[Asterisk-Users] I give up!!
i've just lost $2000 dollars or so on my first commercial asterisk installation .. i'm running a PIV class server, three Digium Wildcard FXO cards, and 10 Grandstream Budgettone SIP phones. The system was to be a PBX for a small company. After over 2 months of pissing about, the client has had his fill of asterisk problems, and asked me to take my equipment out of the building. Obviously, I haven't been paid for anything. The problems I faced were the following : - initially a problem with asterisk crashing totally when there wasn't an extension to ring .. though this was fixed in a subsequent CVS, it was causing downtime. the client has no unix knowledge, and a script I put in to kick in the asterisk when it shut itself down didn't seem to always work. it also reduced the quality of my subsequent callout requests to something on the lines of the phone server is crashed again regardless of what the problem was - a dialplan problem, where one phone was ringing 10 seconds after the others, at the client's request and they were hearing other phones ring and picking up a non-ringing phone (ok, I can't really blame that on asterisk ..) - echo on the lines .. that after much fiddling around with configurations went from terrible to borderline acceptable. To people not used to digital telephony and computer stuff, the echo was VERY annoying. They used to avoid the phones because they said people would not understand them. - no consultative transfer. The closest I got was to park the call, call the other party, tell him a voce which line the call is parked on and then get him to pick up the call. This is, in my opinion, a very basic feature that is missing on asterisk. The park/ pick up sequence proved too difficult for the clients' secretaries to grasp. - I could not get G729 working properly (license paid up, G729 up and running). In the absence of a manual, the fault solving process was something like ask a question on the mailing list, get a few answers, go to the client, try it out, fail, go back home, send another question on the mailinglist with about 48 hours for each iteration. I was also appearing a real chimp expermimenting stuff at the clients' office. At this point I decided to cut my losses, retreive the equipment and call it a day. When asterisk is well documented and released in stable releases, I will willingly consider it again. I would be willing to pay for a stable, documented version of asterisk. It is a lovely software, and to begin with I was very enthusiastic about it. I do understand that the support community is helpful, but the current status of things limits asterisk to a hobbyist scenario or at least somewhere where there is an engineer with lots of linux experience and patience online 24 hours to solve problems as they crop up. If anyone would like a couple of second hand FXO boards, contact me. I have already found a home for the grandstreams. cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Budgettone + G729
hi there .. I asked sometime ago regarding getting a Budgettone working with Asterisk over G729. My system is quite simple, Asterisk server with 1 G 729 license installed, and 10 Grandstream phones. Only one of them needs G729, because it's on a remote link via an ADSL bridge. The rest run happily on G711 on a local network. I added the lines disallow=all allow=g729 to the sip.conf entry for that particular phone, but it doesn't seem to have made any difference. If I remove the allow = g729 line, which supposedly should leave no codecs at all, the phone still works, on G711. With the allow=g729 line in place, and switching off all codec options except g729 from the phone, when I try to for example dial extension 1000, which should give me Asterisk's welcome message, I get a message about no matching codecs on the asterisk console, so obviously the g729 codec isn't working. Can somebody help? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgettone + G729
many thanks! it's all pretty much a hit miss process becaues of Asterisk's notorious lack of documentation.. I will try this out at my client tomorrow, fingers crossed. cheers Dave - Original Message - From: T Aksoy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 03, 2003 1:25 PM Subject: RE: [Asterisk-Users] Budgettone + G729 Hi, Unless there has been a recent change, you can't set codecs in the sip.conf on a per-context basis. The way to do what you want is to have the following in the [general] area: disallow=all allow=ulaw allow=alaw allow=g729 Then, set all the codec preferences on the g729 phone to g729. That phone should then only be able to negotiate g729. If this doesn't work then I suspect some sort of g729 codec installation error? Tan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dave Alan Caruana Sent: 03 October 2003 11:49 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Budgettone + G729 hi there .. I asked sometime ago regarding getting a Budgettone working with Asterisk over G729. My system is quite simple, Asterisk server with 1 G 729 license installed, and 10 Grandstream phones. Only one of them needs G729, because it's on a remote link via an ADSL bridge. The rest run happily on G711 on a local network. I added the lines disallow=all allow=g729 to the sip.conf entry for that particular phone, but it doesn't seem to have made any difference. If I remove the allow = g729 line, which supposedly should leave no codecs at all, the phone still works, on G711. With the allow=g729 line in place, and switching off all codec options except g729 from the phone, when I try to for example dial extension 1000, which should give me Asterisk's welcome message, I get a message about no matching codecs on the asterisk console, so obviously the g729 codec isn't working. Can somebody help? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call parking -- what was the key combination?
hi great gurus of asterisk :) could somebody remind me the key combination to send a call into the parking queue ? while you're at it, are there any other key combinations I should know?? eg. put a call on hold etc. thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call parking -- what was the key combination?
what i'm asking is what is the key sequence you have to dial for the transfer .. it was something like *7# if I remember well, I know I had it working, but the client lost the paper I wrote it on for him, and I can't trace the email I got it from! cheers Dave - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 05, 2003 3:11 PM Subject: Re: [Asterisk-Users] call parking -- what was the key combination? To park a call you simply transfer the call into extension 700 (this is the default and can be changed).. To get the call back you just dial the parked location.. If you are using an IP phone this is a problem becasue it will not tell you the location of the parked call so you will not know where to collect it from.. hi great gurus of asterisk :) could somebody remind me the key combination to send a call into the parking queue ? while you're at it, are there any other key combinations I should know?? eg. put a call on hold etc. thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RedHat Distribution
Redhat 9 works fine unless you really need G729 working on H323 in which case the only solution seems to be chanh323, which only works with G729 support on Redhat 8 .. I found out the hard way :) cheers Dave - Original Message - From: Ernest W. Lessenger To: [EMAIL PROTECTED] Sent: Tuesday, September 02, 2003 5:45 PM Subject: Re: [Asterisk-Users] RedHat Distribution At 12:29 PM 9/2/2003 +0100, you wrote: I'm new in * and I would like to know what version of the Linux kernel or RedHat Distribution do you recomend.Redhat 9 works perfectly. Install with the kernel sources and devel libraries, and the developers software, i.e. gcc, and upgrade to most recent rpms before making asterisk.--Ernest
[Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. I already have the G729 codec installed, and configured with 1 license. Can anyone give me the correct sip.conf commands (or whatever I need) to get the budgettones working over G729? many thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
no .. flash key can do a blind transfer, and that's about it. the only way to do a consultative transfer (ie. speak to the person you are transferring to, and then transfer) is by parking the call .. i've heard that this is pretty much the definitive situation from what i've been reading on this list. if anyone knows better, i'd be happy to know! cheers Dave - Original Message - From: Daniel ANDRE [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 5:53 PM Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems .. Hello, Have you succeded to use flash key to do call transfert? Regards, Daniel Dave Alan Caruana a écrit: well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. I already have the G729 codec installed, and configured with 1 license. Can anyone give me the correct sip.conf commands (or whatever I need) to get the budgettones working over G729? many thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
I was referring to the transfer button (sorry) what again is the way you are using to transfer calls ? so far what i'm doing is after accepting a call, parking it .. then phoning the guy who wants the call and telling him the call is parked on 701 for example .. cheers Dave - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 7:08 PM Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems .. Parking the call is a problem becasue you will not hear the parked call location (because its a blind transfer into the parked call).. The only solution I could think of is to call the person you want to transfer to on the second line, then go back to the first line and blind transfer the call.. (the person you are transfering to will have to hang up after you have spoken to them) What is the process for transfering with the flash button?? I have always used the transfer button and the redial/send button.. no .. flash key can do a blind transfer, and that's about it. the only way to do a consultative transfer (ie. speak to the person you are transferring to, and then transfer) is by parking the call .. i've heard that this is pretty much the definitive situation from what i've been reading on this list. if anyone knows better, i'd be happy to know! cheers Dave - Original Message - From: Daniel ANDRE [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 5:53 PM Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems .. Hello, Have you succeded to use flash key to do call transfert? Regards, Daniel Dave Alan Caruana a écrit: well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. I already have the G729 codec installed, and configured with 1 license. Can anyone give me the correct sip.conf commands (or whatever I need) to get the budgettones working over G729? many thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
has anyone got G729 and SIP working together? some config examples would help :) since I need to do this at a client where I don't really have internet access, or the will to root around mailing lists with the client breathing down my neck! thsnk Dave - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 7:13 PM Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems .. now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. Rememeber G.711 is 64Kbps without overhead.. Actual bandwidth required is somewhere around 87Kbps.. On a 128Kbps line if anything else uses the line you will get clicks and breaks in the transmission.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems ..
