Re: [Asterisk-Users] Call recording
It will be great man, I am looking forward for your patch Hemant - Original Message - From: Fettahlioglu, Mahmut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 06, 2003 4:42 AM Subject: RE: [Asterisk-Users] Call recording I also did a change to asterisk to be able to record a channel irrespective of the application it is currently in. You can start/stop recording by calling an application from the dialplan or sending a message using manager. I'll be submitting a patch to Mark shortly. -Original Message- From: Michiel Betel [mailto:[EMAIL PROTECTED] Sent: Thursday, 6 March 2003 7:33 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Call recording I don't know (haven't tried myself) but Kostya V. Ivanov's 'R' patch to the dial application (december 2002) might be of help for you. Check the archives for Barge (Intrusion) Capabilities. It might be some manual work to apply after all the allmost daily CVS changes but worth a try! Michiel Betel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian J. Schrock Sent: woensdag 5 maart 2003 17:55 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call recording Hello, How would I go ahead a record all phone calls into and out of my asterisk server. I know the legality issues behind it, but I could always play a recording to let people know they will be recorded. Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-537-2817 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017 ___ 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] Dev kit and poor audio quality
Hi All... I spent many hours playing with Asterisk after my dev kit arrived today. One thing I noticed right off is that the audio quality of the voice mail menu items is quite poor. I thought this was a little odd, since the quality of the recorded messages was quite good. I am running on quite a low end PC, a P2 266MHz, just for testing. Is this an issue confined to the dev kit or perhaps the low end PC? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisico ATA licence
I can buy a new ATA186 here, but it is sold with a 1-port user license UK, for euro 192, but does that license stop me from using both ports? I can't read the license agreement till I buy the thing, so I don't know what i'm buying... Michiel -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Cisico ATA licence
the license is needed only with cisco callmanager. so you can ignore it and use both ports with asterisk ;-) -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Michiel Betel Inviato: giovedì 6 marzo 2003 10.43 A: [EMAIL PROTECTED] Oggetto: [Asterisk-Users] Cisico ATA licence I can buy a new ATA186 here, but it is sold with a 1-port user license UK, for euro 192, but does that license stop me from using both ports? I can't read the license agreement till I buy the thing, so I don't know what i'm buying... Michiel -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP INVITEs borked with iconnecthere
I don't think it is a filter (SIP uses UDP, so telneting to the port (which uses TCP) doesn't mean much. There is a problem right now with receiving incoming calls from iconnecthere/d3. Something on d3's end is messed up. It hits the asterisk server, and then cancels the connection. I'm stumped. On Wed, 2003-03-05 at 23:12, [EMAIL PROTECTED] wrote: John, A heads-up: iconnect has apparently put up a filter against my IP address, for whichever reason (apparently they don't like people using asterisk?). I've sent them an email and am pursuing this also through sales side (I'm about to make a resale deal with them), so hopefully tomorrow I'll find out just what they don't like about asterisk. To check if they put up a filter, just do a telnet 213.137.73.141 5060 If you get connection refused, its something else. if it times out, you've been filtered. -alex On Wed, 5 Mar 2003, John Todd wrote: Symptoms: when calling my iconnect phone number (13033913323 in my bogus example below) from my cell phone, I can see that the call makes it to my asterisk server, and my phones even ring once as * passes the call through during the 180 Ringing period. However, it seems that iconnecthere.com cannot see my 100 Trying and 180 Ringing messages, as they continue to send INVITES to me. After two seconds, they either give up or error out and send a CANCEL message. To further increase my suspicions of something weird in the ability to see my replies, they send 11 CANCEL messages over the period of 30 seconds, despite my 200 OK replies. Notes: 204.31.11.32 resolves to asterisk.something.com. 204.31.11.35 is my ATA-186. Neither the domains nor the IP addresses are real, except when referencing iconnect servers. 