Re: [asterisk-users] Zopier Client
On 4/8/2009 1:19 PM, Gregory Malsack wrote: Does anyone have any first-hand experience with the Zoiper Business version softphone? If so what has been your experience with it? Thanks, Greg I am not a very heavy user of it either, but I'm a semi-regular user, and I like it a lot. It's the most stable and usable IAX2/SIP soft-phone I have used, and I've used at least a dozen of them before finally settling on Zoiper, and then Zoiper-Biz. I don't use some of the fancier features, but what I do use, always works as expected. Call quality is very good too. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe through DeadAGI has changed to return -1 on Hangup
I have a support call AGI script that has been working
flawlessly for a couple of years now. It dumps the customer into a
MeetMe conference room, then dials a bunch of support engineers,
and connects anyone who accepts the call into the conference room.
The conference room is recorded. After the support call is over,
the recording is emailed to a list for quality control, etc.
It stopped working correctly on Jun 25th. Roughly on that date, I
upgraded to Asterisk 1.2.20 (I'm now on 1.2.23, and it hasn't
worked correctly on any version since 1.2.19).
What happens now is that when the MeetMe is exited normally (all
participants hang up), the AGI script simply stops executing. I
see no error messages on the CLI. I turned on agi debug, and I
see that MeetMe is returning res=-1. That is not supposed to
happen with DeadAGI (if I understand correctly), and it didn't
used to happen.
If I exit the MeetMe with the #, then I correctly get res=0,
and the script indeed continues to process correctly.
It seems to me that since 1.2.20, and continuing through today's
1.2.23, DeadAGI is behaving like AGI on a hangup of MeetMe.
Can anyone else confirm this, and if so, let me know what I can do
to revert it? This is the entire diff of the current app_meetme.c
with the one from 1.2.19, and it seems too innocuous to be the
culprit, but of course, it _is_ a hangup, so perhaps it's as
simple as reverting this one change?!?
[EMAIL PROTECTED] asterisk]# diff /usr/src/asterisk/apps/app_meetme.c
/usr/src/asterisk-1.2.19/apps/app_meetme.c
40c40
ASTERISK_FILE_VERSION(__FILE__, $Revision: 69894 $)
---
ASTERISK_FILE_VERSION(__FILE__, $Revision: 59360 $)
1299,1302d1298
/* If the channel wants to be hung up,
hang it up */
if (ast_check_hangup(chan))
break;
And here is the entire diff from res_agi.c:
[EMAIL PROTECTED] asterisk]# diff res/res_agi.c
/usr/src/asterisk-1.2.19/res/res_agi.c
44c44
ASTERISK_FILE_VERSION(__FILE__, $Revision: 71656 $)
---
ASTERISK_FILE_VERSION(__FILE__, $Revision: 54771 $)
572c572,579
ast_playstream(fs);
---
res = ast_playstream(fs);
if (res) {
fdprintf(agi-fd, 200 result=%d endpos=%ld\n,
res, sample_offset);
if (res = 0)
return RESULT_SHOWUSAGE;
else
return RESULT_FAILURE;
}
625c632,639
ast_playstream(fs);
---
res = ast_playstream(fs);
if (res) {
fdprintf(agi-fd, 200 result=%d endpos=%ld\n,
res, sample_offset);
if (res = 0)
return RESULT_SHOWUSAGE;
else
return RESULT_FAILURE;
}
1106c1120
return res = 0 ? RESULT_SUCCESS : RESULT_FAILURE;
---
return res;
Thanks in advance!
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe through DeadAGI has changed to return -1 on Hangup
Following up on my own post, and not quoting myself (tsk, tsk), I found a forum thread on Google that discussed a similar problem. They claimed it was a SIGHUP being sent to the script when the caller hung up, even though DeadAGI shouldn't get that type of signal. Anyway, it turns out that was my exact problem as well. I inserted a signal handler that ignores SIGHUP and my script now works the way it used to. This is for the next poor soul that trips on this problem... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stanaphone/Asterisk issue: No Audio with SIP to only one provider when switching servers
Alejandro Lengua wrote: Hello, did you got your issue solved? I am suffering of the same issue Hi. I had it off for a few weeks, and then decided to try again, and it just worked. I didn't change a single thing, only uncommented the register statement that I had previously commented. It's been reliable now for the past 2 weeks since I turned it back on. I didn't bother to report here because I didn't have a solution. I guessed that they changed something on their side, since I did report the problem to them when it first happened (though they didn't respond), but, if you're having the problem, perhaps I just got lucky. Sorry to hear you're having the problem, I know how frustrating it is! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Audio with SIP to only one provider when switching servers
I snipped all of the previous data, as I'm trying to boil down this problem to its essence... I turned off the firewall for a few seconds, and still got no audio. For those that will be suspicious, the commands were: shorewall stop shorewall clear tested connection, no audio shorewall start I also have a SIPPhone number, which (obviously), connects via SIP. I called that number from the outside, using one of their Access Numbers, and my phone rang and I heard audio in both directions (this with the firewall back on), so SIP definitely works, just not with StanaPhone. Then I connected from another server that I run, which is behind a NAT router. That server is running 1.2.18 (as is the one that isn't working, but is on a public IP). Audio works perfectly with this one. To my knowledge the only difference between them is that the two servers that work are both Red Hat 9, with Asterisk 1.2.18 built from source. The one that fails is CentOS 5.0, with Asterisk 1.2.18 built from source. Here is a dump of the active channel from the NAT'ed server, which _works_: * SIP Call Direction: Incoming Call-ID: [EMAIL PROTECTED] Our Codec Capability: 1822 Non-Codec Capability: 1 Their Codec Capability: 262 Joint Codec Capability: 262 Format ulaw Theoretical Address:204.147.183.18:5060 Received Address: 204.147.183.18:5060 NAT Support:RFC3581 Audio IP: XX.XX.XX.XX (local) Our Tag:as78cfb201 Their Tag: da6aae9eb017f29b6c9de270fb85c352 SIP User agent: Sippy Original uri: sip:204.147.183.55:1024 Caller-ID: XX Need Destroy: 0 Last Message: Rx: ACK Promiscuous Redir: No Route: sip:204.147.183.18;ftag=da6aae9eb017f29b6c9de270fb85c352;lr=on DTMF Mode: rfc2833 SIP Options:(none) The only things edited above are the Audio IP, which is my correct local (before NAT) server address, and my Caller-ID. Everything else is unchanged. Here is the channel with dead audio: * SIP Call Direction: Incoming Call-ID: [EMAIL PROTECTED] Our Codec Capability: 1542 Non-Codec Capability: 1 Their Codec Capability: 262 Joint Codec Capability: 6 Format ulaw Theoretical Address:204.147.183.18:5060 Received Address: 204.147.183.18:5060 NAT Support:RFC3581 Audio IP: XX.XX.XX.XX (local) Our Tag:as45dbcfef Their Tag: 420bab62c5da9eae42686897ae65a385 SIP User agent: Sippy Original uri: sip:204.147.183.55:1024 Caller-ID: XX Need Destroy: 0 Last Message: Rx: ACK Promiscuous Redir: No Route: sip:204.147.183.18;ftag=420bab62c5da9eae42686897ae65a385;lr=on DTMF Mode: rfc2833 SIP Options:(none) The same two fields are edited above, and both were correct. To my eye, these are identical. Both are selecting ulaw, correctly. I'm stumped. I guess that I didn't do any packet tracing, but I'm not sure what the value of that would be given that it's not a firewall problem... Suggestions welcome! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Audio with SIP to only one provider whenswitching servers
Brad Sumrall wrote: I would not rule your firewall out as the problem! Port 5060 is only the authentication port, the rtp stream is normally 10,000 thru 20,000. Some of your phone may have STUN modules on them. Open 10,000 thru 20,000 and 5060 on the firewall. Stick some holes in it for testing purposes. Verify ports are open with telnet:port number both ways, telnet is your friend. If it works, close the holes up and consult your firewall docs Brad Thanks for the response Brad (and Brian Capouch as well in a separate note!). I was offline all day yesterday, but I can do more testing today. Of course, it's quite possible that it's the firewall. That said, all other providers (including SIP) work, so it would have to be a reasonably tight number of ports that are open to the other providers, and a different set of ports that are closed that StanaPhone is trying to communicate on. Anyway, more testing on the way ;-) BTW, I run Shorewall (which is a cover for IPTABLES), and it usually logs every dropped packet, and I see _no_ rejections in the log file for source IP from StanaPhone and destination UDP ports on my machine. I'm running the same Shorewall rules (different version of Shorewall and different OS on the two linux boxes) on the box that works with StanaPhone... Thanks again to both of you for the responses! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hadar Pedhazur Sent: Wednesday, April 25, 2007 6:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Audio with SIP to only one provider whenswitching servers I have been running Asterisk for years on a machine with a public IP. Most recently, I have been running 1.2.17, from the day it came out, with no (noticeable) problems. Yesterday, I switched over to a new server that is on the same public subnet, one higher than the original server. I built 1.2.17 from source on that machine (as I did on the old server). My firewall on the new machine is configured identically to the old one as well. All of my IAX connections just worked. All but one of my SIP connections just worked as well (which is why I can't believe it's a firewall issue). StanaPhone, which I use for 2 incoming DIDs, registers correctly, and rings my phones correctly when a call comes in. However, once answered, there is dead silence in both directions, on 100% of the calls. There isn't any problem on StanaPhone's side (which has provided a _fantastic_ service ever since I signed up!), because I can connect to them with X-Lite and receive calls with audio. More importantly, if I fire up Asterisk on the old server, it still works!!! I can connect with X-Lite to the new server, so the new server definitely accepts SIP connections, and audio works. It's _not_ a codec problem. I verified that on both the working and non-working servers the connection is established with ulaw on both sides. I have dumped the peer and the channel on both, while the call was active, and they look identical to me, except for the random bits associated with a particular connection. Here are the ones from the machine that fails: *CLI sip show peer XX * Name : XX Secret : Set MD5Secret: Not set Context : default Subscr.Cont. : Not set Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : Expire : -1 Insecure : port,invite Nat : RFC3581 ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : sip.stanaphone.com Addr-IP : 204.147.183.18 Port 5060 Defaddr-IP : 0.0.0.0 Port 0 Def. Username: 12345678 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw) Status : OK (20 ms) Useragent: Reg. Contact : new*CLI sip show channel [EMAIL PROTECTED] * SIP Call Direction: Outgoing Call-ID: [EMAIL PROTECTED] Our Codec Capability: 4 Non-Codec Capability: 1 Their Codec Capability: 4 Joint Codec Capability: 4 Format ulaw Theoretical Address:204.147.183.18:5060 Received Address: 204.147.183.18:5060 NAT Support:RFC3581 Audio IP: AAA.BBB.CCC.DDD (local) Our Tag:as360c7ca5 Their Tag: 0bd46ffd48e4fbffb3a68f13f8ad2599 SIP User agent: Username: 87654321 Peername: 12345678 Original uri: sip:204.147.183.55:1024 Need Destroy: 0 Last Message: Tx: ACK Promiscuous Redir: No Route
[asterisk-users] No Audio with SIP to only one provider when switching servers
I have been running Asterisk for years on a machine with a public IP. Most recently, I have been running 1.2.17, from the day it came out, with no (noticeable) problems. Yesterday, I switched over to a new server that is on the same public subnet, one higher than the original server. I built 1.2.17 from source on that machine (as I did on the old server). My firewall on the new machine is configured identically to the old one as well. All of my IAX connections just worked. All but one of my SIP connections just worked as well (which is why I can't believe it's a firewall issue). StanaPhone, which I use for 2 incoming DIDs, registers correctly, and rings my phones correctly when a call comes in. However, once answered, there is dead silence in both directions, on 100% of the calls. There isn't any problem on StanaPhone's side (which has provided a _fantastic_ service ever since I signed up!), because I can connect to them with X-Lite and receive calls with audio. More importantly, if I fire up Asterisk on the old server, it still works!!! I can connect with X-Lite to the new server, so the new server definitely accepts SIP connections, and audio works. It's _not_ a codec problem. I verified that on both the working and non-working servers the connection is established with ulaw on both sides. I have dumped the peer and the channel on both, while the call was active, and they look identical to me, except for the random bits associated with a particular connection. Here are the ones from the machine that fails: *CLI sip show peer XX * Name : XX Secret : Set MD5Secret: Not set Context : default Subscr.Cont. : Not set Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : Expire : -1 Insecure : port,invite Nat : RFC3581 ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : sip.stanaphone.com Addr-IP : 204.147.183.18 Port 5060 Defaddr-IP : 0.0.0.0 Port 0 Def. Username: 12345678 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw) Status : OK (20 ms) Useragent: Reg. Contact : new*CLI sip show channel [EMAIL PROTECTED] * SIP Call Direction: Outgoing Call-ID: [EMAIL PROTECTED] Our Codec Capability: 4 Non-Codec Capability: 1 Their Codec Capability: 4 Joint Codec Capability: 4 Format ulaw Theoretical Address:204.147.183.18:5060 Received Address: 204.147.183.18:5060 NAT Support:RFC3581 Audio IP: AAA.BBB.CCC.DDD (local) Our Tag:as360c7ca5 Their Tag: 0bd46ffd48e4fbffb3a68f13f8ad2599 SIP User agent: Username: 87654321 Peername: 12345678 Original uri: sip:204.147.183.55:1024 Need Destroy: 0 Last Message: Tx: ACK Promiscuous Redir: No Route: sip:204.147.183.18;ftag=as360c7ca5;lr=on DTMF Mode: rfc2833 SIP Options:(none) Finally, I built 1.2.18 from source today, and everything is working perfectly _except_ for StanaPhone, which continued to connect with no problems, but deliver no audio in either direction. I have no idea what else to try, and would appreciate _any_ guidance. Thanks in advance! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit - EXPLAINED
