Re: [Asterisk-Users] openh323 compile for Asterisk
You need to make sure the path to the openh323 and pwlib libs are in your ld.so.conf (or equivalent) file. On Sep 19, 2004, at 4:12 PM, Trevor Morrison wrote: HI, I have the latest RC2 of Asterisk on a RH 9 non-modified-load box. I have an Avaya IP phone that uses h323, so I am trying to compile h323 into Asterisk. Now, I downloaded pwlib and openh323 tar files and I have compiled this according to the instructions: pwlib: ./configure make opt openh323: ./configure make opt cd asterisk/channels/h323 make cd asterisk make clean make install I am getting an error when I start asterisk with the -cccg that it can't find the libpt_linux_x86_r.so.1.5.2 when it tries to load the h323 channel. I have verified that the file does exists in the he pwlib/lib directory and with a size. I have the path to this directory in my profiles PATH and I included the path in my ld.so.conf file. I am missing something but what? TIA, Trevor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect inbound - so do we know how to fix it
All, This is related to the following bug reports: http://bugs.digium.com/bug_view_page.php?bug_id=0002024 http://bugs.digium.com/bug_view_page.php?bug_id=0002017 This is not an iConnect specific problem, but a chan_sip change. As it turns out, type=user does not seem to work in the latest CVS (since June) for authentication against inbound--at least not in the way the documentation describes. What we have resorted to in our offices is have two type=peer contexts in sip.conf defined, the first being outbound, the second being inbound. Believe it or not, the order of these peers matters as chan_sip appears to take the LAST defined context as the authentication peer if you have a static host or dns name in the host= field. So, to summarize: 1) chan_sip.c has changed recently the authentication against type=peer and type=user 2) Registration statements that used to work now need to have a matching peer in sip.conf, however, the documentation states that chan_sip will first match type=user and if none is found type=peer. In real life testing with CVS-HEAD from August 4, this did not work. 3) What did work is creating an outbound peer with the authentication information such as username, secret, etc. (same as before) and a separate inbound peer with just the context,type=peer, host=, and any other codec preferences etc for the inbound leg. 4) This inbound peer has to be AFTER the outbound context otherwise, chan_sip will authenticate against the outbound peer instead of the inbound peer. 5) NOTE that the syntax of the registration statement has changed slightly as well (see wiki) and may need to be modified. On Aug 9, 2004, at 8:57 PM, Sathya Weerasooriya wrote: Raj, yes your post helped me. Just to complete the whole thing and clarify the problem that was posted by Greg Blakely; First, if there is no outbound iconnect section in sip.conf, my incoming calls work fine (as long as my register statements exist in the top section). But, when I add an outbound section, using either 'peer' or 'friend,' my incoming calls begin to fail again with the '407 Proxy Authentication' error. When there is a context created in SIP.CONF for iconnect outgoing, we should point it correctly to extensions.conf. Reason is now the incoming too land in this context. Thanks Sathya -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Raj Sent: Monday, August 09, 2004 5:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iconnect inbound - so do we know how to fix it May be you can find the solution in my post: http://lists.digium.com/pipermail/asterisk-users/2004-August/ 058014.html Raj --- Vladyslav [EMAIL PROTECTED] wrote: Try to comment out in your sip.conf ;qualify=yes On Mon, 2004-08-09 at 06:52, Sathya Weerasooriya wrote: Just wondering whether we have a resolution to iconnect incoming problem, which started few days ago. Cheers SW -- Best regards Vlad ___ 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 __ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nested/cascading switch statements: possible?