213.137.73.176 is the real IP address of the SIP proxy at iconnecthere.com (deltathree.com) Note that I actually do my REGISTERs against 213.137.73.178, not .176 - not a big deal, but who knows what clues will be helpful. Unsuccessful Asterisk-iconnect-PSTN call: tethereal port 5060 and host 213.137.73.176 Capturing on fxp0 0.00 213.137.73.176 - asterisk.something.com SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 0.001293 asterisk.something.com - 213.137.73.176 SIP Status: 100 Trying 0.039058 asterisk.something.com - 213.137.73.176 SIP Status: 180 Ringing 0.490181 213.137.73.176 - asterisk.something.com SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 0.490497 asterisk.something.com - 213.137.73.176 SIP Status: 100 Trying 1.530125 213.137.73.176 - asterisk.something.com SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 1.530439 asterisk.something.com - 213.137.73.176 SIP Status: 100 Trying 2.070160 213.137.73.176 - asterisk.something.com SIP Request: CANCEL sip:[EMAIL PROTECTED]:5060 2.070461 asterisk.something.com - 213.137.73.176 SIP Status: 200 OK 2.594680 213.137.73.176 - asterisk.something.com SIP Request: CANCEL sip:[EMAIL PROTECTED]:5060 2.595419 asterisk.something.com - 213.137.73.176 SIP Status: 200 OK 3.634908 213.137.73.176 - asterisk.something.com SIP Request: CANCEL sip:[EMAIL PROTECTED]:5060 3.635179 asterisk.something.com - 213.137.73.176 SIP Status: 200 OK 5.674595 213.137.73.176 - asterisk.something.com SIP Request: CANCEL sip:[EMAIL PROTECTED]:5060 5.674889 asterisk.something.com - 213.137.73.176 SIP Status: 200 OK 9.664659 213.137.73.176 - asterisk.something.com SIP Request: CANCEL sip:[EMAIL PROTECTED]:5060 9.664956 asterisk.something.com - 213.137.73.176 SIP Status: 200 OK 13.645471 213.137.73.176 - asterisk.something.com SIP Request: CANCEL sip:[EMAIL PROTECTED]:5060 13.645755 asterisk.something.com - 213.137.73.176 SIP Status: 200 OK 17.635194 213.137.73.176 - asterisk.something.com SIP Request: CANCEL sip:[EMAIL PROTECTED]:5060 17.635502 asterisk.something.com - 213.137.73.176 SIP Status: 200 OK 21.665856 213.137.73.176 - asterisk.something.com SIP Request: CANCEL sip:[EMAIL PROTECTED]:5060 21.666146 asterisk.something.com - 213.137.73.176 SIP Status: 200 OK 25.676487 213.137.73.176 - asterisk.something.com SIP Request: CANCEL sip:[EMAIL PROTECTED]:5060 25.676767 asterisk.something.com - 213.137.73.176 SIP Status: 200 OK 29.666942 213.137.73.176 - asterisk.something.com SIP Request: CANCEL sip:[EMAIL PROTECTED]:5060 29.667231 asterisk.something.com - 213.137.73.176 SIP Status: 200 OK 33.647019 213.137.73.176 - asterisk.something.com SIP Request: CANCEL sip:[EMAIL PROTECTED]:5060 33.647305 asterisk.something.com - 213.137.73.176 SIP Status: 200 OK A successful ATA-186 to iconnect session, no Asterisk server involved: 1338.126534 213.137.73.176 - 204.31.11.35 SIP/SDP Request: INVITE sip:[EMAIL
[Asterisk-Users] MSN Messenger Versions
Hey all, Just wanted to know what versions of MSN Messenger people have working with Asterisk. I had 4.5 or so working but with the new 5.whatever it seems to be a pain in the ass. Thanks.. Regards, Jamie Carl Email: [EMAIL PROTECTED] PH: +61-414-365-466 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Tormenta ISA E1 card
[EMAIL PROTECTED] wrote: Hi,Steve, My ISA E1 card does not work yet. I would like to know if there are other difference bettween then two kind of ISA card? The crystals, the framer chips and the jumpers are the only differences. What are the symptons of it not working? Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MSN Messenger Versions
I seem to have misplaced my copy. Isn't that always the way. Anyone know where I can get an older 4.x version from? Regards, Jamie Carl Email: [EMAIL PROTECTED] PH: +61-414-365-466 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of William X Walsh Sent: Thursday, 6 March 2003 11:50 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MSN Messenger Versions 5 has no SIP support, go back to 4.6 or 4.7 On Thu, 2003-03-06 at 04:20, Jamie Carl wrote: Hey all, Just wanted to know what versions of MSN Messenger people have working with Asterisk. I had 4.5 or so working but with the new 5.whatever it seems to be a pain in the ass. Thanks.. Regards, Jamie Carl Email: [EMAIL PROTECTED] PH: +61-414-365-466 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- William Walsh [EMAIL PROTECTED] Jabber: [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
[Asterisk-Users] asterisk-oh323-0.5.1
How can you use DTMF detection with netmeeting ? Do you have to set something in configuration file ? Regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323-0.5.1
Rattana BIV wrote: How can you use DTMF detection with netmeeting ? Do you have to set something in configuration file ? In oh323.conf set: inBandDTMF=yes Michael. Regards Rattana ___ 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] Hardware requirements.