Juergen K. Zick wrote: But slowly, we are getting completely off-topic on this list. I doubt that changing to static IP will solve to decribed problem, because it is a line mismatch problem on the physical layer of the connection. And these will not go away unless you change the wiring ! Understood. Hadar, I would suggest to try my wiring first before you take other action to buy something. Also, while testing the line with BellSouth, I would ask for BERT-tests in the ATM-layer loop of your DSL.connection while your father has no phone talk on the POTS side and then with a running phone talk on on his phone. OK. Thanks! He is planning on calling Bellsouth soon, so this information is great, and much appreciated! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit
Bob Chiodini wrote: I'm a Bellsouth DSL user in FL too. Here, the filter has a DSL/modem jack and a POTS jack. So if a phone and modem share the same wall plate the filter does the split. Interesting, I'm pretty sure that when they installed it in his apartment, they put in the Y cable, so it's definitely a supported Bellsouth configuration. I don't think connecting the DSL modem directly the loop is wise. That's assuming that the filter actually filters something on the DSL port and that the modem does not have a built-in filter. My modem is a Westell. http://en.wikipedia.org/wiki/DSL_filter Thanks. One other possibility, the ringing is causing packet loss (UDP) that the HT486 is not handling very well. Normal TCP traffic would generally recover. The streaming audio test should confirm the loss. Does the HT486 have any kind of logging? I don't know about the logging, but you might be correct with regard to the packet loss. Rich Adamson conjectured that's it's a firewall issue, and it certainly feels like that. Last night, it occurred to me that perhaps an answered POTS line causes the modem to request a new DHCP lease, meaning, it changes it's IP address. If that were the case, it would explain the behavior I'm seeing, namely that we can continue to hear him, because the HT can still find the remote end, but the remote end can no longer find him... I don't know how easily I can verify that (remotely, I'm not sure I can talk my Dad through that one ;-), but perhaps I can prove that theory one way or another... I'm still testing VOIP (read newbie) and have not run across this scenario. I'll add it to my list of things to test. Welcome! Bob... On Mon, 2006-05-08 at 15:23 -0400, Hadar Pedhazur wrote: Juergen K. Zick wrote: Well, I have no idea how DSL lines are connected in the US but what happens to a normal Internet connection when the phone is being picked up ? Test scenario could be that your Dad is listening to an Internet radio station or other audio stream and then being called Great idea! It's possible that there is a hiccup when the phone gets picked up, which a streaming audio connection might feel as well, in which case Bellsouth would have to acknowledge the problem ;-). Thanks, I'll have him test that. BTW, how are the real phones and the answering machine being connected ? Is the HT in front of them in the POTS line ? They are separated. The answering machine is in another room, connected to a normal phone jack, using a DSL filter to assure it doesn't get the noise of the DSL line. The HT is connected to the DSL modem, and there are no POTS lines connected to the FXO back-up port on the HT. In other words, the HT has only the WAN (to DSL) and LAN (to PC) ports connected, and an analog phone (a GE 5.8ghz handset) connected to it's FXS port. The only other possible connection problem (which I think I tested and rejected as a problem two months ago) is that there is a Y splitter coming out of the jack with a filter for the real phone, and no filter for the DSL modem (which is in front of the HT). I am reasonably sure that I had him remove the Y cable, and plug the DSL modem directly into the jack, and it still failed. However, I'll retry that again too now :-). Thanks Juergen! --Juergen I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line connected on the same line. He has the appropriate filters on every jack that has a phone connected to it, and he even replaced one or two of them (when I thought that was the problem). I sent him a HandyTone GS-486 (HT), configured to connect back to my Asterisk server. He only has a single computer in his apartment, so it's connected into the HT, and the HT is connected into the DSL modem. He can make and receive calls on the HT, and the quality is excellent. If he's speaking via the HT (meaning a VoIP-only call) and the real phone rings, everything continues fine (temporarily). If the real phone is answered, either by a person, or by the answering machine (which is in another room, connected to a filter on another jack), then the audio on the Asterisk conversation becomes _one way_. My father can be heard _perfectly_ by the remote side of the conversation, but he can hear nothing. When the POTS line is hung up, then both sides of the VoIP call go dead (audio-wise). Of course, he can now redial a VoIP call, and both sides work perfectly... At first, I couldn't imagine that it was anything other than a bad filter, but other than replacing the filter (which didn't help), nothing else stops working. He can continue to use the Internet connection on his PC just fine, and I can continue to hear him speak over the VoIP connection
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit - EXPLAINED
Replying to my own post (and my most recent follow-up). I have now confirmed 100% that the DSL modem gets a _new_ IP address every time his real phone gets answered, or hung up! This (of course) disrupts the audio coming from to him, since the sending machine (Asterisk in my case), no longer has the correct IP address to send to him. I lowered his registration from the default 1 hour to 1 minute, so after we're disconnected, I can see that he's re-registering with a new IP address, each and every time :-(. I told him to call Bellsouth and ask about a Static IP address, but I don't know if they offer it, or how much they charge. While this one isn't solved, it's at least explained. Thanks to everyone who responded! Hadar Pedhazur wrote: I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line connected on the same line. He has the appropriate filters on every jack that has a phone connected to it, and he even replaced one or two of them (when I thought that was the problem). I sent him a HandyTone GS-486 (HT), configured to connect back to my Asterisk server. He only has a single computer in his apartment, so it's connected into the HT, and the HT is connected into the DSL modem. He can make and receive calls on the HT, and the quality is excellent. If he's speaking via the HT (meaning a VoIP-only call) and the real phone rings, everything continues fine (temporarily). If the real phone is answered, either by a person, or by the answering machine (which is in another room, connected to a filter on another jack), then the audio on the Asterisk conversation becomes _one way_. My father can be heard _perfectly_ by the remote side of the conversation, but he can hear nothing. When the POTS line is hung up, then both sides of the VoIP call go dead (audio-wise). Of course, he can now redial a VoIP call, and both sides work perfectly... At first, I couldn't imagine that it was anything other than a bad filter, but other than replacing the filter (which didn't help), nothing else stops working. He can continue to use the Internet connection on his PC just fine, and I can continue to hear him speak over the VoIP connection with no problems either, so the Internet connection has not been lost. I have to admit to being completely clueless as to what to even look for, so _any_ advice as to things to test for would be appreciated. As I said at the top, I can reproduce this 100% of the time, so I can easily setup any debugging environment in advance, and trigger the problem at will, etc. Thanks in advance! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit - EXPLAINED
I combined my reply to both Alex and Joe into this one note :-) Alex Robar wrote: I'm still curious as to WHY it's getting a new IP everytime an incoming POTS call comes in. If I were you, I'd be asking Bellsouth why this happens instead of getting a static IP. A static IP may not even solve your issue too. If the problem is that a POTS call disconnects the modem and causes PPPoE authentication to re-occur, then you'll still see a VoIP call disconnect when this happens, even if the same IP is received when the DSL connection is re-established. I agree with your assessment. I will certainly have him do that. That said, I believe that a static IP will still solve the problem (albeit not correctly), because the receiving audio continues to work, so as long as the one side is finding the other, even though the session may be disconnected and restarted, a static IP will still be found correctly on the new session. Alex Joe Greco wrote: No, a static IP address isn't likely to solve the real problem. What's probably happening is that the transitions on the line are causing havoc with the DSL, and the DSL modem is restarting the session. This is not supposed to happen, but sometimes does. Have him call Bellsouth and tell them his Internet stops working when he picks up or hangs up the phone. They'll most likely get it fixed. ... JG See above, I am sure you are both correct. Thanks! On 5/9/06, *Hadar Pedhazur* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Replying to my own post (and my most recent follow-up). I have now confirmed 100% that the DSL modem gets a _new_ IP address every time his real phone gets answered, or hung up! This (of course) disrupts the audio coming from to him, since the sending machine (Asterisk in my case), no longer has the correct IP address to send to him. I lowered his registration from the default 1 hour to 1 minute, so after we're disconnected, I can see that he's re-registering with a new IP address, each and every time :-(. I told him to call Bellsouth and ask about a Static IP address, but I don't know if they offer it, or how much they charge. While this one isn't solved, it's at least explained. Thanks to everyone who responded! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit - EXPLAINED
Juergen K. Zick wrote: HI, well, that was what I expected in my posting yesterday. For me, your wiring looks strange. Here in Germany, we have spiltters connected to the incoming line which have two outputs: A high pass filter output for the DSL signal and a low pass output with DC pass-through for the POTS signal. the DSL output is being connected to the DSL-modem and the POTS output will feed your internal POTS wiring. The only jack that has both a phone and the DSL connector indeed has a splitter on it, provided by Bellsouth. Therefore, there is _NO_ filter needed on each POTS outlet, because there is nothing to be filtered out on your internal line anymore. You may be correct. I am definitely _not_ familiar enough with DSL. However, 5 years ago, I had a DSL line in my apartment, and I was specifically told by the installation tech that I needed a filter on _any_ jack that had a real phone connected to it. That may not have been necessary, or perhaps isn't necessary any longer, or perhaps varies by provider, but that's what I was told at the time, and that's what I did (with no problems). The filters on the phone jacks that didn't have the modem connected were not splitters, just single filters. Seen from my German wiring knowlegde, your cabling is wrong and causes the interruptions on the DSL service. That's definitely possible, just not my personal (single point!) experience. Don`t you have something like a spiltter available ? It should be the _ONLY_ filter on your incoming line and then the DSL-modem and the POTS phone should be connected to it ... OK, it would be easy for him to remove the other filters temporarily and test again. Thanks! --Jürgen Replying to my own post (and my most recent follow-up). I have now confirmed 100% that the DSL modem gets a _new_ IP address every time his real phone gets answered, or hung up! This (of course) disrupts the audio coming from to him, since the sending machine (Asterisk in my case), no longer has the correct IP address to send to him. I lowered his registration from the default 1 hour to 1 minute, so after we're disconnected, I can see that he's re-registering with a new IP address, each and every time :-(. I told him to call Bellsouth and ask about a Static IP address, but I don't know if they offer it, or how much they charge. While this one isn't solved, it's at least explained. Thanks to everyone who responded! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIPcall over DSL circuit - EXPLAINED
