Re: [Asterisk-Users] Teliax IAX problems -- Asterisk doesn't see answer
I think I ran in to this problem a while back as well. I'm also running a CVS version of Asterisk. I talked to David and he switched me to SIP from their gateway to their Asterisk proxy which solved the issue. On Mon, 2005-10-17 at 22:05 -0500, Rob Fugina wrote: On 10/17/05, Rich Adamson [EMAIL PROTECTED] wrote: I've been trying to diagnose the same problem with teliax, and it seems to be a jitterbuffer problem. Since turning it off, we've not had a problem. My guess is that teliax servers are not current code, or they've modified the code for some reason. The tech that I corresponded with suggested 'gsm' was the culprit, but I had the same issues with g729, ilbc, etc. Try jitterbuffer=no in the appropriate section of your iax.conf and restart asterisk. Well, that did it. Here's hoping that teliax sees the light soon... If having no jitter buffer becomes a problem, I won't hesitate to route my calls through somebody else... Thanks for the help1 Rob ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Sipura 3000
I tried this with CVS Head as of today and I get:
-- Executing Dial(SIP/705-0c37, SIP/g10/768||T) in new stack
Jun 21 11:21:09 WARNING[24473]: chan_sip.c:1742 create_addr: No such
host: g10
Jun 21 11:21:09 NOTICE[24473]: app_dial.c:977 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
== Everyone is busy/congested at this time
(1:0/0/1)
Also, do you use group=, or callgroup= in sip.conf for CVS head, do
you know?
The reason why I decided to give this a try is I think CVS head broke
how SIP unavailable vs. busy replies are handled. When the first SPA-3K
is inuse, it reports Service Unavailable, which Asterisk now thinks is
busy, and my custom script no longer tries the next SPA-3K because of
it. :(
-- Executing Dial(SIP/705-bd2f, SIP/500/768|20|T) in new
stack
-- Called 500/768
-- Got SIP response 503 Service Unavailable back from
192.168.1.190
-- SIP/500-4b1d is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
How does a script know the difference between an actual busy signal when
dialing out, compared to the SPA3K just being in-use?
On Fri, 2005-06-17 at 06:51 -0600, Rich Adamson wrote:
If I have multiple Sipura 3000 device how can I dial out properly? I
can receive call without any problem and that's working really well.
Caller ID is shown and when someone call he get's the welcome message
the same way I have it configure with the X100P card. I don't seem to
have any echo problem with the Sipura 3000 (but I do with X100P cards)
My main concern is for outgoing call. Can I create a group like I did
in zaptel for Sipura 3000 device? Like if the FXO port of the first
Sipura 3000 is busy it will switch to the second and if second is
also busy then to the third one, and all the way until all the Sipura
3000 are in used before saying that there's no line left?
The only configs I saw on the wiki were with 1 Sipura 3000 but I
couldn't find anything on how to setup multiple Sipura 3000 devices
in asterisk for outgoing calls.
If I understood what you're trying to accomplish, try something like
this.
In sip.conf, define each spa3k something like this:
[3021] ; PSTN side of SPA3000
type=friend
host=dynamic
username=3021
secret=myspa1
context=from-sip
canreinvite=no
group=17
pickupgroup=2
deny=0.0.0.0/0.0.0.0
permit=216.21.194.0/255.255.255.0
and be sure to include group=17 in each spa3k definition.
Then in extensions.conf, use a dial statement like this:
exten = _9.,1,Dial(SIP/g17/${EXTEN:1}
Pick whatever group number you want instead of =17 in the above.
If I recall correctly, you can have up to 32 groups (or something
like that).
When the spa3k first hit the market, someone recommended using port
5060 and 5061 in the spa definitions. I have never had to do that
with any spa3k. Rather, I leave both the fxs and fxo definitions
in the spa3k default to 5060 and use different userid secrets
for the fxs and fxo definitions. The above definition for x3021
is the actual one in use right now, which functions correctly.
I've added the group=17 in the above as an example; I don't
actually use that right now (for different reasons).
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Re: [Asterisk-Users] Recommended Network Latency
Define happily? Jitter is obviously important, but latency is too. For day-to-day business calls, 250ms is a little high. Both parties will definitely notice it. In my experience you will find yourself talking over one another quite often. Even with 100ms this continues to happen from time to time. There is no doubt that it can be done, but if your latency ever exceeds 300ms, people tend to get frustrated and it could really start to cause problems. Once your agents realize there is a 250ms latency, and get used to it, it might not be so bad. But personally, it wouldn't be acceptable to me. On Fri, 2005-05-27 at 16:25 +0100, Tony Hoyle wrote: Waldo Rubinstein wrote: I'm planning on setting up some remote agents and before doing so, I did some simple PING tests to measure latency. The average latency I got was 250ms. Does anyone have experience in terms of quality of calls when there is such high latency? Can anyone comment? Latency isn't the issue - you could happily carry on a call over a 2000ms latency.. satellite links can introduce this easily. Your problem is jitter (ping stability). Some software will calculate this for you - eg. mtr gives the standard deviation for its ping history (mine hovers around the 0.4 mark over 12 hops which is really stable AFAIK). You can reduce jitter using good QoS, but it's better to have it as low as possible to start with. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Overheard conversation. Comments please !
I've actually ran in to something similar. Though luckily it hasn't happened in a while. I've gotten at least 4 complaints of an in-progress call all of a sudden being able to hear (but not speak to) another conversation that is in progress. I've never been able to track it down, but I'm running 1.0.6 WITHOUT chan_spy, and SPA-2000's with analog phones. I agree, it is very disturbing, but I'm not 100% sure it is Asterisk related. On Thu, 2005-04-14 at 20:13 +0100, Asterisk wrote: I've just been informed of a disturbing event at our call center. We run 20+ agents taking inbound and outbound calls through the queue system, and use chan_spy to monitor ongoing conversations. My supervisor received a phone call from another call center (nothing to do with us, in fact they are 200 miles away) stating that they overheard a conversation between ourselves and one of our customers that we were speaking to at the time. He was able to give reference numbers and names, (and financial circumstances) so he obviously did hear this conversation. We are running CVS head as of 10 days ago, using TE410p on 32channel ISDN primary line. Has anyone else ever heard of something like this happening ? My boss is going apeshit talking about the DPA (data protection act) and wants answers like yesterday. Quite frankly, I have no idea on where to start to look for something like this. Julian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sipura 3000 - Call Leg/Transaction Does Not Exist - only happens sometimes
Yes, I used to see this problem from time to time as well. I'm not 100% sure, but I think it was caused by someone dialing in to my Asterisk box, then hanging up at the menu and the SPA3K not detecting the hangup, or at least not sending it to Asterisk. But when either another call comes in, or a outgoing call is made, the SPA3K happily accepts it and works as normal, leaving Asterisk to think there are two calls going through the SPA3K, when it fact only one is, thus the Call Leg/Transaction Does Not Exist message. The way I resolved the issue was to put a absolute timeout in my menu (which I should have always had anyways) and upgraded the firmware of the SPA3K. Since then I haven't noticed the issue, but it hasn't negatively affected anything so I'm happy with it. On Sun, 2005-04-10 at 19:47 +0100, Chris Stenton wrote: Here is my problem I have an incoming call on the FXO port of the Sipura 3000 this goes to my asterisk box (running CVS - head) and then by default goes back to the sipura 3000 on the fxs tel port. Every 5 or so calls when the call is picked up on the fxs port telephone I just get beeping and on the asterisk console it comes up with Got SIP response 481 Call Leg/Transaction Does Not Exist back from xx.xx.xx.xx Anyone else had this problem? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Liveviop problem
Speaking of LiveVOIP and there west coast server, is anyone actually using it? I was told by there sales people several months ago that they have a west coast server (and gave me its IP to check routes to it) but it wasn't quite ready yet. Since I signed up though they pretty much stopped replying to emails altogether. On Wed, 2005-04-06 at 13:35 -0700, Wiley Siler wrote: Run this from the CLI... iax2 show registry Do you see an entry that matches your LiveVoIP server IP (east or west coast) and is it registered? W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrejus Stavickis Sent: Wednesday, April 06, 2005 1:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Liveviop problem Hi, I'm just curious if someone had/has a problem with livevoip. When I try to make an outgoing call, I receive: -- Called username:secret@217.160.244.186/x037378896 Apr 2 16:47:21 WARNING[10153]: chan_iax2.c:5546 socket_read: Call rejected by 217.160.244.186: No authority found The username,secret and first 5 digits of the phone is modified in this log. I tried to call Livevoip, they said send us an e-mail and I did, but no response whatsoever for about a week now. Sincerely, --Andy x6722 Outsourcing is akin to making a skyscraper taller by taking material from its lower floors. --Byron Katz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Liveviop problem
217.160.244.186 is the IP address (why they don't use hostnames is beyond me) I'm currently using to make outgoing calls with but that server is definitely not on the west coast. Here are the last few hops I get from a traceroute: 9 chcgildtgr00.bb.telus.com (154.11.11.18) 93.696 ms 83.336 ms 83.615 ms 10 Telia.chcgildtgr00.bb.telus.com (154.11.3.82) 84.364 ms 94.895 ms 83.646 ms 11 nyk-bb1-pos0-3-0.telia.net (213.248.80.154) 101.400 ms 100.787 ms 101.550 ms 12 nyk-i3-geth1-1.telia.net (213.248.82.146) 101.380 ms 101.349 ms 101.834 ms 13 schlund-100973-nyk-i3.c.telia.net (213.248.83.26) 101.289 ms 101.527 ms 101.832 ms 14 a0nycc1.gw-core-a.nyc.schlund.net (217.160.229.72) 102.157 ms 101.023 ms 101.763 ms 15 217.160.244.186 (217.160.244.186) 102.099 ms 101.485 ms 102.079 ms Looks like its in New York City somewhere to me. Which is on the exact opposite end of the country from me. On the other hand, LiveVOIP sales gave me this IP: 69.90.232.186 when I signed up stating it is there west coast server located in Seattle. A traceroute to that is only 8 hops and 40ms away from me. However if I simply switch IPs in my .conf file, outgoing calls fail. On Thu, 2005-04-07 at 08:26 -0700, Wiley Siler wrote: I use the West Coast server. It is located in San Jose. IP Address: 217.160.244.186 As to the replies, I usually get good replies by sending my questions to [EMAIL PROTECTED] Have also had good responses from [EMAIL PROTECTED] BTW - When I signed up I got an email that had all of my connection info in it. Took me a little bit to figure out and I had to request help several times. If you are still ahving trouble with this, contact me off list and we can Go through the setup registration. Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Benoit Sent: Thursday, April 07, 2005 7:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Liveviop problem Speaking of LiveVOIP and there west coast server, is anyone actually using it? I was told by there sales people several months ago that they have a west coast server (and gave me its IP to check routes to it) but it wasn't quite ready yet. Since I signed up though they pretty much stopped replying to emails altogether. On Wed, 2005-04-06 at 13:35 -0700, Wiley Siler wrote: Run this from the CLI... iax2 show registry Do you see an entry that matches your LiveVoIP server IP (east or west coast) and is it registered? W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrejus Stavickis Sent: Wednesday, April 06, 2005 1:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Liveviop problem Hi, I'm just curious if someone had/has a problem with livevoip. When I try to make an outgoing call, I receive: -- Called username:secret@217.160.244.186/x037378896 Apr 2 16:47:21 WARNING[10153]: chan_iax2.c:5546 socket_read: Call rejected by 217.160.244.186: No authority found The username,secret and first 5 digits of the phone is modified in this log. I tried to call Livevoip, they said send us an e-mail and I did, but no response whatsoever for about a week now. Sincerely, --Andy x6722 Outsourcing is akin to making a skyscraper taller by taking material from its lower floors. --Byron Katz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone, VoIPJet, circuit (fast) busy question
If I recall correctly Fast Busy basically means the destination number is not busy (regular busy) but your provider most likely is either over loaded, or has some other issues. I've been getting busy signals with Nufone pretty regularly over the last few days, and there email support is not responding as usual. There front page also says they are no longer accepting new customers due to system upgrades, maybe that has something to do with it. Who knows... On Fri, 2005-04-01 at 09:09 +0400, Jean-Michel Hiver wrote: I've noticed that nufone returns 'circuit busy' messages FAST (when it does) while this tends to take a while with VoIPJet. I've also seen 'circuit fast busy' message - what is the difference between the two? Thanks, Jean-Michel. -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Livevoip still no DTMF?