it's 512k/128k actually ... :) Dave - Original Message - From: Ing. Angel Gomez [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 9:23 PM Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems .. WipeOut . wrote: now .. i have one slight problem left .. although most of my SIP phones are on a LAN connection with the asterisk server, there are two phones which are at a remote office bridged to my LAN via a 128k point to point ADSL .. these do not seem to be working well, you do hear speech but the remote person (dialled over PSTN through an X100P) hears it low and garbled .. I am assuming it's due to the delays in stuffing 64kbits (of g711) over a 128k link and was thinking of switching to G729. Rememeber G.711 is 64Kbps without overhead.. Actual bandwidth required is somewhere around 87Kbps.. On a 128Kbps line if anything else uses the line you will get clicks and breaks in the transmission.. Mmmmh, Does not ADSL stands for ASIMETRIC Digital Subscriber Line ? So if its a 128 Kbps ADSL... what is your Asimetry relation ? You might be having 128 Kbps in one way but 64 Kbps ( Or less ) on the other... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and ECHO
I tried specifying rxgain txgain, copied the format some some message on asterisk-users Result was asterisk bombed out didn't even load due to not being able to understand the config file .. what's the exact syntax that works?? cheers Dave - Original Message - From: Fredrik Hedberg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 02, 2003 9:21 AM Subject: Re: [Asterisk-Users] SIP and ECHO What have you specified as rx and txgain? Simon McAuliffe wrote: I've been having the same problem too, except for me it only happens occasionnally. I'm not 100% sure of this, but it seems that for very local calls (eg across the city) I never get echo. For calls that go longer distance (say 500km or more), or through some closer call centres, I'm getting the echo. I don't get the echo on an analogue POTS connection to the same places (it is clearly only happening on our asterisk system). This might indicate some link between echo cancellation and delayed audio, but if so, its sensitive to very small delays. The echo can only be heard at our end, there is no trace of it at the other end. I'm using ATAs doing SIP to Asterisk and through a PRI connection to a Telco. Echo cancellation is turned on and showing as activated on the Zap channels. Echo cancellation is also enabled on the ATAs. - Original Message - From: Brian J. Schrock [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 29, 2003 3:16 AM Subject: [Asterisk-Users] SIP and ECHO Hello, I have read the information on echo and SIP in the FAQ and I have scoured the mailing list for possible solutions, but as yet I have not been able to get rid of this echo. I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed into an asterisk server. If I call between the Sip Phone (Budgettone-100) and the 4 FXS ports everything sounds great. If I call out to the PSTN through the FXO cards I get horrible echo, I have even been able when talking loud enough to get a horrible feedback loop going. I have tried 4 different echo cancellers in the Makefile for the Zap drivers and nonoe of them changed the situation. I have echocancel = (Any where from 1 - 256, I have tried alot of different values), and I have echocanelwhenbridged = yes.I only hear the echo start when the call gets bridged onto the outgoing PSTN lines. Is there anything I can do? Brian J. Schrock ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Fredrik Hedberg Telavox ABDirect: +46 46 6220013 Lilla torg 1Phone: +46 46 622 S-211 34 MalmoMobile: +46 70 3323033 SwedenWeb: www.telavox.se ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and ECHO
hi .. i have the exact same problem you have .. seems to be related to Budgettone phones in my prob. I *tried* selling an asterisk exchange to a client and today he phoned telling me he is very unsatisfied I risk being thrown out .. suggestions would be welcome! i've tried *everything* that has gone in your correspondence on the list, and a few of my own .. no luck! seems a hardware problem. arghhh!!! the other budgettone problem is it won't do a consultative transfer ie. you answer an incoming call, speak to someone else on a different extension and then pass the call .. only way I have found to "emulate" that is using call parking which is VERY messy!! well .. maybe it's consoling to know you are not alone! cheers, Dave - Original Message - From: Daniel ANDRE To: [EMAIL PROTECTED] Sent: Friday, August 29, 2003 5:56 PM Subject: Re: [Asterisk-Users] SIP and ECHO Hello,Brian West a écrit: I get no echo on mine.. but you can check to make sure your line isn't reversed. A reverse wired jack can do that.I don't think so but I have tested reversed and it doesn't solve my echo problemDaniel bkw On Thu, 28 Aug 2003, Brian J. Schrock wrote: I can minimize doing those tricks, but I cannot seem to get it to go away. On Thu, 2003-08-28 at 11:33, Dan wrote: - Original Message - From: "Brian J. Schrock" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, August 28, 2003 6:16 PM Subject: [Asterisk-Users] SIP and ECHO Hello, I have read the information on echo and SIP in the FAQ and I have scoured the mailing list for possible solutions, but as yet I have not been able to get rid of this echo. I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed into an asterisk server. If I call between the Sip Phone (Budgettone-100) and the 4 FXS ports everything sounds great. If I call out to the PSTN through the FXO cards I get horrible echo, I have even been able when talking loud enough to get a horrible feedback loop going. I have tried 4 different echo cancellers in the Makefile for the Zap drivers and nonoe of them changed the situation. I have echocancel = (Any where from 1 - 256, I have tried alot of different values), and I have echocanelwhenbridged = yes.I only hear the echo start when the call gets bridged onto the outgoing PSTN lines. Is there anything I can do? Brian J. Schrock Hi, For me: rxgain=0.8 txgain=0.8 in zapata conf do the trick. Now the echo is allmost inexistant. Maybe the sound is not very strong but the quality is very good. I have the default echo canceller (no modification in the source files). Tested with a lot of SIP phones (ATA (G.711), X-Lite(GSM), SJ_phone(G.711), Cisco 79x0) and one X100P card. BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com
Re: [Asterisk-Users] Why doesnt anyone reply me ?
maybe because your email seems ot be encoded within an attachment? try sending plaintext! cheers Dave - Original Message - From: Armand A. Verstappen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 25, 2003 7:02 PM Subject: Re: [Asterisk-Users] Why doesnt anyone reply me ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Grandstream power supplies..
anything that supplies a reasonably straight 5v should work .. do not send *more* than that into the phone which is what most unregulated supplies will do .. the supply which comes with the grandstream seems to be a nice switchmode one. cheers Dave - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 12, 2003 2:09 PM Subject: [Asterisk-Users] OT: Grandstream power supplies.. Hi, Quick question to all the electronics gurus out there.. I unpacked my second GS phone yesterday (had it for about a month!) and set it up.. This morning the power supply is dead.. I have looked for a new one online (In the UK using Maplin let me know if you know a better place.) becasue it would probbaly take too long to get one sent from China or the US and I need to get that station operational again.. There seem to be many choices for power supplies.. Looking on the bottom of the broken one it is a 5VDC 400mA output.. When I looked online for a new one the choices are for regulated, unregulated and switch mode power supplies with the regulated and switch mode ones being VERY much more expensive than the unregulated ones.. Which kind would do the job?? Also most of the variable output adapters are 4.5v or 6v, Not the required 5v.. Any help would be appreciated.. Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers
my error .. the cards are X100P which is why I wrote FXO. The Grandstream phones are on a LAN, the * server connects to the phonelines via the X100P cards. When I call from the Grandstream phones onto the PSTN there is a VERY big amount of echo, ie. I can hear myself in the earpiece. cheers Dave - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 05, 2003 8:50 AM Subject: Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. The TDMx00P cards are FXS cards.. :) 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3 seconds delayed to the speech .. there is no echo on incoming voice, just an echo of my own voice as I speak. What are you using to connect to the PSTN?? X100P, T100P, E100P, I4L, Chan_Capi 2nd question: using a grandstream phone asterisk, if I hear another phone ringing, how can answer it from the phone infront of me? eg. if extension 6003 is ringing, and i have phone number 6004, how can I answer it ? You need to setup call groups, search through the archives cos I rememeber a thread on this a short while ago.. 3rd question: can someone give me some starter hints to configure call parking ? I haven't managed to find a direct way to transfer a call from phone to phone except using blind transfer and I want the person initiating the transfer to speak to the receiving person before actually passing the call. As far as I know there is no facility to do a consultative transfer on the GS phones.. Only a blind transfer.. Maybe it will come later.. can anybody help please ? cheers Dave A Caruana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers
could you send me the exact syntax for rxgain / txgain? I think that might help towards my problem becuase i'm having to turn the handset volume all the way up .. thanks Dave - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 05, 2003 9:45 AM Subject: Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers my error .. the cards are X100P which is why I wrote FXO. The Grandstream phones are on a LAN, the * server connects to the phonelines via the X100P cards. When I call from the Grandstream phones onto the PSTN there is a VERY big amount of echo, ie. I can hear myself in the earpiece. cheers Dave An echo at the begining of a call is normal as the * and phone trains themselves but this should dissappear after about 30 seconds to 1 min.. So my only suggesttions are.. First make sure you have echocancel=yes and echocancelwhenbridged=yes in your zapata.conf.. If that doesn't help try lowering the volume on the sip handset and play with the rxgain= and txgain= in zapata.conf for the X100P's.. Other than that I don't really know what else you can try.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Issue
you have to make /etc/zaptel.conf and /etc/asterisk/zapata.conf match on the same type of signalling .. should work then :) cheers Dave - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 08, 2003 12:24 AM Subject: [Asterisk-Users] Newbie Issue Hi All, I recently purchased the Asterisk Developer's Kit (TDM) to try out Asterisk. After following the directions in the Digium's FAQ topic entitled Q. How do I configure my TDM40B and X100P?, I'm receiving the following error: WARNING[1074428608]: File chan_zap.c, Line 6748 (load_module): Ignoring rxwink ERROR[1074428608]: File chan_zap.c, Line 6692 (load_module): Unknown signalling method 'fxs_ks # X100P' ERROR[1074428608]: File chan_zap.c, Line 4793 (mkintf): Signalling requested is FXO Loopstart but line is in FXS Kewlstart signalling ERROR[1074428608]: File chan_zap.c, Line 6498 (load_module): Unable to register channel '1' WARNING[1074428608]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1074428608]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! Any ideas? Jeff Gunther Intalgent Technologies voice: +1 703.444.4404 fax: +1 703.444.2304 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers
I tried putting in txgain=100% rxgain=100% and zaptel wouldn't load telling me I had wrong parameters in my zaptel.conf i'll try again with txgain=5.0 but my setup is at a client so each time a day passes and i have to go round to the client just to try things out ... it's a bit annoying! my 2c .. when is there going to be some concerted effort at documenting some stuff? today I discovered by change that you can dial # to transfer to extension .. surely these are stuff that could be put down in writing somewhere ? cheers Dave - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 05, 2003 7:42 PM Subject: Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers Don't use %'s with txgain/rxgain for txgain=5% is equal to txgain=5.0 and that might be too much On Tue, 5 Aug 2003, WipeOut . wrote: could you send me the exact syntax for rxgain / txgain? I think that might help towards my problem becuase i'm having to turn the handset volume all the way up .. thanks Dave You can use either a percentage or a number IIRC.. Somthing like.. rxgain=5% txgain=5% or rxgain=0.4 txgain=0.4 and I thing that you can use negative values as well.. I am not sure what the minimum and maximum values are I use percntages.. Hope that helps.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers
hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3 seconds delayed to the speech .. there is no echo on incoming voice, just an echo of my own voice as I speak. 2nd question: using a grandstream phone asterisk, if I hear another phone ringing, how can answer it from the phone infront of me? eg. if extension 6003 is ringing, and i have phone number 6004, how can I answer it ? 3rd question: can someone give me some starter hints to configure call parking ? I haven't managed to find a direct way to transfer a call from phone to phone except using blind transfer and I want the person initiating the transfer to speak to the receiving person before actually passing the call. can anybody help please ? cheers Dave A Caruana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP calls cause segmentation fault
does anyone of the programmers know if this has been fixed in a more recent CVS version? should I redownload and recompile? cheers Dave - Original Message - From: Adam Donnison [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 1:18 AM Subject: Re: [Asterisk-Users] SIP calls cause Segmentation Fault I actually found this same thing, and traced it down to app_dial.c line 190. It doesn't explicitly check for a valid chan before trying to use it and it segfaults when it does a strlen on a chan entity. I simply put a check in that winner was non-zero before comparing it to o-chan: if (winner winner == o-chan) Adam Dave Alan Caruana wrote: I have an asterisk installation at a client, it's quite simple. Basically it's an asterisk downloaded from CVS about a week ago, with 3 Zaptel FXO cards (the digium ones) and 10 Grandstream Budgettone SIP phones ... Every now and then, especially when a call is ringing and not picked up immediately, Asterisk quits with a segmentation fault error. IT seems quite inexplicable, my dialplan is a modification of the sample one that came with Asterisk, and I haven't touched that many other conf files actually. Any way I can get this debugged? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Adam Donnison email: [EMAIL PROTECTED] Saki Computer Services Pty. Ltd. 93 Kallista-Emerald Roadphone: +61 3 9752 1512 THE PATCH VIC 3792AUSTRALIAfax: +61 3 9752 1098 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP calls cause Segmentation Fault
Mark, the server has already been installed at a client and the only access to internet I have is from behind a NAT therefore I cannot give you access to log into the server. Also, I do not have an IRC client on the machine, and the closest windows machine is 4 floors away. What is the procedure to extract debug data I can send you please ? thanks Dave - Original Message - From: Mark Spencer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 31, 2003 8:11 PM Subject: Re: [Asterisk-Users] SIP calls cause Segmentation Fault Yes, find me on #asterisk so I can login. Be sure you're generating cores and running on very latest CVS. Mark On Thu, 31 Jul 2003, Dave Alan Caruana wrote: I have an asterisk installation at a client, it's quite simple. Basically it's an asterisk downloaded from CVS about a week ago, with 3 Zaptel FXO cards (the digium ones) and 10 Grandstream Budgettone SIP phones ... Every now and then, especially when a call is ringing and not picked up immediately, Asterisk quits with a segmentation fault error. IT seems quite inexplicable, my dialplan is a modification of the sample one that came with Asterisk, and I haven't touched that many other conf files actually. Any way I can get this debugged? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP calls cause Segmentation Fault
Thanks Adam, I will try it out. cheers Dave - Original Message - From: Adam Donnison [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 1:18 AM Subject: Re: [Asterisk-Users] SIP calls cause Segmentation Fault I actually found this same thing, and traced it down to app_dial.c line 190. It doesn't explicitly check for a valid chan before trying to use it and it segfaults when it does a strlen on a chan entity. I simply put a check in that winner was non-zero before comparing it to o-chan: if (winner winner == o-chan) Adam Dave Alan Caruana wrote: I have an asterisk installation at a client, it's quite simple. Basically it's an asterisk downloaded from CVS about a week ago, with 3 Zaptel FXO cards (the digium ones) and 10 Grandstream Budgettone SIP phones ... Every now and then, especially when a call is ringing and not picked up immediately, Asterisk quits with a segmentation fault error. IT seems quite inexplicable, my dialplan is a modification of the sample one that came with Asterisk, and I haven't touched that many other conf files actually. Any way I can get this debugged? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Adam Donnison email: [EMAIL PROTECTED] Saki Computer Services Pty. Ltd. 93 Kallista-Emerald Roadphone: +61 3 9752 1512 THE PATCH VIC 3792AUSTRALIAfax: +61 3 9752 1098 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP calls cause Segmentation Fault
I have an asterisk installation at a client, it's quite simple. Basically it's an asterisk downloaded from CVS about a week ago, with 3 Zaptel FXO cards (the digium ones) and 10 Grandstream Budgettone SIP phones ... Every now and then, especially when a call is ringing and not picked up immediately, Asterisk quits with a segmentation fault error. IT seems quite inexplicable, my dialplan is a modification of the sample one that came with Asterisk, and I haven't touched that many other conf files actually. Any way I can get this debugged? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stupid questions ..
Sip phones on the system are Grandstream Budgettone 100's. Was assuming it wouldn't be phone specific :) they have flash key which is meant to send a DTMF. thanks for the help with the dial string. Dave - Original Message - From: Low, Adam [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 29, 2003 11:28 AM Subject: RE: [Asterisk-Users] stupid questions .. 1. what's the sequence to press on a SIP phone to transfer a call to another extension. Which SIP phone? Soft/hard ? Phone specific ... 2. what's the same thing if you want to hold an incoming call, speak to the other extension, then pass the call? Which SIP phone? Soft/hard ? Phone specific ... 3. what's the extensions.conf syntax to dial two SIP extensions at once? Separate the dial peer with a as follows: exten = 13646,1,Dial(SIP/4840SIP/4841) many thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stupid questions ..
oh ok ;) just understood!! call transfer is something the phone does and asterisk picks up, not some sequence you send directly to asterisk, hence from the Grandstream manual :) thanks very much for pointing it out! cheers Dave - Original Message - From: Dave Alan Caruana [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 29, 2003 11:24 AM Subject: [Asterisk-Users] stupid questions .. just three stupid questions I need to ask .. 1. what's the sequence to press on a SIP phone to transfer a call to another extension. 2. what's the same thing if you want to hold an incoming call, speak to the other extension, then pass the call? 3. what's the extensions.conf syntax to dial two SIP extensions at once? many thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk user guide ..
Is there any such thing is a userguide for asterisk from an enduser point of view ie. what to do to transfer a call etc ? I've looked through all the official documentation and nothing exists, and trying to install an ASterisk at a client can't even explain how to transfer a call to another extension! Even some basic help would be welcome! cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] interfacing asterisk with a legacy PBX
hi .. i require to interface asterisk to a 60 line analog PBX in a hotel. I was thinking of giving Asterisk a couple of PBX lines interfaced through cards, and then place outgoing calls through SIP/H323 and a DSL connection. analog extension lines -- analog pbx --asterisk -- SIP -- termination I do not need incoming calls to the lines. My question is this : if I take 2 of the existing analog extension lines can I interface these through Wildcard FXO cards on the Asterisk? Is the signalling the same? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_H323, G729 (minor problem)
ok ... I removed the dtmfmode=inband from the h323.conf file which resulted in the error messages vanishing .. ya I thought ... alas DTMF tones sent to an IVR at the other end of the connection do not work either!!! My incoming calls are coming from PSTN lines through an E1 so DTMF must be inline .. THe (thousands of) error messages aren't really a problem, just annoying. Dave - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 15, 2003 4:28 PM Subject: Re: [Asterisk-Users] Chan_H323, G729 (minor problem) You're trying to detect inband dtmfs from the codec stream. Martin On Tue, 15 Jul 2003, Dave Alan Caruana wrote: hi .. I have finally managed to get Chan_H323 G729 working flawlessly, thanks to some help from Jerry McNamara. For those out there who are stuck with the same problem the procedure is : 1. install on RedHat 8.0 and nothing else (RH9 doesn't work!) 2. Install asterisk, zaptel etc. the normal way 3. Compile Pwlib oH323 with versions taken from nufone's site (http://www.nufone.net/downloads) since the latest versions do not have support for G729. Remember to set the environment versions as described in the Readme files. 4. Modify the makefile of chan_h323 (which is in /usr/src/asterisk/channels/h323) to re-enable the G729 code. 5. in h323.conf put in allow=g729 and you should have a working configuration .. now for my question .. during G729 calls I am getting repeatedly the message WARNING[311314]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames this scrolls up the screen at a very high rate of knots.. the call is unaffected and goes through normally. Is this something wrong? normal? can it be fixed/suppressed? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_H323, G729 (minor problem)
hi .. I have finally managed to get Chan_H323 G729 working flawlessly, thanks to some help from Jerry McNamara. For those out there who are stuck with the same problem the procedure is : 1. install on RedHat 8.0 and nothing else (RH9 doesn't work!) 2. Install asterisk, zaptel etc. the normal way 3. Compile Pwlib oH323 with versions taken from nufone's site (http://www.nufone.net/downloads) since the latest versions do not have support for G729. Remember to set the environment versions as described in the Readme files. 4. Modify the makefile of chan_h323 (which is in /usr/src/asterisk/channels/h323) to re-enable the G729 code. 5. in h323.conf put in allow=g729 and you should have a working configuration .. now for my question .. during G729 calls I am getting repeatedly the message WARNING[311314]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames this scrolls up the screen at a very high rate of knots.. the call is unaffected and goes through normally. Is this something wrong? normal? can it be fixed/suppressed? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 + G729 + Go2Call
I am trying the exact same thing and getting a message
-- Called h323:[EMAIL PROTECTED]
== No one is available to answer at this time
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (incoming, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
could I see your conf files? the entry in extensions.conf
and the relevant sections of h323.conf please?
cheers
Dave
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 10, 2003 11:31 PM
Subject: Re: [Asterisk-Users] OH323 + G729 + Go2Call
I get an IVR when I use chan_h323 and Digiun's G.729.