John, If I remember correctly, we accidentally did something like this and caused an infinite loop between a few of our servers (resulting in an IAX Out of Trunk Space message), but if Server C doesn't point back to Server A, then it should work... On Jul 26, 2004, at 9:30 AM, John Todd wrote: I'm lazy and decided that I didn't have the time to hack up a few servers for testing, which I'll probably have to do if nobody answers this mail, but it's late in this part of the world and I'm tired. :-) I'm wondering if the use of the switch statement can be nested, such that system A has a switch statement that points dialplan inclusion to server B, and server B has in that context a switch statement which points to server C. Therefore, server A knows about extensions on server C only indirectly, through server B, though the media path of course may actually end up going directly A-C (if it's IAX or SIP.) Google, the Wiki, nor my mail archives had anything that obviously matched my search keywords, though I can swear I've seen this question on the list before. It sounds too good to be true, and I know what that means... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] High latency from Europe, 500-800ms.
You are right to suspect codec issues here. What codec are you using at the various endpoints? Make sure that the Asterisk box is set up with the correct codecs in the conf files, otherwise it will try to transcode and this will often cause bad audio quality like you mentioned. If you're using G729, make sure that you don't have any rTt or other options in dial enabled, otherwise, Asterisk will proxy the media. I've read in previous threads that the jitter buffer is broken in iax and we tried with and without and it was much better without. On Mar 19, 2004, at 8:57 PM, [EMAIL PROTECTED] wrote: Maybe I'm not articulating myself well. The 7940 on the same network in Europe *works great*, no problems, sound is perfect, even with the higher latency. If I take that 7940 and have it connect to a *local* Asterisk server, which connects to the states, it sucks. The 7940 though, connecting directly to the states, works great. Bill Not all sat connections are one way. But the issue with sat connections is *drumroll* latency! As the signal is beeing relayed over the sattelite this will cause latency. Also if the sat service is not providing enough downstream it's bad too. I would definately look into getting your network straighend out first. There are many factors. Is your connection shared? What speeds? Let say it like that if you have people on your local lan using bandwith or running peer 2 peer filesharing stuff this will take away your upstream speed. Do some tests. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323 translator
Hi, The patches also did not help us and in fact created some new problems. The old chan_h323 could pass on early audio and provider messages, but after the patch, this capability is gone and the channel only rings and rings while the provider is sending the message. We've had no problems with the existing chan_h323 other than that it doesn't return the right indication state to Asterisk, so Asterisk can't branch for busy versus congestion. But this is obviously only for our setup. On Mar 19, 2004, at 9:12 AM, Marian Durkovic wrote: On Thu, Mar 18, 2004 at 12:22:57PM -0500, Billy Huddleston wrote: I just tried this, and it's not working for me.. I can't call a 2600 or a CCM... What version of OpenH323 and PWLIB did you all use? Are you able to call those without the patches? If not, the patches won't help you, since you probably have some other problem.. M. - Original Message - From: Marian Durkovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 18, 2004 10:35 AM Subject: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323 translator Hi all, in an effort to create a SIP - H.323 translator we've found and fixed several problems in H.323 channel. These inlcude: for SIP-H.323 calls - no ringback tone - ringback not related to H.323 events - one-way audio with Cisco CallManager - incorrect Caller ID for H.323-SIP calls - not able to establish call with Cisco IOS 12.3(4)T - ringback not related to SIP events - no support for 183 Call Progress - incorrect Caller ID Please find the patches against aterisk 0.7.2 release below. M. - - Marian Durkovic network manager Slovak Technical University Tel: +421 2 524 51 301 Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351 812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED] - - --- --- Marian Durkovic network manager Slovak Technical University Tel: +421 2 524 51 301 Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351 812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED] --- --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Sipura click click bad quality
See http://bugs.digium.com/bug_view_page.php?bug_id=0001195 and also http://bugs.digium.com/bug_view_page.php?bug_id=0001220 On Mar 16, 2004, at 1:18 PM, Senad Jordanovic wrote: Miguel Cavazos wrote: if it was related to the dsl line i would notice my other phones such as grandstream and the ones on zap cards with the same problem im only having this issue with sipura. Sure... When did this start. I am using sipura devices with no such problem. (I have other problems with it though ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 call return code handling
