[Asterisk-Users] msn authentication
hi guys! i'm going to share a workaround forauthentication from msn messenger, you have to change two lines in chan_sip.c msn messenger is known to look for the correct realm in authentication, therefore, change the realm in chan_sip.c, line 2061 and line 2910 (release 0.4.0) i hopethe realm can be parsed from extensions.conf in the next release... ~kelvin =)
AW: [Asterisk-Users] Wildcard E100P resellers in Europe ?
Here you can get the reseller data -- http://www.digium.com/index.php?menu=resellers Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Nicolas Cartron Gesendet: Freitag, 11. Juli 2003 09:53 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] Wildcard E100P resellers in Europe ? All, I'd like to buy an E100P Wildcard frm Digium, but i prefer to buy it in Europe (costs, ...). Could somebody point me to an european reseller ? Thanks in advance. -- Nicolas Cartron [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] Wildcard E100P resellers in Europe ?
On 11/07/03 at 09:52, Nicolas Cartron ([EMAIL PROTECTED]) wrote : All, I'd like to buy an E100P Wildcard frm Digium, but i prefer to buy it in Europe (costs, ...). Could somebody point me to an european reseller ? Thanks in advance. I'm totally dumb, i saw that there was a 'reseller' section on the website. Sorry for disturbing. -- Nicolas Cartron [EMAIL PROTECTED] pgp0.pgp Description: PGP signature
[Asterisk-Users] Cisco 7960s
Cisco should really be ashamed of this product... While it is physically well constructed, and has excellent sound quality along with a very pleasant user interface, the device has SERIOUS stability issues, unless you run your network with an iron fist... Quite by accident, while configuring my Asterisk system to connect to a Cisco 7960 via SIP in a standard office PBX type arrangement, I discovered something interesting... By screwing around with both the source IP address of a SIP message, along with certain IP addresses in the SIP message itself, it's quite easy to crash the Cisco. In short, it would betrivial to DOS (by forcing continuous crashes and the subsequent reboots) any Cisco 7960 that you can route UDP packets to... Matt HardemanPaperSoft
Re: [Asterisk-Users] Channel Bank configuration
On Thu, 2003-07-10 at 23:12, Marty Mastera wrote: Steven: Thank you for your response. I do want to continue using the Adit for the data... In terms of routing the FXO, I would either want to route 1-10 from the Adit to *, or 1-12 (not sure how much I would gain/lose from having the fax and analog phone jacks direct from Adit or via *)... For the desk phones, there are only a maximum of 6 extensions, and the Adit is equipped with an 8 port FXS card. So I think what you were recommending is routing 1-10 from Adit to *, then map the 6 extensions back from * to the Adit FXS card? The Adit is provider owned and on their side of the dmark (as far as I know), so I realize that I would not be able to configure it in this case and would have to have them do it. Do this sound accurate? Sounds accurate. You may get lucky and they will let you route some channels back to the ADIT. BTW, in your other message, you ask about phones. Your NEC phones won't be usable. Transfer and hold are dealt with via the flash and dial an extension. CallerID is normal caller id, so for these 3 functions you just need a nice looking phone to put on the desk. I found some ATT phones that I like for my office at about $30. Of course the other thing you will have as an option is cordless phones. From my ADIT at home I now have a large collection of cordless phones attached. Thank ebay for the half of my phone system not produced by Digium. I'll try and get some pictures of my home system up on my webpage tomorrow. On 10 Jul 2003 20:58:13 -0500 Steven Critchfield wrote: On Thu, 2003-07-10 at 18:33, Marty Mastera wrote: Hello, I don't have any experience with channel banks and would appreciate any feedback on my theory outlined below: We have a single T1 entering the building with channels 1-12 being voice lines and 13-24 being a 768k internet connection. This T1 terminates to an Adit 600 (T1-1). Here's what I know. Channels 11-12 go out the Adit 600's 25-pair connector to a wiring block (and eventually to 2 fax machines - I assume this is to have the fax machines bypass the currently installed phone switch). The data comes out the Router card on the Adit and into our network. The currently installed phone system is an NEC NEAX2000 IVS box which is connected by CAT5 to the Adit 600's T1-2 port. (I am assuming that voice channels 1-10 are mapped to the Adit T1-2 and getting to the NEC this wayThere is also a 25 pair cable leaving the NEC and terminating on a wiring block for the desk phones. So my assumption is that channels 1-10 are mapped as FXO onto the Adit's 2nd T1 port, channels 11-12 to the Adit's 25-pair connector and 13-24 to the router card for data. This leaves the NEC box to handle the FXS (and hence why it is directly connected to the phones). When I replace the NEC box with an * box/T100P, I'm thinking that I will have to map Channels 1-12 to the T1-2 port and map the 8 Adit FXS channels on the T1-2 port to the Adit's 25-pair for the Adit's FXS capabilitythen run that T1 into the T100P and configure * to route between the FXO and FXS channels appropriately. Does this sound right? I'm trying to understand if the channel bank uses the T1 from the Adit for both FXO and FXS channels! This depends on what you are allowed to do and what you wish to accomplish. First: Who owns the ADIT? a. If it is your telco, then you need to find out where they think the dmark is. Basically you need to know if you can configure the ADIT. b. If you own the ADIT, then you can proceed. Second: You need to think about what you want do with it. Linux with a T100P can be your router, and therefore you can take the T1 from the wall and plug it straight into the T100P and configure linux to be your router. Then you need to bring your lines back out to go to phones. How many extension phones do you need to use, and how many FXS ports do you have on the ADIT? You mentioned only dropping 2 lines out of the ADIT currently, and you mentioned 8 FXS channels. Sounds like you will probably need more FXS cards. The downside of using linux to be your router is that you will need a second T100P or a T400P card to have enough channels to get stuff done. If you want to continue using the ADIT router, then you can route the 10 channels off the ADIT to asterisk, and then you can route up to 14 channels back to the ADIT to be combined with the 2 you already have being dropped off to the FAX machine. This means you will need to make sure you have at least 2 FXS cards. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip CANCEL or BYE when picking up a call ?
I think this is the same problem I was having yesterday. I still have to go back through my config files and find out exactly what I did, but I should be doing that sometime today. I'll let you know when I find out. Leif. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brancaleoni Matteo Sent: Thursday, July 10, 2003 2:28 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sip CANCEL or BYE when picking up a call ? Ok. I've noticed a thing: when you ring a sip phone, and hangup before it answer, asterisk sends a CANCEL to the phone to abort the current operation (in this case, the INVITE). and this's correct according to rfc. But now... when a sip phone A is ringed from a phone B , and that call from B is picked up by the phone C via *8 , asterisk sends 'BYE' to the phone A ( C B are bridged ok). But according to rfc, that's wrong, since 'BYE' must be sent to release an active call . The right thing to do is to send a CANCEL to A, since we want to abort the pending INVITE. I'm right ? That's a bug in asterisk ? I've found that using the budgetones phone. They'll go crazy if a INVITE is aborted by a BYE instead of a CANCEL. Matteo. -- Matteo Brancaleoni Powered by RedHat Linux 8.0 Linux User #153521 -BEGIN GEEK CODE BLOCK- Version: 3.12 GS d? s:- a- C+++ UL P+ L+++ E- W+++ N++ o K- w-- O- M-- V-- PS PE- Y PGP++ t 5 X+ R tv- b++ DI D+ G e h! r++ y --END GEEK CODE BLOCK-- ___ 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] 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] mod tor2 takes 20-30% from CPU (20-30% System)
Hi all, i have a E400P in my P III 1,4 GHz machine. When i start the tor2 driver (modprobe tor2) then i can see (with top) that the System takes 20 - 30 % CPU usage. Is this normal ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wait and user input..