Once again, combining multiple responses. I have a suspicion that this might be bad netiquette, but I hope no one minds too much (just tell me if it is, and I'll stop!). Alexander Lopez wrote: After reading this post, I feel that the problem is, a. Bad Westell, b. Bad loop. Bell will run a loop test from the DSLAM 'out' to your site. While they are doing this, call from another phone to make sure the test fails, or shows what the problem is. Good idea (specifically the part of calling while they are testing!). Juergen K. Zick wrote: Well, to avoid a misunderstanding see the following drawing: /---DSL-MODEM-HT-PC H| | +--+ Inet-PHONE from BellSouth (DSL over POTS) ---| SPLITTER | +--+ L| \-- | | answering POTS machine phone (maybe you have to reformat it into COURIER font) It's depending on the calling in your father`s flat but on the incoming line you should have only _ONE_ device, the SPLITTER !!! Thanks Juergen. Now I see where we were miscommuncating. It's typical in the US to have many phone jacks in one house/apt all connected to the same number. So, while the DSL and one phone are connected to one of the jacks _exactly_ as in your diagram above, there are other phones in other rooms, connected directly to the wall jack, and I was under the impression that they need separate filters. Alexander Lopez wrote: BellSouth will provide Static IPs for home users, staring with the Extreme product. (3MBit) and up. No extra charge for this, included in the package. Cool. Good to know! Andrew Kohlsmith wrote: If the PPPoE session were interrupted all audio would stop. This is not the case. That's exactly what I thought as well, but I had no other explanation for the effect that I am seeing. We need to see packet dumps on both the transmitting side (just before the DSL modem) and on the receiving side (far end Asterisk box). I'm guessing that there is something funny with the ATA or DSL modem, but I'm dumbfounded as to a solution. If that's straightforward to set up, I'm game to try it (I have no idea what I would need to do, and if it has to be on my Dad's PC as well, it will be a lot tougher to get it going ;-). Anyway, my Dad was game to call Bellsouth. Whenever he gets them, and gets an answer (if there is one), I'll report back for completeness sake. Thanks again to everyone! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit
I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line connected on the same line. He has the appropriate filters on every jack that has a phone connected to it, and he even replaced one or two of them (when I thought that was the problem). I sent him a HandyTone GS-486 (HT), configured to connect back to my Asterisk server. He only has a single computer in his apartment, so it's connected into the HT, and the HT is connected into the DSL modem. He can make and receive calls on the HT, and the quality is excellent. If he's speaking via the HT (meaning a VoIP-only call) and the real phone rings, everything continues fine (temporarily). If the real phone is answered, either by a person, or by the answering machine (which is in another room, connected to a filter on another jack), then the audio on the Asterisk conversation becomes _one way_. My father can be heard _perfectly_ by the remote side of the conversation, but he can hear nothing. When the POTS line is hung up, then both sides of the VoIP call go dead (audio-wise). Of course, he can now redial a VoIP call, and both sides work perfectly... At first, I couldn't imagine that it was anything other than a bad filter, but other than replacing the filter (which didn't help), nothing else stops working. He can continue to use the Internet connection on his PC just fine, and I can continue to hear him speak over the VoIP connection with no problems either, so the Internet connection has not been lost. I have to admit to being completely clueless as to what to even look for, so _any_ advice as to things to test for would be appreciated. As I said at the top, I can reproduce this 100% of the time, so I can easily setup any debugging environment in advance, and trigger the problem at will, etc. Thanks in advance! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit
Jerry Jones wrote: I would guess either the DSL itself is bad or perhaps the dsl Modem. perhaps calling Bellsouth would be helpful? Does other Internet traffic get interrupted also? The rest of the Internet traffic is _not_ interrupted, and his voice continues to be heard, so even that part of the connection continues uninterrupted. I'm not sure how to diagnose whether the modem is bad or not, given that other traffic (and half of this traffic) continues to flow over it. I can only imagine calling Bellsouth and being told we don't support VoIP unless you buy our VoIP service Thanks Jerry! On May 8, 2006, at 1:42 PM, Hadar Pedhazur wrote: I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line connected on the same line. He has the appropriate filters on every jack that has a phone connected to it, and he even replaced one or two of them (when I thought that was the problem). I sent him a HandyTone GS-486 (HT), configured to connect back to my Asterisk server. He only has a single computer in his apartment, so it's connected into the HT, and the HT is connected into the DSL modem. He can make and receive calls on the HT, and the quality is excellent. If he's speaking via the HT (meaning a VoIP-only call) and the real phone rings, everything continues fine (temporarily). If the real phone is answered, either by a person, or by the answering machine (which is in another room, connected to a filter on another jack), then the audio on the Asterisk conversation becomes _one way_. My father can be heard _perfectly_ by the remote side of the conversation, but he can hear nothing. When the POTS line is hung up, then both sides of the VoIP call go dead (audio-wise). Of course, he can now redial a VoIP call, and both sides work perfectly... At first, I couldn't imagine that it was anything other than a bad filter, but other than replacing the filter (which didn't help), nothing else stops working. He can continue to use the Internet connection on his PC just fine, and I can continue to hear him speak over the VoIP connection with no problems either, so the Internet connection has not been lost. I have to admit to being completely clueless as to what to even look for, so _any_ advice as to things to test for would be appreciated. As I said at the top, I can reproduce this 100% of the time, so I can easily setup any debugging environment in advance, and trigger the problem at will, etc. Thanks in advance! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit
Juergen K. Zick wrote: Well, I have no idea how DSL lines are connected in the US but what happens to a normal Internet connection when the phone is being picked up ? Test scenario could be that your Dad is listening to an Internet radio station or other audio stream and then being called Great idea! It's possible that there is a hiccup when the phone gets picked up, which a streaming audio connection might feel as well, in which case Bellsouth would have to acknowledge the problem ;-). Thanks, I'll have him test that. BTW, how are the real phones and the answering machine being connected ? Is the HT in front of them in the POTS line ? They are separated. The answering machine is in another room, connected to a normal phone jack, using a DSL filter to assure it doesn't get the noise of the DSL line. The HT is connected to the DSL modem, and there are no POTS lines connected to the FXO back-up port on the HT. In other words, the HT has only the WAN (to DSL) and LAN (to PC) ports connected, and an analog phone (a GE 5.8ghz handset) connected to it's FXS port. The only other possible connection problem (which I think I tested and rejected as a problem two months ago) is that there is a Y splitter coming out of the jack with a filter for the real phone, and no filter for the DSL modem (which is in front of the HT). I am reasonably sure that I had him remove the Y cable, and plug the DSL modem directly into the jack, and it still failed. However, I'll retry that again too now :-). Thanks Juergen! --Juergen I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line connected on the same line. He has the appropriate filters on every jack that has a phone connected to it, and he even replaced one or two of them (when I thought that was the problem). I sent him a HandyTone GS-486 (HT), configured to connect back to my Asterisk server. He only has a single computer in his apartment, so it's connected into the HT, and the HT is connected into the DSL modem. He can make and receive calls on the HT, and the quality is excellent. If he's speaking via the HT (meaning a VoIP-only call) and the real phone rings, everything continues fine (temporarily). If the real phone is answered, either by a person, or by the answering machine (which is in another room, connected to a filter on another jack), then the audio on the Asterisk conversation becomes _one way_. My father can be heard _perfectly_ by the remote side of the conversation, but he can hear nothing. When the POTS line is hung up, then both sides of the VoIP call go dead (audio-wise). Of course, he can now redial a VoIP call, and both sides work perfectly... At first, I couldn't imagine that it was anything other than a bad filter, but other than replacing the filter (which didn't help), nothing else stops working. He can continue to use the Internet connection on his PC just fine, and I can continue to hear him speak over the VoIP connection with no problems either, so the Internet connection has not been lost. I have to admit to being completely clueless as to what to even look for, so _any_ advice as to things to test for would be appreciated. As I said at the top, I can reproduce this 100% of the time, so I can easily setup any debugging environment in advance, and trigger the problem at will, etc. Thanks in advance! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit
Rich Adamson wrote: Hadar Pedhazur wrote: Jerry Jones wrote: I would guess either the DSL itself is bad or perhaps the dsl Modem. perhaps calling Bellsouth would be helpful? Does other Internet traffic get interrupted also? The rest of the Internet traffic is _not_ interrupted, and his voice continues to be heard, so even that part of the connection continues uninterrupted. I'm not sure how to diagnose whether the modem is bad or not, given that other traffic (and half of this traffic) continues to flow over it. I can only imagine calling Bellsouth and being told we don't support VoIP unless you buy our VoIP service Not likely to be a bad modem. More likely is something like NAT tables getting in the way (eg, firewall, nat box, or whatever). If its just your voice that he can't hear, its likely to be the firewall/nat box on his end as it doesn't know which udp port the inbound rtp traffic is going to use. It definitely feels like that's what's happening, but it seems strange that it's kicked off by an incoming POTS call. What would make the firewall change what it thinks it's talking to already? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit
Alex Robar wrote: I'll lean this way too. I had a DSL line from Bell Canada in Kingston, Ontario, and an incoming call on that line to the POTS phones would cause VoIP traffic to become completely unintelligble. The VoIP call would have to be re-established to fix things. A quick call to Bell had a technican out to check the lines, and put a fix in place for me. I was afraid of doing that, unless I specifically explain that it's a VoIP thing, because otherwise, if the tech asks what was interrupted, I won't be able to show anything else... Thanks for the suggestion! Alex Robar On 5/8/06, *Jerry Jones* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I would guess either the DSL itself is bad or perhaps the dsl Modem. perhaps calling Bellsouth would be helpful? Does other Internet traffic get interrupted also? On May 8, 2006, at 1:42 PM, Hadar Pedhazur wrote: I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line connected on the same line. He has the appropriate filters on every jack that has a phone connected to it, and he even replaced one or two of them (when I thought that was the problem). I sent him a HandyTone GS-486 (HT), configured to connect back to my Asterisk server. He only has a single computer in his apartment, so it's connected into the HT, and the HT is connected into the DSL modem. He can make and receive calls on the HT, and the quality is excellent. If he's speaking via the HT (meaning a VoIP-only call) and the real phone rings, everything continues fine (temporarily). If the real phone is answered, either by a person, or by the answering machine (which is in another room, connected to a filter on another jack), then the audio on the Asterisk conversation becomes _one way_. My father can be heard _perfectly_ by the remote side of the conversation, but he can hear nothing. When the POTS line is hung up, then both sides of the VoIP call go dead (audio-wise). Of course, he can now redial a VoIP call, and both sides work perfectly... At first, I couldn't imagine that it was anything other than a bad filter, but other than replacing the filter (which didn't help), nothing else stops working. He can continue to use the Internet connection on his PC just fine, and I can continue to hear him speak over the VoIP connection with no problems either, so the Internet connection has not been lost. I have to admit to being completely clueless as to what to even look for, so _any_ advice as to things to test for would be appreciated. As I said at the top, I can reproduce this 100% of the time, so I can easily setup any debugging environment in advance, and trigger the problem at will, etc. Thanks in advance! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit
Andres wrote: They are separated. The answering machine is in another room, connected to a normal phone jack, using a DSL filter to assure it doesn't get the noise of the DSL line. The HT is connected to the DSL modem, and there are no POTS lines connected to the FXO back-up port on the HT. In other words, the HT has only the WAN (to DSL) and LAN (to PC) ports connected, and an analog phone (a GE 5.8ghz handset) connected to it's FXS port. The only other possible connection problem (which I think I tested and rejected as a problem two months ago) is that there is a Y splitter coming out of the jack with a filter for the real phone, and no filter for the DSL modem (which is in front of the HT). How about a problem with the 5.8Ghz phone itself? Is the regular POTS line on another 5.8Ghz phone? Two phones on the same frequency on different calls might be causing severe interference. I can tell you everytime a call came into my house on the 2.4 Ghz POTS phone, my Linksys WRT54G would drop the wireless signal going to the laptop, so multiple devices on the same frequency don't play well together. Nope. All of the phones are normal land lines, so there's no interference at the wireless level. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Audio on Local Machine, Remote works fine