Hi, Has anyone out there had any luck with channel return codes with chan_h323? It seems that the h323 return codes are in the channel driver for for h.323 debug messages, but for some reason, there is no distinction between busy and congestion returning to Asterisk, so it's not possible to tell if the call terminated because of the line was busy or because the call was unsuccessful for another reason. Perhaps I'm missing something. Otherwise, the chan_h323 works like a charm. If anyone has any experience with distinguishing between busy and congestion (or for that matter any other return code), I'd be most grateful for some pointers. Thanks! ___ 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] H.323 and G729: Another sad tale
THERE IS AN INCONSISTENCY IN THE README FILE THAT IS OUT OF DATE: Follow the instructions on line below and do NOT issue a make clean install in asterisk/channels/h323 as indicated elsewhere, just issue a make and then in /usr/src/asterisk (or wherever you source is), issue a make install and this will work. To compile this code: Issue a make in the asterisk/channels/h323 directory, then go back to the Asterisk source top level directory and issue a make install. On Friday, November 7, 2003, at 02:51 AM, Anton L. Kapela wrote: Jeremy McNamara said: 2. /openh323, make clean, make opt 3. /asteriks/channels/h323, make clean, make install, and it is got error about no chan_h323.o exists. and the make install is failed. You haven't read the README And I quote: To compile this code: Issue a make in the asterisk/channels/h323 directory, then go back to the Asterisk source top level directory and issue a make install. I suspect that he indeed did read the README. In fact, I just (for kicks) cvs checked out a fresh copy of libpri, zaptel, and asterisk this evening. I checked my h.323 channel README, and I see: Example commands to make sure everything gets cleaned and then rebult in proper order: cd /path/to/pwlib make clean opt cd /path/to/openh323 make clean opt cd asterisk/channels/h323 make clean install For some reason, doing as instructed on various sites (digium.com, for one) you'll be pulling stale CVS code. Or, for some reason, your updates to what's in CVS aren't actually working. I wish I understood CVS more to better research this. --Tk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
I can also confirm chan_h323 and g.729 work well to 5300s, but we had to build on RH8 not RH9. Haven't tried 5300 to Asterisk except via SIP which works fine--even to i4l interfaces. On Friday, October 31, 2003, at 01:57 AM, Jeremy McNamara wrote: John Todd wrote: I've done some reviewing of the archives for G729 and H323 experiences. The landscape of that query isn't pretty - lots of pleas for help, and nor do I see too many answers. I have a pending bid that requires some data before I can implement * on this particular solution. My question is perhaps a slightly differently worded one than has been asked before, but it may be the case that it is the same question as I have seen already posted (with no 'definitive' answer): Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be terminated on Asterisk systems and sent out Zap interfaces? Yes, g729r8 If the answer is Yes, then are there any specific patches I will need? Which of the two H323 drivers works? Both? Of course, I assume that the G729 licenses from Digium are required for each active channel. Others seem to have massive issues with chan_h323 and G.729, but i've dealt a dozen or so 5300s of which I haven't had any trouble whatsoever, with nothing other than the code that is currently in the cvs. However, I have only terminated calls from Asterisk to the 5300, never from the 5300 to Asterisk. If Asterisk is going to be encoding G.729, yes you will need licenses from Digium. Jeremy McNamara P.S. I'm biased and cannot comment about that other driver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:no sound after 5sec
Our experience with the Budget Tones 101have been poor as well. The devices seem to die after a day or two (even with the new firmware) and then need to be rebooted. On occasion, the device needs to be literally unplugged and plugged back in as even the reset doesn't work. There are some nice features, but we have all but given up on them for a production environment. Relative to the Cisco ATAs and other devices we are using, the price/performance ratio is not there, particularly from a support cost perspective. If they get the thing to be more stable then we will reconsider them. On Wednesday, October 15, 2003, at 07:00 AM, rnc Info Lists wrote: Do you have a 100 or 101? You have indicated different models in your postings. Were you able to get Call Transfer and Call Waiting working with your Asterisk system and other phones? Which version of the Grandstream firmware do you use? There most recent on their website this weekend was at least 2 version numbers higher than what came on my phone in August. Think that they are making improvements rather frequently. Robert On Wed, 15 Oct 2003, Jon Pounder wrote: The Grandstream 101 I'm using is a piece of junk but I don't have the same problem with it. What don't you like about the grandstream ? (I am not looking to flame you, but was considering buying and if there are problems would rather find out beforehand) Nothing works. Call transfer and call waiting, in particular. (Well, almost nothing; vm notification does work) There is no place to plug in a headset, and since I do a fair amount of tech support and longish conference calls, that's a big deal for me. However, keep in mind that I have an old, no-longer-manufacturered model (the Budgetone 100). Don't take my frustration with my outdated phone as a sign that you should dismiss Grandstream out of hand - I just don't like my 100. -- JustThe.net Internet Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] register = w/MD5?
