[Asterisk-Users] Sip One way audio
I've got a telecommuter working out of her home office, using a Snom 200 phone, what happens is occassionally her phone will loose audio one way.She will be talking on a call that was incoming to her extension, and all of a sudden the caller can not hear her any longer, she can here the caller fine, this has happened with both Sip-Sip calls, and calls that have come in over our PSTN circuits. The really odd thing is while troubleshooting with her yesterday I was using the one way audio to talk to her and do some packet captures, and she was using an instant message client to communicate back to me, but after being in the call for a while (didn't note exact times) the audio came back. At first I thought this was a nat issue, and she is using Bellsouth DSL, so I had her change the dsl modem so it shares its IP address with the phone. Restarting the phone results in the phone getting the public IP address assigned via DHCP. This did not solve the issue. I've experimented with the nat settings, and the canreinvite settings but haven't had much sucess so far. I have suspicions that the cut-outs might be occuring either after the phone has been registered for a certain amount of time (possibly 1 hour) or when she has been talking for a certain amount of time (possibly 5 minutes), I'm not certain of that behavior so it may be a red herring further use of the phone will allow me to firm up if either of those statements is true. Any suggestions would be greatly appreciated!Thank YouPaul M. OsterHere are the relevant portions of my sip.conf file...[general]port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind tocontext = incomingcall ; Default for incoming callstos=lowdelaydisallow=allallow=alawallow=gsmallow=ulaw [104]accountcode=vsllctype=friendcontext=employeeusername=104secret=**redacted**host=dynamicqualify=yesreinvite=nocanreinvite=no[EMAIL PROTECTED],[EMAIL PROTECTED] callgroup=1pickupgroup=1dtmfmode=rfc2833;nat=no ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 911 Service Providers
Who is everyone contracting with for 911 services with the upcoming FCC deadline? I've got a few feelers out there working on this issue, but no real solid leads yet. ___ 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] Automon Question
I've got automon up and recording calls on demand from information I found in the list archives, however instead of ending up with one monolithic file, I've got a -in and -out version of the files in my monitor directory? Anyone have suggestions how I could end up with a monolithic file that does what I want? ___ 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] Re: Automon Question
Doh, I should have kept trying for about 10 minutes longer before I sent that email, the trick is to ensure you have sox (and possibly soxmix) installed on the Asterisk box. figured I'd answer my own question should someone else need the answer or possibly just for the next guy searching the archives! On Tue, 15 Mar 2005 20:39:51 -0600, Paul Oster [EMAIL PROTECTED] wrote: I've got automon up and recording calls on demand from information I found in the list archives, however instead of ending up with one monolithic file, I've got a -in and -out version of the files in my monitor directory? Anyone have suggestions how I could end up with a monolithic file that does what I want? ___ 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] Passing additional information to an AGI via a call file
I have a desire to incorporate asterisk into some of my network monitoring. I would like to use the outgoing calling features to connect a phone (on-call cell phones) to an agi script which can provide some information to the called party. Ultimately I would like to pass 2 pieces of information to the agi, first an integer that is representative of a problem, and second a unix timestamp of the time the problem occurecd, so that my AGI could string together the phrases Error 17 Occured on Monday Feb 28 2005 at 10:59 a.m. The actual agi is pretty much done I just need to figure out how to pass the two new pieces of information to it from the .call file. Any suggestions, or resrources I should be reading? ___ 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] Local Outbound Calls on PRI
I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had a central office tech from the CLEC insert a monitor in the distribution point on the switch and observe call flow. According to the tech call flow appears proper, and he was able to tell me the number I was calling from and the number I attempted to dial. He then placed a PRI test-set at the distribution point in the switch and successfully made and terminated a variety of calls from that point. He then took the PRI test set out to my physical location and did the same test, made and received local and long distance calls on one of my trunks. After thinking about this last night I decided to re-update my Asterisk installation to the latest bleeding edge CVS version and re-test from my test extension with the exact same results. I successfully make long distance calls, successfully receive any calls, but the local calls originated from the SIP phone (SNOM200 and Mediatrix2102) fail with a busy signal that seems to originate from the CLEC's switch. Any suggestions? Thanks in advance. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Local Outbound Calls on PRI