Not all of the * wait commands respond to dtmf whilst playing back. Couldn't you use the Background application to play the music? That does respond to dtmf whilst playback is in progress. Iain --On Friday, July 11, 2003 10:52 am + WipeOut . [EMAIL PROTECTED] wrote: Hi.. How do you accept user input while waiting or playing moh? My Dialplan is as follows.. ring,ring,.. Hello thanks for calling blah blah... Please enter the extention number blah blah... WaitMusicOnHold(10) If no input pass call to operator.. The problem is that the user has to input the extension while they are being told what to do.. any input during Wait or WaitMusicOnHold is ignored... 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
[Asterisk-Users] Compile Problems with gcc 3.3
Hi, after quite some time doing nothing with asterisk I downloaded the current cvs version. Building this on a SuSE 8.2 System with gcc 3.3 i ran into an unpleasant snag: pbx.c:581: warning: comparison between signed and unsigned pbx.c: In function `pbx_substitute_variables_temp': pbx.c:765: warning: comparison between signed and unsigned pbx.c:812: warning: comparison between signed and unsigned pbx.c: In function `pbx_builtin_hangup': pbx.c:4017: internal compiler error: Segmentation fault The last line realy stopped me cold :-( changing the -O level made no difference. There is a Bug filed with gcc that seems to fit here but it could also be a prob with my machine? Anybody else with the same problem? What is the last gcc version that works for you? G! UK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wait and user input..
Two issues with using Background.. 1) Background plays GSM files while moh uses MP3.. 2) Background will play the whole file before moving on so I would need to make a file that will play for the number of seconds that I want to wait for user input..where as WaitMusicOnHold has a configurable timeout.. Thanks anyway.. Not all of the * wait commands respond to dtmf whilst playing back. Couldn't you use the Background application to play the music? That does respond to dtmf whilst playback is in progress. Iain --On Friday, July 11, 2003 10:52 am + WipeOut . [EMAIL PROTECTED] wrote: Hi.. How do you accept user input while waiting or playing moh? My Dialplan is as follows.. ring,ring,.. Hello thanks for calling blah blah... Please enter the extention number blah blah... WaitMusicOnHold(10) If no input pass call to operator.. The problem is that the user has to input the extension while they are being told what to do.. any input during Wait or WaitMusicOnHold is ignored... 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 -- __ 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
Re: [Asterisk-Users] ISDN PRI E1 configuration with E100P
Very sorry about the previous mail, heres the mail again, hi Everyone, We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output, *CLI == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 . . . -- B-channel 31 successfully restarted on span 1 but, when we make a call to this E1 from outside, it gives the following error, WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not found? WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 1 WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not found? WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 2 does anybody hav an idea on this? our zaptel.conf is, #E100p card span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf, ;E100p card switchtype=EuroISDN signalling=pri_cpe pridialplan=unknown context=incoming group = 2 channel = 1-15,17-31 Thanks inadvance, Surajee --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mgcp problems
I strange error messages when using mgcp and ata186 . This session is simply dial into 600 demo extension - echo test ... Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 29 OK to 10.0.1.19:2427 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '0' -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 306 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 2149c6df R: hu(N), hf(N), D/[0-9#*](N) to 10.0.1.19:2427 -- MGCP asked to indicate -1 'UNKNOWN' condition on channel MGCP/aaln/[EMAIL PROTECTED] -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 307 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 2149c6df R: hu(N), hf(N), D/[0-9#*](N) to 10.0.1.19:2427 -- MGCP asked to indicate -1 'UNKNOWN' condition on channel MGCP/aaln/[EMAIL PROTECTED] -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 308 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 2149c6df R: hu(N), hf(N), D/[0-9#*](N) to 10.0.1.19:2427 -- Executing Playback(MGCP/aaln/[EMAIL PROTECTED], demo-echotest) in new stack -- Modified aaln/[EMAIL PROTECTED] with new mode: sendrecv on callid: 7d4b8e932149c6df Posting Request: MDCX 309 aaln/[EMAIL PROTECTED] MGCP 1.0 C: 7d4b8e932149c6df What is the -1 'UNKNOWN' condition on channel ? Is it correct mgcp packet ? -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia phone/fax +7(0732) 727172, http://www.comlink.ru ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960s
Sounds like a security issue. Verify the issue and email [EMAIL PROTECTED] Cisco will take a look at it once it hits bugtraq I am sure. Erik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matthew Hardeman Sent: Friday, July 11, 2003 3:21 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960s Cisco should really be ashamed of this product... While it is physically well constructed, and has excellent sound quality along with a very pleasant user interface, the device has SERIOUS stability issues, unless you run your network with an iron fist... Quite by accident, while configuring my Asterisk system to connect to a Cisco 7960 via SIP in a standard office PBX type arrangement, I discovered something interesting... By screwing around with both the source IP address of a SIP message, along with certain IP addresses in the SIP message itself, it's quite easy to crash the Cisco. In short, it would be trivial to DOS (by forcing continuous crashes and the subsequent reboots) any Cisco 7960 that you can route UDP packets to... Matt Hardeman PaperSoft ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to find IP address???