You have to specify it in sip.conf in the entry for the UA/proxy for which you want to register with. For example, if you want to register with x.com sip.conf [Any alias name here for x.com] hostname=x.com auth=md5 On Friday, September 12, 2003, at 08:47 PM, Eric Wieling wrote: Does anyone know how you specify MD5 auth on a register = line? -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming calls from IAXTEL over NAT
Don't know if this will help, but have you seen: http://www.junghanns.net/asterisk/page12.html ? On Friday, September 12, 2003, at 03:06 AM, Jamie Carl wrote: Hey all, I was playing around with IAXTEL last nite and have outgoing calls working a treat. I'm sure I woke a few people up in the US with my annoying test calls. :) Anywayz, incoming calls are a different matter. I have a NAT firewall my * box is sitting behind and the server 'appears' to have registered correctly with IAXTEL. Thing is, when I try and call my 1700 number to 'loop' back to myself, I get no incoming requests. I've also tried calling the 1700 number from FWD using X-Lite (which works to other numbers) and I get no incoming requests either. Just Marks voice telling me the person i'm calling is either unregistered or unavailable. Is there anything I need to do to my firewall to get incoming calls to work with IAXTEL? Or should it just 'work'. I've also tried forwarding my IAX port (5036) to my * box without success. Anyone done something similar that can point me in the right direction? Thanx... Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 + SIP
Hi Shaun and anyone else looking for Cisco images, I don't know what the support contract would cost on a 7960 for the Cisco TAC, but for the ATA186 it's a great, big $8/year. This gives you access to the Cisco TAC, images, and support team which do a fantastic job of follow-through. So I would recommend calling them and asking about the support contract for your particular phone. If you're too cheap to pay the fees, you can always find a service provider that supports the device and do a tftp cross-grade, upgrade, etc. of the firmware. iConnect, Nikotel, etc all support the ATAs, but not the 7960s. The only catch is if you bought the device second-hand. Then there is the chance that your device is ineligible for support. On Friday, September 12, 2003, at 08:11 AM, Shaun Ewing wrote: Hello again, After doing some searching of the list archives, I came across a message by John Todd posted back in July () To cut a long story short, to be able to use SIP on my phone, I need to P0S30203.bin image. Is there anyway of getting this image without getting a Cisco SMARTnet agreement? -Shaun - Original Message - From: Shaun Ewing [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 12, 2003 3:58 PM Subject: [Asterisk-Users] Cisco 7960 + SIP Hello all, I know this isn't strictly Asterisk, but I'm sure that there are more people here using the Cisco 7960 w/ SIP, so I thought I'd post here. I've just bought a Cisco 7960 phone to use with Asterisk. It came with the CallManager image on it. I've got the 4.4 SIP images (P0S3-04-4-00). If I put P0S3-04-4-00 in the OS79XX.TXT file, the phone downloads this fine (watching TFTP server debug). It then proceeds to request P0S3-04-.bin. I don't know why. Naturally this file isn't found. I tried renaming the file to P0S3-04-.bin. The phone then downloads around 80% before aborting. I hope somebody might be able to shed some light on the situation. Any help would be greatly appreciated. Thanks, Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New cvs compile; basic operational question, please.