I have a LiveVOIP toll-free DID and DTMF works fine. I haven't had any complaints so far. Though, I do get a dead air (call doesn't reach my asterisk box) when I dial the number on rare occasions. A simple re-dial and it usually works. Ring back also doesn't work, but I got around that by using MOH to play a ringing sound, so its not a big deal for me. I don't specify any DTMF mode in my iax.conf either. On Thu, 2005-03-31 at 10:33 -0800, Brian Litzinger wrote: I read in the archives a number of discussions about livevoip, DID, and DTMF not working. However, no resolutions. I just setup a livevoip DID and indeed the DTMF does not work. The same asterisk context works via broadvoice and via direct dialing in to the asterisk server via SIP. Just no DTMF with calls via livevoip. I'm running Asterisk CVS-v1-0-03/06/05-23:15:12 -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No ringback over IAX - LiveVoip
Using the 'r' option doesn't seem to make a difference for me. To work around this issue, I'm using the 'm' option to play MOH. However it just occurred to me, does anyone have a .mp3 recording of a phone ringing for 30+ seconds? I could play that instead of regular music and it would probably work not too bad. On Fri, 2005-03-11 at 21:48 +0100, Peter Svensson wrote: On Fri, 11 Mar 2005, Wiley Siler wrote: I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Perhaps they expect you to provide audioable progress information inband on the reverse channel? I.e. use the 'r' option on the dial command etc. That is the way some isdn lines etc work. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommendation for dialplan in case of DDoS atta cks?
On Mon, 2005-02-28 at 14:20 -0600, Kristian Kielhofner wrote: His suggestion was basically the same thing, only in mine you would dial an extension to activate DDOS mode instead of running the database put from the command line. How about monitoring your hosts with iax2/sip show peers and parsing that output with a cron job? The ping thing looks like it would be more of a problem than anything else. OR you could run Snort and have it detect the DDOS somehow... Not a snort expert, but it has to be doable. Are these inbound or outbound calls? (both?) I am pretty confused about all of this... Isn't this what qualify=latency (ie: qualify=200) in your iax/sip.conf files is for? If the latency exceeds 200ms, Asterisk will automatically disable the link, and you can easily use a fail-over method in your dialplan. I think something like isChanAvail() might work for that. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spandsp 0.0.2pre6 configure fails sanity check.
I get a C++ preprocessor sanity check error when trying to configure spandsp 0.0.2pre6: checking for dlfcn.h... yes checking how to run the C++ preprocessor... /lib/cpp configure: error: C++ preprocessor /lib/cpp fails sanity check See `config.log' for more details. Config.log: -- configure: failed program was: | /* confdefs.h. */ | | #define PACKAGE_NAME | #define PACKAGE_TARNAME | #define PACKAGE_VERSION | #define PACKAGE_STRING | #define PACKAGE_BUGREPORT | #define PACKAGE spandsp | #define VERSION 0.0.2 | #ifdef __cplusplus | void exit (int); | #endif | #define STDC_HEADERS 1 | #define HAVE_SYS_TYPES_H 1 | #define HAVE_SYS_STAT_H 1 | #define HAVE_STDLIB_H 1 | #define HAVE_STRING_H 1 | #define HAVE_MEMORY_H 1 | #define HAVE_STRINGS_H 1 | #define HAVE_INTTYPES_H 1 | #define HAVE_STDINT_H 1 | #define HAVE_UNISTD_H 1 | #define HAVE_DLFCN_H 1 | /* end confdefs.h. */ | #ifdef __STDC__ | # include limits.h | #else | # include assert.h | #endif |Syntax error configure:5144: error: C++ preprocessor /lib/cpp fails sanity check See `config.log' for more details. I'm running Mandrake 10.0, with the follow gcc packages installed: gcc-3.3.2-9mdk gcc-cpp-3.3.2-9mdk Any ideas how to fix this? Thanks. -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 insists on not using port 4569??
For some reason, starting just today, 1 out 3 of my asterisk servers is having issues calling 1 other server. The only issue I see is that when it registers with the problem server it is using port 1027, not 4569. ie: Registered to 'Server 1', who sees us as 'Server 2':1027 Server 1 then proceeds to timeout trying to register with Server 2. The way I have each server registering with one another is this: Server 1- Server 2 - Server 3 *Server 1 is behind NAT, with port 4569 forwarded to it of course. Server 2- Server 1 - Server 3 Server 3- Server 1 - Server 2 Now: Server 1 can call Server 2, and 3 Server 2 can only call Server 3 Server 3 can call Server 1, and 2 So, even though Server 3 registers with Server 1, on port 4569, it can call it fine (So NAT is obviously working). However Server 2 insists on registering to Server 1 with port 1027, and it cannot call server 1. Server 1: *CLI iax2 show registry Host UsernamePerceived Refresh State 'Server 2':4569 west Unregistered 60 Timeout 'Server 3':4569 west 'Server 1':4569 60 Registered Server 2: *CLI iax2 show registry Host UsernamePerceived Refresh State 'Server 1':4569 central 'Server 2':1027 60 Registered 'Server 3':4569 central 'Server 2':4569 60 Registered Server 3: --- *CLI iax2 show registry Host UsernamePerceived Refresh State 'Server 2':4569 east 'Server 3':4569 60 Registered 'Server 1':4569 east 'Server 3':4569 60 Registered This just started happening today, nothing that I am aware of changed. All servers are running stock Asterisk v1.0.2. Any ideas what would cause Server 2 to just start using some weird port? IAX2 debug info output on Server 1: -- Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 000 Type: IAX Subclass: LAGRQ Timestamp: 30014ms SCall: 4 DCall: 0 ['Server 2':4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 9ms SCall: 6 DCall: 0 ['Server 2':1027] USERNAME: central REFRESH : 60 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00010ms SCall: 8 DCall: 6 ['Server 2':1027] AUTHMETHODS : 3 CHALLENGE : 182440343 USERNAME: central Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00074ms SCall: 6 DCall: 8 ['Server 2':1027] USERNAME: central REFRESH : 60 MD5 RESULT : 4c3666e4e091836fb626f79923fed5dc Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00058ms SCall: 8 DCall: 6 ['Server 2':1027] USERNAME: central DATE TIME : 160803087 REFRESH : 60 APPARENT ADDRES : IPV4 'Server 2':1027 Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00058ms SCall: 6 DCall: 8 ['Server 2':1027] Tx-Frame Retry[001] -- OSeqno: 004 ISeqno: 000 Type: IAX Subclass: LAGRQ Timestamp: 30014ms SCall: 4 DCall: 0 ['Server 2':4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00014ms SCall: 1 DCall: 0 ['Server 2':4569] USERNAME: westbank REFRESH : 60 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 3ms SCall: 3 DCall: 0 ['Server 2':1027] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 3ms SCall: 1 DCall: 3 ['Server 2':1027] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 3 DCall: 1 ['Server 2':1027] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-2000 Dropped calls
Don't run LISa on the same network as any SPA-2000 or SPA-3000. (maybe even any Sipura device?) On Wed, 2004-12-01 at 14:57 -0700, Bryan Mannos wrote: What was the fix? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-2000 Dropped calls
Not sure. The only issue I was having was the devices themselves rebooting at very predictable times. Not just one device either, all 4 that I have on my network would reboot all at once. Only one way to find out though. Disable lisa and see if it helps? On Wed, 2004-12-01 at 18:35 -0600, Brian Roy wrote: On Wed, 01 Dec 2004 16:01:16 -0800, Mike Benoit [EMAIL PROTECTED] wrote: Don't run LISa on the same network as any SPA-2000 or SPA-3000. (maybe even any Sipura device?) I have a problem with mine locking up, but not while talking. When it sits idle for a period of time I come back to it and it's dead. No ping, nothing. I have to unplug the power to get it back to life. Could this be symtompatic of the lisa thing? I do have lisa on a lot of my *nix boxes, but none of them appear to have it running (ala ps -aux |grep lisa). -Chuji ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-2000 Dropped calls
=8445985,realm=asterisk,nonce=47af5efb,uri= sip:[EMAIL PROTECTED],algorithm=MD5,response=1a025630f2e7b5a94fa227ae79fef7a 1 Contact: 8445985 sip:[EMAIL PROTECTED]:5060;expires= User-Agent: Sipura/SPA2000-2.0.11(g) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura 13 headers, 0 lines Using latest request as basis request Sending to 192.168.0.20 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK-46bc8b4 From: 8445985 sip:[EMAIL PROTECTED];tag=c864004bd9b6bbbdo0 To: 8445985 sip:[EMAIL PROTECTED];tag=as174ef08c Call-ID: [EMAIL PROTECTED] Seq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.20:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK-46bc8b4 From: 8445985 sip:[EMAIL PROTECTED];tag=c864004bd9b6bbbdo0 To: 8445985 sip:[EMAIL PROTECTED];tag=as174ef08c Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:[EMAIL PROTECTED]:5060;expires=3600 Date: Tue, 30 Nov 2004 04:29:03 GMT Content-Length: 0 to 192.168.0.20:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Sip read: REGISTER sip:192.168.0.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-572e66a From: 8445983 sip:[EMAIL PROTECTED];tag=342babdb37a0856do1 To: 8445983 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER Max-Forwards: 70 Authorization: Digest username=8445983,realm=asterisk,nonce=5a8a4fbd,uri= sip:[EMAIL PROTECTED],algorithm=MD5,response=31e27e17f7a0f434cc636ce16a3e48e 9 Contact: 8445983 sip:[EMAIL PROTECTED]:5061;expires=3600 User-Agent: Sipura/SPA2000-2.0.11(g) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura 13 headers, 0 lines Using latest request as basis request Sending to 192.168.0.20 : 5061 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-572e66a From: 8445983 sip:[EMAIL PROTECTED];tag=342babdb37a0856do1 To: 8445983 sip:[EMAIL PROTECTED];tag=as7ec12bce Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.20:5061 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-572e66a From: 8445983 sip:[EMAIL PROTECTED];tag=342babdb37a0856do1 To: 8445983 sip:[EMAIL PROTECTED];tag=as7ec12bce Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:[EMAIL PROTECTED]:5061;expires=3600 Date: Tue, 30 Nov 2004 04:29:03 GMT Content-Length: 0 to 192.168.0.20:5061 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED]:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK40a138cf From: asterisk sip:[EMAIL PROTECTED];tag=as40e64aad To: sip:[EMAIL PROTECTED]:5061 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 Messages-Waiting: no Voice-Message: 0/1 (no NAT) to 192.168.0.20:5061 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms pbx*CLI Sip read: SIP/2.0 200 OK To: sip:[EMAIL PROTECTED]:5061;tag=a2f4e6f3d386c535i1 From: asterisk sip:[EMAIL PROTECTED];tag=as40e64aad Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK40a138cf Server: Sipura/SPA2000-2.0.11(g) Content-Length: 0 8 headers, 0 lines Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' pbx*CLI Any ideas would be of GREAT help Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-2000 Dropped calls