Jeremy McNamara
Dave Alan Caruana wrote:
hi ..
i've just installed and licensed an instance of the G729 codec.
I am trying to connect through asterisk to Go2Call server ..
According to their info it involves dialling extension 729 on
voip01.go2call.com, to get the IVR.
my extensions.conf shows :
exten = s,2,Dial(OH323/h323:[EMAIL PROTECTED])
which I think is correct, I have G729 enabled in the OH323.conf
file and it seems to be using it ..
connection is not established, I have pasted a dump file below ..
anyone knows what's wrong ? i'm beyond my level of
asterisk knowledge at this point :(
thanks
Dave
- Original Message -
From: root [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 10, 2003 10:11 PM
0:00.006OpenH323 Wrapper OpenH323 Wrapper Version 0.0alpha0 by
inAccess Networks (www.inaccessnetworks.com) on Unix Linux
(2.4.20-8-i686)
at 2003/7/10 22:10:37.181
0:00.008OpenH323 Wrapper H323 Created endpoint.
0:00.008H323 Cleaner H323 Started cleaner thread
0:00.009OpenH323 Wrapper H323 Started listener
Listener[ip$*:1720]
0:00.010 H323 Listener:81249e8 H323 Awaiting TCP connections on port
1720
0:00.011OpenH323 Wrapper H323UDP Binding to interface:
0.0.0.0:5000
0:00.011OpenH323 Wrapper H323 Added capability: G.729{hw} 1
0:00.012OpenH323 Wrapper H323 Added capability:
UserInput/hookflash 2
0:00.012OpenH323 Wrapper H323 Added capability:
UserInput/basicString 3
0:00.012OpenH323 Wrapper H323 Added capability: UserInput/dtmf
4
0:00.012OpenH323 Wrapper H323 Added capability:
UserInput/RFC2833 5
0:05.829 ThreadID=0x495be540 H323 Making call to:
h323:[EMAIL PROTECTED]:1720
0:05.831 ThreadID=0x495be540 H323 Added capability: G.729{hw} 1
0:05.831 ThreadID=0x495be540 H323 Added capability:
UserInput/hookflash
2
0:05.831 ThreadID=0x495be540 H323 Added capability:
UserInput/basicString 3
0:05.831 ThreadID=0x495be540 H323 Added capability: UserInput/dtmf 4
0:05.831 ThreadID=0x495be540 H323 Added capability: UserInput/RFC2833
5
0:05.832 ThreadID=0x495be540 H323 Found capability: G.729{hw} 1
0:05.832 ThreadID=0x495be540 H323 Found capability:
UserInput/hookflash
2
0:05.832 ThreadID=0x495be540 H323 Found capability:
UserInput/basicString 3
0:05.832 ThreadID=0x495be540 H323 Found capability: UserInput/dtmf 4
0:05.832 ThreadID=0x495be540 H323 Found capability: UserInput/RFC2833
5
0:05.833 ThreadID=0x495be540 RFC2833 Handler created
0:05.833 ThreadID=0x495be540 H323 Added capability: G.729A{hw} 1
0:05.833 ThreadID=0x495be540 H323 Created new connection:
ip$localhost/12098
0:05.834 H225 Caller:8131128 H225 Started call thread
0:06.043 H225 Caller:8131128 H323TCP Started connection:
host=216.52.153.206:1720, if=217.168.168.5:5004, handle=64
0:06.044 H225 Caller:8131128 H225 Sending Setup PDU
0:06.044 H225 Caller:8131128 H225 Check for Fast start by local
endpoint
0:06.044 H225 Caller:8131128 H245 Default OnSelectLogicalChannels,
FastStartDisabled
0:06.046 H225 Caller:8131128 H225 Sending PDU: setup
0:06.047 H225 Caller:8131128 H225 Reading PDUs: callRef=12098
0:06.288 H225 Caller:8131128 H225 Receiving PDU: callProceeding
0:06.288 H225 Caller:8131128 H225 Handling PDU: CallProceeding
callRef=12098
0:06.289 H225 Caller:8131128 H225 Set protocol version to 3 and
implying H.245 version 5
0:06.289 H225 Caller:8131128 H225 Set remote party name:
216.52.153.206
0:06.465 H225 Caller:8131128 H323TCP Started connection:
host=216.52.153.206:29709, if=217.168.168.5:5005, handle=65
0:06.465 H225 Caller:8131128 H323
InternalEstablishedConnectionCheck: connectionState=AwaitingSignalConnect
fastStartState=FastStartDisabled
0:06.466H245:8131e68 H245 Started thread
0:06.467H245:8131e68 H245 Started control channel
0:06.468H245:8131e68 H245 Sending TerminalCapabilitySet:
outSeq=1
0:06.470H245:8131e68 H245 Sending PDU
[Asterisk-Users] OH323 + G729 + Go2Call
hi ..
i've just installed and licensed an instance of the G729 codec.
I am trying to connect through asterisk to Go2Call server ..
According to their info it involves dialling extension 729 on
voip01.go2call.com, to get the IVR.
my extensions.conf shows :
exten = s,2,Dial(OH323/h323:[EMAIL PROTECTED])
which I think is correct, I have G729 enabled in the OH323.conf
file and it seems to be using it ..
connection is not established, I have pasted a dump file below ..
anyone knows what's wrong ? i'm beyond my level of
asterisk knowledge at this point :(
thanks
Dave
- Original Message -
From: root [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 10, 2003 10:11 PM
0:00.006OpenH323 Wrapper OpenH323 Wrapper Version 0.0alpha0 by
inAccess Networks (www.inaccessnetworks.com) on Unix Linux (2.4.20-8-i686)
at 2003/7/10 22:10:37.181
0:00.008OpenH323 Wrapper H323 Created endpoint.
0:00.008H323 Cleaner H323 Started cleaner thread
0:00.009OpenH323 Wrapper H323 Started listener
Listener[ip$*:1720]
0:00.010 H323 Listener:81249e8 H323 Awaiting TCP connections on port
1720
0:00.011OpenH323 Wrapper H323UDP Binding to interface:
0.0.0.0:5000
0:00.011OpenH323 Wrapper H323 Added capability: G.729{hw} 1
0:00.012OpenH323 Wrapper H323 Added capability:
UserInput/hookflash 2
0:00.012OpenH323 Wrapper H323 Added capability:
UserInput/basicString 3
0:00.012OpenH323 Wrapper H323 Added capability: UserInput/dtmf
4
0:00.012OpenH323 Wrapper H323 Added capability:
UserInput/RFC2833 5
0:05.829 ThreadID=0x495be540 H323 Making call to:
h323:[EMAIL PROTECTED]:1720
0:05.831 ThreadID=0x495be540 H323 Added capability: G.729{hw} 1
0:05.831 ThreadID=0x495be540 H323 Added capability: UserInput/hookflash
2
0:05.831 ThreadID=0x495be540 H323 Added capability:
UserInput/basicString 3
0:05.831 ThreadID=0x495be540 H323 Added capability: UserInput/dtmf 4
0:05.831 ThreadID=0x495be540 H323 Added capability: UserInput/RFC2833
5
0:05.832 ThreadID=0x495be540 H323 Found capability: G.729{hw} 1
0:05.832 ThreadID=0x495be540 H323 Found capability: UserInput/hookflash
2
0:05.832 ThreadID=0x495be540 H323 Found capability:
UserInput/basicString 3
0:05.832 ThreadID=0x495be540 H323 Found capability: UserInput/dtmf 4
0:05.832 ThreadID=0x495be540 H323 Found capability: UserInput/RFC2833
5
0:05.833 ThreadID=0x495be540 RFC2833 Handler created
0:05.833 ThreadID=0x495be540 H323 Added capability: G.729A{hw} 1
0:05.833 ThreadID=0x495be540 H323 Created new connection:
ip$localhost/12098
0:05.834 H225 Caller:8131128 H225 Started call thread
0:06.043 H225 Caller:8131128 H323TCP Started connection:
host=216.52.153.206:1720, if=217.168.168.5:5004, handle=64
0:06.044 H225 Caller:8131128 H225 Sending Setup PDU
0:06.044 H225 Caller:8131128 H225 Check for Fast start by local
endpoint
0:06.044 H225 Caller:8131128 H245 Default OnSelectLogicalChannels,
FastStartDisabled
0:06.046 H225 Caller:8131128 H225 Sending PDU: setup
0:06.047 H225 Caller:8131128 H225 Reading PDUs: callRef=12098
0:06.288 H225 Caller:8131128 H225 Receiving PDU: callProceeding
0:06.288 H225 Caller:8131128 H225 Handling PDU: CallProceeding
callRef=12098
0:06.289 H225 Caller:8131128 H225 Set protocol version to 3 and
implying H.245 version 5
0:06.289 H225 Caller:8131128 H225 Set remote party name:
216.52.153.206
0:06.465 H225 Caller:8131128 H323TCP Started connection:
host=216.52.153.206:29709, if=217.168.168.5:5005, handle=65
0:06.465 H225 Caller:8131128 H323
InternalEstablishedConnectionCheck: connectionState=AwaitingSignalConnect
fastStartState=FastStartDisabled
0:06.466H245:8131e68 H245 Started thread
0:06.467H245:8131e68 H245 Started control channel
0:06.468H245:8131e68 H245 Sending TerminalCapabilitySet:
outSeq=1
0:06.470H245:8131e68 H245 Sending PDU: request
terminalCapabilitySet
0:06.472H245:8131e68 H245 Sending MasterSlaveDetermination
0:06.472H245:8131e68 H245 Sending PDU: request
masterSlaveDetermination
0:06.474 H225 Caller:8131128 H225 Receiving PDU: connect
0:06.475 H225 Caller:8131128 H225 Handling PDU: Connect
callRef=12098
0:06.475 H225 Caller:8131128 H225 Set protocol version to 3 and
implying H.245 version 5
0:06.475 H225 Caller:8131128 H225 Set remote party name:
216.52.153.206
0:06.475 H225 Caller:8131128 H225 Received connect PDU.