I am building a system that uses all SIP phones and gatewys external to the * box. Is there any special hardware requirements on the * server other than that needed to run the operating system? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Cisico ATA licence
On Thu, 2003-03-06 at 04:25, Matteo Brancaleoni wrote: the license is needed only with cisco callmanager. so you can ignore it and use both ports with asterisk ;-) Thats wrong according to the debates here and on the FWD mailing list. The unit that Michiel was looking at contains software that connects to the Cisco Call Manager, probably using skinney. What Michiel needs is one with the SIP or H323 software load on it. The units with SIP or H323 loaded on it usually have the license for both ports to use that software. -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Michiel Betel Inviato: giovedì 6 marzo 2003 10.43 A: [EMAIL PROTECTED] Oggetto: [Asterisk-Users] Cisico ATA licence I can buy a new ATA186 here, but it is sold with a 1-port user license UK, for euro 192, but does that license stop me from using both ports? I can't read the license agreement till I buy the thing, so I don't know what i'm buying... Michiel -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Cisico ATA licence
Thanks! Is there a safe way to identify (cisco secret part number or something) what SIP loaded ATA to order, or should I call Cisco? I don't really trust the mailorder company guys to sort it out for me as they probably don't sell that many of these units and will probably go uh??? on me if I start questioning Steven Critchfield said: On Thu, 2003-03-06 at 04:25, Matteo Brancaleoni wrote: the license is needed only with cisco callmanager. so you can ignore it and use both ports with asterisk ;-) Thats wrong according to the debates here and on the FWD mailing list. The unit that Michiel was looking at contains software that connects to the Cisco Call Manager, probably using skinney. What Michiel needs is one with the SIP or H323 software load on it. The units with SIP or H323 loaded on it usually have the license for both ports to use that software. -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Michiel Betel Inviato: giovedì 6 marzo 2003 10.43 A: [EMAIL PROTECTED] Oggetto: [Asterisk-Users] Cisico ATA licence I can buy a new ATA186 here, but it is sold with a 1-port user license UK, for euro 192, but does that license stop me from using both ports? I can't read the license agreement till I buy the thing, so I don't know what i'm buying... Michiel -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [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: R: [Asterisk-Users] Cisico ATA licence
On Thu, 2003-03-06 at 10:44, Florian Overkamp wrote: At 10:20 6-3-2003 -0600, you wrote: On Thu, 2003-03-06 at 04:25, Matteo Brancaleoni wrote: the license is needed only with cisco callmanager. so you can ignore it and use both ports with asterisk ;-) Thats wrong according to the debates here and on the FWD mailing list. The unit that Michiel was looking at contains software that connects to the Cisco Call Manager, probably using skinney. What Michiel needs is one with the SIP or H323 software load on it. The units with SIP or H323 loaded on it usually have the license for both ports to use that software. Incorrect. The load that includes Skinny protocol also speaks MGCP, which is perfectly useable with Asterisk. Besides, as I understand it Cisco won't care that much if you load it up with the SIP/H323 stack either. But no-one will confirm that officially ofcourse. So to keep this from dragging out like the rest of the threads about this have. While they may not prosecute an individual for having loaded an unlicensed stack on the hardware, it is unwise to suggest it in a publicly available and archived list. Do remember here in the US we have to now worry more about John Ashcroft than the company whose software we use/abuse since John can bring charges on his own without the company. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Slight Echo problems