Do you by chance have another Linux box on the same network? One that could be running the LISa daemon (network neighborhood browsers), which often gets installed with Mandrake or KDE. If you do, disable the daemon, the reboots should stop immediately. At least they did for me, and one other person who originally discovered the issue. Let me know if this fixes the problem, and also let Sipura support know that this bug in there firmware has affected another person. Hopefully they'll get around to fixing it sooner. On Tue, 2004-11-30 at 19:27 -0600, Tim Lewis wrote: I am seeing the same thing here. A bad SPA? -- Tim Schacher [EMAIL PROTECTED] 218-844-5985 On Tue, 2004-11-30 at 12:31, Mike Benoit wrote: This sounds similar to the issue I have been dealing with for the past several months. Are the calls just being dropped, or is the SPA rebooting itself? One way to find out is to setup a syslog server, and configure the SPA to send all its debug output to that (make sure you increase the debug level to highest as well). Monitor it for a few hours, and see if you get messages like this: Nov 29 08:20:56 192.168.1.185 System started: [EMAIL PROTECTED], reboot reason:H7372014f Nov 29 08:21:04 192.168.1.190 System started: [EMAIL PROTECTED], reboot reason:H0 Nov 29 08:21:04 192.168.1.190 System started: [EMAIL PROTECTED], reboot reason:H0 Nov 29 08:21:04 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Nov 29 08:21:04 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 If you do, let me know, as I can probably help you out. On Mon, 2004-11-29 at 22:48 -0600, Tim Lewis wrote: Been having a problem with my two Sipura 2000's dropping calls from the SPA-2000 side. Seems the calls are dropped right before the Next Registration time. Calls drop about ever 60 minutes or so. I have dialed from one port to the other and let it sit. After about 60 minutes or so the calls get dropped. System details are below Asterisk ver. CVS-HEAD-11/27/04-23:42:45 RHEL 3 10/100 LAN Static IP address SPA-2000 Software Ver. 2.0.11(g) SPA-2000 Hardware Ver. 2.0.1(563f) sip.conf [8445983] type=friend username=8445983 secret=mypassword nat=0 context=toll-access host=dynamic canreinvite=no reinvite=no allow=ulaw ;allow=alaw mailbox=5983 output from sip debug Sip read: REGISTER sip:192.168.0.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK-b1938413 From: 8445985 sip:[EMAIL PROTECTED];tag=c864004bd9b6bbbdo0 To: 8445985 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Max-Forwards: 70 Contact: 8445985 sip:[EMAIL PROTECTED]:5060;expires= User-Agent: Sipura/SPA2000-2.0.11(g) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura 12 headers, 0 lines Using latest request as basis request Sending to 192.168.0.20 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK-b1938413 From: 8445985 sip:[EMAIL PROTECTED];tag=c864004bd9b6bbbdo0 To: 8445985 sip:[EMAIL PROTECTED];tag=as174ef08c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.20:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK-b1938413 From: 8445985 sip:[EMAIL PROTECTED];tag=c864004bd9b6bbbdo0 To: 8445985 sip:[EMAIL PROTECTED];tag=as174ef08c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=47af5efb Content-Length: 0 to 192.168.0.20:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms pbx*CLI Sip read: REGISTER sip:192.168.0.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-9723ca95 From: 8445983 sip:[EMAIL PROTECTED];tag=342babdb37a0856do1 To: 8445983 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Max-Forwards: 70 Contact: 8445983 sip:[EMAIL PROTECTED]:5061;expires=3600 User-Agent: Sipura/SPA2000-2.0.11(g) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura 12 headers, 0 lines Using latest request as basis request Sending to 192.168.0.20 : 5061 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-9723ca95 From: 8445983 sip:[EMAIL PROTECTED];tag=342babdb37a0856do1 To: 8445983 sip:[EMAIL PROTECTED];tag=as7ec12bce Call-ID: [EMAIL PROTECTED] CSeq: 1
RE: [Asterisk-Users] SPA-2000's rebooting every hour or so...
How often was it rebooting before, do you know? Mine seem to be rebooting almost exactly 1hour apart, which is the registration expire time. I've just recently changed it to 6hrs, so I'll see if that makes a difference. On Fri, 2004-10-15 at 08:26 +0200, [EMAIL PROTECTED] wrote: Hi I have also have the Sipura rebooting itself. I changed the codec from G723.1 to G729 and this seems to have helped fix the problem. I have the latest firmware...2.0.10(e) I think..?? Hope this helpsstrange stuff though. regards Clive On 14 Oct 2004 at 14:48, Mike Benoit wrote: I thought it originally started happening after a firmware upgrade to 2.0.10e, so I downgraded to 2.0.10d, and the problem continued. I'm in the process of moving them to a cooler place and putting a fan on them just to rule out overheating, which I've heard can be a problem. On Thu, 2004-10-14 at 12:36 -0700, Dooz Owings wrote: try to run a firmware update on one and see if it works, just a guess. What all have you tried ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mike Benoit Sent: Thursday, October 14, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SPA-2000's rebooting every hour or so... I realize this is slightly off-topic here, but I know quite a few people on this list use Sipura products. Has anyone else experienced the same rebooting problem I'am? I have about 8 SPA-2000's and about half of them just started rebooting 4-8times/day in the last month or so. (they used to be rock solid) I already emailed Sipura support, but they seem to be on strike as of late. Here is the debug output from just one of the devices: (I've trimmed it for size, it happens more often than what is shown) Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 21:34:14 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:20:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 16:44:57 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 16:44:57 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 18:01:03 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 18:01:03 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 19:21:06 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 19:21:06 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 22:01:05 192.168.1.189 System started: [EMAIL
RE: [Asterisk-Users] SPA-2000's rebooting every hour or so...
I'm using the ulaw codecs, and checking again, I just realized we have one SPA-3000 in the mix behaving exactly like the SPA-2000's. Changing the registration expire time to 6hrs didn't seem to make any noticeable difference unfortunately. On Fri, 2004-10-15 at 09:42 +0200, [EMAIL PROTECTED] wrote: Hi Mine used to reboot on every call Clive On 15 Oct 2004 at 0:15, Mike Benoit wrote: How often was it rebooting before, do you know? Mine seem to be rebooting almost exactly 1hour apart, which is the registration expire time. I've just recently changed it to 6hrs, so I'll see if that makes a difference. On Fri, 2004-10-15 at 08:26 +0200, [EMAIL PROTECTED] wrote: Hi I have also have the Sipura rebooting itself. I changed the codec from G723.1 to G729 and this seems to have helped fix the problem. I have the latest firmware...2.0.10(e) I think..?? Hope this helpsstrange stuff though. regards Clive On 14 Oct 2004 at 14:48, Mike Benoit wrote: I thought it originally started happening after a firmware upgrade to 2.0.10e, so I downgraded to 2.0.10d, and the problem continued. I'm in the process of moving them to a cooler place and putting a fan on them just to rule out overheating, which I've heard can be a problem. On Thu, 2004-10-14 at 12:36 -0700, Dooz Owings wrote: try to run a firmware update on one and see if it works, just a guess. What all have you tried ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mike Benoit Sent: Thursday, October 14, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SPA-2000's rebooting every hour or so... I realize this is slightly off-topic here, but I know quite a few people on this list use Sipura products. Has anyone else experienced the same rebooting problem I'am? I have about 8 SPA-2000's and about half of them just started rebooting 4-8times/day in the last month or so. (they used to be rock solid) I already emailed Sipura support, but they seem to be on strike as of late. Here is the debug output from just one of the devices: (I've trimmed it for size, it happens more often than what is shown) Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 21:34:14 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:20:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL
[Asterisk-Users] SPA-2000's rebooting every hour or so...
: [EMAIL PROTECTED], reboot reason:H0 -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA-2000's rebooting every hour or so...
I thought it originally started happening after a firmware upgrade to 2.0.10e, so I downgraded to 2.0.10d, and the problem continued. I'm in the process of moving them to a cooler place and putting a fan on them just to rule out overheating, which I've heard can be a problem. On Thu, 2004-10-14 at 12:36 -0700, Dooz Owings wrote: try to run a firmware update on one and see if it works, just a guess. What all have you tried ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mike Benoit Sent: Thursday, October 14, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SPA-2000's rebooting every hour or so... I realize this is slightly off-topic here, but I know quite a few people on this list use Sipura products. Has anyone else experienced the same rebooting problem I'am? I have about 8 SPA-2000's and about half of them just started rebooting 4-8times/day in the last month or so. (they used to be rock solid) I already emailed Sipura support, but they seem to be on strike as of late. Here is the debug output from just one of the devices: (I've trimmed it for size, it happens more often than what is shown) Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 21:34:14 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:20:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 16:44:57 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 16:44:57 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 18:01:03 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 18:01:03 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 19:21:06 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 19:21:06 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 22:01:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 22:01:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 23:21:17 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 23:21:17 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 00:41:20 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 00:41:20 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 02:01:20 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 02:01:20 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 03:21:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 03:21:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct
RE: [Asterisk-Users] SPA-2000's rebooting every hour or so...
Actually, one thing that I did notice, which is _really_ odd, is two of the SPA-2000's reboot more often then not at almost exactly the same time as each other, and it seems to happen about 1 hour apart. I have them set to re-register with Asterisk every hour, and I'm wondering if they also try to get an new IP via DHCP at the same time? Notice the different IPs: Oct 14 10:28:27 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 10:28:27 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 10:28:30 192.168.1.190 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 10:28:30 192.168.1.190 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 11:45:43 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 14 11:45:43 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 14 11:46:08 192.168.1.190 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 14 11:46:08 192.168.1.190 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 14 11:47:17 192.168.1.190 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 14 11:47:17 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 14 11:47:17 192.168.1.190 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 14 11:47:17 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 14 11:48:28 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 11:48:28 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 11:48:30 192.168.1.190 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 11:48:30 192.168.1.190 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 13:08:25 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 14 13:08:25 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 14 13:08:33 192.168.1.190 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 13:08:33 192.168.1.190 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 14:28:32 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 14:28:32 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 14:28:35 192.168.1.190 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 14:28:35 192.168.1.190 System started: [EMAIL PROTECTED], reboot reason:H0 On Thu, 2004-10-14 at 12:36 -0700, Dooz Owings wrote: try to run a firmware update on one and see if it works, just a guess. What all have you tried ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mike Benoit Sent: Thursday, October 14, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SPA-2000's rebooting every hour or so... I realize this is slightly off-topic here, but I know quite a few people on this list use Sipura products. Has anyone else experienced the same rebooting problem I'am? I have about 8 SPA-2000's and about half of them just started rebooting 4-8times/day in the last month or so. (they used to be rock solid) I already emailed Sipura support, but they seem to be on strike as of late. Here is the debug output from just one of the devices: (I've trimmed it for size, it happens more often than what is shown) Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 21:34:14 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:20:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED
Re: [Asterisk-Users] SPA-3k outbound calls...
I submitted a bug report (http://bugs.digium.com/bug_view_page.php?
bug_id=0002620) regarding this issue a couple days ago, and it has since
been fixed. You can download the patch from the above link, or wait a
bit and it will probably be applied to the stable CVS branch.