This morning, I received a very strange error message on the Asterisk console. The error occurs when I try to access iconnect WARNING[196621]: File chan_sip.c, Line 386 (__sip_xmit): sip_xmit of 0x80d0854 (len 649) to 213.137.73.178 returned -1: Bad file descriptor I also get this error when I try to reload: WARNING[16384]: File chan_sip.c, Line 5355 (reload_config): Unable to get IP address for BusinessOne.telantek.com, SIP disabled I have not changed anything in my sip.conf file recently. Here is what I have: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls allow=gsm allow=ulaw allow=alaw tos=lowdelay tos=185 ;register= :[EMAIL PROTECTED] ;register= :[EMAIL PROTECTED] [iconnect] type=friend username= password= host=sipauth.deltathree.com ;host=213.137.73.178 canreinvite=no Has anybody experienced this before? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hardware requirements
We have heard several times that 2 E400P's in one box is a current practical limit. But what type of machine would I need (as in CPU, RAM etc) to do this, and really put all those (240) channels to work with AGI scripts or the likes ? We have two e400p boards in a UP (3 Ghz P4 Northwood) box to serve ras users. I see 1000 interrupts per second per board. Load looks kinda funny (on and off between 20 and 100 % system every few seconds), but we can use all 240 channels without any dropping calls or packets as far as I can tell. But then this is only data... Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX G729 Codec
On Fri, 2003-07-11 at 06:40, Simon Woodhead wrote: Our problem was that we all of a sudden would get dropped audio, and I had one user complain of extreme lag occasionally. I didn't have anyone else experience the lag, but the dropped audio would come and go. It sometimes would drop out for a second or so. Sound quality when there was still just perfect. For your link to the Pace Vega Stream, what codec are you using? I would assume it would be more of a problem in codec shifting bits or something, but then again this is a wild guess. Thanks for that. We're using G.729 over H.323, incoming and outgoing. Outgoing works perfectly but on incoming we get the underwater sound periodically. It clicks in randomly but once there the only way to clear it is to end the call and try again. One thing we have thought of is co-loing an * box directly at the Telco and plugging in to their switch directly. We'd then be in control of the VoIP part and know that over IAX it would work fine. Can anyone enlighten me as to how we'd connect to them physically on-site? Would it be a PRI or would there be a different method as the PSTN wouldn't be between us and their switch? You would still use PRI if you need bulk lines. You could use channelized T1, but you get a lot more options with PRI. Currently our phone server is in our colo rack and our phone lines are sent down to us via our data T1 line. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN PRI E1 configuration with E100P
Unfortunatelly if your telco doesn't send you any DID along with the SETUP message you need to have immediate=yes in zapata.conf for those channels. regards Martin On Fri, 11 Jul 2003 [EMAIL PROTECTED] wrote: Very sorry about the previous mail, heres the mail again, hi Everyone, We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output, *CLI == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 . . . -- B-channel 31 successfully restarted on span 1 but, when we make a call to this E1 from outside, it gives the following error, WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not found? WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 1 WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not found? WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 2 does anybody hav an idea on this? our zaptel.conf is, #E100p card span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf, ;E100p card switchtype=EuroISDN signalling=pri_cpe pridialplan=unknown context=incoming group = 2 channel = 1-15,17-31 Thanks inadvance, Surajee --This mail sent through OmniBIS.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] IAX G729 Codec
You would still use PRI if you need bulk lines. You could use channelized T1, but you get a lot more options with PRI. Currently our phone server is in our colo rack and our phone lines are sent down to us via our data T1 line. Thanks Steven. I'll go investiagte that. Cheers, Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mgcp problems
When I connected over two mgcp channels and sending numerical indication to cisco ata it seems hangup one channel (receving ) and generate 'fast busy' tone. I hack chan_mgcp and my threewaycalling works ok! But why indications are sent after I press hookflash on answering end? -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia phone/fax +7(0732) 727172, http://www.comlink.ru ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN PRI E1 configuration with E100P
[EMAIL PROTECTED] wrote: hi Everyone, We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output, *CLI == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 . . . -- B-channel 31 successfully restarted on span 1 but, when we make a call to this E1 from outside, it gives the following error, WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not found? WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 1 WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not found? WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 2 does anybody hav an idea on this? our zaptel.conf is, #E100p card span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf, ;E100p card switchtype=EuroISDN signalling=pri_cpe pridialplan=unknown context=incoming group = 2 channel = 1-15,17-31 Thanks inadvance, Surajee --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users I was having the same problem because : 1 number for E1 and the local PTSN was not sending the DID to select the appropriate extension. Set the immediate=yes into zapata.conf and catch the call into s extension! Thanks to Martin Pycko ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to find IP address???
yep, that happened to me. gethostbyaddr() simply fails here. check that the hostname is present into /etc/hosts I.e. if you hostname is 'pingus' in /etc/hosts u should have 127.0.0.1 pingus localhost.localdomain localhost Matteo. Il ven, 2003-07-11 alle 15:34, Derek Beaumont ha scritto: This morning, I received a very strange error message on the Asterisk console. The error occurs when I try to access iconnect WARNING[196621]: File chan_sip.c, Line 386 (__sip_xmit): sip_xmit of 0x80d0854 (len 649) to 213.137.73.178 returned -1: Bad file descriptor I also get this error when I try to reload: WARNING[16384]: File chan_sip.c, Line 5355 (reload_config): Unable to get IP address for BusinessOne.telantek.com, SIP disabled I have not changed anything in my sip.conf file recently. Here is what I have: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls allow=gsm allow=ulaw allow=alaw tos=lowdelay tos=185 ;register= :[EMAIL PROTECTED] ;register= :[EMAIL PROTECTED] [iconnect] type=friend username= password= host=sipauth.deltathree.com ;host=213.137.73.178 canreinvite=no Has anybody experienced this before? ___ 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] mgcp problems
On Fri, 2003-07-11 at 08:42, Pavel Zheltouhov wrote: When I connected over two mgcp channels and sending numerical indication to cisco ata it seems hangup one channel (receving ) and generate 'fast busy' tone. I hack chan_mgcp and my threewaycalling works ok! But why indications are sent after I press hookflash on answering end? indications are sent to provide a dialtone after flashhook. --Karl -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 + G729 + Go2Call
Dave Alan Caruana wrote: I am trying the exact same thing and getting a message -- Called h323:[EMAIL PROTECTED] This is not a proper command... I have absolutely no clue where those other hacks got the h323: bullshit from. This line works perfectly for me: exten = 555,1,Dial,H323/[EMAIL PROTECTED] Then I have made sure to enable G.729 in h323.conf with allow=g729. Very simple. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN PRI E1 configuration with E100P
yes, now i got the that problem solved, with 'immediate=yes', but now i've faced with another, When I connect the E1 to the cards, the LED lights does not change to green, in E100P, its blinking red (even when there is no E1 plugged, its blinking red) and in E400P, its solid red. (this was the same even without 'immediate=yes') When i start asterisk, the cards starts fine (as we see), givin the following output, *CLI == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 . . . -- B-channel 31 successfully restarted on span 1 after i put 'immediate=yes', i can even call from outside to the PRI E1, and get connected to asterisk, and listen to the prompts played by asterisk -- Accepting call from '062279955' to 's' on channel 1, span 1 -- Executing Wait(Zap/1-1, 2) in new stack -- Executing Answer(Zap/1-1, ) in new stack -- Executing DigitTimeout(Zap/1-1, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(Zap/1-1, 10) in new stack -- Set Response Timeout to 10 -- Executing BackGround(Zap/1-1, demo-congrats) in new stack -- Playing 'demo-congrats' But calling to outside from asterisk fails (with the following errors), we don't know whether this related to the above problem Executing Dial(SIP/802-8a26, Zap/g2/0129063800|20|t) in new stack NOTICE[262160]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type 'Zap' == Everyone is busy at this time -- Executing Congestion(SIP/802-8a26, ) in new stack == Spawn extension (sip, 980129063800, 2) exited non-zero on 'SIP/802-8a26' Following are new zaptel and zapata conf files, zapata.conf [channels] transfer=yes echocancel=yes callprogress=yes immediate=yes ;E100p card switchtype=EuroISDN signalling=pri_cpe pridialplan=unknown context=inbound-pstn group=2 channel = 1-15,17-31 zaptel.conf #E100p card span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 defaultzone=us loadzone=us Thank you, Surajee - Original Message - From: Cristi [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 11, 2003 10:39 PM Subject: Re: [Asterisk-Users] ISDN PRI E1 configuration with E100P [EMAIL PROTECTED] wrote: hi Everyone, We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output, *CLI == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 . . . -- B-channel 31 successfully restarted on span 1 but, when we make a call to this E1 from outside, it gives the following error, WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not found? WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 1 WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not found? WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 2 does anybody hav an idea on this? our zaptel.conf is, #E100p card span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf, ;E100p card switchtype=EuroISDN signalling=pri_cpe pridialplan=unknown context=incoming group = 2 channel = 1-15,17-31 Thanks inadvance, Surajee --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users I was having the same problem because : 1 number for E1 and the local PTSN was not sending the DID to select the appropriate extension. Set the immediate=yes into zapata.conf and catch the call into s extension! Thanks to Martin Pycko ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 + G729 + Go2Call
Jeremy McNamara wrote: Dave Alan Caruana wrote: I am trying the exact same thing and getting a message -- Called h323:[EMAIL PROTECTED] This is not a proper command... I have absolutely no clue where those other hacks got the h323: bullshit from. RTFM. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Client Call Management Application?