0:06.476 H225 Caller:8131128 H245 Started control channel
0:06.476 H225 Caller:8131128 H245 TerminalCapabilitySet already in
progress: outSeq=1
0:06.476 H225 Caller:8131128 H245 MasterSlaveDetermination already
in progress
0:06.476 H225 Caller:8131128 H323
InternalEstablishedConnectionCheck: connectionState=HasExecutedSignalConnect
[Asterisk-Users] chanh323 dialling
what is the format for an h323 entry in the dialplan? can I use chan_h323 without compiling anything else or should I compile oh323? basically what's the best way :) cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re. rtp.c RTP codec 19
hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get Invalid CSeq Number back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RTP.C codec error 19
hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get Invalid CSeq Number back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave (I mistakenly put an re in the title of this email and I think it's been ignored .. reposted) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 prob :)
i'm getting Asterisk to dial an h323 call termination service .. right now getting this message: -- Executing Wait(Zap/1-1, 1) in new stack -- Accepting call from '21382890' to 's' on channel 1, span 1 -- Executing Dial(Zap/1-1, OH323/h323:[EMAIL PROTECTED]) in new stack 5:59.330 H323 Cleaner H323Connection ip$localhost/18729 terminated. ERROR[1230546240]: File chan_oh323.c, Line 704 (oh323_call): H323:0: Could not call h323:[EMAIL PROTECTED] -- Couldn't call h323:[EMAIL PROTECTED] -- Hungup 'H323:0' == Everyone is busy at this time -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' any idea what that can mean ? I have my system currently working through SIP, however every now and then it shows this message -- Got SIP response 481 Invalid CSeq Number back from 216.52.153.207 == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/1-1' and drops the line which is the reason I am trying to use H323 instead, maybe I can get around that problem. Can anyone tell me what it means? thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP disconnecting : response 481
-- Got SIP response 481 Invalid CSeq Number back from 216.52.153.207 == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/1-1' I am getting this error on an outgoing call to a SIP host. The call just disconnects .. is there any way around it ? thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Problem (previous post) .. information might be relevant
regarding my previous post about SIP outgoing calls dropping with an error 481 .. this is my output from a SIP debug. the call dropped occurs at the end. Asterisk is mine, Cisco-SIPGateway is the other end (remote) and not in my control. help :) please!! Dave Signal=0 Duration=250 (no NAT) to 216.52.153.207:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Date: Tue, 08 Jul 2003 22:22:57 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 117 INFO Contact: sip:[EMAIL PROTECTED]:5060 10 headers, 0 lines set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 216.52.153.207, port 5060 Reliably Transmitting: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 118 INFO User-Agent: Asterisk PBX Content-Type: application/dtmf-relay Content-Length: 24 Signal=1 Duration=250 (no NAT) to 216.52.153.207:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Date: Tue, 08 Jul 2003 22:22:58 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 118 INFO Contact: sip:[EMAIL PROTECTED]:5060 10 headers, 0 lines set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 216.52.153.207, port 5060 Reliably Transmitting: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 119 INFO User-Agent: Asterisk PBX Content-Type: application/dtmf-relay Content-Length: 24 Signal=2 Duration=250 (no NAT) to 216.52.153.207:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Date: Tue, 08 Jul 2003 22:22:58 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 119 INFO Contact: sip:[EMAIL PROTECTED]:5060 10 headers, 0 lines set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 216.52.153.207, port 5060 Reliably Transmitting: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 120 INFO User-Agent: Asterisk PBX Content-Type: application/dtmf-relay Content-Length: 24 Signal=5 Duration=250 (no NAT) to 216.52.153.207:5060 set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 216.52.153.207, port 5060 Reliably Transmitting: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 121 INFO User-Agent: Asterisk PBX Content-Type: application/dtmf-relay Content-Length: 24 Signal=6 Duration=250 (no NAT) to 216.52.153.207:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Date: Tue, 08 Jul 2003 22:22:59 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 121 INFO Contact: sip:[EMAIL PROTECTED]:5060 10 headers, 0 lines Retransmitting #1 (no NAT): INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 120 INFO User-Agent: Asterisk PBX Content-Type: application/dtmf-relay Content-Length: 24 Signal=5 Duration=250 to 216.52.153.207:5060 set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 216.52.153.207, port 5060 Reliably Transmitting: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 122 INFO User-Agent: Asterisk PBX Content-Type: application/dtmf-relay Content-Length: 24 Signal=4 Duration=250 (no
[Asterisk-Users] E100P installation sheet
hi .. maybe someone can help me, I seem to have lost the sheet of paper that comes with an E100P card and tells you how to compile the stuff it requires to run. I'm trying to move my Asterisk to a different box and at this time totally stuck. Could someone be kind enough as to mail me a PDF of it ?? many thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P installation sheet
yeah thanks :) i've compiled all OK and still can't get my new installation working .. doesn't seem to recognise the E100P board, even though the modprobe wct1xxp command goes through OK and says the board is found .. anyone have any ideas ? same board was working fine on my old server. Dave - Original Message - From: Tais M. Hansen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 30, 2003 2:29 PM Subject: Re: [Asterisk-Users] E100P installation sheet -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 30 June 2003 14:01, Dave Alan Caruana wrote: I seem to have lost the sheet of paper that comes with an E100P card and tells you how to compile the stuff it requires to run. Could someone be kind enough as to mail me a PDF of it ?? Is this what you're looking for? :) http://www.digium.com/downloads/quick_install_zaptel_asterisk.pdf - -- Regards, Tais M. Hansen ComX -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/AC0z2TEAILET3McRAij4AJ4z9a09G8eBIwjD76mHQhtKnH/aNQCdGvNe LTc4x6WPjvj9ihfe+qxEigE= =1oh9 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P installation sheet
problem solved - forgot to update zaptel.conf stupid me! thanks guys :) Dave - Original Message - From: Dave Alan Caruana [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 30, 2003 2:55 PM Subject: Re: [Asterisk-Users] E100P installation sheet yeah thanks :) i've compiled all OK and still can't get my new installation working .. doesn't seem to recognise the E100P board, even though the modprobe wct1xxp command goes through OK and says the board is found .. anyone have any ideas ? same board was working fine on my old server. Dave - Original Message - From: Tais M. Hansen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 30, 2003 2:29 PM Subject: Re: [Asterisk-Users] E100P installation sheet -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 30 June 2003 14:01, Dave Alan Caruana wrote: I seem to have lost the sheet of paper that comes with an E100P card and tells you how to compile the stuff it requires to run. Could someone be kind enough as to mail me a PDF of it ?? Is this what you're looking for? :) http://www.digium.com/downloads/quick_install_zaptel_asterisk.pdf - -- Regards, Tais M. Hansen ComX -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/AC0z2TEAILET3McRAij4AJ4z9a09G8eBIwjD76mHQhtKnH/aNQCdGvNe LTc4x6WPjvj9ihfe+qxEigE= =1oh9 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk CPU usage
hi there.. I have an asterisk installation with a PRI-E1 card running EuroISDN, installed on a 1GHz Intel Celeron box with 256Mbytes RAM. CPU usage is stuck at 100% all the time, even with no calls going through. Is this the normal ? Running top reveals that the CPU allocation is 99.6% to Asterisk. 13:41:48 up 17:55, 3 users, load average: 1.07, 1.02, 1.00 44 processes: 43 sleeping, 1 running, 0 zombie, 0 stopped CPU states: 0.0% user 100.0% system 0.0% nice 0.0% iowait 0.0% idle Mem: 247188k av, 239664k used,7524k free, 0k shrd, 126572k buff 173676k actv, 0k in_d,2932k in_c Swap: 522104k av,2680k used, 519424k free 67164k cached PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME CPU COMMAND 2569 root 22 0 4712 4712 456 S99.8 1.9 1069m 0 asterisk 7 root 15 0 00 0 SW0.2 0.0 0:24 0 kscand/Normal 1 root 15 0 108 8856 S 0.0 0.0 0:03 0 init etc. Second question is this : my asterisk server is currently configured to receive calls, and immediately forward them to a SIP hosts (an ITSP server in USA) that requires input via an IVR. Sometimes this works fine, but many times the connection just drops while typing in codes. Once a connection is established (ie. got past the IVR stage) the connection never drops .. help welcome :) cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk CPU power requirements
Many thanks, Martin .. worked fine with dtmfmode=info Dave - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 23, 2003 4:32 PM Subject: Re: [Asterisk-Users] Asterisk CPU power requirements You need to find out which way your SIP gateway wants to receive the DTMFs. There are three ways to do that. Read sip.conf.sample. Martin On Mon, 23 Jun 2003, Dave Alan Caruana wrote: hi there, I have an installed working Asterisk server, which I am using to connect to a SIP service abroad. Although I can hear the IVR from the ITSP, I cannot seem to send them digits from my phone. I have also noticed that the CPU usage on my machine is up to 100% constantly and 99.9% of that is going to Asterisk, even when asterisk is just idle and doing nothing at all .. The machine is a Celeron 800 with 256Mb of RAM, and there is a Digium single span E1 card going into it. Is something wrong? or do I just need more CPU power? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1, E100P
hi guys, I have a little problem maybe you can help ... I have an asterisk setup, with an E100P, and an ISDN-PRI 30 channel line from the telco going into it .. the E1 line is OK, because plugged into a Lucent Portmaster 4 it works OK .. plugged into the asterisk box I just get an engaged tone, and asterisk posts this message on screen : WARNING[1167272000]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[1167272000]: File chan_zap.c, Line 5816 (pri_dchannel): Hangup on bad channel 1 With PRI Debug switched on, I get the debug log as per file attached .. I'm in Malta, but the ISDN should be a regular one as in the rest of europe, as far as I know .. for now all i'm trying to do is to get ISDN calls answered, and thrown directly into the asterisk demo, which works fine when contacted over SIP / H323. cheers Dave debuglog.rtf Description: MS-Word document
[Asterisk-Users] out of curiosity ..