I don't even know where to begin. I run a lot of production Asterisk servers, for a couple of years now, with no real problems. We built a brand new box, CentOS 4.2, and installed Asterisk 1.2.4 from source tarball(s). Built fine, and started up fine. Any attempts to do local audio (e.g. a Playback(welcome)) results in complete silence. Worse, the Playback command will hang forever (even if the file is tiny), so it's not just not being heard, it's like the command is waiting to do something. In one specific case (and only in case), I'll hear a 1/2 second burst of audio, like it's about to start, and then dead air. The Record command creates a zero length file if the format is ulaw, and hangs forever after that, and a wav format is always 44 bytes before the hang. If I run the demo-echo-test, I don't hear the prompt, and it hangs on the Playback. OK, now for the weirdness ;-). If I connect this Asterisk to one of our other servers, and dial the echo test on the remote server through this same server, I hear the prompts, and can hear my voice echoed correctly, so this same Asterisk server will happily forward the audio in both directions, it just won't generate it. This is with notransfer=yes, so this Asterisk is staying in the audio stream. I'm stumped, and any help or pointers in the right direction will be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No Audio on Local Machine, Remote works fine
[EMAIL PROTECTED] wrote: Is ztdummy loaded properly? I had a similar problem with a system recently. The machine has a real Digium T1 card in it, so I didn't think to check for a timing source. Since it's a backup machine, the actual T1 line isn't plugged in at the moment, but chan_zap.so definitely starts up correctly. I'll look into this in the morning (running out of the office now :-). Thanks for the suggestion! PaulH - Original Message - From: Hadar Pedhazur [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 02, 2006 8:16 AM Subject: [Asterisk-Users] No Audio on Local Machine, Remote works fine I don't even know where to begin. I run a lot of production Asterisk servers, for a couple of years now, with no real problems. We built a brand new box, CentOS 4.2, and installed Asterisk 1.2.4 from source tarball(s). Built fine, and started up fine. Any attempts to do local audio (e.g. a Playback(welcome)) results in complete silence. Worse, the Playback command will hang forever (even if the file is tiny), so it's not just not being heard, it's like the command is waiting to do something. In one specific case (and only in case), I'll hear a 1/2 second burst of audio, like it's about to start, and then dead air. The Record command creates a zero length file if the format is ulaw, and hangs forever after that, and a wav format is always 44 bytes before the hang. If I run the demo-echo-test, I don't hear the prompt, and it hangs on the Playback. OK, now for the weirdness ;-). If I connect this Asterisk to one of our other servers, and dial the echo test on the remote server through this same server, I hear the prompts, and can hear my voice echoed correctly, so this same Asterisk server will happily forward the audio in both directions, it just won't generate it. This is with notransfer=yes, so this Asterisk is staying in the audio stream. I'm stumped, and any help or pointers in the right direction will be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sixtel
Bill Michaelson wrote: Just curious... Is there anyone out there who has given this outfit money and actually received any service from them? I have accounts with roughly 5 providers, with sixtel being one of the primary ones. I have been using them since 2/2/2005. While there have been occasional glitches, some severe, for the most part their service works extremely well. For a long time, the biggest complaint was receiving DID's in a timely manner, or at all. Relatively recently, they ran a promotion for vanity DID's, and I signed up for two of them, bother of which were provisioned within the time that they claimed they would, and both work well. That said, of all of the providers I use, the most rock-solid in terms of availability and quality is NuFone. I know many people have complained on the lists before about their service, but I have found their responsiveness to be excellent, so not everyone is getting the same results. They are the most expensive of the services I subscribe to (still very cheap!!!), but they are also the most reliable. To be more specific, all of my _wife's_ calls get routed through NuFone, because when they weren't, I never heard the end of it ;-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_icd anyone? on 1.2?
Tyler wrote: Anyone using app_icd? I need to use some of the advanced features that the regular asterisk Queue() application won't provide. Anyone have any configuration examples, etc? Will it work with the current 1.2rc release? I played around with ICD in August. I was generally impressed with the flexibility, but the one thing I wanted, which people claimed was one of the reasons for starting the ICD project, wasn't implemented, delivering an announcement to an Agent _before_ the acknowledge with the #. So, I updated the old patch to chan_agent.c, and moved on. Now to answer your question ;-) Yesterday, I had a few minutes to kill, and I ran make on app_icd with 1.2beta2 (not rc2!). The build failed, and given the above problem (for me) with app_icd, I ignored the problem and moved on. So, from a one-shot test of a make that worked in August against CVS Head from July 31, the make did _not_ work against 1.2. Also, none of the files in the ICD project have been updated in over 7 months, and there hasn't been a single email to the mailing list since September, so I wouldn't count on lots more happening with the project in the near term :-(... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Beta2 problems with DTMF with T option in Dial Command
I was running CVS HEAD from 2005/07/31 until the day that beta2 came out. I installed beta2 on a number of servers without touching anything in /etc/asterisk. Most everything has been working well. One thing that is not is remote DTMF, more specifically, the # key. When I dial voicemail from DIAX, connected directly to the asterisk machine, I can retrieve voicemail. If I have DIAX connected to another asterisk, and dial the extension that connects me back to voicemail on that first box, then after I type the box number, it complains about an incorrect password on the first number that I type, no matter what that is. This is _not_ just a voicemail problem. If I have a DISA statement, with a hard-coded PIN, if DIAX is connected to the box directly, DISA works correctly. If I go through a remote asterisk, DISA fails every time. It _never_ recognizes the #, so it thinks the password has timed out every time. A little digging seems to show that the problem is in the T option to the Dial command which connects the two asterisk boxes. My features.conf file has blindxfer = #7 and atxfer = ##. A single # has been passed through correctly for months. Now, if I remove the T from the Dial command, then the remote voicemail (or DISA) works correctly. A few details: 1) all boxes in this experiment are running 1.2 beta2. 2) all boxes force ULAW codec only 3) if dtmfmode is ever referenced, it is always set to inband. 4) all of this worked in CVS HEAD as of July 31st, 2005. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fredericksburg ZPUG Meeting will have an Asterisk Flavor this Month
I've been asked to forward this announcement to the list. It's a little short notice as the meeting is this Wednesday night. I'm one of the presenters as well :-) From: Gary Poster [EMAIL PROTECTED] Date: October 10, 2005 11:51:10 AM EDT To: zope-announce@zope.org, python-announce-list@python.org, [EMAIL PROTECTED] Subject: Fifth Fredericksburg, VA ZPUG Meeting Please join us October 12, 7:30-9:00 PM, for the fifth meeting of the Fredericksburg, VA Zope and Python User Group (ZPUG). Learn about Python configuration of Asterisk, an open source VOIP! Free food! Rob Page, Zope Corporation CEO and President, will present a technical session on Asterisk [1] installation, configuration and operation. A brief discussion of connections to the public telephone network and internet telephony providers will be presented. Hadar Pedhazur, Zope Corporation Chairman of the Board, will present a technical session on call handling and processing using Python extensions to Asterisk. We will also serve delicious fruit, cheese, and soft drinks. We've had a nice group for all the meetings. Please come and bring friends! We also are now members of the O'Reilly and Apress user group programs, which gives us nice book discounts (prices better than Amazon's, for instance) and the possibility of free review copies. Ask me about details at the meeting if you are interested. General ZPUG information When: second Wednesday of every month, 7:30-9:00. Where: Zope Corporation offices. 513 Prince Edward Street; Fredericksburg, VA 22408 (tinyurl for map is http://tinyurl.com/ duoab). Parking: Zope Corporation parking lot; entrance on Prince Edward Street. Topics: As desired (and offered) by participants, within the constraints of having to do with Python. Contact: Gary Poster ([EMAIL PROTECTED]) [1] From www.asterisk.org: Asterisk is a complete PBX in software. It runs on Linux, BSD and MacOSX and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards- based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response and Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H. 323 (as both client and gateway). Check the Features section for a more complete list. Asterisk needs no additional hardware for Voice over IP. For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsors, Digium�¹. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks as well as a single port FXO card and a one to four-port modular FXS and FXO card. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Sipura SPA 200 Caller ID Problem
Sorry to bug all of you with this, but I have to assume there are a number of Sipura experts here... I have a Sipura SPA 2000 that I've been using for nearly 2 years now. It's flashed up to firmware 3.1.5. On line 1, I no longer get Caller ID (it used to work, and I can't remember when it stopped). On line 2, I always get Caller ID. To my old eyes, _every_ switch on both lines seems configured identically. I can see that calls to Line 1 have the correct Caller ID on the Asterisk CLI, and it displays correctly on my soft phone, so the problem is definitely at the Sipura level. As a hint, I think that I was playing with some attended transfer settings in Asterisk (not on the Sipura), and I may have typed one of the magical *XX codes the went to the Sipura instead of to Asterisk. Anyway, I've tried to type in each one that looked like a candidate for affecting this, and nothing seems to work. Any pointers would be greatly appreciated. Thanks in advance! P.S. I reset the Sipura to factory defaults, and rebuilt from there. It still works on line 2 and not on line 1 :-(. It's not the phone which is a Uniden Tru8866, since all handsets exhibit the same problem... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ICD Features
Peter Svensson wrote: ICD has its own mailinglist at [EMAIL PROTECTED] There is close to zero traffic there as well. I think the authors read it though. Peter, thank you very much for the response (which I snipped), and for the pointer to these (very quiet) lists as well. I just subscribed to both, and perhaps I'll make a post there of my early experience. Ironically, I've written two Python AGI scripts that implement a reasonably sophisticated Support system (bridging customers and support engineers in a MeetMe room). It's working fine, and is much simpler than Queues/Agents and ICD. My only real problem with my current setup is that because I use Call Files to contact the Agents, I have no direct way to cancel ringing phones when the call has been bridged to another channel. In fact, my problem is actually a little more subtle than that. We don't mind multiple support engineers hopping on to the same conference with the customer, as the second person might be more familiar with this customer or problem, etc. What we really want to cancel are the remaining ringing phones for a _particular_ agent, who has already answered a different channel (he picked up his desk phone, we can stop ringing his cell phone and soft phone, etc.). We can't ring all phones in one Dial statement, because a cell phone picking up VM will cancel the other channels in that scenario. We force acknowledgment in order to decide which channel the agent wants to bridge with. Anyway, thanks again! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ICD Features
Peter Svensson wrote: On Wed, 31 Aug 2005, Hadar Pedhazur wrote: My only real problem with my current setup is that because I use Call Files to contact the Agents, I have no direct way to cancel ringing phones when the call has been bridged to another channel. You can use the Manager interface with the Originate command to do that. I think you can get back a call handle with the FastOriginate variant. The handle can be used to call Hangup to cancel the call. Thanks for the suggestion. I'll take a look at that, because if I get that part working, I won't really need ICD except for future, more esoteric requirements ;-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ICD Features
Following up on a thread that I started about Agents/Queue and acknowledging calls before bridging them... Greg Boehnlein said that he was putting his efforts into ICD. I downloaded and installed ICD, and I can get simple queue and agent stuff working fine, and see that this new design is much cleaner and more powerful. That said, in the sample conf files, the acknowledge_call field is labeled as TBD, so it doesn't appear to be implemented yet. A quick scan of the c files shows it being parsed in at least one place (or so I think), but I am also not getting the debug output on the CLI that seems to be in there, so I'm either putting the keyword in the wrong place (I have it in the agent definition context) or that part of the code doesn't get hit. Anyway, the real point of this post is to point out that I am marginally surprised that there is close to zero traffic on this list regarding ICD, and I don't know if that's because no one uses it, no one has any problems with it (including wanting to get the new stuff working), or I'm just on the wrong list (I am not currently subscribed to -dev, but would head over there if this is an active topic on that list). If the authors of ICD are on this list, and prefer a private email dialogue, that would work as well, as I'm willing to be a serious tester of the app. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Updated Patch to chan_agent.c for PREACKANNOUNCE