Title: Message Is there anywork in process towards a Client Call management application that integrates with a SIP phone? I am thinking something along the lines of MXIE, the app thatZultus if offering with their MX1200. http://www.zultys.com/FAQs_MX1200.htm "Does the MX1200 support Instant Messaging, Presence, and other productivity tools?Absolutely. It supports all this with its MXIE (pronounced "mixee"), its client user interface. Not only can MXIE provide instant messaging and presence, it gives users the power of making and receiving calls, central and local address books, buddy lists, chat, call logs, call park and pickup, and voice mail management. MXIE feature set and simple interface (with screen pops and drag and drop of calls) will streamline communications at any desktop. Below is an image of how the client interface looks with one call active, one call on hold, and 2 conversations by instant messaging." Alternatively, has there been any additional progress towards some type TAPI integration, or other methods of getting desktop application like Outlook having the ability of directly placing a call on behalf of the phone? Thanks, Marcus
Re: [Asterisk-Users] Cisco 7960s
I have an open ticket at cisco with status development review; workaround provided. I'm going to remind them of the potential security consequences later today... The tech I've been working with seems very competent, and I suspect this may eventually get dealt with... Matt Hardeman PaperSoft - Original Message - From: Josh Howlett [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 11, 2003 3:30 AM Subject: Re: [Asterisk-Users] Cisco 7960s Cisco and bugtraq need to know this! josh. On Fri, 2003-07-11 at 09:21, Matthew Hardeman wrote: Cisco should really be ashamed of this product... While it is physically well constructed, and has excellent sound quality along with a very pleasant user interface, the device has SERIOUS stability issues, unless you run your network with an iron fist... Quite by accident, while configuring my Asterisk system to connect to a Cisco 7960 via SIP in a standard office PBX type arrangement, I discovered something interesting... By screwing around with both the source IP address of a SIP message, along with certain IP addresses in the SIP message itself, it's quite easy to crash the Cisco. In short, it would be trivial to DOS (by forcing continuous crashes and the subsequent reboots) any Cisco 7960 that you can route UDP packets to... Matt Hardeman PaperSoft -- --- Josh Howlett, Networking Digital Communications, Information Systems Computing, University of Bristol, U.K. 'phone: 0117 928 7850 email: [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] ISDN PRI E1 configuration with E100P
Look in zttool or head /proc/zaptel/[1-5] to see if the spans are in alarms. The leds on your boards might not lit properly. regards Martin On Fri, 11 Jul 2003 [EMAIL PROTECTED] wrote: yes, now i got the that problem solved, with 'immediate=yes', but now i've faced with another, When I connect the E1 to the cards, the LED lights does not change to green, in E100P, its blinking red (even when there is no E1 plugged, its blinking red) and in E400P, its solid red. (this was the same even without 'immediate=yes') When i start asterisk, the cards starts fine (as we see), givin the following output, *CLI == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 . . . -- B-channel 31 successfully restarted on span 1 after i put 'immediate=yes', i can even call from outside to the PRI E1, and get connected to asterisk, and listen to the prompts played by asterisk -- Accepting call from '062279955' to 's' on channel 1, span 1 -- Executing Wait(Zap/1-1, 2) in new stack -- Executing Answer(Zap/1-1, ) in new stack -- Executing DigitTimeout(Zap/1-1, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(Zap/1-1, 10) in new stack -- Set Response Timeout to 10 -- Executing BackGround(Zap/1-1, demo-congrats) in new stack -- Playing 'demo-congrats' But calling to outside from asterisk fails (with the following errors), we don't know whether this related to the above problem Executing Dial(SIP/802-8a26, Zap/g2/0129063800|20|t) in new stack NOTICE[262160]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type 'Zap' == Everyone is busy at this time -- Executing Congestion(SIP/802-8a26, ) in new stack == Spawn extension (sip, 980129063800, 2) exited non-zero on 'SIP/802-8a26' Following are new zaptel and zapata conf files, zapata.conf [channels] transfer=yes echocancel=yes callprogress=yes immediate=yes ;E100p card switchtype=EuroISDN signalling=pri_cpe pridialplan=unknown context=inbound-pstn group=2 channel = 1-15,17-31 zaptel.conf #E100p card span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 defaultzone=us loadzone=us Thank you, Surajee - Original Message - From: Cristi [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 11, 2003 10:39 PM Subject: Re: [Asterisk-Users] ISDN PRI E1 configuration with E100P [EMAIL PROTECTED] wrote: hi Everyone, We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output, *CLI == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 . . . -- B-channel 31 successfully restarted on span 1 but, when we make a call to this E1 from outside, it gives the following error, WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not found? WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 1 WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not found? WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 2 does anybody hav an idea on this? our zaptel.conf is, #E100p card span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf, ;E100p card switchtype=EuroISDN signalling=pri_cpe pridialplan=unknown context=incoming group = 2 channel = 1-15,17-31 Thanks inadvance, Surajee --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users I was having the same problem because : 1 number for E1 and the local PTSN was not sending the DID to select the appropriate extension. Set the immediate=yes into zapata.conf and catch the call into s extension! Thanks to Martin Pycko ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --This mail sent through OmniBIS.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] OH323 + G729 + Go2Call
My driver doesn't need it, why should yours? Jeremy McNamara Michael Manousos wrote: Jeremy McNamara wrote: Dave Alan Caruana wrote: I am trying the exact same thing and getting a message -- Called h323:[EMAIL PROTECTED] This is not a proper command... I have absolutely no clue where those other hacks got the h323: bullshit from. RTFM. 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
Re: [Asterisk-Users] Compile Problems with gcc 3.3
