stumped myself on this one. I had my son rattle off some really great sound bytes for his own extension (busy, after hours, etc) and that was easy to set up with the dial plan. Now I have his actual VM greeting in a .gsm and no idea how to get it into his VM Greeting, I am guessing that these are not stored where the other sounds are, maybe in the database? I looked through the .pdf book, not many helpful hits on google. Help? :) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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 ___ --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] saved.gsm - Voicemail greeting ??
Ok, I am getting closer on how to choose the files played. I missed this in the book, but VoIP-info had this: s: Play nothing. (no flags): Play instructions. su: Play unavailable message. u: Play unavailable message, then instructions. sb: Play busy message. b: Play busy message, then instructions. In all cases, the beep.gsm file will also be played, prior to starting to record. and it will play the file I want if I rename my file to the stock named in the spool directory.. But as I have been making only .gsm's, am I leaving the VM Box's half-baked? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 15 Sep 2006, Nick Ellson wrote: I seem to have stumped myself on this one. I had my son rattle off some really great sound bytes for his own extension (busy, after hours, etc) and that was easy to set up with the dial plan. Now I have his actual VM greeting in a .gsm and no idea how to get it into his VM Greeting, I am guessing that these are not stored where the other sounds are, maybe in the database? I looked through the .pdf book, not many helpful hits on google. Help? :) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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] saved.gsm - Voicemail greeting ??
Trying that now... umm, anyone know what condition makes use of just the name in voicemail, is that part of the directory or something? -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 15 Sep 2006, Bill Gibbs wrote: I assume it will use the files .gsm too? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Friday, September 15, 2006 10:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] saved.gsm - Voicemail greeting ?? Example for mailbox 100 under context default /var/spool/asterisk/voicemail/default/100 -rwx-w 1 asterisk asterisk 442604 Aug 22 16:44 busy.wav -rwx-w 1 asterisk asterisk 44976 Aug 22 16:44 busy.WAV -rwx-w 1 asterisk asterisk 25964 Aug 8 02:17 greet.wav -rwx-w 1 asterisk asterisk 2660 Aug 8 02:17 greet.WAV drwx-w 2 asterisk asterisk 4096 Sep 13 13:45 INBOX drwx-w 2 asterisk asterisk 4096 Sep 13 13:37 tmp -rwx-w 1 asterisk asterisk 168044 Aug 9 17:00 unavail.wav -rwx-w 1 asterisk asterisk 17090 Aug 9 17:00 unavail.WAV drwx-w 2 asterisk asterisk 4096 Sep 1 11:31 Work Those are the files (wav format) that it expects for the voicemail greetings/name announcement. Greet.wav is the name. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Ellson Sent: Friday, September 15, 2006 10:26 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] saved.gsm - Voicemail greeting ?? Hi John, Yes, I followed an example that put all my family sound files in /var/lib/asterisk/sounds/local, which is also where this file is. Now I am trying to figure out how to get the unavailable|name|Busy .gsm's I made loaded into a mailbox without playing my sounds back into a phone ;) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 15 Sep 2006, John covici wrote: Check in /var/spool/asterisk/voicemail/default/extension number for a particular extension, don't know how you want to differentiate after hours, etc. Also, you can put files in /var/lib/asterisk/sounds/custom and do with them what you want. on Friday 09/15/2006 Nick Ellson([EMAIL PROTECTED]) wrote I seem to have stumped myself on this one. I had my son rattle off some really great sound bytes for his own extension (busy, after hours, etc) and that was easy to set up with the dial plan. Now I have his actual VM greeting in a .gsm and no idea how to get it into his VM Greeting, I am guessing that these are not stored where the other sounds are, maybe in the database? I looked through the .pdf book, not many helpful hits on google. Help? :) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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 ___ --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] saved.gsm - Voicemail greeting ??
Yes, it does. :) Ok Then, I guess my issue is solved until I see a glitch because i am lacking the .wav|.WAV versions. Thanks All! -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 15 Sep 2006, Bill Gibbs wrote: I assume it will use the files .gsm too? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Friday, September 15, 2006 10:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] saved.gsm - Voicemail greeting ?? Example for mailbox 100 under context default /var/spool/asterisk/voicemail/default/100 -rwx-w 1 asterisk asterisk 442604 Aug 22 16:44 busy.wav -rwx-w 1 asterisk asterisk 44976 Aug 22 16:44 busy.WAV -rwx-w 1 asterisk asterisk 25964 Aug 8 02:17 greet.wav -rwx-w 1 asterisk asterisk 2660 Aug 8 02:17 greet.WAV drwx-w 2 asterisk asterisk 4096 Sep 13 13:45 INBOX drwx-w 2 asterisk asterisk 4096 Sep 13 13:37 tmp -rwx-w 1 asterisk asterisk 168044 Aug 9 17:00 unavail.wav -rwx-w 1 asterisk asterisk 17090 Aug 9 17:00 unavail.WAV drwx-w 2 asterisk asterisk 4096 Sep 1 11:31 Work Those are the files (wav format) that it expects for the voicemail greetings/name announcement. Greet.wav is the name. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Ellson Sent: Friday, September 15, 2006 10:26 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] saved.gsm - Voicemail greeting ?? Hi John, Yes, I followed an example that put all my family sound files in /var/lib/asterisk/sounds/local, which is also where this file is. Now I am trying to figure out how to get the unavailable|name|Busy .gsm's I made loaded into a mailbox without playing my sounds back into a phone ;) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 15 Sep 2006, John covici wrote: Check in /var/spool/asterisk/voicemail/default/extension number for a particular extension, don't know how you want to differentiate after hours, etc. Also, you can put files in /var/lib/asterisk/sounds/custom and do with them what you want. on Friday 09/15/2006 Nick Ellson([EMAIL PROTECTED]) wrote I seem to have stumped myself on this one. I had my son rattle off some really great sound bytes for his own extension (busy, after hours, etc) and that was easy to set up with the dial plan. Now I have his actual VM greeting in a .gsm and no idea how to get it into his VM Greeting, I am guessing that these are not stored where the other sounds are, maybe in the database? I looked through the .pdf book, not many helpful hits on google. Help? :) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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 ___ --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
[asterisk-users] Features.. phone vs. asterisk?