Make sure that both phones are set to accept the same codecs. The Not
Acceptable Here is usually when the SIP negotiation fails for a common
codec.
Use SIP DEBUG at the CLI to see the transcripts for details. You might
want to use in sip.conf allow and disallow statements for codecs as
well.
On Sunday, September 7, 2003, at 07:04 PM, Rich Adamson wrote:
Can someone offer a hint on what I'm doing wrong with the basic *
config?
Just implemented * for the first time using yesterday's cvs. The
initial
configs are based on John Todd's article at
http://www.onlamp.com/lpt/a/3956,
and using two 7960's for initial testing. When one 7960 calls the
other, I
get the following and the call is dropped:
-- Executing Dial(SIP/3001-ec1c, SIP/3000|20) in new stack
-- Called 3000
-- Got SIP response 488 Not Acceptable Here back from
206.222.193.92
== No one is available to answer at this time
-- Executing VoiceMail(SIP/3001-ec1c, u3000) in new stack
== Parsing '/etc/asterisk/voicemail.conf': Found
-- Playing 'vm/3000/unavail'
My sip.conf looks like:
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about
here
[3000]
type=friend ; This device takes and makes calls
username=3000 ; Username on device
secret=npi2003 ; Password for device
host=dynamic; This host is not on the same IP addr every
time
context=from-sip; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
; voicemailbox has messages in it
[3001]
type=friend ; This device takes and makes calls
username=3001 ; Username on device
secret=npi2003 ; Password for device
host=dynamic; This host is not on the same IP addr every
time
context=from-sip; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
---
and my extensions.conf looks like:
[general]
static=yes ; These two lines prevent the command-line
interface
writeprotect=yes; from overwriting the config file. Leave them
here.
[bogon-calls]
exten = _.,1,Congestion
[from-sip]
exten = 3000,1,Dial(SIP/3000,20)
exten = 3000,2,Voicemail(u3000)
exten = 3000,102,Voicemail(b3000)
exten = 3000,103,Hangup
exten = 3001,1,Dial(SIP/3001,20)
exten = 3001,2,Voicemail(u3001)
exten = 3001,102,Voicemail(b3001)
exten = 3001,103,Hangup
exten = 3999,1,VoicemailMain(${CALLERIDNUM})
Apparently I'm doing something wrong, but since this is my first
attempt
at making * work, I don't actually have a clue what I'm doing (yet).
Asterisk did complile and install clean the first time (on new RH9
system),
and both 7960's are registered. In some attempts to dial, I do receive
vmail announcements, etc, so whatever I've done wrong I'm guessing it
must
be in the above config statements.
If someone would kindly point out my error (and maybe a constructive
comment
about the error so I can learn), if would be greatly appreciated.