On Sat, 2004-10-09 at 23:41 -0400, Jeff owen wrote:
Ok, now since I have inbound working properly the outbound seemed to
have gotten hosed.
In the extensions.conf I have it setup as:
exten = _9NX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
In the sip.conf I have it setup as:
[pstn] SPA-3k PSTN Line
type=friend
context=default
secret=supersecretpassword
port=5061
host=dynamic
dtmfmode=rfc2833
canreinvite=no
nat=no
Which should be correct for inbound and outbound calling, right?
However all I get when I try and dial out is another dial tone and if
I try to dial a number a second time the call will go thru. Kind of
like dialing 98145551212, getting dial tone, then dialing 8145551212
and the call gets connected then.
Im not sure what needs to be set on the SPA-3k to allow calls to be
made by what is passed to it. However, when I look at the SIP debug I
dont even see the number listed as passed to the SPA-3k.
Any ideas on where to look or what to set?
Thanks,
Jeff
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--
Mike Benoit [EMAIL PROTECTED]
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Re: [Asterisk-Users] Re: SPA3000 as a replacement for X100P
Old SPA-3000 firmware versions had issues with bad echo when raising txgains, apparently it has been greatly reduced, if not fixed in the latest firmware. Volume/echo was always an issue on my X100P's, but since I switched to the SPA-3000, the volume is _much_ higher with next to zero echo. 2.0.10f: 21. Improved echo suppressor for FXO port in presence of strong echo 2.0.10d: 17. Improved FXO ECAN to auto-compensate for large SPA to PSTN gain and PSTN to SPA gain settings (so that large gain wil not lead to larger echo) On Sun, 2004-10-10 at 12:16 -0700, Randy Bush wrote: With mine no echo problem, but the sound level is very low... :/ You have to speak higher to be heard... Raise the TxGain setting on the SPA. and get echo randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Status of conference calls at Astricon ?
Is there an IAXtel 1-700 number by chance that can get me in to the conference? On Wed, 2004-09-22 at 05:36 -0400, Jeremy McNamara wrote: Matteo Brancaleoni wrote: so, bring on and demostrate to the world what asterisk can do! I have packed all the necessary gear to stream the Developers Meeting on Friday. I am looking for people with Big Pipe(tm) to get crazy and link multiple Asterisk MeetMe's together. Lets see how much Pipe she can take. ; normal usage until it is abused with MOH! IAX2/[EMAIL PROTECTED]/4569 ; if you plan to link in your own MeetMe use this IAX2/[EMAIL PROTECTED]/4569|qm ; Monitor only and be quiet! Then to truly experience Asterisk's power, join the same conference... ...Using SIP: [EMAIL PROTECTED] ...and using H.323: [EMAIL PROTECTED] However, you can also call 877-677-9648 using room number 4569 to get into the same conference, if you are not 'leet enough to have any VoIP skills. Feel free to abuse the system as necessary. My thugs are ready to break knee caps if someone starts playing games. There have been a few different ideas thrown around on how to integrate the participants in the MeetMe to the developers meeting, I will discuss this with others when the time is appropriate. Hopefully we will be able to take questions from the MeetMe, so start pondering some good questions. If the powers that be let me stream more content to the MeetMe, I will. So you better hang out in the conf all day, what else do you have to do, work? A few random numbers to call: Twisted has lit up 866-251-9527 to do various crazy things, feel free to call him, any time. :) Then 866-498-4988 and 248-724-1710 are routed to my mobile Asterisk box, if it has 'net access. Be prepared to deal with IVRs to find a human. Then, if you are really bored you can call 877-966-6673, but don't expect to talk to a human and you will most likely hangup in disgust, so maybe you shouldn't call this one. Give us a few hours and we will think of some more crazy things to do. Jeremy McNamara p.s. So, I left out MGCP and SCCP, sue me, my legal department is bored ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting SPA-300 to Asterisk
What I did was create an extension that goes to my mainmenu, ie: exten = 7000,1,Goto(mainmenu,s,2) Then I setup the SPA-3000 to dial that extension when a call comes in according to the FAQ entry on Sipura's website: http://www.sipura.com/support/spa3000faq/Section_3.html#4 4: How can I forward all PSTN callers to a VoIP number? A: You can use specify a dial plan to be used by the default PSTN caller with a hot line syntax: (S0:voip_number) where voip_number is replaced with the actual phone number (or sip url) of the VoIP destination. So I used: (S0:7000) Works like a charm. In my experience Sipura has an excellent product, I use both the SPA- 2000 and SPA-3000, and they are both great. I've had _far_ less problems with Sipura products then I have had with Digium X100P cards. Especially when it comes to echo. Sipura's email support is also exceptional, I often get replies to emails within minutes, and they have even implemented feature requests and sent me a beta firmware within days. On Sat, 2004-09-18 at 13:31 -0600, Joseph wrote: On Fri, 2004-09-17 at 23:37, Kristian Kielhofner wrote: Sys. Concept Inc. wrote: How to Connect SPA-3000 to Asterisk so * will answer? After setting up Asterisk on Gentoo the extension.conf contains [demo] context; but my asterisk is not answering? In SPA-3000 in PSTN Line - tab under Dial Plan 1 I have: S0:[EMAIL PROTECTED] Default dial plan is set to 1. My box's IP where Asterisk is running has IP: 10.0.0.101 Line 1 - tab has: SIP Settings Port: 5060 Nat is disabled as both Asterisk and SPA-3000 are behind firewall. What am I missing? -- #Joseph Check this out from Voxilla: http://voxilla.com/forum-viewtopic-t-557.html I checked their forum, actually Voxilla is the place I bought the SPA-3000 unit from. But the configuration instructions are not clear to me, there is a lot of information but none is complete. The SPA-3000 unit might be good but the configuration is nightmare and Sipura manual is only good for reference nothing else in addition they do not offer any support. If I was to recommend any unit, go with Digium cards and stay away from Siupra -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] just-added second X100P
Although it shouldn't make a difference, try: channel = 1-2 As well, did you run ztcfg after you installed the new card? I've found sometimes I've had run ztcfg a couple times before Asterisk would kick in and recognize a new card. On Sat, 2004-08-21 at 02:49 -0500, spectro wrote: I just added a second X100P card to my * server, altough it seems to be working * seems to be ignoring it: zaptel.conf: - fxsks=1-2 loadzone=us defaultzone=us zapata.conf: -- context=inbound-analog signalling=fxs_ks group=1 channel = 1 channel = 2 I created a couple of test extensions: ; test extensions exten = 4390,1,Dial(Zap/g1/4189) exten = 4390,2,Congestion exten = 4391,1,Dial(Zap/1/4189) exten = 4391,2,Congestion exten = 4392,1,Dial(Zap/2/4189) exten = 4392,2,Congestion 4391 works fine, 4392 doesn't: -- Executing Dial(IAX2/[EMAIL PROTECTED]/2, Zap/2/4189) in new stack Aug 21 02:47:36 NOTICE[426002]: app_dial.c:714 dial_exec: Unable to create chann el of type 'Zap' == Everyone is busy/congested at this time I don't know what's wrong, Zap/2 shows fine in the zap channels list: pbx*CLI zap show channels Chan Extension Context Language MusicOnHold pseudoinbound-analog 1inbound-analog 2inbound-analog Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura endpoints
We have about 8 SPA-2000's and one SPA-3000 right now, and going to be ordering about 10 more in the near future. We had one unit that was Dead on Arrival, it wouldn't make an ethernet connection, but it turned out to be just an issue with the pins in the jack on the SPA2K weren't high enough. 1 paperclip and 5 minutes later, I managed to raise the pins and its been working flawlessly for the last couple months. If its an actual hardware issue, you need to get an RMA number from the company that sold it you, or maybe from Sipura, if they do that. Sipura's email support has been some of the best I've ever run across. I've sent them about 5 emails this month, and on average I got accurate, straight to the point, fixed the problem responses in less then 15minutes. Compare this to Digium's email support that took on average ONE WEEK just get a one sentence response that wasn't helpful at all. But to there credit, after about one month of back and forth (and some IRC conversations) they offered to upgrade my hardware at no cost. *Keep in mind that this is only MY experiences, Digium has done great things for the community, and I wish them all the best. I even emailed Sipura a feature request on Aug 6/04 and got a beta firmware implementing it on Aug 12/04. How's that for service? :) There products are great, there service is great, I would recommend them to anyone. Especially the SPA-3000, which I will be using to replace 12 X100P cards with as well. The price is right, Caller*ID works _consistently_, echo is virtually gone, and volume is louder on the SPA- 3000 compared to X100P on the same line. (I'm in Canada too, so its not a country issue) On Fri, 2004-08-20 at 07:43 -0500, Matt Schulte wrote: Anyone have experience with Sipura's? Anyone know if they offer a warranty? Would like opinions on these, good or flame. We bought *one* to test with and it died, can't even get a response from Sipura support. Could anyone recommend another device to replace these? Prefer 1 or 2 port design. Ty :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Received packet with bad UDP checksum
I was just on 70minute call (IAX2 - Internet - IAX2) and during that time I heard several pops, or clicks. Each time it happened, I saw the following message: Aug 19 15:36:36 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Any ideas what causes these, and why they turn in to a pop, instead of just silence, or a missed portion of audio? Thanks Here are the rest of them: - Aug 19 15:36:36 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:36:58 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:37:39 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:38:30 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:38:52 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:40:22 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:42:16 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:46:09 NOTICE[1209019312]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:46:51 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:50:34 NOTICE[1209019312]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:50:43 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:52:46 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:53:06 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:53:35 NOTICE[1209019312]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:53:44 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:54:04 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voice choppy
You need to narrow down the cause, seeing as you tried across a local network and it sounded excellent, I would start looking at your remote network connection. Things to try: - Try not using the VPN, I've heard reports that some VPNs cause major issues with VOIP. I'm not 100% why, but its worth a try turning off the VPN to see if that helps. - I'm not sure if this applies in your case, but make sure silence suppression is DISABLED on your phones. - Grab the latest version of mtr (http://www.bitwizard.nl/mtr/) , and run traceroutes between all end points with it. While the traceroute is running, hit j so it shows you the jitter information. High latencies don't usually degrade _sound_ quality of a call, however it severely degrades _call_ quality. Anything over 300ms round trip really starts to hurt the call quality, but all that really happens is both parties start to talk over one another. You don't normally get chop. Round trip latencies below 100ms I would consider excellent, and virtually unnoticeable during normal conversations. If the chop is related to the network quality, its usually due to packet loss, or jitter. mtr is a great tool for this type of network diagnostics. Hopefully this helps, good luck. On Fri, 2004-08-13 at 14:21 -0400, Dana Nowell wrote: OK, background/config. running * (show version reports 0.9.0) on Mandrake 9.2 (kernel: 2.4.22-32mdk) with a dual 800mhz PIII with 256M Ram 4port FXO digium card, no IRQ sharing I can find (cat /proc/pci cat /proc/interrupts), vmstat reports a minimum of 80+% CPU idle when problem occurs. connect to a Grandstream 101 (GS) via vpn (no nat). Link has 100ms - 150ms ROUND TRIP latency (constant 'ping' during test). Codec is alaw or ulaw. TDM card is plugged to an NEC PBX (old NEAX 2000 IVS) via the analog station card in the PBX. PROBLEM: Establish a call from the GS to an NEC phone (Dterm III) connected to the PBX. The voice quality on the GS sounds good. The voice quality on the NEC gets very choppy (random). A call from the same GS to an internal GS here (same latency, same IP path) is much better (some chop). SOLUTIONS tried to date: Tried a local network test, sound excellent. Tried a 'low latency vpn' test bed to reduce latency (is latency the issue?). Latency down to ~70ms round trip, better but still noticeably choppy. Tried different codecs (ulaw, alaw, iLBC), no impact. Tried various echo cancel/train on/off values, cancel and train on seem best. Tried modifing Samples/TX packet changes, 2 and 64 tried as 'ends of spectrum' values. little noticable impact (2 seems marginally better, still clips, MAY be less often). Tried changing TXgain and RXgain under the assumption audio level clipping occurs between tdm and NEC, some improvement. Substituted an SJPhone on a PC instead of the GS, better, still choppy. Tried the 'demo speech' on an extension, called from Dterm, sound is excellent, tried GS over low latency vpn, choppy, tried SJ over low latency vpn, choppy. (h, does that eliminate the tdm to NEC link as an issue?) Tried internal GS over low latency vpn to SJ (out thru vpn to * back thru vpn to sj), choppy. Current state: I've improved from 'crappy cell call about to disconnect' to 'average to crappy cell call' quality. Not exactly ready for customer use. ASSUMPTIONS: Original assumption was latency, but latency was still below the '150MS one way' values I see quoted for good sound quality. In fact over the low latency vpn the values were less than a third the quoted value. Sometimes I get a high latency and get a drop out, OK expected. Sometimes I get chop at a lower latency than I was getting with 'good' sound. Second assumption was interface between NEC and TDM was clipping, after changes to tx/rx gain the 'demo' speech sounds fine on the Dterm. (Does this remove the interface from the possible bad guys list?) HELP Needed: So given that latency is in the 70 - 100 ms round trip. Given that I've diddled rx/tx gain. Given that I've tried basic echo on/off settings. What's next? Is the 'common' 150 ms one way (150+ms round trip) value a bunch of crap? Do I need some other magic latency goal? Would a 729 codec help? Is there a test I missed? Some other values to twiddle? Is this a 'known issue' I don't know about, fixed in some more recent version? (Yeah, I can just try the CVS-HEAD and hope to get lucky and I probably will as I'm out of options/ideas but KNOWING it is fixed is better than hoping it is fixed) I'm stuck and my forehead is getting flat from pounding it on the wall. Anyone handing out clues? -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000 as a regular Asterisk FXO device?