Just tried it with 7,10, and 11 digit dialing, and got the expected error from the switch, the number you have dialed is not a long distance number, there is no need to dial the digit one before the number... Good suggestion, but that doesn't appear to be the problem. On Fri, 24 Sep 2004 11:25:12 -0400, Scott Lykens [EMAIL PROTECTED] wrote: On Fri, 24 Sep 2004 10:06:06 -0500, Paul Oster [EMAIL PROTECTED] wrote: I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. Any chance you're sending more or fewer digits than the CLEC expects to see from you? I would expect that the tech would have picked up on it when he was watching your call attempts but just in case... ___ Asterisk-Users mailing list [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] Desired Install in MotorHome
I've got a client who would love to be able to leave an asterisk server running sompelace, and achieve telephone connectivity using an IP phone from within his Motorhome in his words I want to be able to work from a mountaintop in Glacier National Park I've done some initial testing, and a SNOM200 SIP phone comes close enough to working that I have not ruled out the idea as completely un-workable. I understand that this is an extreamly hostile environment, the satelite uplink itself introduces too much latency for a standard configuration to work (1500ms) which is most likely where the problems come from. What I'm wondering is if anyone has ever succeded in making a setup like this work. Different protocol (H.323 MCGP etc) or different codecs? Our testing in his driveway revealed the following. 1. Incoming calls achieve a ring almost immediately, when answered there is 1 to 2 seconds delay in the conversation (like a really poor trans-atlantic call) 2. Outgoing calls fail... the phone returns Proxy Authentication Required however a few seconds later when the handset is picked up the inbound leg of the RTP stream is preseant on the phone. Outgoing audio is non-existant 3. Attempts to disconnect from that audio stream fail, the * server is simply not seeing the hangup from the phone. So whats everyones opinion, worth exploring further, or am I wasting my time trying? Thanks in advance. Paul M. Oster ___ Asterisk-Users mailing list [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] Desired Install in MotorHome
Oh, now thats an idea... I'll have to dig up a spare PC and try it... he's got 50amps available, why not burn it :) On Wed, 28 Jul 2004 16:12:52 + (UTC), Peter Corlett [EMAIL PROTECTED] wrote: Paul Oster [EMAIL PROTECTED] wrote: I've got a client who [wants VoIP working over a very high-latency link]. So whats everyones opinion, worth exploring further, or am I wasting my time trying? Can you stick an Asterisk box at his end so you can speak IAX over the link? It may not help with the massive delays (which is going to be inherent in any kind of VoIP over the link) but the signalling should be a lot more reliable. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue Statistics Line
Does anyone know what the various statistics in this line mean? I'm curious what the C:3, A:3 and SL:0.0% withing 0s parts stand for and I have not been able to find any references to it by using search engines. main has 0 calls (max unlimited) in 'rrmemory' strategy (3s holdtime), C:3, A:0, SL:0.0% within 0s Thanks in advance Paul p.s. That line comes from show queues... ___ Asterisk-Users mailing list [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] Random Dropped Called
I've got a 4 port T1 card in my Asterisk box with a PRI from Qwest as my PSTN interface. I'm experiencing random dropped calls on the various SIP devices I have tested. Network connectivity to the SIP devices looks ok, and I have tried a variety of the devices including all of the following. Grandstream 286 Grandstresm 486 Sipura SPA 1000 Mediatrix 2102 Some example lines from my logs which may indicate a problem Jul 15 15:32:41 WARNING[11276]: PRI: !! Got reject for frame 30, retransmitting frame 30 now, updating n_r! Jul 15 17:03:20 WARNING[11276]: PRI: !! Got reject for frame 95, but we only have others! Jul 15 17:07:44 WARNING[11276]: PRI: !! Got reject for frame 124, retransmitting frame 124 now, updating n_r! Jul 15 17:07:44 WARNING[11276]: PRI: !! Got reject for frame 124, retransmitting frame 125 now, updating n_r! Jul 15 17:11:56 WARNING[11276]: PRI: Read on 66 failed: Unknown error 500 Jul 15 23:08:37 WARNING[5126]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 30406 (Response) Jul 16 05:39:08 NOTICE[11276]: PRI got event: 8 on span 1 Jul 16 06:25:04 NOTICE[5126]: Request to schedule in the past?!?! Jul 17 14:43:43 WARNING[11276]: Ring requested on channel 1 already in use on span 1. Hanging up owner. This issue has had me baning my head on my desk for weeks, any information that you may have that could clear this up will be much appreciated. --Paul M. Oster [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] Snom 200 MWI Button