not really asterisk related this, but is it normal for a mail to take so long to be resent through the mailing list server? i'm speaking about 20 minute + delays here .. (or it it only me ?) cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OH323 crashing
hi, does anyone have a problem with OH323 crashing with a segmentation fault whenever anything tries to connect to it ??? are the current CVS versions OK? Would like to speak to someone with a bit of OH323 experience, so if u're in a good mood to help, please do :) cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a little oh323 questoin
this might be a better dump: #0 0x41ec7279 in ast_oh323_new (i=0x810e538, state=0) at chan_oh323.c:1170 #1 0x41ec786e in oh323_request (type=0x4971aa6c OH323, format=8, data=0x4971a91c) at chan_oh323.c:1302 #2 0x0805878f in ast_request (type=0x4971aa6c OH323, format=4, data=0x810e538) at channel.c:1488 #3 0x41d7ba6f in dial_exec (chan=0x810cdc8, data=0x4971aa6c) at app_dial.c:478 #4 0x0806055a in pbx_exec (c=0x810cdc8, app=0x80e5cf0, data=0x4971adac, newstack=1) at pbx.c:393 #5 0x080672b8 in pbx_extension_helper (c=0x810cdc8, context=0x0, exten=0x810cf8c 1304, priority=1, callerid=0x8105250 217.168.168.49, action=135290696) at pbx.c:1125 #6 0x08062292 in ast_pbx_run (c=0x8105f48) at pbx.c:1609 #7 0x08067971 in pbx_thread (data=0x810e538) at pbx.c:1822 #8 0x4003b2b6 in start_thread () from /lib/tls/libpthread.so.0 hope u're still around to help!! (Michael, ie) i've been away from office for 2 days .. cheers Dave - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 6:17 PM Subject: Re: [Asterisk-Users] a little oh323 questoin Dave Alan Caruana wrote: (gdb) bt #0 oss_new (p=0x41be3aa0, state=5) at chan_oss.c:698 #1 0x41bad7d2 in console_dial (fd=1, argc=0, argv=0xbfffe290) at chan_oss.c:902 #2 0x08069b1a in ast_cli_command (fd=1, s=0x41bae940 Console) at cli.c:1006 #3 0x0807b32e in main (argc=1102817156, argv=0x41be3af4) at asterisk.c:496 #4 0x42015574 in __libc_start_main () from /lib/tls/libc.so.6 that's the debug output exactly after it crashed ... This doesn't seem like a call between a SIP an OH323 channel. The crash occurs inside chan_oss. Dave Michael. - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 5:39 PM Subject: Re: [Asterisk-Users] a little oh323 questoin Dave Alan Caruana wrote: doesn't seem to be dumping a core at all if it is, can't find it. Turn it on by running: ulimit -c 100 Michael. Dave - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 5:23 PM Subject: Re: [Asterisk-Users] a little oh323 questoin Dave Alan Caruana wrote: many thanks Michael, i've modified my extensions.conf ... from a softphone i'm dialling SIP/[EMAIL PROTECTED] which is the address of my asterisk installation. Asterisk quits immediately with a segmentation fault .. -- Executing Dial(SIP/217.168.168.49:5060, OH323/[EMAIL PROTECTED]) in new stack Segmentation fault You should provide the backtrace of this core dump. Run: gdb /usr/sbin/asterisk core_file_name From gdb run: bt and sent the output. Michael. help!! :) cheers Dave - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 4:28 PM Subject: Re: [Asterisk-Users] a little oh323 questoin Dave Alan Caruana wrote: hi, just wanted to know what's the proper syntax for an h323 extension. exten = 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207 mailto:SIP/[EMAIL PROTECTED],52,153.207) dials SIP extension 723 on IP 216.52.153.207, what is the h323 equivalent of that ?? Using asterisk-oh323: exten = 555,1,Dial(OH323/[EMAIL PROTECTED]) ; No gatekeeper exten = 555,1,Dial(OH323/216.52.153.207) ; No gatekeeper, ; default extension exten = 555,1,Dial(OH323/723) ; Gatekeeper available cheers Dave Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a little oh323 questoin
I had a very recent version of asterisk, but to be sure just downloaded the latest from CVS and compiled all packages except OH323 which is about 3 days old ... thanks Dave - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 05, 2003 4:33 PM Subject: Re: [Asterisk-Users] a little oh323 questoin Dave Alan Caruana wrote: this might be a better dump: #0 0x41ec7279 in ast_oh323_new (i=0x810e538, state=0) at chan_oh323.c:1170 #1 0x41ec786e in oh323_request (type=0x4971aa6c OH323, format=8, data=0x4971a91c) at chan_oh323.c:1302 #2 0x0805878f in ast_request (type=0x4971aa6c OH323, format=4, data=0x810e538) at channel.c:1488 #3 0x41d7ba6f in dial_exec (chan=0x810cdc8, data=0x4971aa6c) at app_dial.c:478 #4 0x0806055a in pbx_exec (c=0x810cdc8, app=0x80e5cf0, data=0x4971adac, newstack=1) at pbx.c:393 #5 0x080672b8 in pbx_extension_helper (c=0x810cdc8, context=0x0, exten=0x810cf8c 1304, priority=1, callerid=0x8105250 217.168.168.49, action=135290696) at pbx.c:1125 #6 0x08062292 in ast_pbx_run (c=0x8105f48) at pbx.c:1609 #7 0x08067971 in pbx_thread (data=0x810e538) at pbx.c:1822 #8 0x4003b2b6 in start_thread () from /lib/tls/libpthread.so.0 Can't see anything strange around the crash point. Also, from your previous posting, with the other core dump, (it occured inside the oss_new(...) function of chan_oss), it seems that the problem is somewhere else and not in H.323. Do you use the latest CVS code of Asterisk? hope u're still around to help!! (Michael, ie) i've been away from office for 2 days .. cheers Dave Michael. - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 6:17 PM Subject: Re: [Asterisk-Users] a little oh323 questoin Dave Alan Caruana wrote: (gdb) bt #0 oss_new (p=0x41be3aa0, state=5) at chan_oss.c:698 #1 0x41bad7d2 in console_dial (fd=1, argc=0, argv=0xbfffe290) at chan_oss.c:902 #2 0x08069b1a in ast_cli_command (fd=1, s=0x41bae940 Console) at cli.c:1006 #3 0x0807b32e in main (argc=1102817156, argv=0x41be3af4) at asterisk.c:496 #4 0x42015574 in __libc_start_main () from /lib/tls/libc.so.6 that's the debug output exactly after it crashed ... This doesn't seem like a call between a SIP an OH323 channel. The crash occurs inside chan_oss. Dave Michael. - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 5:39 PM Subject: Re: [Asterisk-Users] a little oh323 questoin Dave Alan Caruana wrote: doesn't seem to be dumping a core at all if it is, can't find it. Turn it on by running: ulimit -c 100 Michael. Dave - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 5:23 PM Subject: Re: [Asterisk-Users] a little oh323 questoin Dave Alan Caruana wrote: many thanks Michael, i've modified my extensions.conf ... from a softphone i'm dialling SIP/[EMAIL PROTECTED] which is the address of my asterisk installation. Asterisk quits immediately with a segmentation fault .. -- Executing Dial(SIP/217.168.168.49:5060, OH323/[EMAIL PROTECTED]) in new stack Segmentation fault You should provide the backtrace of this core dump. Run: gdb /usr/sbin/asterisk core_file_name From gdb run: bt and sent the output. Michael. help!! :) cheers Dave - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 4:28 PM Subject: Re: [Asterisk-Users] a little oh323 questoin Dave Alan Caruana wrote: hi, just wanted to know what's the proper syntax for an h323 extension. exten = 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207 mailto:SIP/[EMAIL PROTECTED],52,153.207) dials SIP extension 723 on IP 216.52.153.207, what is the h323 equivalent of that ?? Using asterisk-oh323: exten = 555,1,Dial(OH323/[EMAIL PROTECTED]) ; No gatekeeper exten = 555,1,Dial(OH323/216.52.153.207) ; No gatekeeper, ; default extension exten = 555,1,Dial(OH323/723) ; Gatekeeper available cheers Dave Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP codecs
i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the Go2Call side, not the SJPhone cos I can dial from SJPHone to SJPhone routing through asterisk with no problems. many cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP codecs
i've installed X-lite, can't get it to actually dial a SIP number, seems cryptic compared to SJPhone .. I have a feeling my problems is the codecs within * though, my question was how could I know which codecs * supports, and how to add other ones .. cheers Dave - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, June 06, 2003 2:27 PM Subject: Re: [Asterisk-Users] SIP codecs If you have the package available for download for free from SJLabs, then you only have G.711 codec installed on SJPhone. If you are a developer, you can register for a G.729 codec from SJLabs. BR, Dan P.S. Have you tried X-Lite? It has G.711ulaw, G.711a law, GSM and iLbc. - Original Message - From: Dave Alan Caruana [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, June 06, 2003 3:05 PM Subject: [Asterisk-Users] SIP codecs i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the Go2Call side, not the SJPhone cos I can dial from SJPHone to SJPhone routing through asterisk with no problems. many cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] more about SIP ...