TIA,
Rich
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Re: [Asterisk-Users] RE: Asterisk stops responding
Update to latest CVS and check the bug report that I filed re:DTMF. Your problem could be related. Latest CVS seems to fix the blocking problem for me. On Friday, September 5, 2003, at 01:15 AM, Andres wrote: It happened once again here. This time I called an IVR (SIP to SIP) and upon sending the 1st DTMF tone, * bombed out. The console got filled with these messages (and they wouldn't stop): DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, trying again... DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, trying again... DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, trying again... DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, trying again... DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab lock, trying again.. * stopped responding and I had to kill the process manually. *CLI show version Asterisk CVS-08/22/03-22:24:05 built by [EMAIL PROTECTED] on a i686 running Linux Has anybody else seen this message? Regards, Andres On Thursday 28 August 2003 13:37, Andres wrote: We run Iptel's SER as our SIP Server. All subs register with our SIP Server, but if anyone needs to call the PSTN then the call gets forwared to *. The Request to schedule in the past messages have to do with MOH and I was told it was due to a slow PC. I don't think it is related with Asterisk hanging up. Regards, Andres On Thursday 28 August 2003 13:27, David Harris wrote: Gazing at the console I was able to determine the exact time Asterisk froze. Even with DEBGUG on it did not show anything important. The moment it freezes is when a call from Phone1 tries to connect to a SIP Provider like Iconnect: I have not been able to pin point exactly what event causes the freeze-up but I have been on the console when it has happened. It didn't print out anything interesting. The call I was on cut off. Phone1Our SIP Server---Our AsteriskSIP Provider It was by no means 100% reproducible. Maybe 1 out of 10 calls caused the trouble. Same here except I would say more like 1 out of 100 calls. A bad symptom would be that the command show sip channels would show several calls, even though they had hungup a long time ago. I definitely have this problem. Troubleshooting revealed that the BYE message was not being sent by our SIP Server to the Asterisk server upon hangup. We rectified this and we no longer see those phantom SIP Channels and Aterisk has not froze for about a week. What is your SIP Server what does it do? Maybe I have the same issue with my Cisco Voice Gateway not sending the BYE message sometimes. But would this cause asterisk to freeze? Other symptoms I have are these errors in the asterisk messages log file Aug 27 09:21:00 NOTICE[1081364]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Aug 27 09:21:24 NOTICE[1081364]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Aug 27 09:21:29 NOTICE[1081364]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Aug 27 09:21:35 NOTICE[1081364]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Aug 27 09:22:05 NOTICE[1081364]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Thanks, David Harris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Tones During Call
Is this an ISDN4Linux card? If so, search the archives and you'll see postings and patches on how to correct it. On Wednesday, September 3, 2003, at 03:48 AM, Jay Tyndall wrote: Hi, I am receiving calls via a Netjet-S card on asterisk, and I notice that whenever I am talkimng to someone, if their voice is loud enough, sometimes asterisk generates a DTMF Tone as they speak. that is played to me. (Caller doesn't hear it). Any ideas how to stop this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ENUM, iax,iax2 and h323?
Yes, keep up the good work! On Sunday, August 31, 2003, at 09:24 AM, Brian West wrote: I have added support for enum looks for iax,iax2 and h323. So far in my testing it has worked perfect. (note: you need to strip the + for iax and iax2 calls or they will fail. h323 will accpet the + but I striped it in the below example) *.1.enum.bkw.org. IN NAPTR 2 42 u h323+E2U !^\\+(.*)$!h323:[EMAIL PROTECTED] . Also you could do something like this for iax2: *.8.1.9.1.enum.bkw.org. IN NAPTR 2 42 u iax2+E2U !^\\+(.*)$!iax2:[EMAIL PROTECTED]/\\1! . (gateway would be setup in iax.conf) Would anyone else like to see iax, iax2 and h323 enum lookup support in *? here are my diffs: http://www.bkw.org/~brian/app_enumlookup.diff http://www.bkw.org/~brian/enum.diff Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] App Directory issues-again?