Does anyone have the solution for this. I have a T100P going to a channel bank for all internal extensions. Calls from one internal to another sounds prefect. When placing a outbound call (going out over X100P) there is a very slight echo, almost like the comfort noise is delated by a fraction of a second. Its just a little annoyong, but when moving papers on my desk, it is very noticeable. I turned echo cancellation on for channels 25-26 (X100P), but that doesn't seem to do it. Does the gain affect this in any way? Would a faster computer help? Right now it is running on a 433Mhz celeron. Thanks everyone, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Cisico ATA licence
On Thu, 2003-03-06 at 10:30, Michiel Betel wrote: Thanks! Is there a safe way to identify (cisco secret part number or something) what SIP loaded ATA to order, or should I call Cisco? I don't really trust the mailorder company guys to sort it out for me as they probably don't sell that many of these units and will probably go uh??? on me if I start questioning Everything below is from Richard on the FWD mailing list. I quoted his contribution only. -- Let me have go at trying to help explain this.. There currently exists two (2) different ways someone can buy an ata-186/ata-188 from Cisco. Firstly, you can order product codes ATA186-I1 or ATA186-I2 or ATA188-I1 or ATA188-I1 depending on which model you want and in addition to this you MUST order either SW-SMH-UL-ATA-1P (SIP/MGCP/H.323 license for ATA) or SW-CCM-UL-ANA (Cisco CallManager license for ATA). Note that with the CallManager you need a license for EACH port that has a phone connected to the ATA which is connected to CallManager (ie if both ports talk to callmanager then you need 2 licenses) whereas for the SIP/H.323/MGCP code you only need one license even if you use both ports. Secondly, if you order product codes ATA186-I1-1P-CH1 or ATA186-I2-1P-CH1 or ATA188-I2-1P-CH1 or ATA188-I2-1P-CH1 then this has the license included into the price of the ata. ie in this case you do NOT need to purchase a separate SW-SMH-UL-ATA-1P or SW-CCM-UL-ANA license. Once agin if you use both ports with CallManager you will then need to buy ONE (1) additional SW-CCM-UL-ANA license. On most online stores the reseller lists the manufacturers product code so just check which of the above two product codes you are ordering and you will then know if you need to order a separate license or not. Hope this helps, Richard -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Cisico ATA licence
Well at least we don't have Janet Reno sending tanks in on civilians anymore. Regards MIKE On Thu, 2003-03-06 at 11:59, Steven Critchfield wrote: On Thu, 2003-03-06 at 10:44, Florian Overkamp wrote: At 10:20 6-3-2003 -0600, you wrote: On Thu, 2003-03-06 at 04:25, Matteo Brancaleoni wrote: the license is needed only with cisco callmanager. so you can ignore it and use both ports with asterisk ;-) Thats wrong according to the debates here and on the FWD mailing list. The unit that Michiel was looking at contains software that connects to the Cisco Call Manager, probably using skinney. What Michiel needs is one with the SIP or H323 software load on it. The units with SIP or H323 loaded on it usually have the license for both ports to use that software. Incorrect. The load that includes Skinny protocol also speaks MGCP, which is perfectly useable with Asterisk. Besides, as I understand it Cisco won't care that much if you load it up with the SIP/H323 stack either. But no-one will confirm that officially ofcourse. So to keep this from dragging out like the rest of the threads about this have. While they may not prosecute an individual for having loaded an unlicensed stack on the hardware, it is unwise to suggest it in a publicly available and archived list. Do remember here in the US we have to now worry more about John Ashcroft than the company whose software we use/abuse since John can bring charges on his own without the company. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Problem