I tried a lot of SIP and IAX softphones looking for ones I liked, noticing some have certain features and others did not. For things like call transfer, call park, group pick-up, line presence, and all those kinds of extras I have a bit of confusion on where it is implemented? Are these functions that Asterisk handles and the phone just triggers them with some out-of-band signal or DTMF sequence? Or does some of this rest on the phone itself? (Here is where I would love TFM to R. :) Just having a hard time finding what to read.) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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] music onhold choppy music problems
What player? I found that my system had mpg123 but too new a version and something was seriously hosed with it. I downgraded to the version listed in the install help and it started working. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Sun, 10 Sep 2006, Matt wrote: anyone has any success in using music onhold. even if we have ztdummy installed we still got choppy music. buy our old asterisk 1.0.9 is working, why is that? thanks for any help in advance. Best Regards matt ___ --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] Hrmm.. OK, what am I missing? sendmail: Cannot open mail:25
OK, help.. Am not sure where this is not configured right. I followed the voicemail.conf directions, even tried specifying sendmail -t directly. My sendmail mail log shows: Sep 10 17:19:26 goonie sSMTP[4439]: Unable to locate mail Sep 10 17:19:26 goonie sSMTP[4439]: Cannot open mail:25 Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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] Hrmm.. OK, what am I missing? sendmail: Cannot open mail:25
Hey, that's why i had no idea how to spot the glitch... I added a line in my /etc/hosts file for mail aimed at my SMTP server, all better now. Thanks! Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Sun, 10 Sep 2006, C F wrote: Take this to sendmail list. this is not an asterisk problem. In any case it looks like it's trying to send email to host mail on port 25 and it's failing. Try doing a telnet mail 25 and see what happens. On 9/10/06, Nick Ellson [EMAIL PROTECTED] wrote: OK, help.. Am not sure where this is not configured right. I followed the voicemail.conf directions, even tried specifying sendmail -t directly. My sendmail mail log shows: Sep 10 17:19:26 goonie sSMTP[4439]: Unable to locate mail Sep 10 17:19:26 goonie sSMTP[4439]: Cannot open mail:25 Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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] Softphones IAX vs. SIP, remote connectivity.
Hey all, A previous annoyance with not being able to call out to my brother on FWD from my Asterisk system had me thinking that since I have my own PBX, and that system has it's own 1-to-1 static NAT to the internet, I should be able to act as the provider for him or any of my family, and have them as local extensions of my PBX, right? So I took my laptop to work (using the X-Lite SIP softphone) and watch my ACL logs on my router for any denies to my Asterisk box. As expected udp/5060, then once that was open, a series of randomish udp/1+ requests. My phone registered, and I tried to call one of the phones behind a PAP2. Worked first shot, and just as clear and responsive as it was when I was home. But, the phones at home could not call me, they when to voice mail. I had heard that SIP doesn't survive NAT all that well, and that IAX native phones do a better job. My question is, given my description of how I am set up and what I am trying to accomplish, should I be looking at SIP or is IAX a more robust choice? (I was hoping to get video working as well, h.263 I believe it is). Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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] Softphones IAX vs. SIP, remote connectivity.
Bob, I will up the logs today, have my phone at work with me. (though the Wife and Kids are not up yet ;) Anything specific I should target? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Bob Chiodini wrote: Nick, Anything helpful in the asterisk or system logs. Try bumping up the debug and verbose levels see what shows up on the console. Weird that it would work inbound and not outbound. Bob... On Thu, 2006-09-07 at 04:48 -0700, Nick Ellson wrote: Hey all, A previous annoyance with not being able to call out to my brother on FWD from my Asterisk system had me thinking that since I have my own PBX, and that system has it's own 1-to-1 static NAT to the internet, I should be able to act as the provider for him or any of my family, and have them as local extensions of my PBX, right? So I took my laptop to work (using the X-Lite SIP softphone) and watch my ACL logs on my router for any denies to my Asterisk box. As expected udp/5060, then once that was open, a series of randomish udp/1+ requests. My phone registered, and I tried to call one of the phones behind a PAP2. Worked first shot, and just as clear and responsive as it was when I was home. But, the phones at home could not call me, they when to voice mail. I had heard that SIP doesn't survive NAT all that well, and that IAX native phones do a better job. My question is, given my description of how I am set up and what I am trying to accomplish, should I be looking at SIP or is IAX a more robust choice? (I was hoping to get video working as well, h.263 I believe it is). Nick ___ --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] Softphones IAX vs. SIP, remote connectivity.
Bruce, I *just* tested the XtremePhone, IAX2 softphone. Other than trying to figure out how to get it to send proper CallerID to the other phones, it worked right off, in both directions. Excellent! Perhaps working the IAX2 angle will be less of a hassle, I will go looking for one that does video now. Maybe it's time to buy an IAX2-ATA adaptor and see how well that works over the net. Nick As for the SIP logs, I start Asterisk with -c already, I did a sip debug and tried my call from the house to my remote SIP phone. YIKES!! Gunna take a bit to understand all that, but I think I did see an INVITE, and a CANCEL twice in a row and I did not hit the hang-up switch. So that might explain why no connection is made, and the called gets my voice-mail (according to my wife) -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Bruce Reeves wrote: Nick, I have done what you are talking about as far as being a provider for family members. I used an IAX softphone mainly to eliminate the need for so many holes in the firewall. And secondly because the idefisk IAX softphone allowed me to extract the zip version, configure the phone, and zip the folder up and email it to my family members. So for my mom it was simply unzip the folder and On 9/7/06, Nick Ellson [EMAIL PROTECTED] wrote: Bob, I will up the logs today, have my phone at work with me. (though the Wife and Kids are not up yet ;) Anything specific I should target? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Bob Chiodini wrote: Nick, Anything helpful in the asterisk or system logs. Try bumping up the debug and verbose levels see what shows up on the console. Weird that it would work inbound and not outbound. Bob... On Thu, 2006-09-07 at 04:48 -0700, Nick Ellson wrote: Hey all, A previous annoyance with not being able to call out to my brother on FWD from my Asterisk system had me thinking that since I have my own PBX, and that system has it's own 1-to-1 static NAT to the internet, I should be able to act as the provider for him or any of my family, and have them as local extensions of my PBX, right? So I took my laptop to work (using the X-Lite SIP softphone) and watch my ACL logs on my router for any denies to my Asterisk box. As expected udp/5060, then once that was open, a series of randomish udp/1+ requests. My phone registered, and I tried to call one of the phones behind a PAP2. Worked first shot, and just as clear and responsive as it was when I was home. But, the phones at home could not call me, they when to voice mail. I had heard that SIP doesn't survive NAT all that well, and that IAX native phones do a better job. My question is, given my description of how I am set up and what I am trying to accomplish, should I be looking at SIP or is IAX a more robust choice? (I was hoping to get video working as well, h.263 I believe it is). Nick ___ --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 -- Bruce Nortex Networks ___ --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] Softphones IAX vs. SIP, remote connectivity.