Compiles OK for me on SuSE 8.2 Professional with k_smp-2.4.20-86 and gcc-3.3-23 on a Tyan S2462 with a Asterisk CVS snapshot from 1700UTC. [EMAIL PROTECTED]:~/digium/asterisk md5sum pbx.c c5b9063e18fe10a5f07054061c2ecd18 pbx.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-05/10/03-10:40:13\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\-DBUSYDETECT -c -o pbx.o pbx.c pbx.c: In function `ast_extension_match': pbx.c:562: warning: comparison between signed and unsigned pbx.c: In function `extension_close': pbx.c:581: warning: comparison between signed and unsigned pbx.c: In function `pbx_substitute_variables_temp': pbx.c:765: warning: comparison between signed and unsigned pbx.c:812: warning: comparison between signed and unsigned pbx.c: In function `pbx_builtin_stripmsd': pbx.c:4026: warning: comparison between signed and unsigned pbx.c: In function `load_pbx': pbx.c:4373: warning: comparison between signed and unsigned and the remainder of the make proceeds to completion. (I did a make clean prior to this). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why mp3 (licensing issues) as opposed to Open Source OGG
On Thursday 10 July 2003 03:56 pm, marrandy wrote: Just wondering. http://www.vorbis.com/ Because when mp3 support was written, Ogg Vorbis was not yet complete. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM10B - Dies after a few hours
On Thursday 10 July 2003 01:15 pm, Brad Bergman wrote: Thanks, though I don't see PCI Master Abort. I do get Freshmaker failed register test a few times, and I'm basically lost. ... fa != ff fb != ff fc != ff fd != ff fe != ff Freshmaker failed register test This may sound odd, but try upgrading your power supply to a higher wattage (worked for us). -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why mp3 (licensing issues) as opposed to Open Source OGG
On Friday 11 July 2003 01:19 pm, Tilghman Lesher wrote: On Thursday 10 July 2003 03:56 pm, marrandy wrote: Just wondering. http://www.vorbis.com/ Because when mp3 support was written, Ogg Vorbis was not yet complete. -Tilghman Hello Tilghman. Knew there had to be a good reason. I hope it's added and becomes standard. Thanks. Regards...Martin -- Claret is the liquor for boys; port for men; but he who aspires to be a hero ... must drink brandy. -- Samuel Johnson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 + G729 + Go2Call
Jeremy McNamara wrote: My driver doesn't need it, why should yours? Because I use OpenH323_1.12.0 Jeremy McNamara Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] channel bank configuration
Ok, my channel bank is configured. The signaling is configured for fxsloopstart. The framing is esf. The line is b8zs. When I plug in a crossover cable from the T100P to the channel bank my T1 line status indicator shows a red light. However if I plug a crossover cable straight from the Dmark (PRI) to the channel bank my T1 line status indicator shows green. So I'm thinking my problem is with either my T100P cards/asterisk or the crossover cable but I don't think it's the crossover cable. Any suggestions on what I should do next or how I can trouble shoot it? On 11 Jul 2003, Steven Critchfield wrote: On Fri, 2003-07-11 at 08:59, [EMAIL PROTECTED] wrote: I have a premisys slimline 24 channel fxs channel bank. I'm attempting to get it configured to work with my asterisk server. I have 2 T100P cards in the asterisk box. One is connected to an incoming pri, the second is connected to the channel bank. In my /etc/zaptel.conf file I have the line fxoks=25-48 In my zapata.conf file I have the line signaling=fxo_ks When talking to the company I bought the channel bank from, they wish to know what type of signaling the channel bank will get. Apparently their only familiar with connectling the channel bank straight to the T1 line. What type of signalling do I need to advise them that the channel bank is receiving? He advised me that they commonly see loopstart signaling. When I mentioned Koolstart to him, he was unfamiliar with it. I went with fxoks because that is what someone on the list recommended. All suggestions are appreciated. AJ Koolstart is loopstart with disconnect supervision. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP call from one extention to another
Hi I am trying to call from Linphone on extention 109 to Xlite on extention 108 and I get this error -- to 216.75.167.18:5068 WARNING[98315]: File pbx.c, Line 1133 (pbx_extension_helper): No application 'Dial ' for extension (sip, 108, 1) == Spawn extension (sip, 108, 1) exited non-zero on 'SIP/sergeXlite-be43' - Can you tell me what might be wrong with my setup? Thanks Serge _ Tired of spam? Get advanced junk mail protection with MSN 8. http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 + G729 + Go2Call
So do I. Get a clue Michael Manousos wrote: Jeremy McNamara wrote: My driver doesn't need it, why should yours? Because I use OpenH323_1.12.0 Jeremy McNamara 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] Configuring BudgeTone and ringer over TFTP
I noticed that the BudgeTone (I have the 102) with the latest firmware tries to download a file called cfg.txt (presumably the configuration) and a file called ring.bin (presumably a ringer) from the tftp server. The ring-in sound on the budgetone is the same as the ring-out sound and that is going to be confusing for users. I contacted GrandStream and was informed that both of these formats are available for licensing which sounded a bit odd - does anyone have other info? I really like this phone as an entry-level IP phone. It has great sound and works perfectly with asterisk. For the price it will be hard to beat. Too bad it still has these few usability issues... I'm looking forward to seeing the 102D. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Recording
Can Asterisk automatically record all calls to unique files, like voicemail does with the messages? __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Recording
Sure you just need to use Monitor and Changemonitor apps. A little bit of scripting is a must though to get a unique id eg a current date in seconds. I'm not sure if asterisk has it already. regards Martin On Fri, 11 Jul 2003, Erik Kendall wrote: Can Asterisk automatically record all calls to unique files, like voicemail does with the messages? __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.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] Call Recording
By default the Monitor resource/app uses the channel name as the filename,
but you can override the filename base. A good choice of the filename
base would be the uniqueid for each channel, fortunately the ${UNIQUEID}
channel variable is available.
So from extensions.conf you can do Monitor(wav,${UNIQUEID}) to record
'wav'.
The uniqueid is available in the cdr struct as well, but it isnt used
right now for backwards compatibility. You can edit cdr/cdr_csv.c and
uncomment the: /* #define CSV_LOGUNIQUEID 1 */ line to get it to log the
uniqueid at the end of each entry.
James
On Fri, 11 Jul 2003, Erik Kendall wrote:
Can Asterisk automatically record all calls to unique
files, like voicemail does with the messages?