Hmmm. I am often surprised when I don't get a response to a post that I think would interest at least _one_ person in the community. This one surprised me a little more, since I offered some code ;-). This morning, I just got a bounce notice that it was undelivered, which might explain it, except that I received the original post back through the list, so I don't understand it at all... Anyway, I solved the one bone-headed problem that I describe below, namely why did the agents show up in one DB and not the other. I didn't set the persistent keyword in the agents.conf file (doh...). All of my other questions still apply, as well as my offer to share the code/patch. Original Message Subject: [Asterisk-Users] Updated Patch to chan_agent.c for PREACKANNOUNCE Date: Thu, 18 Aug 2005 16:28:19 -0400 From: Hadar Pedhazur [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com First, many thanks to Greg Boehnlein for his patch to chan_agent.c for adding a preackannounce option. I am running CVS HEAD from 2005/07/31, and the patch failed in a few hunks, since the code was refactored to add in some CASE statements where there were compound if statements before. Anyway, I have successfully updated the patch to work against head as of 3 weeks ago, and would happily share that with anyone who is interested (just drop me a line off list). If a diff is preferable to the full 70k of C, just let me know what the correct options are for creating a diff suitable for patching the asterisk tree. OK, that said, I have a few questions and comments on this topic. This is my first use of the Queue command (very successfully so far), but I am afraid that expanding my use will require further patches, and I would like to verify that first. 1) If I use the syntax: Member = SIP/100 (rather than member = Agent/100, which maps to SIP/100) Then ackcall isn't used at all. In other words, a hard-wired member seems to ignore the agents.conf file completely. Is this the desired behavior? (It isn't for me...) 2) Since agents.conf is a separate file from queues.conf, having multiple queues does _not_ permit multiple preackannounce messages, each tied to a different queue (this strikes me as having better been patched into the Queue command). Similarly, you can't have one queue that has ackcall=yes, and another with ackcall=no. 3) I have the _exact_ same source version of CVS HEAD (from 2005/07/31) running on different servers (after a cvs co, I tar the source so that I can be sure I'm running _identical_ versions). On one machine, when an Agent logs in, I can see it in the DB, database show shows a key of: //Agents/1001 : [EMAIL PROTECTED];1001 On another machine, the DB shows _nothing_, yet the AgentCallbackLogin application works correctly (logging agents in and out), and shows the correct mapping on the CLI during a login. Still, the DB has _no trace_ of the Agents. I can't explain the difference in behavior, and would _love_ to have someone solve that mystery for me. I'm hoping that I am missing something obvious in the interaction between the Queue command and the Agents channel, and that some kind soul here will educate me. Otherwise, I think I might be off to doing more work in C than I ever though I would again in my life ;-). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Updated Patch to chan_agent.c for PREACKANNOUNCE
Greg Boehnlein wrote: Right. It's a botched design and chan_agent's design doesn't lend itself to being very helpful in the process, but that is where it had to go. This is the reason that I dropped work on it, as ICD was a much more intelligent design at the time. Thank you very much for response, it clarified all of my fears ;-) I just checked out the wiki page for ICD. I see that it's a work in progress. At the moment, my patched chan_agent does exactly what I need, given that I only have one queue that needs to be processed. I know I will eventually grow out of that. So, the question is: Is ICD ready to use in a semi-production mode (meaning, once the specific tasks that I want are tested, if I stick to those, is the code stable)?, or, since my need isn't immediate, would I be better off waiting a few months for the code to mature a bit. Just looking for an opinion, not absolution ;-) Thanks again. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Updated Patch to chan_agent.c for PREACKANNOUNCE
First, many thanks to Greg Boehnlein for his patch to chan_agent.c for adding a preackannounce option. I am running CVS HEAD from 2005/07/31, and the patch failed in a few hunks, since the code was refactored to add in some CASE statements where there were compound if statements before. Anyway, I have successfully updated the patch to work against head as of 3 weeks ago, and would happily share that with anyone who is interested (just drop me a line off list). If a diff is preferable to the full 70k of C, just let me know what the correct options are for creating a diff suitable for patching the asterisk tree. OK, that said, I have a few questions and comments on this topic. This is my first use of the Queue command (very successfully so far), but I am afraid that expanding my use will require further patches, and I would like to verify that first. 1) If I use the syntax: Member = SIP/100 (rather than member = Agent/100, which maps to SIP/100) Then ackcall isn't used at all. In other words, a hard-wired member seems to ignore the agents.conf file completely. Is this the desired behavior? (It isn't for me...) 2) Since agents.conf is a separate file from queues.conf, having multiple queues does _not_ permit multiple preackannounce messages, each tied to a different queue (this strikes me as having better been patched into the Queue command). Similarly, you can't have one queue that has ackcall=yes, and another with ackcall=no. 3) I have the _exact_ same source version of CVS HEAD (from 2005/07/31) running on different servers (after a cvs co, I tar the source so that I can be sure I'm running _identical_ versions). On one machine, when an Agent logs in, I can see it in the DB, database show shows a key of: //Agents/1001 : [EMAIL PROTECTED];1001 On another machine, the DB shows _nothing_, yet the AgentCallbackLogin application works correctly (logging agents in and out), and shows the correct mapping on the CLI during a login. Still, the DB has _no trace_ of the Agents. I can't explain the difference in behavior, and would _love_ to have someone solve that mystery for me. I'm hoping that I am missing something obvious in the interaction between the Queue command and the Agents channel, and that some kind soul here will educate me. Otherwise, I think I might be off to doing more work in C than I ever though I would again in my life ;-). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX TO IAX call between two registered servers
John Millican wrote: [snipping] I get the following message on home: Aug 8 18:31:03 WARNING[20598]: chan_iax2.c:6820 socket_read: Call rejected by 69.xxx.xxx.xxx: No authority found and get this message on away Aug 8 18:32:42 NOTICE[16923]: chan_iax2.c:5448 socket_read: Rejected connect attempt from 165.xxx.xxx.xxx home iax.conf [away] type=peer username=away auth=plaintext secret=x host=dynamic context=pap2 dissallow=all allow=ulaw [more snipping] away iax.conf [home] type=peer user=home secret=x host=dynamic context=default [home-in] type=user username=home secret=x context=default [final snipping] any suggestions would be greatly appreciated. Thank you, John M OK, I think your problem is simple, at least I hope so ;-) Specifically, I think you have the wrong syntax in your Dial command. You are dialing: exten = _998, 1, Dial(IAX2/x:[EMAIL PROTECTED]); The above says (to me) to use channel IAX2, username=x, password=home, in context away (in the local iax.conf). I doubt that's what you want! I think you want to dial extension x, with username home (or remote context home, which isn't place to put it), at local context away. So, change your Dial string to: exten = _998, 1, Dial(IAX2/[EMAIL PROTECTED]/x); That might fail too, for the following reasons: In your Dial command, you are specifying user home, calling via your context away (in home's iax.conf). So far, so good. However, in the away context (on home's machine), you are setting the username to away. There is no context away in the iax.conf on the away machine. Perhaps (I'm not sure), the username=away is the line that will interfere with your authorization (after making the change to the Dial command as noted above). Two suggestions: 1) Delete the line that says username=away in the away context in the iax.conf on home. 2) Change it to be username=home (in which case you can change your Dial command to be just IAX2/away/XXX ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sixtel is still alive?
Jay Milk wrote: Is anyone still using them? How's quality now? Availability? I wouldn't pick them up for incoming calls again, but it might be nice to keep an extra outgoing provider on hand, since voipjet and livevoip have been behaving like boxes of chocolate lately. I have been using sixtel for outgoing calls since 2/2/2005. Call quality has been very high (uLaw), and availability of a channel nearly perfect. Thankfully, I have never had to open a trouble ticket. A buddy of mine has been waiting for a vanity 800 DID for 3+ months though :-(. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail Email Bouncing
I have been using Asterisk for a couple of years now. I recently upgraded to CVS HEAD (March 9, 2005). Independently (and perhaps this is the problem) I also upgraded from Postfix 2.0.16 to 2.2.1. Anyway, I just realized this morning that I have not been getting emails when someone leaves me voicemail. The voicemail gets recorded correctly, and gets emailed as well. However, the email bounces with the following in /var/log/messages: Apr 1 10:05:05 zc postfix/pickup[2629]: C73021F8013: uid=0 from=root Apr 1 10:05:05 zc postfix/cleanup[2605]: C73021F8013: message-id=Asterisk-0-55 [EMAIL PROTECTED] Apr 1 10:05:06 zc postfix/cleanup[2605]: C73021F8013: to=unknown, relay=none, delay=1, status=bounced (No recipients specified) My spam filter is tossing the bounce, which is another reason why I didn't notice it for this long. However, one message made it through, and in the attachment to the bounce, the email was addressed correctly (the right To: from voicemail.conf, and the correct default From: address as well). To repeat, this could be a Postfix config error, since that changed in between too, and not necessarily an Asterisk problem. What I'd really like to know is whether there are debugging options I can turn on at the Asterisk level to see exactly what is being sent to Postfix, so that I can clearly rule out one of them as the cause of the problem. Thanks in advance! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail Email Bouncing - SOLVED
Hadar Pedhazur wrote: I have been using Asterisk for a couple of years now. I recently upgraded to CVS HEAD (March 9, 2005). Independently (and perhaps this is the problem) I also upgraded from Postfix 2.0.16 to 2.2.1. Anyway, I just realized this morning that I have not been getting emails when someone leaves me voicemail. The voicemail gets recorded correctly, and gets emailed as well. However, the email bounces with the following in /var/log/messages: Apr 1 10:05:05 zc postfix/pickup[2629]: C73021F8013: uid=0 from=root Apr 1 10:05:05 zc postfix/cleanup[2605]: C73021F8013: message-id=Asterisk-0-55 [EMAIL PROTECTED] Apr 1 10:05:06 zc postfix/cleanup[2605]: C73021F8013: to=unknown, relay=none, delay=1, status=bounced (No recipients specified) My spam filter is tossing the bounce, which is another reason why I didn't notice it for this long. However, one message made it through, and in the attachment to the bounce, the email was addressed correctly (the right To: from voicemail.conf, and the correct default From: address as well). To repeat, this could be a Postfix config error, since that changed in between too, and not necessarily an Asterisk problem. What I'd really like to know is whether there are debugging options I can turn on at the Asterisk level to see exactly what is being sent to Postfix, so that I can clearly rule out one of them as the cause of the problem. Thanks in advance! Responding to my own post, to save people the time to track this down. It was a Postfix problem, not an Asterisk one. The fix was to add a mailcmd in the general context as follows: mailcmd=/usr/sbin/sendmail.postfix -t -oi The -oi is probably unnecessary, but the addition of the .postfix is what did the trick. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FWD and SIPPHONE problems after upgrading toCVSHEAD - VERIFIED
Mike Matthews wrote: This works for me both incoming and outgoing w/Sipphone. Note there is NO username, secret entries in the peer definition. I am using * vers 1.05 register=1747nnn:[EMAIL PROTECTED]/1747xxx ; note:extension in extensions.conf matches for incoming Thank you very much for weighing in, I was getting paranoid that everyone was blacklisting the few posts I make a year :-) I too can get it to work on Stable (versions 1.0.3 and 1.0.6), so I'm not surprised to hear your results. I also had to add an insecure=very, which is disappointing, but since it is hard-coded to a particular IP address, I guess it's not as awful as it could be otherwise. That said, I don't think I dropped the username from the CVS HEAD test, but I did add insecure=very, which still failed. So, I continue to maintain that _something_ has changed in CVS HEAD which makes incoming and outgoing calls fail to SIPPHONE, and _outgoing_ calls via IAX2 fail to FWD (for me), while incoming calls from FWD via IAX2 definitely continue to work for me. For the moment, it remains a mystery... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FWD and SIPPHONE problems after upgrading to CVSHEAD - VERIFIED
Replying to my own post :-(
Yes, I'm top-posting, because no one ever seems to reply to my posts
anyway, I don't want to make you re-read my old post just to find out
what I'm adding.