Hello Michael, I just had both Mom and my brother up as extensions on my Asterisk pbx using IAX2, the Cubix phone for now, but I downloaded and tried several. I loke multiple lines, but a clean GUI is better for my family.. Oh yeah, it worked flawlessly :) I open one port to my server udp/4569 and that was it. I shut the rest off. For remote family, IAX2 will be what I use right now. Anybody see a Video capable version for Windows? The MAC has one, darn it. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Ferguson, Michael wrote: Hi Guys I too am trying to do exactly the same thing in being a provider for family members. My Asterisk server is on a public ip, my home is behind a Watchguard Firebox, my job is also behind a Firebox. I am using a combination of Cisco 7960, Linksys 941 and XTEN Softphone. Sometimes it works and sometimes it does not. You idea on using a IAX2 softphone appears to be what will solve my problem. Thanks very much Post more ideas. 'preciate it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Ellson Sent: Thursday, September 07, 2006 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphones IAX vs. SIP, remote connectivity. Bruce, I *just* tested the XtremePhone, IAX2 softphone. Other than trying to figure out how to get it to send proper CallerID to the other phones, it worked right off, in both directions. Excellent! Perhaps working the IAX2 angle will be less of a hassle, I will go looking for one that does video now. Maybe it's time to buy an IAX2-ATA adaptor and see how well that works over the net. Nick As for the SIP logs, I start Asterisk with -c already, I did a sip debug and tried my call from the house to my remote SIP phone. YIKES!! Gunna take a bit to understand all that, but I think I did see an INVITE, and a CANCEL twice in a row and I did not hit the hang-up switch. So that might explain why no connection is made, and the called gets my voice-mail (according to my wife) -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Bruce Reeves wrote: Nick, I have done what you are talking about as far as being a provider for family members. I used an IAX softphone mainly to eliminate the need for so many holes in the firewall. And secondly because the idefisk IAX softphone allowed me to extract the zip version, configure the phone, and zip the folder up and email it to my family members. So for my mom it was simply unzip the folder and On 9/7/06, Nick Ellson [EMAIL PROTECTED] wrote: Bob, I will up the logs today, have my phone at work with me. (though the Wife and Kids are not up yet ;) Anything specific I should target? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Bob Chiodini wrote: Nick, Anything helpful in the asterisk or system logs. Try bumping up the debug and verbose levels see what shows up on the console. Weird that it would work inbound and not outbound. Bob... On Thu, 2006-09-07 at 04:48 -0700, Nick Ellson wrote: Hey all, A previous annoyance with not being able to call out to my brother on FWD from my Asterisk system had me thinking that since I have my own PBX, and that system has it's own 1-to-1 static NAT to the internet, I should be able to act as the provider for him or any of my family, and have them as local extensions of my PBX, right? So I took my laptop to work (using the X-Lite SIP softphone) and watch my ACL logs on my router for any denies to my Asterisk box. As expected udp/5060, then once that was open, a series of randomish udp/1+ requests. My phone registered, and I tried to call one of the phones behind a PAP2. Worked first shot, and just as clear and responsive as it was when I was home. But, the phones at home could not call me, they when to voice mail. I had heard that SIP doesn't survive NAT all that well, and that IAX native phones do a better job. My question is, given my description of how I am set up and what I am trying to accomplish, should I be looking at SIP or is IAX a more robust choice? (I was hoping to get video working as well, h.263 I believe it is). Nick ___ --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 -- Bruce
RE: [asterisk-users] Softphones IAX vs. SIP, remote connectivity.
You need to MAKE a sample config by configuring your phone first, then ya get a nice little .xml config file you can batch tweak. :) That's what I found out. -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Ferguson, Michael wrote: Thanks but question! In this folder I see: the original Zip file i downloaded - idefisk137.zip addressbook.conf idefisk.conf hostory.txt iaxclient.dll Idefiskmanual.htm idefisk.exe Using Wordpad, I opened addressbook.conf and idefisk.conf but saw no reference to the IP address of my Asterisk server. Where is this info included in the zip file you sent or did you folks have to do the actual config of the softphone? Thanks again From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Thursday, September 07, 2006 1:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphones IAX vs. SIP, remote connectivity. Micheal, I do this with the zip version of idefisk avaliable here : http://asteriskguru.com/tools/idefisk_windows.php I download and extract the files the run the phone and configure the settings and the speed dials, all of which is stored in the folder with the application. I then zip it up and email it with instructions to unzip and run the program. Works great on my thumb drive also. On 9/7/06, Ferguson, Michael [EMAIL PROTECTED] wrote: Bruce, How do you go about accomplishing configuring the phone, zipping it up and sending it over to your family? Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Thursday, September 07, 2006 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphones IAX vs. SIP, remote connectivity. Nick, I have done what you are talking about as far as being a provider for family members. I used an IAX softphone mainly to eliminate the need for so many holes in the firewall. And secondly because the idefisk IAX softphone allowed me to extract the zip version, configure the phone, and zip the folder up and email it to my family members. So for my mom it was simply unzip the folder and On 9/7/06, Nick Ellson [EMAIL PROTECTED] wrote: Bob, I will up the logs today, have my phone at work with me. (though the Wife and Kids are not up yet ;) Anything specific I should target? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Bob Chiodini wrote: Nick, Anything helpful in the asterisk or system logs. Try bumping up the debug and verbose levels see what shows up on the console. Weird that it would work inbound and not outbound. Bob... On Thu, 2006-09-07 at 04:48 -0700, Nick Ellson wrote: Hey all, A previous annoyance with not being able to call out to my brother on FWD from my Asterisk system had me thinking that since I have my own PBX, and that system has it's own 1-to-1 static NAT to the internet, I should be able to act as the provider for him or any of my family, and have them as local extensions of my PBX, right? So I took my laptop to work (using the X-Lite SIP softphone) and watch my ACL logs on my router for any denies to my Asterisk box. As expected udp/5060, then once that was open, a series of randomish udp/1+ requests. My phone registered, and I tried to call one of the phones behind a PAP2. Worked first shot, and just as clear and responsive as it was when I was home. But, the phones at home could not call me, they when to voice mail. I had heard that SIP doesn't survive NAT all that well, and that IAX native phones do a better job. My question is, given my description of how I am set up and what I am trying to accomplish, should I be looking at SIP or is IAX a more robust choice? (I was hoping to get video working as well, h.263 I believe it is). Nick
Re: [asterisk-users] Softphones IAX vs. SIP, remote connectivity.