I'm trying to get the MWI button to work with my Asterisk configuration. The snom is accepting and responding to the Message indications from *, but when I press the MWI button, it is dialing my extension (the one with the voice mail on it). I'm wondering if there is a way to specify what extension to dial to check email in the configuration, either the phone, or * itself. Asterisk Version 1/30/2003 checked out and compiled this evening Snom Version 2.03o (most recent auto-update) Any help would be greatly appreciated. At one point Mark had talked about adding a voicemail= directive in sip.conf on the mailing list at one point, however grepping the code doesn't reveal a feature like that at this time. Anyone have success in getting the MWI button to work on Snoms? If so I would LOVE to hear from you. Paul M. Oster
[Asterisk-Users] Dial / Ring multiple sip channels
I know I can dial multiple channels in sequence exten = 101,1,Dial(SIP/101,10) exten = 101,2,Dial(SIP/102,10) extne = 101,3,Dial(Zap/1/5551212) What the boss would really like is to be able to ring 2 lines simultaneously. exten = 101,Dial(Sip/101,10) Dial(Sip/102,10) so that both extensions ring at the same time... mostly so that he can have the remote phone at his house ring at the same time as his office phone is someone dials his extension or direct line. Is this possible with *, and what would the dialplan look like? I've been googling for about 30 minutes now and haven't seen an example config in the list. Would setting up a queue for the extension I want to multi-ring, and having the 2 phones be permanent members be a better soloution? Paul Free 20 MB Bannerless Domain Hosting, 1000 MB Data Transfer 10 Personalized POP and Web E-mail Accounts, and more. Get It Now At Doteasy.com http://www.doteasy.com/et/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channelbank Recomendation and GS102 question
Hi All. I'm working on an * configuration. We require 8 inbound POTS lines, and CT1 or PRI seems like it will be quite expensive at that level. I've read that a T1 Channelbank plus the T100P would be a (the?) way to go for this situation. What is the recommended channelbank for use in this scenario? From searching the archives I see a lot of suggestions to get a channelbank from ebay. I would prefer to be able to use new products so I can easily duplicate the setup for other branch offices in my company. My second question relatees to the Grandstream phones. When they are a member of a queue group, I get a loud annoying ring in the handset when its in use and another call comes in on the queue. Is there a way to enforce 1 call per phone in sip.conf? Either that or a way to tell the GS102 to return busy when * trys to send them a call. Thanks in advance. Paul M. Oster Free 20MB Web Site Hosting and Personalized E-mail Service! Get It Now At Doteasy.com http://www.doteasy.com/et/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Asterisk overwrite any libraries?
Looks like your box has been compromised. Try ls -l `which ps` You'll probably find an inapropriate date. Whenever I've diagnosed problems like this, I've found badly installed rootkits. To address this on my production machines, I'm going to insruct the router to only allow traffic that is coming from trusted locations to connect to the box anyplace. I really hope I'm wrong about this Costas, but you should probably start verifying your binaries. If your machine has been compromised, a clean install, and patch with all the updated RPMS is a recommended soloution. Paul costas wrote: I am using a brand new RH9.0 installation. I installed Asterisk afterwards so I am not sure if Asterisk caused the problem below. The ps doesn't work. It could also be something else. I also tried installing a some video package. But I thought to ask here first if someone has seen this before. [EMAIL PROTECTED] asterisk]# ps ps: error while loading shared libraries: libproc.so.2.0.6: cannot open shared object file: No such file or directory [EMAIL PROTECTED] asterisk]# which ps /bin/ps Thanks Costas -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Free 20MB Web Site Hosting and Personalized E-mail Service! Get It Now At Doteasy.com http://www.doteasy.com/et/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