I think your extensions.conf entry should look like this. exten = 2111,1,Dial,SIP/18504844535 That is how I do it on my server. On Thursday, March 6, 2003, at 12:33 PM, Eric Wieling wrote: I have a simple problem with sialing a SIP device. I'm SURE it's a syntax problem, but I dunno what it might be. Here are the debug messages: == Accepting call on 'Zap/1-1' (PENSACOLA, FL 8503846785) -- Executing Goto(Zap/1-1, 2111|1) in new stack -- Goto (default,2111,1) -- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED])) in new stack WARNING[12299]: File chan_sip.c, Line 411 (create_addr): No such host: 18504844535) NOTICE[12299]: File app_dial.c, Line 437 (dial_exec): Unable to create channel of type 'SIP' == Everyone is busy at this time -- Executing Answer(Zap/1-1, ) in new stack DEBUG[12299]: File chan_zap.c, Line 1654 (zt_answer): Took Zap/1-1 off hook -- Executing Congestion(Zap/1-1, 5) in new stack Here part of extentions.conf exten = 2111,1,Dial,SIP/[EMAIL PROTECTED] exten = 2111,2,Answer exten = 2111,3,Congestion Here is part of sip.conf [18504844535] type=friend host=dynamic context=default Does anyone have any suggestions? --Eric ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-537-2817 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Debugging
I have debugging on in Asterisk and sip debug. How do I tell what username a SIP client is trying to use to register with Asterisk as? --Eric ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Debugging
It's the From: line. Mark On Thu, 6 Mar 2003, Eric Wieling wrote: I have debugging on in Asterisk and sip debug. How do I tell what username a SIP client is trying to use to register with Asterisk as? --Eric ___ 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] I?m Learning
How i must setup h323 on asterisk? Juan -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Eric Wieling Enviado el: Jueves, 06 de Marzo de 2003 03:56 p.m. Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] SIP Debugging I have debugging on in Asterisk and sip debug. How do I tell what username a SIP client is trying to use to register with Asterisk as? --Eric ___ 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 INVITEs borked with iconnecthere
I don't think it is a filter (SIP uses UDP, so telneting to the port (which uses TCP) doesn't mean much. There is a problem right now with receiving incoming calls from iconnecthere/d3. Something on d3's end is messed up. It hits the asterisk server, and then cancels the connection. I'm stumped. [snip] I'm not convinced it's a problem with deltathree (iconnect). I can make calls directly out of and into my ATA-186 to/from deltathree, and they work fine, so it is the case that some SIP implementations work fine with deltathree's SIP system. Unless there is intentional inability to talk to Asterisk servers, I suspect that something can be changed in Asterisk's implementation of the SIP responses to make it work. Note that I am not saying that it IS asterisk, either, but I would not put either side as the culprit right now. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM sound quality
Thanks a lot!! That fixed it. David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Harragin Sent: Thursday, March 06, 2003 12:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SNOM sound quality Try in sip.config disallow=all allow=ulaw Select 30ms packet size in the snom. John - Original Message - From: David Davis [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 06, 2003 2:40 PM Subject: RE: [Asterisk-Users] SNOM sound quality I have a problem with the snom (200) phones that causes all sound to be 'raspy' or stuttered BUT when I use the same phone and connect to a older version of asterisk, everything sounds fine to the asterisk system (I can check voicemail, use a conf, but can't receive calls or call another snom) I've been told in the irc channel that this is a snom issue, but it seems it can be fixed in asterisk. Maybe * is telling the snom to use the wrong codec? David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus. ___ 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 Response 400
You could turn off reinvite. Mark On Thu, 6 Mar 2003, Eric Wieling wrote: I'm getting the following message: -- Executing Dial(SIP/2111-b825, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/2111-0bd5 answered SIP/2111-b825 -- Attempting native bridge of SIP/2111-b825 and SIP/2111-0bd5 -- Got SIP response 400 Bad Request - 'Malformed/Missing Contact field' back from 172.16.17.1 172.16.17.1 is a Cisco 1750 box using SIP and an FXO card. Any ideas on what might cause this? --Eric ___ 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] SNOM sound quality