HUSHshout I think it was called... -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Blake Krone wrote: Which one has video for the mac? On 9/7/06, Nick Ellson [EMAIL PROTECTED] wrote: Hello Michael, I just had both Mom and my brother up as extensions on my Asterisk pbx using IAX2, the Cubix phone for now, but I downloaded and tried several. I loke multiple lines, but a clean GUI is better for my family.. Oh yeah, it worked flawlessly :) I open one port to my server udp/4569 and that was it. I shut the rest off. For remote family, IAX2 will be what I use right now. Anybody see a Video capable version for Windows? The MAC has one, darn it. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Ferguson, Michael wrote: Hi Guys I too am trying to do exactly the same thing in being a provider for family members. My Asterisk server is on a public ip, my home is behind a Watchguard Firebox, my job is also behind a Firebox. I am using a combination of Cisco 7960, Linksys 941 and XTEN Softphone. Sometimes it works and sometimes it does not. You idea on using a IAX2 softphone appears to be what will solve my problem. Thanks very much Post more ideas. 'preciate it. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Nick Ellson Sent: Thursday, September 07, 2006 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphones IAX vs. SIP, remote connectivity. Bruce, I *just* tested the XtremePhone, IAX2 softphone. Other than trying to figure out how to get it to send proper CallerID to the other phones, it worked right off, in both directions. Excellent! Perhaps working the IAX2 angle will be less of a hassle, I will go looking for one that does video now. Maybe it's time to buy an IAX2-ATA adaptor and see how well that works over the net. Nick As for the SIP logs, I start Asterisk with -c already, I did a sip debug and tried my call from the house to my remote SIP phone. YIKES!! Gunna take a bit to understand all that, but I think I did see an INVITE, and a CANCEL twice in a row and I did not hit the hang-up switch. So that might explain why no connection is made, and the called gets my voice-mail (according to my wife) -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Bruce Reeves wrote: Nick, I have done what you are talking about as far as being a provider for family members. I used an IAX softphone mainly to eliminate the need for so many holes in the firewall. And secondly because the idefisk IAX softphone allowed me to extract the zip version, configure the phone, and zip the folder up and email it to my family members. So for my mom it was simply unzip the folder and On 9/7/06, Nick Ellson [EMAIL PROTECTED] wrote: Bob, I will up the logs today, have my phone at work with me. (though the Wife and Kids are not up yet ;) Anything specific I should target? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Bob Chiodini wrote: Nick, Anything helpful in the asterisk or system logs. Try bumping up the debug and verbose levels see what shows up on the console. Weird that it would work inbound and not outbound. Bob... On Thu, 2006-09-07 at 04:48 -0700, Nick Ellson wrote: Hey all, A previous annoyance with not being able to call out to my brother on FWD from my Asterisk system had me thinking that since I have my own PBX, and that system has it's own 1-to-1 static NAT to the internet, I should be able to act as the provider for him or any of my family, and have them as local extensions of my PBX, right? So I took my laptop to work (using the X-Lite SIP softphone) and watch my ACL logs on my router for any denies to my Asterisk box. As expected udp/5060, then once that was open, a series of randomish udp/1+ requests. My phone registered, and I tried to call one of the phones behind a PAP2. Worked first shot, and just as clear and responsive as it was when I was home. But, the phones at home could not call me, they when to voice mail. I had heard that SIP doesn't survive NAT all that well, and that IAX native
Re: [asterisk-users] Softphones IAX vs. SIP, remote connectivity.
Erm.. I mean LOUDHush and I went to look and the features did not list Video, but the phone was listed in the video softphone section of the catalog search I did. So, I see FullDisclosure with vulnerabilities in IAX2 Video, I see questions asking what happens when you go from SIP video to IAX2.. But I have yet to see a IAX2 video softphone, commercial or otherwise. Is the feature still a bit young? Or maybe to narrow a market? Just not looking forward to setting up SIP again to try out video bewteen relatives over the asterisk server. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Nick Ellson wrote: HUSHshout I think it was called... -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Blake Krone wrote: Which one has video for the mac? On 9/7/06, Nick Ellson [EMAIL PROTECTED] wrote: Hello Michael, I just had both Mom and my brother up as extensions on my Asterisk pbx using IAX2, the Cubix phone for now, but I downloaded and tried several. I loke multiple lines, but a clean GUI is better for my family.. Oh yeah, it worked flawlessly :) I open one port to my server udp/4569 and that was it. I shut the rest off. For remote family, IAX2 will be what I use right now. Anybody see a Video capable version for Windows? The MAC has one, darn it. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Ferguson, Michael wrote: Hi Guys I too am trying to do exactly the same thing in being a provider for family members. My Asterisk server is on a public ip, my home is behind a Watchguard Firebox, my job is also behind a Firebox. I am using a combination of Cisco 7960, Linksys 941 and XTEN Softphone. Sometimes it works and sometimes it does not. You idea on using a IAX2 softphone appears to be what will solve my problem. Thanks very much Post more ideas. 'preciate it. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Nick Ellson Sent: Thursday, September 07, 2006 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphones IAX vs. SIP, remote connectivity. Bruce, I *just* tested the XtremePhone, IAX2 softphone. Other than trying to figure out how to get it to send proper CallerID to the other phones, it worked right off, in both directions. Excellent! Perhaps working the IAX2 angle will be less of a hassle, I will go looking for one that does video now. Maybe it's time to buy an IAX2-ATA adaptor and see how well that works over the net. Nick As for the SIP logs, I start Asterisk with -c already, I did a sip debug and tried my call from the house to my remote SIP phone. YIKES!! Gunna take a bit to understand all that, but I think I did see an INVITE, and a CANCEL twice in a row and I did not hit the hang-up switch. So that might explain why no connection is made, and the called gets my voice-mail (according to my wife) -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Bruce Reeves wrote: Nick, I have done what you are talking about as far as being a provider for family members. I used an IAX softphone mainly to eliminate the need for so many holes in the firewall. And secondly because the idefisk IAX softphone allowed me to extract the zip version, configure the phone, and zip the folder up and email it to my family members. So for my mom it was simply unzip the folder and On 9/7/06, Nick Ellson [EMAIL PROTECTED] wrote: Bob, I will up the logs today, have my phone at work with me. (though the Wife and Kids are not up yet ;) Anything specific I should target? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 7 Sep 2006, Bob Chiodini wrote: Nick, Anything helpful in the asterisk or system logs. Try bumping up the debug and verbose levels see what shows up on the console. Weird that it would work inbound and not outbound. Bob... On Thu, 2006-09-07 at 04:48 -0700, Nick Ellson wrote: Hey all, A previous annoyance with not being able to call out to my
Re: [asterisk-users] Asterisk calling through FWD?