It _seems_ to only pick up the line when the VoIP end answers. At least for me it doesn't stop ringing until I see the log entry in Asterisk say it picked up the call. Now I just have to figure out how to get Asterisk to _not_ override the incoming caller id with the SIP information from sip.conf. The SPA-3000 says it sends the CID from the PSTN through, but Asterisk is just show the SIP extension number. On Wed, 2004-08-04 at 12:12 +1200, Andrew Gordon wrote: Andres wrote: The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk mainmenu context (or ext I guess). Configure an auto-dial number in the SPA to that it corresponds to something in the mainmenu context. Like: PSTN_Caller_Default_DP[2] 2 ; Dial_Plan_2[2](S0:551155) ; When a call comes in the FXO port, the SPA automatically dials 551155 via your Proxy[2] settings.. Does it answer the line first then ring the 551155 or does it ring the 551155 and only pick up the line when the VoIP end answers? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-3000 as a regular Asterisk FXO device?
My SPA-3000 finally arrived and I'm trying to get the FXO port on it to work as if it was a X100P card as far as Asterisk is concerned. I have Asterisk dialing out over the SPA-3000 FXO port no problem. The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk mainmenu context (or ext I guess). Currently the SPA-3000 answers the call, then I hear a modified dial tone, which if I dial any extension + #, it will ring a SIP phone no problem. So now I just need to get it to that automatically. The SPA-3000 User guide shows how to have it automatically forward incoming PSTN calls to its FXS port, but that would normally be a phone, not Asterisk. Any ideas? -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000 as a regular Asterisk FXO device?
Works like a charm Andres. Much appreciated. On Tue, 2004-08-03 at 16:44 -0500, Andres wrote: The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk mainmenu context (or ext I guess). Configure an auto-dial number in the SPA to that it corresponds to something in the mainmenu context. Like: PSTN_Caller_Default_DP[2] 2 ; Dial_Plan_2[2](S0:551155) ; When a call comes in the FXO port, the SPA automatically dials 551155 via your Proxy[2] settings.. -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RAID affecting X100P performance...
Someone mailed me off list and suggested the below: Tuning these [PCI latencies] should allow you to give your TDM cards long burst lengths, and make your IDE devices very premptable... A decent article which has info in PCI latency (and IRQ, etc) is at: http://www-106.ibm.com/developerworks/library/l-hw2.html I tried it out, and though it did seem to help the problem, it was not a 100% fix. I ended up having to go a Athlon 1800 (don't have the mainboard model on hand right now) to solve the problem 100%. I also discovered my SPA-2000's silence suppression was causing a good chunk of choppiness (much more so then any SS should), so I disabled that too. The cutting out is completely gone now. Good luck. On Fri, 2004-07-30 at 15:26 -0700, Florin Andrei wrote: On Wed, 2004-07-21 at 12:14, Mike Benoit wrote: I have a P3-800 with two IDE drives in a software RAID1 configuration. Each drive is on a separate IDE channel. Now anytime there is HD activity, I hear beeps and cutting out on a call using the X100P card. Wow, i'm seeing exactly the same behaviour! AthlonXP/1800, MSI NForce1 mobo, Wildcard TDM400P, soft RAID1 on /boot, soft RAID5 on everything else, Asterisk-1.0-RC1, Linux Fedora 2 fully updated. I'll explore the idea offered by someone else in this thread and shuffle the cards around, trying to put the Wildcard in another PCI bus. -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?
Yes, they seem to have many benefits. No PCI slots are used, echo issues related to mainboards are eliminated, no interrupt sharing/dropping issues, the list goes on. My SPA-3000 was ordered last week and should be here any day now. Once I do some testing on it I'll be sure to write a review comparing it directly to Digium FXO cards for the mailing list. On Tue, 2004-07-27 at 18:52 -0400, Carmi Weinzweig wrote: I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. /carmi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echotraining on T1 circuits
I don't use T1's, only regular lines, but echotraining works with any zaptel interface as far as I know. I would try echotraining=yes and echotraining=800 (if your using a relatively new CVS version). I personally haven't noticed any pause when using echotraining, I think its less then 1 second, but not 100% sure on that. Also, you didn't mention if it was near end echo, or far end echo your hearing. On Wed, 2004-07-21 at 13:24 -0400, mattf wrote: Hello, We just had some new T1s turned up today to replace others that our contract has run out on and we are now getting more echo on the new T1 lines than we had on the old ones. The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they replaced) The problem is that we are getting echo on about 10% of the calls in and out placed on these new T1s compared to less than 1% with echos on the old T1s. I was wondering if anyone with T1s or E1s is using echotraining=yes. All the info on echotraining I've found seems to involve POTS lines not T1s. Also, how much of a pause is made when using echotraining? Here are my current zapata.conf settings for the T1s: [channels] group=1 language=en signalling=em_w usecallerid=yes callerid=asreceived context=default echocancel=yes echocancelwhenbridged=yes rxgain=1.0 txgain=1.0 channel = 1-24 Thanks, MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RAID affecting X100P performance...
I have a P3-800 with two IDE drives in a software RAID1 configuration. Each drive is on a separate IDE channel. Now anytime there is HD activity, I hear beeps and cutting out on a call using the X100P card. I ran the zttest program, and discovered HD activity would drop the accuracy down to between 2% and 50%. However I noticed if I disabled one drive in the RAID1 array, zttest would always report 99.98% or higher. So one drive running works fine, but as soon as I enable the second drive, all hell breaks loose. DMA and 32-bit mode are enabled on both drives as well. I have a backup server with two Promise PCI IDE controllers in it, with 4 drives in a software RAID5 configuration, so just out of curiosity sake, I stuck a X100P card in it and tried running zttest while the RAID was re-syncing. The results were pretty bad. --- Results after 384 passes --- Best: 36.779785 -- Worst: 1.562500 Is this a poor mainboard issue, or is it actually not possible to do IDE software RAID on a machine running Asterisk with X100P cards? Is anyone currently doing it? Thanks. -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RAID affecting X100P performance...
I didn't want to turn this in to a software vs. hardware raid, or IDE vs. SCSI. I was more curious about the PCI bus/interrupt issues and the mainboard. I only have 1 line going in to this asterisk server, so CPU usage is not an issue whatsoever. Even during a raid rebuild. Asterisk's CPU usage doesn't even show up on the charts. zttest and the odd complaint is the only way to tell if the problem is occurring. I've actually done some more testing since my post and discovered the problem happens even with a SINGLE HD in the box. If I work the drive hard enough, zttest starts reporting accuracy down to 80%. With two drives I can get it as low as 2%. I did a quick test with a Celeron 2.4Ghz, and brand new mainboard and couldn't seem to reproduce an accuracy drop below 99.98%. So it seems like the X100P (possibly other Digium cards too?) is _very_ picky about the mainboard its used with. Not only have I had echo issues that are specific to some mainboards, I've now also discovered beeps or cutting out with some mainboards. The two don't seem to be linked either. Since the mainboard that is experiencing cutting out is one that has had the least amount of echo so far. However I do wonder if the cutting out has a negative effect on the echo canceler or echo training. Really working two HD's on the mainboard (RAID or not) really brings the problem to light, but the problem is even visible (though not near as bad) with a single HD. I would be interested in seeing if other people can reproduce low zttest accuracy rates with their mainboards. zttest is in the zaptel/ directory, and you can run it while Asterisk happily chugs along handling calls. What I usually do is run zttest in one window, then in another window run updatedb or find / /dev/null or hdparm -t /dev/hda or all of them at the same time. The harder and longer you work the drives the better. If the problem exists on your system, you'll see the zttest accuracy drop below 99.98% at some points. On Wed, 2004-07-21 at 16:25 -0700, Kevin P. Fleming wrote: Scott Laird wrote: That hasn't been my experience at all. Frankly, I've never seen a cheap ($3k) hardware RAID controller that can touch software RAID's performance on Linux, especially in challenging setups, like RAID-5. Sure, software RAID eats more CPU, but most PCs have CPU to spare these days. Would you rather eat 10% of one of your Xeon CPUs to get 200 MB/sec or 100% of an Intel 960 to get 15 MB/sec? While this is certainly true, in the context of Asterisk you also have to consider the extra PCI bus usage for all this data going back and forth to the drives while the RAID parity/mirror stuff is being done. In the context of Asterisk, where disk I/O is either logging or voicemail, buying a 3ware card and a pair of IDE drives seems like a decent business decision. Yep, it works very well. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RAID affecting X100P performance...