I have _not_ solved the problem, but I reverted briefly to 1.0.3, and
I can indeed call to FWD without any problems. This is with _no
changes_ to the iax.conf between the two, so something in the recent
CVS HEAD has caused me to be able to receive calls from FWD (via
IAX2), but no longer call FWD.
I can't believe this is only happening to me, but apparently, it must
be...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hadar
Pedhazur
Sent: Thursday, March 03, 2005 5:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] FWD and SIPPHONE problems after upgrading to
CVSHEAD
I have been successfully connected (incoming and outgoing)
to FWD for a very long time. A few months ago, I changed
from SIP-based FWD service to IAX2-based, and that went fine
as well, both incoming and outgoing.
At the time, I was running Asterisk 1.0.3 Stable.
I rarely use the service, so other than noticing that I was
always successfully registered to FWD, I didn't make or
receive calls recently.
2 weeks ago, I upgraded to CVS HEAD:
Asterisk CVS-HEAD-02/21/05-09:07:50
Still didn't make or receive calls to FWD since the upgrade,
but everything else has worked flawlessly (including sixTel,
NuFone, etc.). All my softphones (SIP and IAX2) and
Sipura-2000's work perfectly too.
On to the problem... A few days ago, I signed up for an
account with SIPPhone. When I did a sip reload, which had
the register statement, I immediately got a call welcoming
me, so I thought everything was fine. It wasn't.
I have been unable to make any calls to sipphone, and even
though the registration appears to work (and my.sipphone.com
shows me as online), all calls to my number actually claim
that I am unavailable, and go directly to voicemail.
Before I show my configs and CLI output, a few more
background data points:
I can successfully connect to sipphone with their own
download of X-Lite (pre-configured), and I can set a profile
in SJPhone by hand and it works too, both incoming and
outgoing, so I have the correct password, etc.
Today, I tested outgoing calls on FWD (actually to use the
peering to test incoming on sipphone), and my calls to FWD
are failing now as well. Incoming from FWD (via IAX2) still
works correctly. Worse, I also tried to go back to SIP-based
outgoing to FWD, and I get the same error as I do for
sipphone, so now I am starting to suspect that it's Asterisk
CVS HEAD that's possibly the problem...
Finally, the machine that is connected to both FWD and
SIPPHONE is on a public static IP address, so there are no
NAT issues involved here, and no STUN services needed
either.
OK, here is the sip.conf entry:
register=1747XXX:[EMAIL PROTECTED]/4321
[proxy01.sipphone.com]
type=peer
;auth=md5
secret=YYY
username=1747XXX
fromuser=1747XXX
fromdomain=proxy01.sipphone.com
host=proxy01.sipphone.com
nat=no
qualify=no
canreinvite=no
disallow=all
allow=ulaw
;context=default
;callerid=Hadar Pedhazur 1747XXX
(The above has been variously named sipphone, sipphone-out
and now proxy01.sipphone.com, all with the same exact
result! Also, the above has been tried with auth=md5
uncommented as well, and also no password, and
insecure=vary, etc.)
Now extensions.conf:
; Dial SIPPhone with a prefix of 76
exten = _76.,1,SetCallerID(${SIPPHONENUM})
exten = _76.,2,SetCIDName(Hadar Pedhazur)
exten = _76.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
OK, here's the output from a call:
-- Called [EMAIL PROTECTED]
-- Got SIP response 500 I'm terribly sorry, server error occured
(1/SL) back from 198.65.166.131
-- SIP/proxy01.sipphone.com-78d5 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Notice that at the end of the Got SIP response line, is
the correct IP address of their server, so it's finding the
correct server. As mentioned above, if I switch FWD to call
via SIP, I get the same _exact_ error message, but from
FWD's correct IP address rather than SIPPhone. This seems
very suspicious to me...
Finally, just for completeness, here is the CLI output for
attempting to call FWD via IAX2. This used to work, though I
can't say when it started failing:
-- Called fwd-gw/612
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
-- IAX2/fwd-gw-4 is busy
I have called _many_ times, and every time I get an instant
is busy in the CLI, and I can receive calls without a
problem, so I don't think it's that they really are busy.
For now, I'm more interested in fixing the SIPPhone problem,
and if that ends up working, and doesn't shed light on the
FWD problem, I'll move on to that. Of course, PITA that it
would be, my next move if no one here can help will be to
restore my settings from a few weeks back (yes, I back up
[Asterisk-Users] FWD and SIPPHONE problems after upgrading to CVS HEAD
I have been successfully connected (incoming and outgoing)
to FWD for a very long time. A few months ago, I changed
from SIP-based FWD service to IAX2-based, and that went fine
as well, both incoming and outgoing.
At the time, I was running Asterisk 1.0.3 Stable.
I rarely use the service, so other than noticing that I was
always successfully registered to FWD, I didn't make or
receive calls recently.
2 weeks ago, I upgraded to CVS HEAD:
Asterisk CVS-HEAD-02/21/05-09:07:50
Still didn't make or receive calls to FWD since the upgrade,
but everything else has worked flawlessly (including sixTel,
NuFone, etc.). All my softphones (SIP and IAX2) and
Sipura-2000's work perfectly too.
On to the problem... A few days ago, I signed up for an
account with SIPPhone. When I did a sip reload, which had
the register statement, I immediately got a call welcoming
me, so I thought everything was fine. It wasn't.
I have been unable to make any calls to sipphone, and even
though the registration appears to work (and my.sipphone.com
shows me as online), all calls to my number actually claim
that I am unavailable, and go directly to voicemail.
Before I show my configs and CLI output, a few more
background data points:
I can successfully connect to sipphone with their own
download of X-Lite (pre-configured), and I can set a profile
in SJPhone by hand and it works too, both incoming and
outgoing, so I have the correct password, etc.
Today, I tested outgoing calls on FWD (actually to use the
peering to test incoming on sipphone), and my calls to FWD
are failing now as well. Incoming from FWD (via IAX2) still
works correctly. Worse, I also tried to go back to SIP-based
outgoing to FWD, and I get the same error as I do for
sipphone, so now I am starting to suspect that it's Asterisk
CVS HEAD that's possibly the problem...
Finally, the machine that is connected to both FWD and
SIPPHONE is on a public static IP address, so there are no
NAT issues involved here, and no STUN services needed
either.
OK, here is the sip.conf entry:
register=1747XXX:[EMAIL PROTECTED]/4321
[proxy01.sipphone.com]
type=peer
;auth=md5
secret=YYY
username=1747XXX
fromuser=1747XXX
fromdomain=proxy01.sipphone.com
host=proxy01.sipphone.com
nat=no
qualify=no
canreinvite=no
disallow=all
allow=ulaw
;context=default
;callerid=Hadar Pedhazur 1747XXX
(The above has been variously named sipphone, sipphone-out
and now proxy01.sipphone.com, all with the same exact
result! Also, the above has been tried with auth=md5
uncommented as well, and also no password, and
insecure=vary, etc.)
Now extensions.conf:
; Dial SIPPhone with a prefix of 76
exten = _76.,1,SetCallerID(${SIPPHONENUM})
exten = _76.,2,SetCIDName(Hadar Pedhazur)
exten = _76.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
OK, here's the output from a call:
-- Called [EMAIL PROTECTED]
-- Got SIP response 500 I'm terribly sorry, server error occured
(1/SL) back from 198.65.166.131
-- SIP/proxy01.sipphone.com-78d5 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Notice that at the end of the Got SIP response line, is
the correct IP address of their server, so it's finding the
correct server. As mentioned above, if I switch FWD to call
via SIP, I get the same _exact_ error message, but from
FWD's correct IP address rather than SIPPhone. This seems
very suspicious to me...
Finally, just for completeness, here is the CLI output for
attempting to call FWD via IAX2. This used to work, though I
can't say when it started failing:
-- Called fwd-gw/612
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
-- IAX2/fwd-gw-4 is busy
I have called _many_ times, and every time I get an instant
is busy in the CLI, and I can receive calls without a
problem, so I don't think it's that they really are busy.
For now, I'm more interested in fixing the SIPPhone problem,
and if that ends up working, and doesn't shed light on the
FWD problem, I'll move on to that. Of course, PITA that it
would be, my next move if no one here can help will be to
restore my settings from a few weeks back (yes, I back up
religiously :-), and see if 1.0.3 will just work.
Thanks in advance to any kind soul who has some insight!
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Migrating from CVS HEAD to Stable 1.0.3?
I am sorry to ask such a simple questions. I have been using Asterisk successfully for well over a year now on three servers. I was using CVS HEAD, and the last time I updated was sometime back in July. I decided to switch to the recent stable 1.0.3. I built zaptel, libpri and asterisk, and installed them in that order. All installations reported success. (I stopped asterisk before installing any of them...) When I started up safe_asterisk (and connected to the console), the first error I got was that iaxprov.conf wasn't found. I copied the sample from there to /etc/asterisk and it then got a little further. The last message I see is that it found phone.conf, and then it dies with an Error 1. I am sorry but I don't have the exact error message in front of me, and I had to revert quickly so that my phone would work. When migrating (I don't know if it's downgrading or upgrading) from a July CVS HEAD to 1.0.3, do I need to do anything special, like: 1) Add, change or delete any of my existing conf files in /etc/asterisk, or should they just work? 2) Remove the modules from the old build before doing the install (I assumed that they would just be overwritten, but perhaps that isn't the case...)? 3) Anything else?!? Again, if this appears in any docs, I really apologize, but a quick skim of the doc directory didn't seem to contain a file that seemed to cover this situation... Thanks in advance for any guidance! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Forward voicemail to *remote* voice mailbox?
Brian Capouch wrote:
I've googled a bunch and looked at the Wiki, but so far if it's
out there I can't find it.
I would like for a user to be able to forward to a remote
mailbox, if that can be done without any AGI-ish fancies.
I'm going to bet that it can't be done with standard dialplan
logic, but thought I would ask here before starting to think up
something as an alternative.
Howdy Brian, very long time no speak :-)
Yes, it can be done easily, I do it all the time.
Here is my macro for doing voicemail on my local servers, which
really want to leave the voicemail on my main server:
[macro-pbxvm];
exten = s,1,Dial(${PBX4ALL}/${ARG2},10); OK, save the voicemail there
exten = s,2,Voicemail(u${ARG1}); PBX timed out, save it
here...
exten = s,3,Hangup ; We're done
The variable PBX4ALL defines a standard IAX2/[EMAIL PROTECTED] thingy, and
ARG2 is the correct mailbox to dial on the other end. If it can't get
through, it leaves the vm on the local server.
On the other end, there is a specific extension (for each person)
which goes directly to vm.
Hope this helps!