___
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Re: [Asterisk-Users] Configuring BudgeTone and ringer over TFTP
Also budgetone tries to get a ring.bin from the tftp server. but never seen one and don't know the format. Perhaps that's a ringer file ;) ? Matteo. Il ven, 2003-07-11 alle 21:44, John Laur ha scritto: I noticed that the BudgeTone (I have the 102) with the latest firmware tries to download a file called cfg.txt (presumably the configuration) and a file called ring.bin (presumably a ringer) from the tftp server. The ring-in sound on the budgetone is the same as the ring-out sound and that is going to be confusing for users. I contacted GrandStream and was informed that both of these formats are available for licensing which sounded a bit odd - does anyone have other info? I really like this phone as an entry-level IP phone. It has great sound and works perfectly with asterisk. For the price it will be hard to beat. Too bad it still has these few usability issues... I'm looking forward to seeing the 102D. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator - IT services Website : http://www.espia.it Email : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Sound via Sip Phone
Hi, I just setup a box with RH 9, and latest asterisk via CVS. The box as a T100P card in it that is currently hooked up to nothing. I did have the sample configs in place via make samples, and the only change I made was to add an entry to sip.conf for my Cisco 7960. When I dial 1000 to get to the main greeting I hear nothing, though the command line output looks fine to me. Any ideas? -- Executing Goto(SIP/306-8509, default|s|1) in new stack -- Goto (default,s,1) -- Executing Wait(SIP/306-8509, 1) in new stack -- Executing Answer(SIP/306-8509, ) in new stack -- Executing DigitTimeout(SIP/306-8509, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(SIP/306-8509, 10) in new stack -- Set Response Timeout to 10 -- Executing BackGround(SIP/306-8509, demo-congrats) in new stack -- Playing 'demo-congrats' == Spawn extension (default, s, 5) exited non-zero on 'SIP/306-8509' Thanks, Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Recording
I uncommented the variable and recompiled *, but I
can't seem to figure out how to add
Monitor(wav,${UNIQUEID}) to my extensions.conf file.
How would I monitor the incoming zap channel in the
following extensions.conf?
[incoming]
exten = s,1,Zapateller(answer|nocallerid)
exten = s,2,Wait(1)
exten = s,3,Background(monitor-main)
exten = s,5,Dial,Zap/2|20
exten = s,6,VoiceMail,u1000
Thanks for your help,
Erik
--- James Golovich [EMAIL PROTECTED] wrote:
By default the Monitor resource/app uses the channel
name as the filename,
but you can override the filename base. A good
choice of the filename
base would be the uniqueid for each channel,
fortunately the ${UNIQUEID}
channel variable is available.
So from extensions.conf you can do
Monitor(wav,${UNIQUEID}) to record
'wav'.
The uniqueid is available in the cdr struct as well,
but it isnt used
right now for backwards compatibility. You can edit
cdr/cdr_csv.c and
uncomment the: /* #define CSV_LOGUNIQUEID 1 */ line
to get it to log the
uniqueid at the end of each entry.
James
On Fri, 11 Jul 2003, Erik Kendall wrote:
Can Asterisk automatically record all calls to
unique
files, like voicemail does with the messages?
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[Asterisk-Users] audio pause/delay problems
[I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP-PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost. After that it seems that either my client (linphone) or Asterisk doesn't want to resynchronize -- what gets played back is all voice packets as they have been received. This creates an increasing lag in the conversation and the only way I've found to fix it is to disconnect and reconnect again. Is anyone else seeing this? Is it linphone's fault, or is it expected behavior? Now, I have tried running another * on my side of the link. The setup then becomes: linphone - * - internet (IAX2) - * - PSTN (or echo). I'm testing with the echo application (GSM used everywhere) and I'm getting the same thing: everything seems to work, but sooner or later there is an audio pause and the delay grows. It never gets back to normal. I've had it grow to as much as 10s. What makes it even more surprising is the network performance. I've had ping running in the background, same TOS settings, 10 packets per second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 with 0% loss! That's a pretty good network. So where do the pauses and delays come from? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What does callerid= in sip.conf do?
Hi since callerid= in sip.conf doesn't set the Caller ID, I suppose it must be there for some other reason. Is this a not-yet-working feature for future releases of Asterisk? If not, what does it actually do? thanks regards bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Weird experience with MOH
Hi I thought I share this one, just in case this is an indication of some bug ... When I was trying to use music on hold at first, I didn't bother to copy any music into /var/lib/asterisk/mohmp3 since there was a sample- hold.mp3 in there which played just fine in a standalone MP3 player. But after uncommenting one of the lines in musiconhold.conf and doing reload on the console, there was only silence when putting a caller on hold. Somebody told me I may have the wrong mp3 app (321 vs 123) while I was getting busy with something else and so I put this aside. Although I found that I did have mpg123 installed. Yesterday, I copied some music files into /var/lib/asterisk/mohmp3 in anticipation that I would get this to work eventually and to my surprise, putting a caller on hold now plays the music. I have no idea why it didn't work at first, but it would seem that for some unknown reason, Asterisk didn't like the sole sample-hold.mp3 file. rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP immediate hangups with latest CVS
I've been banging my head on this for several hours, and I have no idea what's going on. Maybe there is a very simple result, and I've been looking too hard at this this evening. This is a brand new system, and I'm wondering if there have been SIP bugs introduced in the latest CVS that are preventing from working what should be a stupendously simple test. - Cisco 7960 (non-NATed) - RH 8.0 - Asterisk CVS update as of ~8:00 PM EDT - full make clean; make install on [asterisk,zaptel,libpri] - 2ghz box with E1 card (that's pretty much not part of the equation) I have boiled the configuration down to an extremely (_extremely_) simple setup, and it does not work. SIP calls from the 7960 are hanging up almost immediately, with no audio getting through. It seems that the hangup happens just after the moment that the 7960 sends the ACK message (judging from the debug below, at least.) I have verified that demo-congrats is there, as my original problem stemmed from strange behavior with Zap dialing, and I kept simplifying, so this is the culmination of winnowing down the options to the most basic config. The same phone works flawlessly with other lines that are configured on it to other * servers. Here is my entire relevant configuration. It's as simple as you can get, really. I dial 14109850123 (as a test number - it matches the _1X. list) and I get an almost instant hangup. --- ;sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls dtmfmode=rfc2833 allow=all [3015321510] type=friend username=3015321510 secret=fluffernutter host=dynamic context=from-sip allow=all --- ;extensions.conf [general] static=yes writeprotect=yes [from-sip] exten = _1X.,1,SetCallerID(3015321510) exten = _1X.,2,Answer exten = _1X.,3,Playback(demo-congrats) exten = h,1,Hangup exten = t,1,Hangup exten = i,1,Hangup --- Other strange notes: - quite often, when launching with -gcd I get a segfault. I have the cores, if anyone is interested. - I have almost identical systems (same