On Thu, 2004-07-22 at 12:56 +1200, wrote: I would be interested in seeing if other people can reproduce low zttest accuracy rates with their mainboards. zttest is in the zaptel/ directory, and you can run it while Asterisk happily chugs along handling calls. What I usually do is run zttest in one window, then in another window run updatedb or find / /dev/null or hdparm -t /dev/hda or all of them at the same time. The harder and longer you work the drives the better. If the problem exists on your system, you'll see the zttest accuracy drop below 99.98% at some points. Okay, I have 1 X100P and 4FXS modules (TDM400) and running zttool under normal load gets about 99.98. I have 1 hard disk, no raid. If however I run updatedb on another console I get drops down to 33.508301, but still spends most of the time between 99 and 100. I'll have to remember not to run updatedb during business hours (actually it's run from cron in the middle of the night anyway). What other task would an asterisk system do that would use so much disk though? Maybe recording every conversation on a fully loaded quad T1 box... It's not just that, I've found I'll get cutting out even when the disk isn't really being used. The test seems to show that your mainboard can't handle the load. I'm still not 100% sure every mainboard doesn't do the same thing, but I couldn't re-create it on a brand new Celeron 2.4ghz box. The Celeron was the only system I had that I couldn't get the accuracy to drop below 99.98%, a older P3-800, and a older Celeron 533 all had major issues. Have you had any complaints of cutting out? It can be _very_ minor, and usually not noticeable during normal conversations. In the minor cases it may sound like just parts of words go missing but the person can still make out the word in the end. Also I have a hunch it _may_ be contributing to DTMF tones not being recognized in rare cases. I've had the odd complaint that people have to sometimes press buttons twice before my menu will clue in. Try being on a call that traverses the X100P card, or the TDM400 for that matter I'm interested if it has the same issue, and run the same test. See if you notice any cutting out. -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reverse Hold feature prototype...
I have no idea what this really should be called, so for lack of a
better name, I called it reverse hold. Hopefully someone else can make
use of it, or even make it better, as its the first thing of its kind
I've made for asterisk.
Like most people, I'm very busy, so when I call other companies, sitting
on hold really sucks. If you have speaker phone, its not so bad, but
then you have to sit there and listen to the company's horrible choice
in music.
The solution: Reverse Hold.
When your on hold, blind transfer the call to the special extension.
This extension repeats Press 1 to be connected to the caller, do not
hangup constantly. Once the remote party presses 1, the original
extension is dialed back. No more listening to hold music, or having a
phone stuck to your ear forever! ;)
It confuses people at first, but for the most part it works great.
This will definitely need some tweaking to fit in to your own setup.
Extensions.conf
;This is used in case you manually set the callerid
;in your extensions.conf, ie: Nufone
;It saves the originating EXT for use later on.
;I tried getting the src channel, and just dialing it back,
;but it didn't work out so well.
[macro-set-callerid]
exten = s,1,setGlobalVar(SRC_EXT=${CALLERIDNUM})
exten = s,2,SetCallerID(${ARG1})
exten = s,3,SetCIDName(${ARG2})
[reverse-hold]
exten = s,1,GotoIf($[${SRC_EXT} = ]?99:2)
exten = s,2,NoOp(Reverse Hold from: ${SRC_EXT})
exten = s,3,AbsoluteTimeout,600 ;10 mins, then force a callback to the
originating ext as a reminder the call is still on hold
exten = s,4,Wait,2
exten = s,5,Background(reverse-hold-repeat)
exten = s,6,Goto(s,4) ;Loop
exten = s,99,Macro(set-callerid,55,Foo Company) ;If SRC_EXT
isn't set already, this will set it
exten = i,1,Goto(s,4) ; Keep repeating if they don't press 1
exten = t,1,Goto(s,4)
exten = 1,1,NoOp(Reverse Hold party dialed 1, calling back: ${SRC_EXT})
exten = 1,2,Goto(reverse-hold,999,1)
exten = T,1,NoOp(Reverse Hold timeout, calling back: ${SRC_EXT})
exten = T,2,Goto(reverse-hold,999,1)
exten = 999,1,SetCallerID(${EXTEN})
exten = 999,2,SetCIDName(Reverse-Hold)
exten = 999,3,AbsoluteTimeout,0
exten = 999,4,Goto(extensions,${SRC_EXT},1)
exten = 999,5,Goto(s,4)
exten = 899,1,Goto(reverse-hold,s,1)
Ideally something like this would become its own application, or tied in
to something like parkedcalls. But for now, the above works relatively
well. The biggest drawback is there is no way to get back to call on
hold until it times out.
Enjoy.
--
Mike Benoit [EMAIL PROTECTED]
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Re: [Asterisk-Users] spa-3000 review?
Dameon and Wolfgang, Have either of you experienced echo when making a call from the FXS port to the FXO port on the SPA-3000? Thanks -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 calls through IAXTEL.com
I've been getting the same issue with toll free numbers over IAXTEL for
the last 4-5 days. I contacted Digium support (IAXTEL's website says to)
on July 9th, and all I got back was We will look in to it.
I haven't heard anything since.
On Tue, 2004-07-13 at 08:38 -0400, Steve Woolley wrote:
I created an account at IAXTEL.com to route 1-700-XXX- calls
through. IAXTEL.com gave me a number (example) of 700-555-6226. I have
made the following changes to my:
/etc/asterisk/extensions.conf:
[iaxtel700]
exten =
_81700XXX,1,Dial(IAX2/myusername:[EMAIL PROTECTED]/${EXTEN:1})
exten =
_81800NXX,1,Dial(IAX2/myusername:[EMAIL PROTECTED]/${EXTEN:1})
/etc/asterisk/iax.conf:
[general]
port=5036
bandwidth=high
disallow=all
allow=gsm
tos=0x18
register = myusername:[EMAIL PROTECTED]
[guest]
type=user
context=guest
[iaxtel]
type=peer
context=inbound-analog
auth=rsa
inkeys=iaxtel
[iaxtel-outbound]
type=peer
username=swoolley
secret=gl0bal
host=iaxtel.com
The good news is that dialing 700-XXX- numbers (at Digium) works
great.
I however have two problems:
1) if I dial 800 numbers, like (800)555-1212, I get a bunch of silence
and the following in my log:
-- Starting simple switch on 'Zap/97-1'
-- Executing NoOp(Zap/97-1, ) in new stack
-- Executing Goto(Zap/97-1, intern-post|818005551212|1) in new
stack
-- Goto (intern-post,817005556226,1)
-- Executing Dial(Zap/97-1,
IAX2/myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack
-- Called myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Call accepted by 69.73.19.178 (format GSM)
-- Format for call is GSM
-- IAX2[iaxtel-outbound]/3 stopped sounds
The call never seems to go through.
2) Not knowing any other way to test, I have simply picked up my
asterisk SIP and analog phones and dialed my own 700 number
(700)555-6226 to which I get a bunch of silence and the following in my
log:
-- Executing NoOp(Zap/97-1, ) in new stack
-- Executing Goto(Zap/97-1, intern-post|817005556226|1) in new
stack
-- Goto (intern-post,817005556226,1)
-- Executing Dial(Zap/97-1,
IAX2/myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack
-- Called myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Call accepted by 69.73.19.178 (format GSM)
-- Format for call is GSM
-- IAX2[iaxtel-outbound]/2 stopped sounds
-- Hungup 'IAX2[iaxtel-outbound]/2'
But I do get a:
-- Registered to '69.73.19.178', who sees us as 63.143.35.201:4569
When asterisk is starting up so I belive I am registered.
Can I simply not dial my own 700 number from the same asterisk PBX as
a test or do I have some real problem?
--
Steve Woolley
IT Manager
ADS Telecom, Inc.
59 Skyline Drive
Suite 1250
Lake Mary, Florida 32746
Phone: (407)682-6226 x1110
Fax: (407)682-3455
Cell: (321)229-5311
[EMAIL PROTECTED]
www.adstelecom.com
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--
Mike Benoit [EMAIL PROTECTED]
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Re: [Asterisk-Users] UPDATE - Echo cancellation, when softwaredoesn't cut it. Whats next?
That doesn't explain why a incoming call from a land line has nearly no echo, while an outgoing call to the same land line has echo. Also it has always been near end echo I'm hearing, and prior to upgrading the mainboard/CPU I heard echo when calling the same cell phones. On Mon, 2004-07-12 at 02:06 -0400, Anton wrote: No it points to Cell phone companies having better hardware echo cancellation on their lines, also cell phones themselves have a hardware echo can built in. - Original Message - From: Mike Benoit [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 12, 2004 1:52 AM Subject: Re: [Asterisk-Users] UPDATE - Echo cancellation, when softwaredoesn't cut it. Whats next? Here's an update on my progress for all who are interested. After carrying out many more hours of testing, the only thing that made a significant difference was changing the mainboard/CPU of my asterisk server. My original Asterisk server was a Celeron 533 with 128mb ram. Now, keep in mind, even with 5 channels in use at a time, the CPU usage was always minimal, the load never went above 0.2 that I saw. First I upgraded to a brand new Celeron 2.4Ghz with 128mb ram, and immediately noticed an improvement in the echo. Incoming calls now have virtually no echo. I have to really try to hear it. Outgoing calls still have echo, but after about 30 seconds it mostly goes away during normal conversation. Still not 100% acceptable though. However it is a huge improvement. The weird part though is this. Outgoing calls to cell phone numbers (tried 3 different ones) have virtually no echo. Outgoing calls to land line numbers do seem to have echo. I then downgraded my asterisk server to a P3-800 with 128mb ram, and I didn't notice any difference from the Celeron 2.4. So in my case just upgrading a Celeron 533 to P3-800 made a noticeable difference, but anything more than that did not. What I did notice is in /proc/interrupts, the P3-800 displays: IO-APIC-level wcfxo Whereas I believe the older Celeron 533 displayed: XT-PIC wcfxo So my guess is its not the CPU speed at all, just the way interrupts are handled. So the two questions remain. 1. Why do incoming calls have nearly no echo (sound great), and outgoing calls are bad during the first 30 seconds, and okay (but not good) after that. 2. Why do outgoing calls to cell phone numbers sound great? Seeing as an outgoing call to a land line has echo, but the same land line calling in has virtually no echo, does this point the finger at Asterisk code having issues? On Wed, 2004-06-30 at 16:36 -0700, Mike Benoit wrote: Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none acceptable solutions in my situation. Playing with rx/tx gain values just seemed to quiet the voice down and along with that the echo happened to be less noticeable. I could almost get the echo to disappear with a low enough rx/tx gain, but then the voice could barely be heard, or DTMF tones stopped working. So whats the next step? I only get echo when dialing over the PSTN. Using Nufone to dial a PSTN number results in absolutely zero echo. Do I put in a request for a Telco technician to come out and take a look at the lines? One page on the Wiki says: Most of the telco's have technicians with the equipment necessary to help find the problem if the problem really is their outside plant. However, getting to that person can be a real challenge. Any suggestions on ways to overcome the challenge of getting the right technician on the phone? Thanks. -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPDATE - Echo cancellation, when software doesn't cut it. Whats next?