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Leave one call to pick up another
Andrew Thompson wrote: Eric Wieling wrote: How is this different from the way standard call waiting works when provided from your telco? Um, he actually has two phone lines, not just one that he's flash-ing back and forth between. If he hangs up the line, does the second call not continue ringing? I take it once he hangs up the line both calls are gone? I can't help with any solution, but I can add my voice describing another symptom of this exact problem, in direct response to your last question. I have two lines, each handled by a Digium X100P card. If I am on the phone (whether I initiated a call, or received one, whether it uses one of the POTS lines or whether it's a VoIP call), if another call comes in, I hear the Call Waiting signal. If I simply hang up the current call, I lose _both_ calls. Meaning, the phone does not start ringing with the pending call any longer. This is _not_ the same behavior that I had with the same exact phone, when it was connected to the POTS line directly. Hanging up on the current call yields a ringing for the second call, after a second or two delay... P.S. If I flash the call, I can indeed speak to the second caller, and bounce back and forth between the calls, so I get the same behavior that the original poster (Brian Capouch) described. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ZAPRTC question(s)
Much snipping along the way :-) Tony Mountifield wrote: Actually, I have used Zaprtc quite successfully. The only reason you have to disable kernel RTC support is because Zaprtc is actually a *replacement* for the standard RTC module. It provides the same facilities, but includes extra parts for Zaptel use. Thanks for the very clear explanation. You've given me the confidence to try this too. Hopefully, I'll get to it over the weekend, and get the same results that you are having :-). Thanks again for taking the time to respond, especially to a 5-day old post! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAPRTC question(s)
I have a system with no Digium hardware in it (two others with 2 X100P cards in each of them as well). I'm interested in using MeetMe in the one without the hardware (it works great in the ones with the hardware). I can't use ztdummy, because the system has usb-ohci drivers, rather than usb-uhci. I have read the little there is about zaprtc, and I am wondering whether there is a downside in turning off RTC support in the kernel, and recompiling. Are there other things that might break if I do this (it simply feels like a more drastic step than the ztdummy approach)? (I am running Red Hat 9.0) Finally, and this will show my complete naivete for linux programming, I am curious as to why no one has written a timer that simply hooks the standard kernel installed RTC? From the rtc.txt file in the Documentation directory of the kernel source, it seems that one can hook the interrupt and get the clock ticks delivered via interrupt directly to your c code. Isn't that what is needed to get a stable timing device in *? Just curious, as I'm sure that it's way more sophisticated than that... Thanks in advance. P.S. The system with no Digium hardware in it is in a colo facility that is 250 miles from my house, and besides, I don't have physical access to the machine. So, it would be painful, and expensive, for me to arrange for a Digium card to be installed in the machine, and it would be used for nothing other than the clock, since there are no other interfaces available for me to plug into the card. This was just to nip the why don't you just pony up for a Digium card? responses :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX Followup
No one responded to the original, nor to my followup. So, here I am again following up my own followup :-) I was speaking with a colleague of mine today who is running * at his office and at home. He told me that he was using iaxcomm and couldn't hear any sounds. I told him that I had the same problem with DIAX (after it was working flawlessly for a month). He told me that he did a cvs up, rebuilt *, and the problem went away. Instantly, I realized I was bitten by the same cvs bug. I had a working system, I did a cvs up because I thought I had a transcoding problem after I bought some g729 licenses, and the next time I ran DIAX, it failed. At the time, I didn't make the connection that I had done a cvs up that day, because it never occurred to me that something as simple as a direct IAX connection would be broken by updating *. Anyway, after updating to cvs from today, DIAX is once again working great for me, and I thought I'd share that tidbit with anyone else who might be silently pulling their hair out wondering what's wrong in their config that used to work... Hadar Pedhazur wrote: Anyway, in my P.S. yesterday (the main post was on Codec problems), I described a situation where any IAX softphone was registering successfully, and then having zero sounds heard on either side of the call. Here is an iax2 debug output from a DIAX call to a local * server, dialing the extension that goes directly to the demo application. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec Voodoo
All three of my servers are at the same level of cvs checkout (within minutes of each other), I believe from March 22, 2004. All of my calls to NuFone are using GSM, though I allow iLBC as well. Thanks for the response, I was beginning to think my questions were invisible :-) Andres wrote: Are you making calls out to Nufone or simply from one of your servers to another? We noticed this problem when we upgraded one of our servers to the latest CVS and left another one with an older version. Seems that the latest changes with rtp.c need to be applied everywhere. When we upgraded all servers then the audio returned to normal but the connection with Nufone started sounding horrible. We had to roll back to the older version of rtp.c to get back the good audio with Nufone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAX Followup
Anyway, in my P.S. yesterday (the main post was on Codec problems), I described a situation where any IAX softphone was registering successfully, and then having zero sounds heard on either side of the call. Here is an iax2 debug output from a DIAX call to a local * server, dialing the extension that goes directly to the demo application. AsteriskHouse*CLI iax2 debug IAX2 Debugging Enabled Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 1ms SCall: 22150 DCall: 0 [10.251.1.2:4569] VERSION : 2 CALLING NUMBER : XXX-XXX- CALLING NAME: Hadar Pedhazur FORMAT : 2 CAPABILITY : 2 USERNAME: hadar CALLED NUMBER : DNID: CALLED CONTEXT : from-hadar Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 1655939482ms SCall: 4 DCall: 22150 [10.251.1.2:4569] AUTHMETHODS : 2 CHALLENGE : 133911739 USERNAME: hadar Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 000 Type: VOICE Subclass: 2 Timestamp: 00010ms SCall: 22150 DCall: 0 [10.251.1.2:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00010ms SCall: 4 DCall: 22150 [10.251.1.2:4569] Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00020ms SCall: 22150 DCall: 4 [10.251.1.2:4569] MD5 RESULT : 91f6cc1e25fasd0bb43c22d366e4dcd4 -- Accepting AUTHENTICATED call from 10.251.1.2, requested format = 2, actual format = 2 -- Executing Goto([EMAIL PROTECTED]/4, default|s|1) in new stack -- Goto (default,s,1) -- Executing Wait([EMAIL PROTECTED]/4, 1) in new stack Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 003 Type: IAX Subclass: ACCEPT Timestamp: 1658659482ms SCall: 4 DCall: 22150 [10.251.1.2:4569] FORMAT : 2 Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 1658659482ms SCall: 22150 DCall: 4 [10.251.1.2:4569] -- Executing Answer([EMAIL PROTECTED]/4, ) in new stack Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: ANSWER Timestamp: 2227876761ms SCall: 4 DCall: 22150 [10.251.1.2:4569] -- Executing DigitTimeout([EMAIL PROTECTED]/4, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout([EMAIL PROTECTED]/4, 10) in new stack -- Set Response Timeout to 10 -- Executing BackGround([EMAIL PROTECTED]/4, demo-congrats) in new stack Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 003 Type: VOICE Subclass: 2 Timestamp: 2227876762ms SCall: 4 DCall: 22150 [10.251.1.2:4569] -- Playing 'demo-congrats' (language 'en') Tx-Frame Retry[001] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: ANSWER Timestamp: 2227876761ms SCall: 4 DCall: 22150 [10.251.1.2:4569] Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 2227876761ms SCall: 22150 DCall: 4 [10.251.1.2:4569] Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: PING Timestamp: 04356ms SCall: 22150 DCall: 4 [10.251.1.2:4569] Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: PONG Timestamp: 04356ms SCall: 4 DCall: 22150 [10.251.1.2:4569] Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 04356ms SCall: 22150 DCall: 4 [10.251.1.2:4569] Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: HANGUP Timestamp: 06930ms SCall: 22150 DCall: 4 [10.251.1.2:4569] CAUSE : Dumped Call Tx-Frame Retry[-01] -- OSeqno: 005 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 06930ms SCall: 4 DCall: 22150 [10.251.1.2:4569] == Spawn extension (default, s, 5) exited non-zero on '[EMAIL PROTECTED]/4' -- Hungup '[EMAIL PROTECTED]/4' AsteriskHouse*CLI iax2 no debug ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 International Termination
Humble apologies for using list space for this. The message is actually for Stephen Karrington. I wrote a lengthy reply to you directly (Stephen), but it was bounced by your spam filter. If you are interested in seeing it, please contact me directly, and let me know how else to forward that email to you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec Voodoo
I have three * servers that all talk to each just fine, and all talk to other * servers (like NuFone, VoicePulse, etc.). I have hard-phones connected to Sipura SPA-2000s on two of the * servers via a local network connection. The third * server only gets connected to remotely, both with IAX and SIP softphones, and with a roaming Sipura with hard-phones. The setup works well. All of the * servers communicate exclusively with GSM between themselves (and also to NuFone and VoicePulse). The quality is pretty good. The local hard phones are using g711 uLaw (since I think that the X100P cards I believe use uLaw by default as well, but I could be way off on that assumption). Codec transcoding from uLaw to GSM seems to work just fine. From a couple of people who post regularly on this list, I have heard that they have great success with iLBC (and some with Speex as well). I think that NuFone prefers iLBC as well, though it works remarkably well for me with GSM. I did some experiments in forcing my * servers to communicate with each other only with iLBC. When I do that, and can see that they are indeed using iLBC, the quality is horrible. There is long stutter, like every sound is being stretched out. I purchased g729 licenses from Digium for all three servers as well. Using g729 on the Sipura devices yielded no better quality than the built-in g726. However, when I made two * servers communicate only with g729, the quality was marginally better than iLBC, and ridiculously worse than GSM. This was surprising to me. All of this is with a very recent cvs checkout of *, done this past Monday the 22nd I believe. Last point is that if I turn jitterbuffer on (with =yes), then I never hear _any sound_ whatsoever, but there are _no errors_ on either side of the channel. I can see on the CLI that voicemail prompts are being played (for example), but I can't hear anything on either side. Turning jitterbuffer=no immediately restores sound, but the quality only sounds good with GSM. What I don't understand is how some people have success with iLBC, and I don't. I also noticed one or two posts from people that claim that GSM isn't working for them, yet it works really well for me. Are there any settings that I am unaware of (other than the standard allow/disallow directives) that I should be tweaking to make these other codecs work as I understand they should? P.S. One last piece of voodoo, just if anyone knows the answer to this. On occasion, I use DIAX to connect to the remote * server. It works very well, and is the best of the IAX softphones (IMHO). Yesterday, it was working just fine. Today, from a different location (both yesterday and today behind NAT, just from different networks), it connects fine, but I have zero sounds and zero errors. There were _no_ changes to the server or the software setup in between. In the past, I have had trouble using X-Lite to this particular * server. Today, when DIAX wasn't working (neither was iaxcomm, it's not a specific DIAX problem), I tried X-Lite again, and it worked flawlessly... The last bit of info on this is that one of the other * servers is on the same lan as the DIAX client, but on different machines. Both are coming from the same NAT router though. The * machine is in the DMZ, so all packets that are sent to the public side are routed directly to *, and that part works perfectly. I don't know if DIAX is clashing with * packets, but I know this has worked in the past (though it's been 2 weeks since I've tried, and I did cvs up the * server since it last worked...). Thanks in advance to any brave soul who tackles some or all of these questions/issues! :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playback Volume for Record Application
The Asterisk Demo prompts come through loud and clear on any phone that I use to call in on. When someone leaves me voicemail, it also comes through loud and clear. When I use the Record application and then use the recorded file in a Playback or Background application, it is very soft (clear, but the volume is cranked way down). This is true for all format types (I've tried wav, WAV and gsm). Also, from a number of different input devices (headset using an DIAX, real phone connected on the lan to *, etc.). Just curious whether there is anything I can do to improve the volume, other than recording the prompts in another application and dropping in the resulting files (it's obviously ultra-convenient to be able to change prompts just by dialing the correct extension) :-). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback Volume for Record Application
Thanks Steve. Right after I sent my note, it occurred to me that I could create a dummy voicemail account, and point my current Record extension to that, and use Voicemail to record the higher volume version. I just did one quick test, unscientific at best, and I think the above works reasonably well. Instead of leaving a message, I actually created a dummy vm user, and used VoicemailMain to record a busy.wav file, which I then moved to a normal sound file, and used the Playback application. It sounded enough louder to me than the original Record application made it. Steven Critchfield wrote: I've not experienced the low volume when using record, but I have when using the record command in AGI. Upon looking around to see what it was that caused voicemail to be of such a passable volume, but not our AGI apps, I noticed that there is a section in format_wav.c that will bit shift the audio data up to increase volume. When I changed our application from recording in GSM to recording in WAV, it fixed some of our volume problems. Not necessarily a solution to your problem but a few data points to help sort the problem out. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phantom problem authenticating with RSA?