hardware, same MB, etc.) churning away with no problems with slightly older revs of code *CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Sat, 12 Jul 2003 03:24:34 GMT CSeq: 101 INVITE User-Agent: CSCO/4 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Content-Type: application/sdp Content-Length: 247 Accept: application/sdp Remote-Party-ID: 3015321510 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 15975 21108 IN IP4 128.151.224.33 s=SIP Call c=IN IP4 128.151.224.33 t=0 0 m=audio 19364 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 14 headers, 11 lines Using latest request as basis request Sending to 128.151.224.33 : 5060 (non-NAT) Found audio format 0 Found audio format 8 Found audio format 18 Found audio format 101 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 2147483647, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them - 1, combined - 1 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED];tag=as74174b76 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=2c9c06be Content-Length: 0 to 128.151.224.33:5060 Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED];tag=as74174b76 Call-ID: [EMAIL PROTECTED] Date: Sat, 12 Jul 2003 03:24:34 GMT CSeq: 101 ACK Content-Length: 0 8 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Sat, 12 Jul 2003 03:24:34 GMT CSeq: 102 INVITE User-Agent: CSCO/4 Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=3015321510,realm=asterisk,uri=sip:64.33.1.8,response=4a9e7d0429571ec4047634179fc43f2d,nonce=2c9c06be,algorithm=md5 Expires: 180 Content-Type: application/sdp Content-Length: 247 Remote-Party-ID: 3015321510 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 15975 21108 IN IP4 128.151.224.33 s=SIP Call c=IN IP4 128.151.224.33 t=0 0 m=audio 19364 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000
Re: [Asterisk-Users] SIP immediate hangups with latest CVS
I had this a while back, and set canreinvite=no, and it fixed it. -d At 08:42 PM 7/11/2003 -0700, you wrote: I've been banging my head on this for several hours, and I have no idea what's going on. Maybe there is a very simple result, and I've been looking too hard at this this evening. This is a brand new system, and I'm wondering if there have been SIP bugs introduced in the latest CVS that are preventing from working what should be a stupendously simple test. - Cisco 7960 (non-NATed) - RH 8.0 - Asterisk CVS update as of ~8:00 PM EDT - full make clean; make install on [asterisk,zaptel,libpri] - 2ghz box with E1 card (that's pretty much not part of the equation) I have boiled the configuration down to an extremely (_extremely_) simple setup, and it does not work. SIP calls from the 7960 are hanging up almost immediately, with no audio getting through. It seems that the hangup happens just after the moment that the 7960 sends the ACK message (judging from the debug below, at least.) I have verified that demo-congrats is there, as my original problem stemmed from strange behavior with Zap dialing, and I kept simplifying, so this is the culmination of winnowing down the options to the most basic config. The same phone works flawlessly with other lines that are configured on it to other * servers. Here is my entire relevant configuration. It's as simple as you can get, really. I dial 14109850123 (as a test number - it matches the _1X. list) and I get an almost instant hangup. --- ;sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls dtmfmode=rfc2833 allow=all [3015321510] type=friend username=3015321510 secret=fluffernutter host=dynamic context=from-sip allow=all --- ;extensions.conf [general] static=yes writeprotect=yes [from-sip] exten = _1X.,1,SetCallerID(3015321510) exten = _1X.,2,Answer exten = _1X.,3,Playback(demo-congrats) exten = h,1,Hangup exten = t,1,Hangup exten = i,1,Hangup --- Other strange notes: - quite often, when launching with -gcd I get a segfault. I have the cores, if anyone is interested. - I have almost identical systems (same hardware, same MB, etc.) churning away with no problems with slightly older revs of code *CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Sat, 12 Jul 2003 03:24:34 GMT CSeq: 101 INVITE User-Agent: CSCO/4 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Content-Type: application/sdp Content-Length: 247 Accept: application/sdp Remote-Party-ID: 3015321510 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 15975 21108 IN IP4 128.151.224.33 s=SIP Call c=IN IP4 128.151.224.33 t=0 0 m=audio 19364 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 14 headers, 11 lines Using latest request as basis request Sending to 128.151.224.33 : 5060 (non-NAT) Found audio format 0 Found audio format 8 Found audio format 18 Found audio format 101 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 2147483647, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them - 1, combined - 1 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED];tag=as74174b76 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=2c9c06be Content-Length: 0 to 128.151.224.33:5060 Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED];tag=as74174b76 Call-ID: [EMAIL PROTECTED] Date: Sat, 12 Jul 2003 03:24:34 GMT CSeq: 101 ACK Content-Length: 0 8 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 128.151.224.33:5060 From: 3015321510 sip:[EMAIL PROTECTED];tag=0002b9eb0ef4012c1a228361-11beb479 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Sat, 12 Jul 2003 03:24:34 GMT CSeq: 102 INVITE User-Agent: CSCO/4 Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=3015321510,realm=asterisk,uri=sip:64.33.1.8,response=4a9e7d0429571ec4047634179fc43f2d,nonce=2c9c06be,algorithm=md5 Expires: 180 Content-Type: application/sdp Content-Length: 247 Remote-Party-ID: 3015321510 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 15975 21108 IN IP4 128.151.224.33
[Asterisk-Users] Hook Flash INFO messages
Here is a question that needs a few opinions... Recently we installed a couple of FXS gateways into a site with aSIP Proxy/Softswitchother than Asterisk. One of the things that the users on this site need to do is receive calls on single line phones on the FXS gateways and then hookflash and transfer them to other SIP users. We found that the FXS units, true to their nature asVoIP gateways, saw the hookflash and passed a SIP INFO (event hookflash) back to the Proxy. The Proxy sent this message on to the calling SIP phone which replied that this "feature is not implemented." The gateway manufacturer says that theProxy should process the INFO packet, place the calling endpoint on hold (as a PBX would), stream dialtone to the gateway prompting the user to dial the digits indicating the destination to whom the calling party should be transferred, and then do a transfer. The Proxy manufacturer says that the gateway should see the hookflash,Hold the caller locally (as a SIP phone would), and give new dialtone to the single line phone prompting the user to dial the digits digits indicating the destination to whom the calling party should be transferred, and then send a complete transfer sequence to the Proxy. My question is, how would Asterisk handlea situation like this? Are there any opinions as to how this scenario should be handled? Sean
Re: [Asterisk-Users] audio pause/delay problems