Is this GSM or ulaw? You can force ulaw by doing: disallow=all allow=ulaw in your /etc/asterisk/sip.conf in the general section. Mark On Thu, 6 Mar 2003, David Davis wrote: I have a problem with the snom (200) phones that causes all sound to be 'raspy' or stuttered BUT when I use the same phone and connect to a older version of asterisk, everything sounds fine to the asterisk system (I can check voicemail, use a conf, but can't receive calls or call another snom) I've been told in the irc channel that this is a snom issue, but it seems it can be fixed in asterisk. Maybe * is telling the snom to use the wrong codec? David ___ 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] Inexpensive VoIP phones
Hi All... I was wondering if someone could recomend an inexpensive VoIP phone. It need not be fancy. I need a few extensions for places where I need no computers. Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco SIP Weirdness (1750, not ATA)
exten = 2111,1,Dial(SIP/[EMAIL PROTECTED]) exten = 2111,2,Voicemail(u2111) exten = 2111,3,Hangup exten = 2111,100,Voicemail(b2111) exten = 2111,101,Hangup Needs to be 102 and 103... If that doesn't work, find me on IRC. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT working outbound with Asterisk and ATA-186 phones
I still can't get Window Messenger SIP to work, and the SNOM 100 doesn't seem to work through NAT without STUN. -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roderick Montgomery Sent: Thursday, March 06, 2003 6:40 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT working outbound with Asterisk and ATA- 186 phones Thanks to Mark and John! With Mark's nat=1 changes yesterday and John's instructions, I can now call outbound from my NATted ATA-186! Works beautifully. Good job, guys! rm - Roderick Montgomery [EMAIL PROTECTED] URL:http://thecomplex.com/ the fool stands only to fall, but the wise trip on grace... [Sarah Masen] - ___ 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 problems with iconnecthere
This may be slight off topic, but perhaps it has relevance: My iconnecthere account no longer works for inbound calls through NAT using the standard configuration that they provide on their website. I have sent them a message, but I believe it will be flushed down the toilet by the first-tier support people. When I call my iconnect number, it goes directly to voicemail. There aren't even any packets sent to my NAT/ATA-186 device at all; it's as if they didn't see my REGISTER request, despite their OK reply to my REGISTERs. Asterisk has different symptoms for inbound calls from iconnect. See my previous email. In short, they INVITE me to the call, but don't seem to hear my replies and then dump the call into voicemail after timing out. This is not on NAT; this is all clear-IP transactions. If anyone from iconnecthere or deltathree is lurking on the list, please see if you can figure out what is fubar'ed there, if anything. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MSN Messenger Versions
Thanx heaps everyone!! --Jamie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wade Weppler Sent: Friday, 7 March 2003 1:24 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] MSN Messenger Versions I'll try to be a little more specific: http://www.microsoft.com/exchange/downloads/2000/IMClient47.asp This one works for me. -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: Thursday, March 06, 2003 8:45 AM To: [EMAIL PROTECTED]; Jamie Carl Subject: Re: [Asterisk-Users] MSN Messenger Versions microsoft.com On Thursday 06 March 2003 14:20, Jamie Carl wrote: I seem to have misplaced my copy. Isn't that always the way. Anyone know where I can get an older 4.x version from? Regards, Jamie Carl Email:[EMAIL PROTECTED] PH: +61-414-365-466 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of William X Walsh Sent: Thursday, 6 March 2003 11:50 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MSN Messenger Versions 5 has no SIP support, go back to 4.6 or 4.7 On Thu, 2003-03-06 at 04:20, Jamie Carl wrote: Hey all, Just wanted to know what versions of MSN Messenger people have working with Asterisk. I had 4.5 or so working but with the new 5.whatever it seems to be a pain in the ass. Thanks.. Regards, Jamie Carl Email:[EMAIL PROTECTED] PH: +61-414-365-466 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- William Walsh [EMAIL PROTECTED] Jabber: [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 -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. ___ 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] Cisco SIP Weirdness (1750, not ATA)