Here's an update on my progress for all who are interested. After carrying out many more hours of testing, the only thing that made a significant difference was changing the mainboard/CPU of my asterisk server. My original Asterisk server was a Celeron 533 with 128mb ram. Now, keep in mind, even with 5 channels in use at a time, the CPU usage was always minimal, the load never went above 0.2 that I saw. First I upgraded to a brand new Celeron 2.4Ghz with 128mb ram, and immediately noticed an improvement in the echo. Incoming calls now have virtually no echo. I have to really try to hear it. Outgoing calls still have echo, but after about 30 seconds it mostly goes away during normal conversation. Still not 100% acceptable though. However it is a huge improvement. The weird part though is this. Outgoing calls to cell phone numbers (tried 3 different ones) have virtually no echo. Outgoing calls to land line numbers do seem to have echo. I then downgraded my asterisk server to a P3-800 with 128mb ram, and I didn't notice any difference from the Celeron 2.4. So in my case just upgrading a Celeron 533 to P3-800 made a noticeable difference, but anything more than that did not. What I did notice is in /proc/interrupts, the P3-800 displays: IO-APIC-level wcfxo Whereas I believe the older Celeron 533 displayed: XT-PIC wcfxo So my guess is its not the CPU speed at all, just the way interrupts are handled. So the two questions remain. 1. Why do incoming calls have nearly no echo (sound great), and outgoing calls are bad during the first 30 seconds, and okay (but not good) after that. 2. Why do outgoing calls to cell phone numbers sound great? Seeing as an outgoing call to a land line has echo, but the same land line calling in has virtually no echo, does this point the finger at Asterisk code having issues? On Wed, 2004-06-30 at 16:36 -0700, Mike Benoit wrote: Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none acceptable solutions in my situation. Playing with rx/tx gain values just seemed to quiet the voice down and along with that the echo happened to be less noticeable. I could almost get the echo to disappear with a low enough rx/tx gain, but then the voice could barely be heard, or DTMF tones stopped working. So whats the next step? I only get echo when dialing over the PSTN. Using Nufone to dial a PSTN number results in absolutely zero echo. Do I put in a request for a Telco technician to come out and take a look at the lines? One page on the Wiki says: Most of the telco's have technicians with the equipment necessary to help find the problem if the problem really is their outside plant. However, getting to that person can be a real challenge. Any suggestions on ways to overcome the challenge of getting the right technician on the phone? Thanks. -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does the SPA-3000 get rid of echo that the X100P can't?
After trying everything under the sun to get rid of echo on my X100P, I'm curious if anyone managed to solve the echo issues by switching to a SPA-3000? As well, if you have multiple SPA-3000's, can you create dial-out groups similar to the Dial(ZAP/g1) functionality? Thanks. -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't transfer with Zap and SPA-2000
It doesn't look like your using t or T in your Dial command. The Wiki on voip-info.org will explain those flags. On Wed, 2004-06-30 at 20:05 -0700, Seth Mattinen wrote: I am having trouble getting transfers to work when a zap channel is part of the call. I have a couple SPA-2000's and some X100P cards as my setup. This is what I'm trying: Dial number from phone: -- Executing Dial(SIP/206-2c61, Zap/1/###) in new stack Currently on call: -- Called 1/### Press flash to place call on hold with SPA-2000: -- Hungup 'Zap/1-1' As soon as I press the flash button on my SPA-2000 connected phone, the zap channel hangs up and the call is disconnected. The same procedure works fine between SIP channels (flash to hold, or transfer). I'm hoping this is just a config problem and not a general defect, because it seems odd to me that I can't transfer calls when a zap channel is the other end of the call I want to transfer or place on hold. sip.conf: [206] type=friend username=206 secret=blah host=dynamic context=from-sip reinvite=no canreinvite=no disallow=all allow=ulaw nat=0 zapata.conf: [channels] language=en signalling=fxs_ks usecallerid=no echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=2.5 txgain=7.0 busydetect=yes busycount=8 faxdetect=incoming context=inbound-analog1 channel = 1 context=inbound-analog2 channel = 2 -- Seth experientia docet Mattinen [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next?
Obviously the less I spend the better. But if we have to, a few thousand more I guess. The problem I have is that this setup is more of a trial run. Once it works, I'm going to be cloning slightly smaller setups to 9 other cities. But they are pretty small, 1 or 2 lines and 2-4 phones in each location. I will only be using POTS lines in each location. The current setup works great besides the echo, and some of the information I've read point to the Telco being the issue. If thats the case, I should in theory be able to get them to fix the problem. (though I could be dreaming) On Wed, 2004-06-30 at 22:42 -0500, Daniel Jimenez wrote: Mike Benoit wrote: So whats the next step? How much money are you willing to put in the project? Are you talking POTS lines or a PRI? If this is a serious project and you'd really like to clear it up I'd look at a Cisco device (maybe one of the newer rackmount 1700s) with FXO ports or a Serial interface for PRI. You can use h.323 or SIP to communicate with the device. -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-2000, call for help testing echo issues...
In my hunt to track down my echo issues, I tried disabling all echo cancellation, suppression, adaption, on my SPA-2000 (Advanced section of the config, under Line 1/2). Then calling from one local extension to another. (SPA-2000 Line1, to Line2 on the same device) I was pretty shocked with the results, the echo was HORRIBLE! I even tried 3 different analog phones. Now, once I turned the echo cancellation back on, I tried a few tests to see if I could hear any echo. The test was saying: test, check, and hitting my desk with a single finger fairly hard. When I would say test and check I would try all sorts of different voice tones, volumes, etc... I found I could still hear echo when doing this. When I hit the table with my finger (while still holding the phone to my ear and not talking of course), you could _really_ notice the echo. I could hear the hit perfectly clear coming back through the phone. As well if you change the Input/Output gains under the Regional section, cranking them both up to 5 or so, you could easily hear the echo, and the SPA-2000's poor attempts to cancel it. I'm curious to know if anyone else using SPA-2000's have the same issues. I wonder if when calls are made from SPA-2000's to PSTN numbers through Asterisk, asterisk is just amplifying the SPA-2000's own echo somehow. Any test results are welcome, I would be very interested if other people are unable to replicate my results. -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo -when software doesn't cut it.
Yes, I will definitely let everyone know what/if I find anything. Currently I've spent some time talking with Rich, and done many more tests under his guidance, so far we are leaning towards a Asterisk problem, not the PSTN. On Thu, 2004-07-01 at 18:27 -0400, [EMAIL PROTECTED] wrote: Kindly post it back here when you find something, IF you find something, as Ive been fighting the exact same problem here, and followed the same route you guys did. Ive given up at this point. At this juncture, Im convinced the telco is sending it out intentionallyor space aliens are bringing it...its only on the PSTN lines...Versleazon in my casebecause if this problem isn't curable, then asterisk is dead in the water as far as Im concerned. I simply can not use a system that sounds like people talking are in a tunnel. At this point, this is the only point of failure that kills the whole idea of using this as a real switch. Now, as far as echo cancellation, Ive looked as some of the docs, like the motorola paper, http://e-www.motorola.com/files/dsp/doc/white_paper/PTECANWP.pdf and some purely theoretical stuff, but their math is beyond me. Unless someone has developed a whizz-bang improvement over the stuff I read, the point of transiting from a two wire to a 4 wire model will always cause echo, period, done, abandon the idea, lets go do something else. The limiting factor I see, is our software echo handling doesn't have the computational horsepower attached (no cpu, or stealing cpu cycles from main cpu) to be effective. Does this mean another piece of expensive hardware? Dedicated proprietary chips like the Moto? I'm thinking so. SO, This is it, critical point of failure...cant go on, at least I cant use it. Yes, * does a lot of other stuff just fine, but this is the immovable object, at least for me.. This is something it must do and do well or its over. Ive been holding my breath, hoping something pops up, but it hasn't. At 09:21 7/1/2004, you wrote: Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none acceptable solutions in my situation. Playing with rx/tx gain values just seemed to quiet the voice down and along with that the echo happened to be less noticeable. I could almost get the echo to disappear with a low enough rx/tx gain, but then the voice could barely be heard, or DTMF tones stopped working. So whats the next step? I only get echo when dialing over the PSTN. Using Nufone to dial a PSTN number results in absolutely zero echo. Do I put in a request for a Telco technician to come out and take a look at the lines? One page on the Wiki says: Most of the telco's have technicians with the equipment necessary to help find the problem if the problem really is their outside plant. However, getting to that person can be a real challenge. Any suggestions on ways to overcome the challenge of getting the right technician on the phone? Mike, Contact me off list and let's see if we can isolate the issue. Can't tell from the words you've used what steps you've gone through to date. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next?
Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none acceptable solutions in my situation. Playing with rx/tx gain values just seemed to quiet the voice down and along with that the echo happened to be less noticeable. I could almost get the echo to disappear with a low enough rx/tx gain, but then the voice could barely be heard, or DTMF tones stopped working. So whats the next step? I only get echo when dialing over the PSTN. Using Nufone to dial a PSTN number results in absolutely zero echo. Do I put in a request for a Telco technician to come out and take a look at the lines? One page on the Wiki says: Most of the telco's have technicians with the equipment necessary to help find the problem if the problem really is their outside plant. However, getting to that person can be a real challenge. Any suggestions on ways to overcome the challenge of getting the right technician on the phone? Thanks. -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Customized Call Parking
This sounds like it should be relatively simple to do in theory. Couldn't you just create specific extensions that set the MusicOnHold context (to play your different announcements) then transfers the call to the parkext in parking.conf? However, what do you want to happen when the announcement is finished? If you park the call there is no way to know when the announcement is finished so you can return to the call later (if thats what you want to do). Perhaps just creating a sort of menu you transfer the calls to, that plays the announcement of your choice, then when its done, it could automatically transfer the call back to the originating extension, or park the call, or put it on hold, or hang up, etc... I think what your looking for is definitely possible without writing a custom application for it. On Tue, 2004-06-29 at 15:06 +0600, Adnan Shah wrote: Hi ! I need a solution to park incoming calls to an extension of my choice where a special announcement is played, park subsequent calls to specific pools so that they listen to announcements of my choice. any ideas ? Shah. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap X100P oscillation
I wonder if your issue and mine are related somehow. I have a asterisk server with 4 FXO cards in it, and when a call comes in one ZAP channel, then dials out another, I hear what could be described as a steam engine starting up. It starts off kinda slower/ quiet, then quickly (in about 2-4 seconds) completely over powers the line. The only way I could stop it was by adjusting the gains. rxgain=-8.5 txgain=4 Seemed to do the trick. As did: rxgain=-6.5 txgain=1 An rxgain of even -8.0 or -6.0 in either case would result in this steam engine sound. -8.5 or -6.5 would make it go away completely. I'm using a CVS checkout from yesterday, and I tried with both echotraining=800 and turning echo cancellation off completely. Neither made any difference. It would be really nice to be able to use a positive rxgain value. I haven't tried with the echo app, but using just one FXO card works fine with almost any rx/txgain value. As soon as the call utilizes two FXO card at the same time, the steam engine sound occurs. On Mon, 2004-06-28 at 16:26 +0100, Whisker, Peter wrote: Has anyone seen this problem before? I have a server with a single X100P card. The audio level is a low, but if I raise the gain to more than -2db (Rx + Tx) it starts to oscillate in an echo test. Not at a high frequency but with a noise that is best described as a steam engine starting up. It then starts to clip and crackle. If I bring the gain down to Rx=-2.0 and Tx=0.0 or lower then it settles down but it is very very quiet. I have tried the latest CVS Head with echotraining=800 set and also complied with the aggressive echo cancelling, but nothing seems to help. Ideas welcome! Many thanks Peter Whisker This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3-way calling woes... Nasty static and inconsistent flash detection?