I have three * servers that are inter-connected, registering with each other. Up until yesterday I was authenticating all three with MD5, and all was working fine. Yesterday I switched to RSA, and everything is working as well. I can see AUTHENTICATED messages on the console if one of the servers is restarted and reconnects, etc. Everything is working fine with calls being passed between them as well (which is why I labeled the subject Phantom problem). However, whenever a call is initiated between the servers I see the following NOTICE message: -- Called [EMAIL PROTECTED]/2001 -- Called [EMAIL PROTECTED]/2001 Mar 18 07:46:19 NOTICE[1150528304]: chan_iax2.c:3507 authenticate: No way to send secret to peer 'XX.XX.XX.XX' (their methods: 4) Mar 18 07:46:19 NOTICE[1150528304]: chan_iax2.c:3507 authenticate: No way to send secret to peer 'YY.YY.YY.YY' (their methods: 4) -- SIP/sipura-4b82 is ringing -- Call accepted by XX.XX.XX.XX (format ULAW) -- Format for call is ULAW -- IAX2[remote1]/3 stopped sounds -- Call accepted by YY.YY.YY.YY (format ULAW) Method 4 is RSA, which is what I have in all of the iax.conf files (below). The call shown above was successfully answered by a sipura device connected to remote2, so I am not having an authentication problem which is causing a problem at the user experience level, but this seems like something is still mis-configured on my part. Here are the iax.conf entires: on the local machine: [remote2] context=remote2-in type=friend host=remote2.com ; not the real name... auth=rsa inkeys=remote2 outkey=local [remote1] context=remote1-in type=friend host=remote1.com ; not the real name... auth=rsa inkeys=remote1 outkey=local on the remote1 machine: [remote2] context=remote2-in type=friend host=remote2.com auth=rsa inkeys=remote2 outkey=remote1 [local] context=local-in type=friend host=local.com auth=rsa inkeys=local outkey=remote1 on the remote2 machine: [local] context=from-local type=friend auth=rsa inkeys=local outkey=remote2 host=dynamic callgroup=1 pickupgroup=1 qualify=5 [remote1] context=from-local type=friend auth=rsa inkeys=remote1 outkey=remote2 host=dynamic callgroup=1 pickupgroup=1 qualify=5 Finally, since both local and remote1 are technically behind NAT firewalls, and remote2 is on a public IP address, I have register statements in both local and remote1 iax.conf files, and that's why the entries in remote2 have host=dynamic for those machines. I think that the qualify=5 statements are ignored in the iax.conf file, and I will remove them, but since they're in there now, I wanted to show the complete entries. Here are the register statements: on remote1: register = remote1:[EMAIL PROTECTED] on local: register = local:[EMAIL PROTECTED] Any help would be appreciated. Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hang-ups when using IAX
Darrin, I had a similar (though not identical) problem. The solution in my case was to add notransfer=yes in the iax.conf context for the IAX softphone. It's possible that the hand off to attempt a native transfer for you is failing because one of the servers is behind a NAT router. Anyway, it's worth a quick test. Darrin Johnson wrote: I have two Asterisk systems running in my environment. In between the two there is a router running NAT. One server services extensions 90XX and the other extensions 95XX. Both boxes are running Red Hat 9 with version 0.7.2 Asterisk. I am running IAX and registering an IAX softphone to each server so two IAX clients with one registered as a 90XX number to the 90XX server and one registered to the 95XX server with a 95XX number. A call is initiated from the client registered to the 90XX server to the client registered on the 95XX server. The call is completed successfully but then after about 30 seconds to a minute the initiating client complains that the remote user (95XX client) hung-up. The 95XX client has the connection still open and live until the hang-up button is manually clicked. The debug in Asterisk shows that the 90XX server records a remote hang-up, but the 95XX server does not record anything until the hang-up button is pushed from the 95XX client. Does anyone have any ideas as to why I would be getting a hang-up after about 30 seconds when using IAX in this type of scenario? I have tried multiple clients with the same result which is implying there must be a problem server-to-server. Thanks much for your help! Darrin Johnson Systems Engineer IS Domain Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec Translation Problem on IAX Softphones - Incoming Only
Philipp von Klitzing was kind enough to send the solution to me off list, and I want to save others from responding as well, since his solution worked! Here's what he wrote: Add this to the [hadar] context and reload the server: disallow=all allow=gsm Only after seeing this, did I realize how obvious it is. Clearly, the IAX softphones can only handle GSM, and by not overriding the default setting of allowing ulaw, I was telling * that it was OK to change the format from GSM to ULAW since the Sipura wanted ULAW. Doh. On a separate note, I was having trouble getting cut off after 20-30 seconds on calls from DIAX to other IAX2 servers, but not from my hard phones connected via Sipuras. A quick google of the list solved that problem too, by adding: notransfer=yes to the same context above. I think that's because my softphone is behind NAT, on the same segment that another * server is behind NAT as well, or something similar to that. Anyway, all is right with the world again :-) Thanks Philipp! - Lots of snipping, just summary problem and conf files left in... Hadar Pedhazur wrote: Just to pendatically repeat myself, the same exact devices work fine if DIAX initiates the call (even if it ends up passing over a real Zap channel to the Sipura), and if the Sipura initiates the call and DIAX picks it up, the above is the error. Here are the relevant sections of the iax.conf file: [general] port=5036 bandwidth=high disallow=all allow=ulaw allow=alaw allow=gsm allow=iLBC tos=lowdelay ; ; Connect to the apartment * server ; [apartment] context=apartment-in type=friend host=apartment.X.com secret=mysecret auth=md5 [hadar] type=friend context=from-hadar secret=mysecret auth=md5 host=dynamic callerid=Hadar Pedhazur (XXX) XXX- mailbox=100 In the apartment server iax.conf file, I have an identical set of declarations in the [general] section, and nearly identical sections for the devices there (meaning, no one is overriding the codecs defined in general). I have played with a number of permutations of playing with the codecs in general as well as the individual ones, and the above works best for everything except a call _answered_ by DIAX. Any pointers would be greatly appreciated! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Parsing a variable, or rather Splitting a variable
The wiki page on ChanIsAvail says: Thus, if you are going to use the value of AVAILCHAN, you need to strip the session ID off. I understand why this is necessary (from seeing what is returned in the CLI), but I don't see any commands that can accomplish this since the session id after the - is of variable length (so you can't use : notation, or StripLSD, etc.). I would like to avoid making an external AGI call if possible, and I'm hoping I'm just missing something obvious here. Thanks in advance! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Parsing a variable, or rather Splitting a variable
Derek, this is great, thanks much! I can definitely make this work for
me, but in practice, my problem is a little more complex, and on the
off chance that there is a solution, I'll state it. Otherwise, I can
work around it (I think).
Here's an example of what my input looks like (in reality, it's even
worse than this, because there are 3 or 4 variable length channels):
ChanIsAvail(SIP/2001IAX2/[EMAIL PROTECTED])
So, a fixed length substring won't really work for me. Of course,
worst case scenario here I can test the substring first for whether it
starts with SIP or IAX2 (for the above), and then branch accordingly.
That's not too bad (logic-wise), but it adds a number of steps to the
dialplan that perhaps could be avoided if there was a
Split(String,'-',1) (for example, to pick out the first token)...
Thanks again, since I'm now all set even if my particular needs can't
be met directly. :-)
Derek Bruce wrote:
given:
exten = 2001,1,ChanIsAvail(SIP/2001SIP/3001)
exten = 2001,2,SubString,ToDial=${AVAILCHAN}|0|8
exten = 2001,3,Dial(${ToDial},20)
you know that your dialstring will be 'SIP/2001', your technology prefix
('SIP/' 4 characters) length plus your extention length (4 characters) = 8
characters... a fixed length...
for a variable length extention use:
exten = _1.,1,ChanIsAvail(SIP/$EXTEN})
exten = _1.,2,SubString,ToDial=${AVAILCHAN}|0|${LEN(EXTEN)}+4 )
exten = _1.,3,Dial(${ToDial},20)
- Original Message -
From: Hadar Pedhazur [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, March 09, 2004 3:25 PM
Subject: [Asterisk-Users] Parsing a variable, or rather Splitting a variable
The wiki page on ChanIsAvail says:
Thus, if you are going to use the value of AVAILCHAN, you need to
strip the session ID off.
I understand why this is necessary (from seeing what is returned in
the CLI), but I don't see any commands that can accomplish this since
the session id after the - is of variable length (so you can't use
: notation, or StripLSD, etc.).
I would like to avoid making an external AGI call if possible, and I'm
hoping I'm just missing something obvious here.
Thanks in advance!
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Parsing a variable, or rather Splitting a variable
Perfect! Thanks!
Derek Bruce wrote:
Well, after thinking about it some more... try this:
exten = s,1,ChanIsAvail(SIP/2001IAX2/[EMAIL PROTECTED])
exten = s,2,cut,ToDial=${AVAILCHAN},1
exten = s,3,Dial(${ToDial},20)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec Translation Problem on IAX Softphones - Incoming Only
This is my first post to the list, and while I am sorry that I have a problem that I need to bring to the list, I have been a very satisfied reader/lurker on the list, which has saved me from asking lots of questions so far :-). Apologies in advance for the length... I am new to *, but am already hooked, and have a reasonably complicated setup working nearly perfectly. I have three separate * servers under my control. One is on a static publically available server (no boards in it, so it's more like a switch). The other two are behind NAT'ed routers, and each have two Digium X100P cards in them, with Sipura 2000 connected to the phones for each card. I have ITSP accounts for outgoing with VoicePulse and NuFone as well. I never know where I am going to be, so I set up the servers for multiple ringing. If a call comes in to any of the three * servers, the appropriate line(s) ring in all locations at the same time. This works fantastically well. In addition to this, I badly want to use an IAX softphone as well, since I already stare at my laptop screen for 12+ hours every day, with a headset on (listening to MP3s :-), so it would take less time to answer the call on the laptop than to switch headsets to the phone. Now to the problem... If I use Sipura connected phones, everything works flawlessly, every time. If I initiate a call from the softphone (my current favorite is DIAX, latest version), then all is fine as well. I get in trouble if I _answer_ a call with any IAX softphone, that came in on a real Zap channel. It's most definitely a codec translation problem, but I can dial _out_ from DIAX via a Zap channel, and have that call answered and be able to communicate, so I'm not sure exactly what the problem is. First, the specific error: Mar 8 18:03:20 NOTICE[1209277232]: channel.c:1097 ast_read: Dropping incompatible voice frame on IAX2[apartment]/4 of format GSM since our native format has changed to ULAW I get hundreds of these in rapid succession until I kill the call. Just to pendatically repeat myself, the same exact devices work fine if DIAX initiates the call (even if it ends up passing over a real Zap channel to the Sipura), and if the Sipura initiates the call and DIAX picks it up, the above is the error. Here are the relevant sections of the iax.conf file: [general] port=5036 bandwidth=high disallow=all allow=ulaw allow=alaw allow=gsm allow=iLBC tos=lowdelay ; ; Connect to the apartment * server ; [apartment] context=apartment-in type=friend host=apartment.X.com secret=mysecret auth=md5 [hadar] type=friend context=from-hadar secret=mysecret auth=md5 host=dynamic callerid=Hadar Pedhazur (XXX) XXX- mailbox=100 In the apartment server iax.conf file, I have an identical set of declarations in the [general] section, and nearly identical sections for the devices there (meaning, no one is overriding the codecs defined in general). I have played with a number of permutations of playing with the codecs in general as well as the individual ones, and the above works best for everything except a call _answered_ by DIAX. Any pointers would be greatly appreciated! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