I changed 100 to 102 and 101 to 103 and the same thing happens. On Thu, Mar 06, 2003 at 05:46:22PM -0600, Mark Spencer wrote: exten = 2111,1,Dial(SIP/[EMAIL PROTECTED]) exten = 2111,2,Voicemail(u2111) exten = 2111,3,Hangup exten = 2111,100,Voicemail(b2111) exten = 2111,101,Hangup Needs to be 102 and 103... If that doesn't work, find me on IRC. Mark ___ 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] Re: Tormenta ISA E1 card
If I comment all lines of zaptel.conf,after I run the ztcfg,the warning in the zttool is OK, that is to say,no warning. If I leave the zaptel.conf's content as: span=1,0,0,ccs,hdb3 span=2,0,0,ccs,hdb3 then the RED warning will appear. I I connect a loop back line into span1's RJ45 socket,then the warning become RED/YELLOW. if I config the span1 as PRI_NET and span2 as PRI_CPE and connect them with a cable ,after I run the asterisk with -c,I could see the Tor channels are started. But when I dial the extenion based on zaptel's channel,I always got the information that the channel is busy. (if I change the chip and crystal to T1's,then all is OK!) Regards john Steve Underwood writes: [EMAIL PROTECTED] wrote: Hi,Steve, My ISA E1 card does not work yet. I would like to know if there are other difference bettween then two kind of ISA card? The crystals, the framer chips and the jumpers are the only differences. What are the symptons of it not working? Regards, Steve ___ 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] X100P question about odd behavior
Hi All... I have installed a single X100P card in my PC and am playing with Asterisk. The wire I plugged into the X100P has two POTS lines on it, wired on the RJ45 in the normal way. I am getting odd behavior. It seems when I dial out that the X100P dials both lines at the same time. I have two questions. First, I see that the X100P is only a single channel. Does this mean that I can only use one POTS line with it? When I installed it I thought that it would support two POTS lines. I guess I thought this because it has an ordinary phone jack that had 4 little metal fingers in it. Is it possible that the X100P is really dialing both lines at the same time and if so is there a way to stop this? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P question about odd behavior
First, I see that the X100P is only a single channel. Does this mean that I can only use one POTS line with it? When I installed it I thought that it would support two POTS lines. I guess I thought this because it has an ordinary phone jack that had 4 little metal fingers in it. Is it possible that the X100P is really dialing both lines at the same time and if so is there a way to stop this? I used the single line (two wire) phone cord that shipped with the X100P. I've noticed this issue with some modems also. If I use a two line (four wire) phone cord they will bleed over onto the second line. Randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P question about odd behavior
On Thursday 06 March 2003 21:43, Jim Archer wrote: Hi All... I have installed a single X100P card in my PC and am playing with Asterisk. The wire I plugged into the X100P has two POTS lines on it, wired on the RJ45 in the normal way. 1. It's not two POTS lines. The second port is a pass-through port. 2. RJ45 is 4 pair. The port on the back of the X100P is 1 pair -- RJ11. I am getting odd behavior. It seems when I dial out that the X100P dials both lines at the same time. That's understandable; if you've connected two lines together, the X100P will take the line off hook. It has no way of knowing that you've plugged two lines together. I have two questions. First, I see that the X100P is only a single channel. Does this mean that I can only use one POTS line with it? When I installed it I thought that it would support two POTS lines. I guess I thought this because it has an ordinary phone jack that had 4 little metal fingers in it. Yes; it's only one channel and it can only handle one line at once. Is it possible that the X100P is really dialing both lines at the same time and if so is there a way to stop this? Don't connect two lines together via the pass-through port? -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip or H332 or *?