I thought maybe my configs would have been a good idea to post:
iax.conf:
[general]
bindport=4569
bindaddr=10.0.0.20
bandwidth=medium
disallow=lpc10
allow=gsm
jitterbuffer=no
forcejitterbuffer=no
register = 776754:snipped@iax2.fwdnet.net
allow=ulaw
tos=lowdelay
autokill=yes
[iaxfwd]
type=user
context=fromiaxfwd
auth=rsa
inkeys=freeworlddialup
Extensions.conf
[globals]
FWDNUMBER=776754
FWDCIDNAME=Nick Ellson
FWDPASSWORD=snipped
FWDRINGS=SIP/4003
FWDVMBOX=4003
[default]
include = mainmenu
exten = _393.,1,SetCIDNum(${FWDNUMBER})
exten = _393.,2,SetCallerID,${FWDCIDNAME}
exten =
_393.,3,Dial(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN:3},60,r)
exten = _393.,4,Congestion
[fromiaxfwd]
exten = ${FWDNUMBER},1,Dial(${FWDRINGS},20,r)
exten = ${FWDNUMBER},2,Voicemail,u${FWDVMBOX}
exten = ${FWDNUMBER},102,Voicemail,b${FWDVMBOX}
-
And I am not sure if something changed but now I get:
-- Executing SetCIDNum(SIP/4003-4dcc, 776754) in new stack
-- Executing SetCallerID(SIP/4003-4dcc, Nick Ellson) in new stack
-- Executing Dial(SIP/4003-4dcc,
IAX2/776754:[EMAIL PROTECTED]/XX|60|r) in new stack
-- Called 776754:[EMAIL PROTECTED]/XX
-- IAX2/fwd-gw-5 is circuit-busy
Sep 4 15:47:54 WARNING[28513]: chan_iax2.c:7013 socket_read: Call rejected by
192.246.69.186: No authority found
Sep 4 15:47:54 NOTICE[28513]: chan_iax2.c:1601 iax2_destroy: Avoiding IAX
destroy deadlock
-- Hungup 'IAX2/fwd-gw-5'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion(SIP/4003-4dcc, ) in new stack
== Spawn extension (default, 393XX, 4) exited non-zero on 'SIP/4003-4dcc'
The No Authority found I think is new? I am going to figure out how to
increase the logging, but does anyone see an obviuos boo-boo?
Nick
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Mon, 4 Sep 2006, Michael Graves wrote:
I had similar troubleat first. Try not specifying CID. As I recall FWD is
sensitive to this.
Michael
fwd: 54245
On Sun, 3 Sep 2006 22:26:35 -0700 (PDT), Nick Ellson wrote:
Hi all,
I have been researching a dialing problem I am having with FWD. I followed
their IAX2 config notes, and I can receive calls from my brother from FWD,
and all the echo tests, call me services work. But I cannot call him.
-- Executing SetCallerID(SIP/4003-9de6, Nick Ellson) in new
stack
-- Executing Dial(SIP/4003-9de6,
IAX2/776754:scrubbed@iax2.fwdnet.net/snipped|60|r) in new stack
-- Called 776754:scrubbed@iax2.fwdnet.net/snipped
-- Call accepted by 192.246.69.186 (format ulaw)
-- Format for call is ulaw
-- IAX2/192.246.69.186:4569-2 is busy
-- Hungup 'IAX2/192.246.69.186:4569-2'
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing Congestion(SIP/4003-9de6, ) in new stack
== Spawn extension (default, 393number snipped, 3) exited non-zero on
'SIP/4003-9de6'
This is pretty much just what a few others from the FWD forums have posted
with no real response.
Has any one of you also had this problem with FWD?
Nick
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
___
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Re: [asterisk-users] Asterisk calling through FWD?
Hi Michael,
I tried what you had said and then tried calling you, and it worked. Then
I called my brother and while I did not get the error, I still got the
busy message i was getting before I borked my config trying too many
ideas ;)
So, any other 6 digit FWD users willing to take a call from me? Just so I
can eliminate the call string?
My two back to back calls..
*CLI
-- Executing SetCallerID(SIP/4003-508e, Nick Ellson) in new stack
-- Executing Dial(SIP/4003-508e,
IAX2/776754:snip@iax2.fwdnet.net/5 digits|60|r) in new stack
-- Called 776754:snip@iax2.fwdnet.net/5 digits
-- Call accepted by 192.246.69.186 (format ulaw)
-- Format for call is ulaw
-- IAX2/192.246.69.186:4569-2 answered SIP/4003-508e
-- Hungup 'IAX2/192.246.69.186:4569-2'
== Spawn extension (default, 3935 digits, 2) exited non-zero on
'SIP/4003-508e'
-- Executing SetCallerID(SIP/4003-5d5e, Nick Ellson) in new stack
-- Executing Dial(SIP/4003-5d5e,
IAX2/776754:snip@iax2.fwdnet.net/6 digits|60|r) in new stack
-- Called 776754:snip@iax2.fwdnet.net/6 digits
-- Call accepted by 192.246.69.186 (format ulaw)
-- Format for call is ulaw
-- IAX2/192.246.69.186:4569-3 is busy
-- Hungup 'IAX2/192.246.69.186:4569-3'
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing Congestion(SIP/4003-5d5e, ) in new stack
== Spawn extension (default, 3936 digits, 3) exited non-zero on
'SIP/4003-5d5e'
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Mon, 4 Sep 2006, Michael Graves wrote:
Nick,
Now I remember why this doesn't work. It's the caller ID settings. The syntax
you use is older and makes two separate calls
exten = _393.,1,SetCIDNum(${FWDNUMBER})
exten = _393.,2,SetCallerID,${FWDCIDNAME}
This won't work for some reason. As soon as I changed my settings to:
exten = _393.,1,SetCallerID,${FWDCIDNAME}
exten = _393.,2,Dial(IAX2/54245:[EMAIL PROTECTED]/${EXTEN:3},60)
exten = _393.,3,Congestion
Then it worked. The SetCIDNum function broke it. I can't say why, only that I
inquired with folk at FWD who told me that it was most definitely at my end.