This is my setup: SPA-2000 - Asterisk - X101P (x4) - PSTN 3-way calling works fine if I use flash and dial just local extensions. Or even if I use flash and dial one local extension, and one remote party over the PSTN. However, as soon as I dial from my SPA-2000 out over the PSTN, and hit flash the call hangs-up about 50% of the time. The other 50% of the time it puts the call on hold and gives me a dial tone. Now if I call a second number that goes over the PSTN, and try to connect all 3 parties (hitting flash again) it works, but I get this terrible static on the line that blasts everyone in the ear making it impossible to talk. As soon as one of the remote parties hangs up, the static stops and I can continue to talk to that one person just fine. Here is a log of the output of the inconsistent flash detection, with my commentary: -- Executing Macro(SIP/705-49a2, trunklocal|7680989) in new stack -- Executing SetAccount(SIP/705-49a2, local) in new stack -- Executing Dial(SIP/705-49a2, Zap/g1/7680989||t) in new stack -- Called g1/7680989 -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 answered SIP/705-49a2 *** Testing the flash button a few times, making sure it puts the call on hold, which it did. -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Hungup 'Zap/1-1' *** Intentionally ended the call, to immediately call back and try again. == Spawn extension (macro-trunklocal, s, 3) exited non-zero on 'SIP/705-49a2' in macro 'trunklocal' == Spawn extension (local, 7680989, 1) exited non-zero on 'SIP/705- 49a2' -- Executing Macro(SIP/705-37c1, trunklocal|7680989) in new stack -- Executing SetAccount(SIP/705-37c1, local) in new stack -- Executing Dial(SIP/705-37c1, Zap/g1/7680989||t) in new stack -- Called g1/7680989 Jun 25 16:56:12 WARNING[1191312304]: app_dial.c:338 wait_for_answer: Unable to forward frame -- Hungup 'Zap/1-1' *** Pressed flash, it hung up. So called back to try it again. == Spawn extension (macro-trunklocal, s, 3) exited non-zero on 'SIP/705-37c1' in macro 'trunklocal' == Spawn extension (local, 7680989, 1) exited non-zero on 'SIP/705- 37c1' -- Executing Macro(SIP/705-bb8f, trunklocal|7680989) in new stack -- Executing SetAccount(SIP/705-bb8f, local) in new stack -- Executing Dial(SIP/705-bb8f, Zap/g1/7680989||t) in new stack -- Called g1/7680989 -- Zap/1-1 is ringing -- Hungup 'Zap/1-1' *** Pressed flash, it hung up. == Spawn extension (macro-trunklocal, s, 3) exited non-zero on 'SIP/705-bb8f' in macro 'trunklocal' == Spawn extension (local, 7680989, 1) exited non-zero on 'SIP/705- bb8f' I don't have any logs of the 3-way in progress when I was getting the static. The static only happens when the two outgoing channels are ZAP. I wonder if the static has anything to do with the echo cancellation just committed in to CVS that I'm using? I'll have to test this theory when I get a chance tomorrow. I'm running CVS Head from yesterday. As well the SPA-2000's are running firmware 2.0.8, and there are no IRQ sharing conflicts. So, any ideas why Asterisk is sometimes putting the call on hold when flash is hit, and other times just hanging up? The flash button obviously works. As well, what would cause the nasty static when a 3-way call with 2 of the parties connected over the PSTN finally does work? Notable lines in my zapata.conf file: context=mainmenu signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes musiconhold=default faxdetect=both echocancel=yes echocancelwhenbridged=yes echotraining=800 -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax - Fast carrier training failed
Good question Steve. My setup is basically: Fax Machine - PSTN - X100P - Asterisk - RxFax I'm not even sure, does Asterisk do encoding if its not sending the call to a SIP device, or over IAX? In the mean time I configured Asterisk to send faxes to a SIP extension (SPA-2000) and it receives faxes just fine. Though for some reason the fax machine doesn't know the line is ringing, but if I manually answer, the fax comes through just fine. This setup was: Fax Machine - PSTN - X100P - Asterisk - SPA-2000 (alaw) - Fax Machine One other thing to note, I have two FXO cards in the Asterisk box, connected to two PSTN lines, I dialing both numbers to send the fax, and the Coarse carrier frequency varied quite a bit between both lines. One line as I showed you was 1832 and the other line was 1785.79. Both lines resulted in the same outcome though, no fax was received. On Fri, 2004-06-18 at 00:25 +0800, Steve Underwood wrote: Hi Mike, To get something like: Coarse carrier frequency 1832.96 (4) Training error 927.702492 Training failed (convergence failed) something is horribly wrong. The carrier should be 1700Hz, not 1832.96Hz :-) Do you have a codec mismatch, or are you using a codec other than u-law or A-law? Sometimes the slow control messages will get through the wrong modem, but the fast modem for the images never will. Regards, Steve Mike Benoit wrote: I'm trying to send a fax to my asterisk box, however shortly after connecting the fax machine reports a communication error and hangs up. Below is the error I get from RxFax: Fast carrier training failed Nothing is written to the file system as far as the tiff is concerned. Any ideas on how to fix this? Thanks Sending Fax Machine: HP PSC 2000 Series 2210 Printer/Fax/Copier/Scanner Asterisk: CVS-HEAD-06/14/04-15:43:15 SpanDSP: spandsp-0.0.1k FXO Card: X100P Interesting zapata.conf lines: --- echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=10.5 txgain=-4.5 faxdetect=incoming Log: --- Jun 16 21:45:40 NOTICE[1199700912]: callerid.c:239 callerid_feed: Caller*ID failed checksum Jun 16 21:45:43 NOTICE[1199700912]: chan_zap.c:4951 ss_thread: Got event 2 (Ring/Answered)... -- Executing Answer(Zap/2-1, ) in new stack -- Executing Wait(Zap/2-1, 1) in new stack -- Executing DigitTimeout(Zap/2-1, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(Zap/2-1, 10) in new stack -- Set Response Timeout to 10 -- Executing SetMusicOnHold(Zap/2-1, default) in new stack -- Executing SetCallerID(Zap/2-1, ) in new stack -- Executing SetCIDName(Zap/2-1, ) in new stack -- Executing BackGround(Zap/2-1, thank-you-for-calling) in new stack -- Playing 'thank-you-for-calling' (language 'en') -- Redirecting Zap/2-1 to fax extension == Spawn extension (mainmenu, fax, 0) exited non-zero on 'Zap/2-1' -- Executing Goto(Zap/2-1, fax|699|1) in new stack -- Goto (fax,699,1) -- Executing Macro(Zap/2-1, faxreceive) in new stack -- Executing SetVar(Zap/2-1, FAXFILE=/var/spool/asterisk- fax/1087447538.0.tif) in new stack -- Executing SetVar(Zap/2-1, [EMAIL PROTECTED]) in new stack -- Executing RxFAX(Zap/2-1, /var/spool/asterisk-fax/1087447538.0. tif) in new stack Changed from phase 0 to 1 Slow carrier up Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up DCS: 83 00 c6 f0 80 80 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Minimum scan line time for higher resolutions: T15.4 = T7.7 Get at 9600 Changed from phase 3 to 5 Fast carrier up Coarse carrier frequency 1832.96 (4) Training error 927.702492 Training failed (convergence failed) Fast carrier training failed -- Executing System(Zap/2-1, /usr/local/sbin/mailfax /var/spool/ asterisk-fax/1087447538.0.tif [EMAIL PROTECTED] ) in new stack TIFFOpen: /var/spool/asterisk-fax/1087447538.0.tif: Cannot open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com
Re: [Asterisk-Users] RxFax - Fast carrier training failed
up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up DCS: 83 00 c6 f0 80 80 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Minimum scan line time for higher resolutions: T15.4 = T7.7 Get at 9600 Changed from phase 3 to 5 Fast carrier up Coarse carrier frequency 1654.21 (4) Fast carrier down Fast carrier up Coarse carrier frequency 1700.11 (66) Training error 6.866837 Training succeeded (constellation mismatch 13.592167) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 2 Start rx page CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Fast carrier down Fast carrier up Coarse carrier frequency 1700.11 (66) Training error 15.050116 Training succeeded (constellation mismatch 18.914549) Fast carrier trained Page 1 of /var/spool/asterisk-fax/1087533965.3.tif: 2316 rows received 0 total bad rows 0 max consecutive bad rows Rx page end detected Changed from phase 5 to 3 Slow carrier up Slow carrier down Slow carrier up EOP: 2f EOP with final frame tag In state 5 Changed from phase 3 to 4 MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up DCN: fb DCN with final frame tag In state 8 Disconnecting Changed from phase 3 to 7 Changed from phase 7 to 8 -- Executing System(Zap/2-1, /usr/local/sbin/mailfax /var/spool/ asterisk-fax/1087533965.3.tif [EMAIL PROTECTED] ) in new stack ESP Ghostscript 7.07.2: Unrecoverable error, exit code 1 On Thu, 2004-06-17 at 13:33 -0700, Mike Benoit wrote: Good question Steve. My setup is basically: Fax Machine - PSTN - X100P - Asterisk - RxFax I'm not even sure, does Asterisk do encoding if its not sending the call to a SIP device, or over IAX? In the mean time I configured Asterisk to send faxes to a SIP extension (SPA-2000) and it receives faxes just fine. Though for some reason the fax machine doesn't know the line is ringing, but if I manually answer, the fax comes through just fine. This setup was: Fax Machine - PSTN - X100P - Asterisk - SPA-2000 (alaw) - Fax Machine One other thing to note, I have two FXO cards in the Asterisk box, connected to two PSTN lines, I dialing both numbers to send the fax, and the Coarse carrier frequency varied quite a bit between both lines. One line as I showed you was 1832 and the other line was 1785.79. Both lines resulted in the same outcome though, no fax was received. On Fri, 2004-06-18 at 00:25 +0800, Steve Underwood wrote: Hi Mike, To get something like: Coarse carrier frequency 1832.96 (4) Training error 927.702492 Training failed (convergence failed) something is horribly wrong. The carrier should be 1700Hz, not 1832.96Hz :-) Do you have a codec mismatch, or are you using a codec other than u-law or A-law? Sometimes the slow control messages will get through the wrong modem, but the fast modem for the images never will. Regards, Steve Mike Benoit wrote: I'm trying to send a fax to my asterisk box, however shortly after connecting the fax machine reports a communication error and hangs up. Below is the error I get from RxFax: Fast carrier training failed Nothing is written to the file system as far as the tiff is concerned. Any ideas on how to fix this? Thanks Sending Fax Machine: HP PSC 2000 Series 2210 Printer/Fax/Copier/Scanner Asterisk: CVS-HEAD-06/14/04-15:43:15 SpanDSP: spandsp-0.0.1k FXO Card: X100P Interesting zapata.conf lines: --- echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=10.5 txgain=-4.5 faxdetect=incoming Log: --- Jun 16 21:45:40 NOTICE[1199700912]: callerid.c:239 callerid_feed: Caller*ID failed checksum Jun 16 21:45:43 NOTICE[1199700912]: chan_zap.c:4951 ss_thread: Got event 2 (Ring/Answered)... -- Executing Answer(Zap/2-1, ) in new stack -- Executing Wait(Zap/2-1, 1) in new stack -- Executing DigitTimeout(Zap/2-1, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(Zap/2-1, 10) in new stack -- Set Response Timeout to 10 -- Executing SetMusicOnHold(Zap/2-1, default) in new stack -- Executing SetCallerID(Zap/2-1
[Asterisk-Users] Asterisk Management Interface... Do you want one?
Hi All, I realize there has been much talk about Asterisk web interfaces in the past, and there are even a few created such as vmail.cgi and Astweb. However, coming from the perspective of trying to convince upper management of a large multinational company to convert its entire phone system over to Asterisk, the question keeps coming up... Is it easy to manage, configure, and monitor in real-time? I think Asterisk's .conf files are pretty straight forward, but they still require that a person knows what they are doing. In a large company, things like changing passwords on voice mail boxes, changing greetings, monitoring support center call volume, viewing CDR logs and such is all very important. Having some sort of GUI interface to do this is essential. So, my question to the list is this: If there was a first class web based product that offered complete Asterisk (single and multiple server) configuration and monitoring features, together with fine grained access control so any user could login and carry out specific actions, would you be willing to pay for such a beast, and how much? (specify currency please. ;) Additionally, what would be the killer feature(s) that you feel this product should have? -- Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