I would rather say take whatever is cheap and make it work;) h.323 does work pretty well with *, but if you are using a GK or GW which wants you to authenticate your IP Phone then be carefull H.235 is still not supported in *. Hemant - Original Message - From: Andres Tello Abrego [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 07, 2003 12:56 AM Subject: [Asterisk-Users] Sip or H332 or *? Ok, So, I'm willing to isntall ip phones, I have, to choise: snom 200 cisco phones alcatel phones... snom 200, not in mexico,must import. cisco phone, in mexico, quite expensive, protocl to use; sip alcatel, in mexico, no so expensive, protocol to use: h323 Asteris, which protocol support better? Sip o H323? Any advice is welcome... :) ___ 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] X100P question about odd behavior
I can't speak for the x100p in particular but when a phone jack in an arrangement like that has 4wires (2pair) the second pair is normally just passed straight through from the line to the phone jack. This may be used to supply power to the phones for lighted dials etc, supply end of line resistors to ensure circuit integrity, pass a second line straight through on the same cable, or various other scenarios. Normally though you will only find single pair cable supplied with the device as the patch cable so using that effectively limits your options on the second pair right away. Again I have not actually tested the x100p, but I would assume in offhook mode the phone jack is disconnected from the line jack, but I could be wrong here, I have seen devices that handle this both ways (jacks in parallel and jacks interrupted when off hook.) The usual way to tell is there are 2 relays in the type that disconnect the phone jack from the line jack when off hook (one is the disconnector, and one is the hookswitch), but that is just a clue, not a guarantee. At 10:32 PM 3/6/2003 -0600, you wrote: On Thursday 06 March 2003 21:43, Jim Archer wrote: Hi All... I have installed a single X100P card in my PC and am playing with Asterisk. The wire I plugged into the X100P has two POTS lines on it, wired on the RJ45 in the normal way. 1. It's not two POTS lines. The second port is a pass-through port. 2. RJ45 is 4 pair. The port on the back of the X100P is 1 pair -- RJ11. I am getting odd behavior. It seems when I dial out that the X100P dials both lines at the same time. That's understandable; if you've connected two lines together, the X100P will take the line off hook. It has no way of knowing that you've plugged two lines together. I have two questions. First, I see that the X100P is only a single channel. Does this mean that I can only use one POTS line with it? When I installed it I thought that it would support two POTS lines. I guess I thought this because it has an ordinary phone jack that had 4 little metal fingers in it. Yes; it's only one channel and it can only handle one line at once. Is it possible that the X100P is really dialing both lines at the same time and if so is there a way to stop this? Don't connect two lines together via the pass-through port? -Tilghman ___ 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 or H332 or *?
chan_h323 can deal with H.235 with no problem Jeremy McNamara Hemant Kumar wrote: I would rather say take whatever is cheap and make it work;) h.323 does work pretty well with *, but if you are using a GK or GW which wants you to authenticate your IP Phone then be carefull H.235 is still not supported in *. Hemant - Original Message - From: "Andres Tello Abrego" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 07, 2003 12:56 AM Subject: [Asterisk-Users] Sip or H332 or *? Ok, So, I'm willing to isntall ip phones, I have, to choise: snom 200 cisco phones alcatel phones... snom 200, not in mexico,must import. cisco phone, in mexico, quite expensive, protocl to use; sip alcatel, in mexico, no so expensive, protocol to use: h323 Asteris, which protocol support better? Sip o H323? Any advice is welcome... :) ___ 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] X100P question about odd behavior
Jim: An RJ-11 has 4 pins and is the most commonly used POTS physical interface 1 2 3 4 A standard one line POTS will be on pins 23, and if you have a dual line wire, then the other line is mapped to 14? If both your lines are on 23, then X100P will dial out on both lines. Do you have a RJ-11 breakout box handy? If so, you should be able to isolate where the lines are getting shorted. -- Mirza Wasim Baig | Principal Consultant | Convergence Business Systems VOX: +92(51)282-0628 x7400 | FAX: +92(51)282-0621 | IAX: (700)282-0628 On Fri, 7 Mar 2003, Jim Archer wrote: terrible with numbers) and not a RJ-45. Both line 1 and line 2, each a POTS line, are both on that wire, so I can plug it into my 2 line analog phone. I have plugged nothing into the pass through port (which is labeled phone). When I plug this into the X100P it seems to dial out on both. So it seems to be shorting the lines together. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