Feel free to call my fwd number. It rings at my desk. If I'm there I answer but
you may just get VM.
Michael
On Mon, 4 Sep 2006 16:06:42 -0700 (PDT), Nick Ellson wrote:
I thought maybe my configs would have been a good idea to post:
iax.conf:
[general]
bindport=4569
bindaddr=10.0.0.20
bandwidth=medium
disallow=lpc10
allow=gsm
jitterbuffer=no
forcejitterbuffer=no
register = 776754:snipped@iax2.fwdnet.net
allow=ulaw
tos=lowdelay
autokill=yes
[iaxfwd]
type=user
context=fromiaxfwd
auth=rsa
inkeys=freeworlddialup
Extensions.conf
[globals]
FWDNUMBER=776754
FWDCIDNAME=Nick Ellson
FWDPASSWORD=snipped
FWDRINGS=SIP/4003
FWDVMBOX=4003
[default]
include = mainmenu
exten = _393.,1,SetCIDNum(${FWDNUMBER})
exten = _393.,2,SetCallerID,${FWDCIDNAME}
exten = _393.,3,Dial(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN:3},60,r)
exten = _393.,4,Congestion
[fromiaxfwd]
exten = ${FWDNUMBER},1,Dial(${FWDRINGS},20,r)
exten = ${FWDNUMBER},2,Voicemail,u${FWDVMBOX}
exten = ${FWDNUMBER},102,Voicemail,b${FWDVMBOX}
-
And I am not sure if something changed but now I get:
-- Executing SetCIDNum(SIP/4003-4dcc, 776754) in new stack
-- Executing SetCallerID(SIP/4003-4dcc, Nick Ellson) in new stack
-- Executing Dial(SIP/4003-4dcc, IAX2/776754:[EMAIL
PROTECTED]/XX|60|r) in new stack
-- Called 776754:[EMAIL PROTECTED]/XX
-- IAX2/fwd-gw-5 is circuit-busy
Sep 4 15:47:54 WARNING[28513]: chan_iax2.c:7013 socket_read: Call rejected by
192.246.69.186: No authority found
Sep 4 15:47:54 NOTICE[28513]: chan_iax2.c:1601 iax2_destroy: Avoiding IAX
destroy deadlock
-- Hungup 'IAX2/fwd-gw-5'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion(SIP/4003-4dcc, ) in new stack
== Spawn extension (default, 393XX, 4) exited non-zero on
'SIP/4003-4dcc'
The No Authority found I think is new? I am going to figure out how to
increase the logging, but does anyone see an obviuos boo-boo?
Nick
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Mon, 4 Sep 2006, Michael Graves wrote:
I had similar troubleat first. Try not specifying CID. As I recall FWD is
sensitive to this.
Michael
On Sun, 3 Sep 2006 22:26:35 -0700 (PDT), Nick Ellson wrote:
Hi all,
I have been researching a dialing problem I am having with FWD. I followed
their IAX2 config notes, and I can receive calls from my brother from FWD,
and all the echo tests, call me services work. But I cannot call him.
-- Executing SetCallerID(SIP/4003-9de6, Nick Ellson) in new
stack
-- Executing Dial(SIP/4003-9de6,
IAX2/776754:scrubbed@iax2.fwdnet.net/snipped|60|r) in new stack
-- Called 776754:scrubbed@iax2
Re: [asterisk-users] Asterisk calling through FWD?
-- Call accepted by 192.246.69.186 (format ulaw) -- Format for call is ulaw -- IAX2/192.246.69.186:4569-3 is busy -- Hungup 'IAX2/192.246.69.186:4569-3' == Everyone is busy/congested at this time (1:1/0/0) -- Executing Congestion(SIP/4003-5d5e, ) in new stack == Spawn extension (default, 3936 digits, 3) exited non-zero on 'SIP/4003-5d5e' heh.. i got that all the time when i had the 8xx #'s routed thru FWD.. maybe 50-60% of the calls actually went thru incoming did from ipkall using fwd seems to work ok most of the time I am checking now to see if my Brother actually set up his voice mail, I wonder if that is the issue foe me tonight now that I have the dialer going out with no errors. Now the call to a 5 digit FWD number when first shot.. Ugh... Still fun though :) Nick ___ --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 calling through FWD?