John == John Todd [EMAIL PROTECTED] writes: John For what it's worth, I have noticed the same problem, but I think John the problem is in IAX2, since my long-haul portions of the John diagram were over IAX2, while my SIP clients are almost always John sitting on the same LAN as the Asterisk server. I have noticed these problems both in this kind of setup and in a SIP call to a remote Asterisk server. John What codec were you testing with over IAX2? GSM. --J. [I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP-PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost. After that it seems that either my client (linphone) or Asterisk doesn't want to resynchronize -- what gets played back is all voice packets as they have been received. This creates an increasing lag in the conversation and the only way I've found to fix it is to disconnect and reconnect again. Is anyone else seeing this? Is it linphone's fault, or is it expected behavior? Now, I have tried running another * on my side of the link. The setup then becomes: linphone - * - internet (IAX2) - * - PSTN (or echo). I'm testing with the echo application (GSM used everywhere) and I'm getting the same thing: everything seems to work, but sooner or later there is an audio pause and the delay grows. It never gets back to normal. I've had it grow to as much as 10s. What makes it even more surprising is the network performance. I've had ping running in the background, same TOS settings, 10 packets per second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 with 0% loss! That's a pretty good network. So where do the pauses and delays come from? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users John ___ Asterisk-Users John mailing list [EMAIL PROTECTED] John 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] Weird experience with MOH
If you're on a RedHat system, mpg321 is installed by default, and is symlinked to as mpg123... So, it can easily look like you have mpg123, but you really have mpg321... Sorry if you checked for that, and I've offended, but just thought I'd offer. Matt Hardeman PaperSoft - Original Message - From: BK [address only for mailing lists] [EMAIL PROTECTED] To: Asterisk List [EMAIL PROTECTED] Sent: Friday, July 11, 2003 10:09 PM Subject: [Asterisk-Users] Weird experience with MOH Hi I thought I share this one, just in case this is an indication of some bug ... When I was trying to use music on hold at first, I didn't bother to copy any music into /var/lib/asterisk/mohmp3 since there was a sample- hold.mp3 in there which played just fine in a standalone MP3 player. But after uncommenting one of the lines in musiconhold.conf and doing reload on the console, there was only silence when putting a caller on hold. Somebody told me I may have the wrong mp3 app (321 vs 123) while I was getting busy with something else and so I put this aside. Although I found that I did have mpg123 installed. Yesterday, I copied some music files into /var/lib/asterisk/mohmp3 in anticipation that I would get this to work eventually and to my surprise, putting a caller on hold now plays the music. I have no idea why it didn't work at first, but it would seem that for some unknown reason, Asterisk didn't like the sole sample-hold.mp3 file. rgds bk ___ 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] ISDN PRI E1 configuration with E100P
here are the outputs, for, zttool, Alarms Span RED LSS Wildcard T100P T1/PRI Board card 0 for head /proc/zaptel/1 Span 1: WCT1/0 LSS Wildcard T100P T1/PRI Board card 0 HDB3/CCS/CRC4 RED 1 WCT1/0/1 ClearChannel 2 WCT1/0/2 ClearChannel 3 WCT1/0/3 ClearChannel 4 WCT1/0/4 ClearChannel 5 WCT1/0/5 ClearChannel 6 WCT1/0/6 ClearChannel 7 WCT1/0/7 ClearChannel 8 WCT1/0/8 ClearChannel 9 WCT1/0/9 ClearChannel any idea? Surajee - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 12, 2003 12:41 AM Subject: Re: [Asterisk-Users] ISDN PRI E1 configuration with E100P Look in zttool or head /proc/zaptel/[1-5] to see if the spans are in alarms. The leds on your boards might not lit properly. regards Martin On Fri, 11 Jul 2003 [EMAIL PROTECTED] wrote: yes, now i got the that problem solved, with 'immediate=yes', but now i've faced with another, When I connect the E1 to the cards, the LED lights does not change to green, in E100P, its blinking red (even when there is no E1 plugged, its blinking red) and in E400P, its solid red. (this was the same even without 'immediate=yes') When i start asterisk, the cards starts fine (as we see), givin the following output, *CLI == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 . . . -- B-channel 31 successfully restarted on span 1 after i put 'immediate=yes', i can even call from outside to the PRI E1, and get connected to asterisk, and listen to the prompts played by asterisk -- Accepting call from '062279955' to 's' on channel 1, span 1 -- Executing Wait(Zap/1-1, 2) in new stack -- Executing Answer(Zap/1-1, ) in new stack -- Executing DigitTimeout(Zap/1-1, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(Zap/1-1, 10) in new stack -- Set Response Timeout to 10 -- Executing BackGround(Zap/1-1, demo-congrats) in new stack -- Playing 'demo-congrats' But calling to outside from asterisk fails (with the following errors), we don't know whether this related to the above problem Executing Dial(SIP/802-8a26, Zap/g2/0129063800|20|t) in new stack NOTICE[262160]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type 'Zap' == Everyone is busy at this time -- Executing Congestion(SIP/802-8a26, ) in new stack == Spawn extension (sip, 980129063800, 2) exited non-zero on 'SIP/802-8a26' Following are new zaptel and zapata conf files, zapata.conf [channels] transfer=yes echocancel=yes callprogress=yes immediate=yes ;E100p card switchtype=EuroISDN signalling=pri_cpe pridialplan=unknown context=inbound-pstn group=2 channel = 1-15,17-31 zaptel.conf #E100p card span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 defaultzone=us loadzone=us Thank you, Surajee - Original Message - From: Cristi [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 11, 2003 10:39 PM Subject: Re: [Asterisk-Users] ISDN PRI E1 configuration with E100P [EMAIL PROTECTED] wrote: hi Everyone, We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output, *CLI == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 . . . -- B-channel 31 successfully restarted on span 1 but, when we make a call to this E1 from outside, it gives the following error, WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not found? WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 1 WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not found? WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 2 does anybody hav an idea on this? our zaptel.conf is, #E100p card span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf, ;E100p card switchtype=EuroISDN signalling=pri_cpe pridialplan=unknown context=incoming group = 2 channel = 1-15,17-31 Thanks inadvance, Surajee --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users I was having the same problem because : 1 number for E1 and the local PTSN was not sending the DID to select the appropriate extension. Set the immediate=yes
Re: [Asterisk-Users] Hook Flash INFO messages
On Fri, 2003-07-11 at 22:12, Sean P. Robertson wrote: Here is a question that needs a few opinions... Recently we installed a couple of FXS gateways into a site with a SIP Proxy/Softswitch other than Asterisk. One of the things that the users on this site need to do is receive calls on single line phones on the FXS gateways and then hookflash and transfer them to other SIP users. We found that the FXS units, true to their nature as VoIP gateways, saw the hookflash and passed a SIP INFO (event hookflash) back to the Proxy. The Proxy sent this message on to the calling SIP phone which replied that this feature is not implemented. The gateway manufacturer says that the Proxy should process the INFO packet, place the calling endpoint on hold (as a PBX would), stream dialtone to the gateway prompting the user to dial the digits indicating the destination to whom the calling party should be transferred, and then do a transfer. The Proxy manufacturer says that the gateway should see the hookflash, Hold the caller locally (as a SIP phone would), and give new dialtone to the single line phone prompting the user to dial the digits digits indicating the destination to whom the calling party should be transferred, and then send a complete transfer sequence to the Proxy. My question is, how would Asterisk handle a situation like this? Are there any opinions as to how this scenario should be handled? Asterisk currently only handles dtmf INFO messages. --Karl Sean -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