I added the line allow G723.1 in my sip.conf general config, and from a bridge connection which gives silence, I have progressed to the error message below, and the call gets rejected. help!! Dave ps. 217.168.168.49 : soft sipphone, i'm trying SJphone Pingel Instant Expressa [EMAIL PROTECTED] : Go2Call SIP gateway -- Executing Dial(SIP/217.168.168.49:5060, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] WARNING[1240577216]: File channel.c, Line 1711 (ast_channel_make_compatible): No path to translate from SIP/216.52.153.207-2e12(1) to SIP/217.168.168.49:5060(4) -- SIP/216.52.153.207-2e12 answered SIP/217.168.168.49:5060 WARNING[1240577216]: File channel.c, Line 1711 (ast_channel_make_compatible): No path to translate from SIP/217.168.168.49:5060(4) to SIP/216.52.153.207-2e12(1) WARNING[1240577216]: File app_dial.c, Line 606 (dial_exec): Had to drop call because I couldn't make SIP/217.168.168.49:5060 compatible with SIP/216.52.153.207-2e12 == Spawn extension (default, 1303, 1) exited non-zero on 'SIP/217.168.168.49:5 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a little oh323 questoin
hi, just wanted to know what's the proper syntax for an h323 extension. exten = 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207) dials SIP extension 723 on IP 216.52.153.207, what is the h323 equivalent of that ?? cheers Dave
Re: [Asterisk-Users] a little oh323 questoin
NOTICE[1232188736]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type 'H323' the default channel type created in the startup is OH323, but how do I specify which extension number (723, in this case) it dials to ?? cheers again Dave - Original Message - From: Erik Anderson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 4:03 PM Subject: RE: [Asterisk-Users] a little oh323 questoin exten = 555,1,Dial(H323/216,52,153.207) Erik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dave Alan Caruana Sent: Tuesday, June 03, 2003 8:27 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] a little oh323 questoin hi, just wanted to know what's the proper syntax for an h323 extension. exten = 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207) dials SIP extension 723 on IP 216.52.153.207, what is the h323 equivalent of that ?? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a little oh323 questoin
(gdb) bt #0 oss_new (p=0x41be3aa0, state=5) at chan_oss.c:698 #1 0x41bad7d2 in console_dial (fd=1, argc=0, argv=0xbfffe290) at chan_oss.c:902 #2 0x08069b1a in ast_cli_command (fd=1, s=0x41bae940 Console) at cli.c:1006 #3 0x0807b32e in main (argc=1102817156, argv=0x41be3af4) at asterisk.c:496 #4 0x42015574 in __libc_start_main () from /lib/tls/libc.so.6 that's the debug output exactly after it crashed ... Dave - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 5:39 PM Subject: Re: [Asterisk-Users] a little oh323 questoin Dave Alan Caruana wrote: doesn't seem to be dumping a core at all if it is, can't find it. Turn it on by running: ulimit -c 100 Michael. Dave - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 5:23 PM Subject: Re: [Asterisk-Users] a little oh323 questoin Dave Alan Caruana wrote: many thanks Michael, i've modified my extensions.conf ... from a softphone i'm dialling SIP/[EMAIL PROTECTED] which is the address of my asterisk installation. Asterisk quits immediately with a segmentation fault .. -- Executing Dial(SIP/217.168.168.49:5060, OH323/[EMAIL PROTECTED]) in new stack Segmentation fault You should provide the backtrace of this core dump. Run: gdb /usr/sbin/asterisk core_file_name From gdb run: bt and sent the output. Michael. help!! :) cheers Dave - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 4:28 PM Subject: Re: [Asterisk-Users] a little oh323 questoin Dave Alan Caruana wrote: hi, just wanted to know what's the proper syntax for an h323 extension. exten = 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207 mailto:SIP/[EMAIL PROTECTED],52,153.207) dials SIP extension 723 on IP 216.52.153.207, what is the h323 equivalent of that ?? Using asterisk-oh323: exten = 555,1,Dial(OH323/[EMAIL PROTECTED]) ; No gatekeeper exten = 555,1,Dial(OH323/216.52.153.207) ; No gatekeeper, ; default extension exten = 555,1,Dial(OH323/723) ; Gatekeeper available cheers Dave Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a beginner's SIP question .. (further to previous mailing)
-- Executing Dial("SIP/sipphone-b6e6", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED] --
SIP/216.52.153.207-ab35 answered SIP/sipphone-b6e6 --
Attempting native bridge of SIP/sipphone-b6e6 and
SIP/216.52.153.207-ab35
is what shows up on the console window
...
thanks again :)
Dave
- Original Message -
From:
Dan
To: [EMAIL PROTECTED]
Sent: Friday, May 30, 2003 7:50 PM
Subject: Re: [Asterisk-Users] a
beginner's SIP question ..
Hi Dave,
If you have registered theSIP phone with
Asterisk, then you must have a line like:
exten = 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207)
in extensions.conf file
Then call 555 from the SIP phone to access the
destination.
BR,
Dan
- Original Message -----
From:
Dave Alan
Caruana
To: [EMAIL PROTECTED]
Sent: Friday, May 30, 2003 6:21
PM
Subject: Re: [Asterisk-Users] a
beginner's SIP question ..
I have included a dump of the debug info
...
what I am trying to do is route a call from
sipphone 217.168.168.49
through asterisk 217.168.168.51 onto a gateway
[EMAIL PROTECTED]
If i dial direct from the sip phone to the
gateway it works fine .. so
I do not think there is any incompatibility
there.
Calls don't go through though ...
please help!!!
cheers
Dave
*CLI -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-eca2 answered
SIP/217.168.168.49:5060 -- Attempting native bridge of
SIP/217.168.168.49:5060 and SIP/216.52.153.207-eca2WARNING[1125329600]:
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-1418 answered
SIP/217.168.168.49:5060 -- Attempting native bridge of
SIP/217.168.168.49:5060 and SIP/216.52.153.207-1418WARNING[1125329600]:
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Request) -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-11ed answered
SIP/217.168.168.49:5060 -- Attempting native bridge of
SIP/217.168.168.49:5060 and SIP/216.52.153.207-11edWARNING[1125329600]:
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Request)
- Original Message -
From:
Dan
To: [EMAIL PROTECTED]
Sent: Thursday, May 29, 2003 8:15
PM
Subject: Re: [Asterisk-Users] a
beginner's SIP question ..
Hi,
Check to have a common set of
codecs.
If X-Lite is used and at the other end is a
phone without GSM support, then it doesn't work.
Try to disable GSM on the soft phone (if
X-Lite).
BR,
Dan
- Original Message -
From:
Dave Alan
Caruana
To: [EMAIL PROTECTED]
Sent: Thursday, May 29, 2003 9:01
PM
Subject: [Asterisk-Users] a
beginner's SIP question ..
I am trying to get asterisk to dial this
address :
sip:[EMAIL PROTECTED]
Using a softphone on my PC
(217.168.168.49)
it dials immediately and I get a voice
prompt ..
I have configured an extension, 1303 on
asterisk,
modifying the demo configuration
:
exten = 1303,1,Dial(SIP/[EMAIL PROTECTED])
When from my softphone I dial
sip:[EMAIL PROTECTED]
on the console I get :
-- Executing
Dial("SIP/sipphone-97b6", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.20
Re: [Asterisk-Users] a beginner's SIP question .. (further!)
more about the same problem ...
i've been playing around and got to this error
message which seems relevant ..
*CLI dial 1303 --
Executing Dial("OSS/dsp", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED] --
SIP/216.52.153.207-1fb9 answered OSS/dsp Console call has been
answered NOTICE[1232188736]: File rtp.c, Line 326 (ast_rtp_read):
Unknown RTP codec 19 receivedNOTICE[1232188736]: File rtp.c, Line 326
(ast_rtp_read): Unknown RTP codec 19 receivedNOTICE[1232188736]: File rtp.c,
Line 326 (ast_rtp_read): Unknown RTP codec 19 receivedNOTICE[1232188736]:
File rtp.c, Line 326 (ast_rtp_read): Unknown RTP codec 19
receivedKilled
am I right in thinking i need a different codec to
connect to the sip host I want to
connect to? where do codecs come from?
many cheers
Dave
- Original Message -
From:
Dan
To: [EMAIL PROTECTED]
Sent: Friday, May 30, 2003 7:50 PM
Subject: Re: [Asterisk-Users] a
beginner's SIP question ..