Hi all, I have been researching a dialing problem I am having with FWD. I followed their IAX2 config notes, and I can receive calls from my brother from FWD, and all the echo tests, call me services work. But I cannot call him. -- Executing SetCallerID(SIP/4003-9de6, Nick Ellson) in new stack -- Executing Dial(SIP/4003-9de6, IAX2/776754:scrubbed@iax2.fwdnet.net/snipped|60|r) in new stack -- Called 776754:scrubbed@iax2.fwdnet.net/snipped -- Call accepted by 192.246.69.186 (format ulaw) -- Format for call is ulaw -- IAX2/192.246.69.186:4569-2 is busy -- Hungup 'IAX2/192.246.69.186:4569-2' == Everyone is busy/congested at this time (1:1/0/0) -- Executing Congestion(SIP/4003-9de6, ) in new stack == Spawn extension (default, 393number snipped, 3) exited non-zero on 'SIP/4003-9de6' This is pretty much just what a few others from the FWD forums have posted with no real response. Has any one of you also had this problem with FWD? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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 Linksys PAP2 ATA
Hi Corey, I spend 2 hours with REALLY bad docs on how to Unlock the PAP2 Vonage I got from Office Max. I bought it the second I saw a glimpse of an articaly that it could be turned back into an NA. Anyone want to try this? The nes ones one the shelf in my area had 3.1.3 code already, but if you put together a few seperate How-To's ya can get a really simple procedure and the files to clean out Vonage and I can say my kids LOVE the new phones in their rooms to play house with ;) (So yeah, end result was actually 30 seconds and two reboots on the unit and it's Vonage Free and happily on my Asterisk network) So now I am looking at the Linksys SPA3000 to use as a poor-man's FXO port. It looks like this is an easier task doc's wise. Anyone set this up? Nick If anyone wants the final steps I used on the PAP2 lemme know. -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Sat, 2 Sep 2006, Cory Andrews wrote: Nick - the Linksys PAP2's that you see in retail stores like Fry's, BestBuy, WalMart, etcare all locked devices. They are provisioned and locked down for use only with specific VoIP providers like Vonage, Packet8, Broadvoice, etc. If you want an unlocked, unprovisioned PAP2, you need to purchase the PAP2-NA or PAP2T-NA (Current Model). Search Google using either of these part numbers and you will find many only merchant who offer the unlocked version, which will work with Asterisk. Regards, Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 800.398.VoIP X3402 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Nick Ellson [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, September 01, 2006 10:12 PM Subject: [asterisk-users] Asterisk Linksys PAP2 ATA I was loonking for an easy off the shelf ATA to get two analog phones up on Asterisk. I am not yet ready to by a full 4 port digium card until My wife can see this work with FWD and a real phone :) I see that Fry's sells the Linksys PAP2, which appears to be a SIP adaptor? I have found no posts on it being able to log into Asterisk. Any one tried this? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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 Linksys PAP2 ATA
Hey Bob, I think the SPA31-2 is the new guy on the block. Only $10 more too mail order. $86 was the best I saw. So I have the PAP2 with two cheapy $4 wall phones mounted in the kids room, they are calling each other and my laptop.. Only issue so far is that to call one PAP2 from the other there is a 10 sec delay before the ringback/ring occurs.. and a 3 5 year old can have an entire conversation before the phone even rings. ;) Calling from my X-Lite soft phone to the PAP2 is nearly instant. But it does have my wife actually jazzed about having two more phones where she works in the house so she can join the fun.. Score! A free pass to buy more toys! Another PAP2 and a SPA3102 for me Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Sat, 2 Sep 2006, Bob Chiodini wrote: Nick, I've used a SPA3000. There seems to be a later model from Linksys, hopefully it works better. I had some severe echo problems due to my distance from the CO. The SPA3000 never could seem to compensate. The older software worked better, but it never passed muster with the wife. Went to a Digium TDM11B, no problems. There are plenty of mini-howtos on the web to set up a SPA3000. If you are close to your CO and the price is right, give it a try. I think I read somewhere that at up to 7000 feet between the CO and the SPA acceptable results are possible. I'm at about 18000 feet. Bob... Nick Ellson wrote: Hi Corey, I spend 2 hours with REALLY bad docs on how to Unlock the PAP2 Vonage I got from Office Max. I bought it the second I saw a glimpse of an articaly that it could be turned back into an NA. Anyone want to try this? The nes ones one the shelf in my area had 3.1.3 code already, but if you put together a few seperate How-To's ya can get a really simple procedure and the files to clean out Vonage and I can say my kids LOVE the new phones in their rooms to play house with ;) (So yeah, end result was actually 30 seconds and two reboots on the unit and it's Vonage Free and happily on my Asterisk network) So now I am looking at the Linksys SPA3000 to use as a poor-man's FXO port. It looks like this is an easier task doc's wise. Anyone set this up? Nick If anyone wants the final steps I used on the PAP2 lemme know. ___ --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] Blind transfer 3/4 digits
I have just noticed that my X-Lite soft phones don't dial 3-4 digit extensions without first dialing it in the display and then hitting send. So tthat is an issue with the phone you think? Ok, I'll start there for the inter digit timeout, see if there is a certain dial string lenth before it will transmit. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Sat, 2 Sep 2006, Kevin Smith wrote: Dialing a number and transferring a number are two different things. And no offense, you are not really providing a lot of details along with your problem. So you can dial the numbers but not transfer from one to the other. What does the CLI say when you try the transfer? That would provide a lot of information that could clue you in to what is going on. What type of phones are you using? Some phones have the ability to pattern match and wait for a certain number of seconds before sending the number to asterisk. For example. On our Polycom phones a user has 3 seconds (between digits) to enter in 10 digits. This could be where most of your problem is. My guess the problem lies with the Phones, not Asterisk form the information you provided. Kevin Ronald Wiplinger wrote: David Gagnon wrote: Ronald, You seem to be a little bit angry about VoIP. If so, I could give you my old Nortel system. Does this would make you happy? David David, I am not angry about VoIP, but please send my your old Nortel system ! I just do not understand why I can DIAL 601 and 6014, but not use blind transfer. Is the question too difficult? I am sure there is somewhere a switch to say, wait two seconds (as for dialing) before you assume it is a complete number. It is also strange that snom phone can do it correct, because it uses the ok key. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 2 septembre 2006 04:20 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits Anthony Rodgers wrote: With respect, the problem is with your numbering plan.. This answer is therefore totally nonsense !!! (With all respect!!!) Both answers have actually not lead to any step further, but to more messages. I use to refer to such answers as NON-ANSWERS. Please only reply if and really only if you know a solution for the problem! Thanks for your understanding. bye Ronald - again, I am not angry at all. WHERE do you see a problem in the numbering plan? I see the problem in ASTERISK, because it does not wait for the last digit!!! Where can I set that it waits for it? The beauty on voip IS that you can have different length and overlapping, bye Ronald CP On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote: I found a problem in blind transfer: I have an extension number 601 and I have an extension 6014 If I get a call on 615 (snom) and transfer to 6014 it works, since snom requires me to hit ok If I get a call on 601 and transfer to 6014, than 601 will get the busy signal and I hang up as usually with transfer. Howerver the caller get the announcements: I could not get that, What could be the problem ? bye Ronald ___ --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 --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0635-4, 2006/09/01 Tested on: 2006/9/2 ¤U¤È 03:52:00 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.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 ___ --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 Linksys PAP2 ATA