Hi Dave,
If you have registered theSIP phone with
Asterisk, then you must have a line like:
exten = 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207)
in extensions.conf file
Then call 555 from the SIP phone to access the
destination.
BR,
Dan
- Original Message -----
From:
Dave Alan
Caruana
To: [EMAIL PROTECTED]
Sent: Friday, May 30, 2003 6:21
PM
Subject: Re: [Asterisk-Users] a
beginner's SIP question ..
I have included a dump of the debug info
...
what I am trying to do is route a call from
sipphone 217.168.168.49
through asterisk 217.168.168.51 onto a gateway
[EMAIL PROTECTED]
If i dial direct from the sip phone to the
gateway it works fine .. so
I do not think there is any incompatibility
there.
Calls don't go through though ...
please help!!!
cheers
Dave
*CLI -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-eca2 answered
SIP/217.168.168.49:5060 -- Attempting native bridge of
SIP/217.168.168.49:5060 and SIP/216.52.153.207-eca2WARNING[1125329600]:
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-1418 answered
SIP/217.168.168.49:5060 -- Attempting native bridge of
SIP/217.168.168.49:5060 and SIP/216.52.153.207-1418WARNING[1125329600]:
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Request) -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-11ed answered
SIP/217.168.168.49:5060 -- Attempting native bridge of
SIP/217.168.168.49:5060 and SIP/216.52.153.207-11edWARNING[1125329600]:
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Request)
- Original Message -
From:
Dan
To: [EMAIL PROTECTED]
Sent: Thursday, May 29, 2003 8:15
PM
Subject: Re: [Asterisk-Users] a
beginner's SIP question ..
Hi,
Check to have a common set of
codecs.
If X-Lite is used and at the other end is a
phone without GSM support, then it doesn't work.
Try to disable GSM on the soft phone (if
X-Lite).
BR,
Dan
- Original Message -
From:
Dave Alan
Caruana
To: [EMAIL PROTECTED]
Sent: Thursday, May 29, 2003 9:01
PM
Subject: [Asterisk-Users] a
beginner's SIP question ..
I am trying to get asterisk to dial this
address :
sip:[EMAIL PROTECTED]
Using a softphone on my PC
(217.168.168.49)
it dials immediately and I get a voice
prom
Re: [Asterisk-Users] a beginner's SIP question ..
sorry i'm sending so many emails, I always think of
something
exactly after i've pressed Send .. please be
patient with me :)
I also have OH323 installed, supposedly correctly,
and the same
gateway I want to connect to on SIP also supports
H323, however
i do not know what the dialcommand line for
H323 is .. i'm trying
exten = 1304,1,Dial(OH323/216.52.153.206)
;ring
but I actually want to dial extension 723 on the
remote end,
so this is surely not right.. current messages i'm
getting
from Asterisk are these :
*CLI dial 1304 --
Executing Dial("OSS/dsp", "OH323/216.52.153.206") in new
stack*CLI
0:03.623
H323 Cleaner H323 Connection ip$localhost/9771
terminated.ERROR[1232188736]: File chan_oh323.c, Line 610 (oh323_call):
H323:0: Could not call 216.52.153.206. -- Couldn't call
216.52.153.206 -- Hungup 'H323:0' == Everyone is
busy at this time
help *very* welcome ;)
cheers
Dave
- Original Message -
From:
Dan
To: [EMAIL PROTECTED]
Sent: Friday, May 30, 2003 7:50 PM
Subject: Re: [Asterisk-Users] a
beginner's SIP question ..
Hi Dave,
If you have registered theSIP phone with
Asterisk, then you must have a line like:
exten = 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207)
in extensions.conf file
Then call 555 from the SIP phone to access the
destination.
BR,
Dan
- Original Message -----
From:
Dave Alan
Caruana
To: [EMAIL PROTECTED]
Sent: Friday, May 30, 2003 6:21
PM
Subject: Re: [Asterisk-Users] a
beginner's SIP question ..
I have included a dump of the debug info
...
what I am trying to do is route a call from
sipphone 217.168.168.49
through asterisk 217.168.168.51 onto a gateway
[EMAIL PROTECTED]
If i dial direct from the sip phone to the
gateway it works fine .. so
I do not think there is any incompatibility
there.
Calls don't go through though ...
please help!!!
cheers
Dave
*CLI -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-eca2 answered
SIP/217.168.168.49:5060 -- Attempting native bridge of
SIP/217.168.168.49:5060 and SIP/216.52.153.207-eca2WARNING[1125329600]:
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-1418 answered
SIP/217.168.168.49:5060 -- Attempting native bridge of
SIP/217.168.168.49:5060 and SIP/216.52.153.207-1418WARNING[1125329600]:
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Request) -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-11ed answered
SIP/217.168.168.49:5060 -- Attempting native bridge of
SIP/217.168.168.49:5060 and SIP/216.52.153.207-11edWARNING[1125329600]:
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Request)
- Original Message -
From:
Dan
To: [EMAIL PROTECTED]
Sent: Thursday, May 29, 2003 8:15
PM
Subject: Re: [Asterisk-Users] a
beginner's SIP question ..
Hi,
Check to have a common set of
codecs.
If X-Lite is used and at the other end is a
phone without GSM support, then it doesn't work.
Try to disable GSM on the soft phone (if
X-Lite).
BR,
Dan
- Original Message -
From:
Dave Alan
Caruana
To: [EMAIL PROTECTED]
Sent: Thursday, May 29, 2003 9:01
PM
Subject: [Asterisk-Users] a
beginner's SIP question ..
I am trying to get asterisk to dial this
address :
sip:[EMAIL PROTECTED]
Using a
Re: [Asterisk-Users] a beginner's SIP question ..
I have included a dump of the debug info
...
what I am trying to do is route a call from
sipphone 217.168.168.49
through asterisk 217.168.168.51 onto a gateway [EMAIL PROTECTED]
If i dial direct from the sip phone to the gateway
it works fine .. so
I do not think there is any incompatibility
there.
Calls don't go through though ...
please help!!!
cheers
Dave
*CLI -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED] --
SIP/216.52.153.207-eca2 answered SIP/217.168.168.49:5060
-- Attempting native bridge of SIP/217.168.168.49:5060 and
SIP/216.52.153.207-eca2WARNING[1125329600]: File chan_sip.c, Line 404
(retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1) exited
non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File chan_sip.c,
Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED] --
SIP/216.52.153.207-1418 answered SIP/217.168.168.49:5060
-- Attempting native bridge of SIP/217.168.168.49:5060 and
SIP/216.52.153.207-1418WARNING[1125329600]: File chan_sip.c, Line 404
(retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1) exited
non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File chan_sip.c,
Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Request) -- Executing
Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED] --
SIP/216.52.153.207-11ed answered SIP/217.168.168.49:5060
-- Attempting native bridge of SIP/217.168.168.49:5060 and
SIP/216.52.153.207-11edWARNING[1125329600]: File chan_sip.c, Line 404
(retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 1 (Response) == Spawn extension (default, 1303, 1) exited
non-zero on 'SIP/217.168.168.49:5060'WARNING[1125329600]: File chan_sip.c,
Line 404 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Request)
- Original Message -
From:
Dan
To: [EMAIL PROTECTED]
Sent: Thursday, May 29, 2003 8:15
PM
Subject: Re: [Asterisk-Users] a
beginner's SIP question ..
Hi,
Check to have a common set of
codecs.
If X-Lite is used and at the other end is a phone
without GSM support, then it doesn't work.
Try to disable GSM on the soft phone (if
X-Lite).
BR,
Dan
- Original Message -
From:
Dave Alan
Caruana
To: [EMAIL PROTECTED]
Sent: Thursday, May 29, 2003 9:01
PM
Subject: [Asterisk-Users] a beginner's
SIP question ..
I am trying to get asterisk to dial this
address :
sip:[EMAIL PROTECTED]
Using a softphone on my PC
(217.168.168.49)
it dials immediately and I get a voice prompt
..
I have configured an extension, 1303 on
asterisk,
modifying the demo configuration :
exten = 1303,1,Dial(SIP/[EMAIL PROTECTED])
When from my softphone I dial
sip:[EMAIL PROTECTED]
on the console I get :
-- Executing
Dial("SIP/sipphone-97b6", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-7c3b answered SIP/sipphone-97b6
-- Attempting native bridge of SIP/sipphone-97b6 and
SIP/216.52.153.207-7c3b
but on my headset all I get is silence .. the
call doesn't drop though.
What am I doing wrong ?
many thanks,
Dave
[Asterisk-Users] a beginner's SIP question ..
I am trying to get asterisk to dial this address
:
sip:[EMAIL PROTECTED]
Using a softphone on my PC
(217.168.168.49)
it dials immediately and I get a voice prompt
..
I have configured an extension, 1303 on
asterisk,
modifying the demo configuration :
exten = 1303,1,Dial(SIP/[EMAIL PROTECTED])
When from my softphone I dial
sip:[EMAIL PROTECTED]
on the console I get :
-- Executing
Dial("SIP/sipphone-97b6", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED] --
SIP/216.52.153.207-7c3b answered SIP/sipphone-97b6 --
Attempting native bridge of SIP/sipphone-97b6 and
SIP/216.52.153.207-7c3b
but on my headset all I get is silence .. the call
doesn't drop though.
What am I doing wrong ?
many thanks,
Dave