Hey Bob, Just tested the PAP2, yes a # sends right away. I am looking for why, still new at the dial plan stuff.. this is the default.. Should I be looking for a way to have the PAP2 NOT deal with dialing and let Asterisk handle it? (*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.) -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Sat, 2 Sep 2006, Bob Chiodini wrote: Nick, I know some adults that can have an entire conversation in the same amount of time. Does pressing the # key speed up dialing? If so look for a timer in the PAP config or tell the kids to press #. IIRC the spa3k had something similar, but never did much in-house dialing. $86 is a pretty good price. I paid more than that for the spa3000 6 months ago. Bob... Nick Ellson wrote: Hey Bob, I think the SPA31-2 is the new guy on the block. Only $10 more too mail order. $86 was the best I saw. So I have the PAP2 with two cheapy $4 wall phones mounted in the kids room, they are calling each other and my laptop.. Only issue so far is that to call one PAP2 from the other there is a 10 sec delay before the ringback/ring occurs.. and a 3 5 year old can have an entire conversation before the phone even rings. ;) Calling from my X-Lite soft phone to the PAP2 is nearly instant. But it does have my wife actually jazzed about having two more phones where she works in the house so she can join the fun.. Score! A free pass to buy more toys! Another PAP2 and a SPA3102 for me Nick ___ --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 Linksys PAP2 ATA
Ok, I found the Interdigit short timer (3 secs) and Interdigit long timer (sure enough, 10 secs) So, what I have seen is that when a dial plan hits a match, it fires without looking for more digits.. The interdigit short delay is in effect, but the long timer hits ya when you are trying to find that page agaain in the phone book because it fell off the counter.. ;) Maybe a dial-plan on the PAP2 can send digits direct to Asterisk.. not sure. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Sat, 2 Sep 2006, Bob Chiodini wrote: Nick, I know some adults that can have an entire conversation in the same amount of time. Does pressing the # key speed up dialing? If so look for a timer in the PAP config or tell the kids to press #. IIRC the spa3k had something similar, but never did much in-house dialing. $86 is a pretty good price. I paid more than that for the spa3000 6 months ago. Bob... Nick Ellson wrote: Hey Bob, I think the SPA31-2 is the new guy on the block. Only $10 more too mail order. $86 was the best I saw. So I have the PAP2 with two cheapy $4 wall phones mounted in the kids room, they are calling each other and my laptop.. Only issue so far is that to call one PAP2 from the other there is a 10 sec delay before the ringback/ring occurs.. and a 3 5 year old can have an entire conversation before the phone even rings. ;) Calling from my X-Lite soft phone to the PAP2 is nearly instant. But it does have my wife actually jazzed about having two more phones where she works in the house so she can join the fun.. Score! A free pass to buy more toys! Another PAP2 and a SPA3102 for me Nick ___ --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 Linksys PAP2 ATA
Hi Tim, The dial plan trick worked great. Added |40[01]x| to my plan and 4000-4019 connect instantly from the PAP2 :) Added it to my X-Lite as well, and worked there too. Thanks! -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Sat, 2 Sep 2006, Tim St. Pierre wrote: You have to set in in the PAP2. When using SIP, it has to send an invite with the number it wants to be connected to. The Sipura has to know a complete number to send - it can't send it in pieces. You need to make the dialplan in the Sipura match what you have programmed in Asterisk. Ie. My extensions are 51XX, and 52XX, so in the Sipura dialplan, I added 5[12]XX - this will match any of my extensions, and complete the call. This can be a problem if you use direct 10 digit dialing, and dial to an area code beginning with 51 or 52. You could get around this (if it's a likely issue) by prefixing a 9 to the 10 digit patterns, or inserting a . (I think) to make it wait for another digit. -Tim On September 2, 2006 20:43, Nick Ellson wrote: Hey Bob, Just tested the PAP2, yes a # sends right away. I am looking for why, still new at the dial plan stuff.. this is the default.. Should I be looking for a way to have the PAP2 NOT deal with dialing and let Asterisk handle it? (*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.) -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [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
Solved: sigh Re: [asterisk-users] MOH help needed with fresh install
I'm sorry all... But I knew it would take me asking the questions before the answer would present itself.. I found a reference to the version of mpg123 needing to be r not s and that was my problem. Had to load mpg123 from src and fix a few typos in the makefile, but it plays very nice now. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Thu, 31 Aug 2006, Nick Ellson wrote: I have been reading the archives through google and I see mention several times that the MOH with the default class (set to quietmp3) still plays the 3 default mp3's at seriously high volume. I read that the most common issue with MOH is the timing and for non card users that ztdummy should be loaded. I did load ztdummy. (this is a Gentoo 2.6.17 build on a Intel Server Board with 2 dual core pentium 4's) The multi processor note along with the Kernel 2.6 notes and addendums say that ztdummy no longer needs, nor can I use USB kernel modules and that ztdummy will be using a RTC. I can't tell from the blaring din from my softphones if the music is out of timing, or just plain too loud. Any Gentoo Portage users running Asterisk MOH that can help me determine what I can do? I have tried using MPG123 with no difference (no errors on the console suggesting that did try /usr/bin/mpg123 ) Thanks in advance for any help, Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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] Asterisk Linksys PAP2 ATA
I was loonking for an easy off the shelf ATA to get two analog phones up on Asterisk. I am not yet ready to by a full 4 port digium card until My wife can see this work with FWD and a real phone :) I see that Fry's sells the Linksys PAP2, which appears to be a SIP adaptor? I have found no posts on it being able to log into Asterisk. Any one tried this? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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] MOH help needed with fresh install
I have been reading the archives through google and I see mention several times that the MOH with the default class (set to quietmp3) still plays the 3 default mp3's at seriously high volume. I read that the most common issue with MOH is the timing and for non card users that ztdummy should be loaded. I did load ztdummy. (this is a Gentoo 2.6.17 build on a Intel Server Board with 2 dual core pentium 4's) The multi processor note along with the Kernel 2.6 notes and addendums say that ztdummy no longer needs, nor can I use USB kernel modules and that ztdummy will be using a RTC. I can't tell from the blaring din from my softphones if the music is out of timing, or just plain too loud. Any Gentoo Portage users running Asterisk MOH that can help me determine what I can do? I have tried using MPG123 with no difference (no errors on the console suggesting that did try /usr/bin/mpg123 ) Thanks in advance for any help, Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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