Hi, I've seen some postings on the Directory application, but haven't seen too many resolution postings. Has anyone experienced where the Directory app doesn't even answer when called? For example, using the config below, dialing 899 results in just a continual ringing sound. extensions.conf exten = 899,1,Directory(local) exten = 899,2,Hangup [local] exten = 8000,1,Dial(SIP/8000) ..various extensions defined voicemail.conf [local] 8000 = 1234,John Morris,[EMAIL PROTECTED] ...etc... Is there a config problem or are others having this issue too? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail2 patches
I haven't tried the patches, but they sounds very useful! My 2 cents... BTW, there have been some recent bug fixes to Voicemail2, so you might want to test them against recent CVS (8/16 or later) On Sunday, August 17, 2003, at 11:16 AM, Brad Bergman wrote: Yeah, I haven't really had time the last couple of weeks to follow up on this. I'd be happy to have any of the patch in CVS if that were so desired, but before heading down that road I'd be curious to know if anyone besides me has had a successful time actually using the patches. At the very least there is a bit of fine tuning for me to do in the patch, and I don't have matching prompts, just my own voice for now. Brad On August 16, 2003 02:13 pm, Brian West wrote: A few weeks ago Brad posted his patches to the mailing list: http://www.universaltime.org/~brad/vmail/ But I can't find his email address... does anyone happen to have his address. I hope he would be willing to see if Mark wanted to add those options to the CVS. I think you need to fill out a disclaimer and post it to bugs.digium.com bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN
Stefano, I've come across this problem as well using SIP devices and asterisk. As far as I can tell, the IVR systems are deliberately not answering in order to not pay the telco for call charges. Ironically, they are sending audio before they answer the call. Depending on what device you are using, you may or may not receive the audio on your phone. For example, using a Cisco ATA186 via SIP and Asterisk, I can dial pretty much any IVR system and even if they don't answer, I can hear the audio. However, passing DTMF is another issue. The only reliable connection type that I've found is using IAXTEL. IAXTEL successfully passes DTMF to IVR systems (using the ATA186 via SIP) before the call is answered. Using other VoIP systems (Cisco GW, local alternative telco calling card, etc.) I've not been able to pass DTMF before the call is answered. There is a kind of chicken and egg problem here. The IVR system doesn't answer the call until after the user has selected the first level menu prompts. However, if it is only a DTMF system (no voice recognition) and the user can't pass DTMF, then the call will never go answered and time out. Using the same connection methods and different SIP devices, however, yields different results: 1) Grandstream Budgetone 102 - According to their tech support, early audio is not supported in the current firmware. The phone keeps ringing and ringing. 2) Voicefinder GW - Using my office *, early audio is successfully processed and with IAXTEL, DTMF can be passed. However, using the EXACT SAME configuration of Asterisk (same CVS, same drivers except i4L) at a machine hosted at a data center, no audio, just ringing and ringing. I suspect this is a NAT issue, but the Cisco ATA works in this exact configuration and NAT. 3) Same symptons for 2 other ATA devices. Early audio works in one case, not in the other. Interoperability of equipment, codecs, etc. all add up to some things not working correctly in certain cases. I guess this is why VoIP hasn't become mainstream--yet... On Wednesday, August 6, 2003, at 05:10 AM, James Sizemore wrote: Tones are to short. Stefano Finetti wrote: Mark, I'm now able to send proper DTMF tones checking on the isdn driver and using rfc2833 as dtmf mode for sip.conf and phones. But there is a question that i think only you can check and answer: Why * often when calling via outside line some number that has IVR systems, doesn't recognize the answer? It stays there, waiting, even if i'm sure the other side of the line has answered the call (tried in the same time from * and using my mobile phone). I can't figure out what kind of problem can be, I encountered it in many * installation... -- Stefano Finetti ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Fwd: FW: [Asterisk-Users] SIP NAT question
Just in case other people on the list have this problem... Begin forwarded message: From: George Lin [EMAIL PROTECTED] Date: Thu Aug 14, 2003 6:54:46 AM Europe/Budapest To: Paul Cheng [EMAIL PROTECTED] Subject: RE: FW: [Asterisk-Users] SIP NAT question Dear Paul, Thanks for the suggestion. It works now. Thank you very much. George Lin -Original Message- From: Paul Cheng [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 13, 2003 2:54 PM To: George Lin Subject: Re: FW: [Asterisk-Users] SIP NAT question What kind of router do you have? That makes a huge difference! Try the qualify first and the restart Asterisk and wait for the SIP UAs to register. Then run Asterisk in command line (asterisk -cr) and do a sip show peers. You should see each UA and then their status (hopefully they say OK (x ms)). Now try dial each extension to see if that worked. If the problem still exists, then e-mail me again with your router type and we can go from there. On Wednesday, August 13, 2003, at 11:58 PM, George Lin wrote: Dear Paul, Thanks for the note. SO what should I configure the router at my office router ?? I will add qualify=yes in each entry at sip.conf. In our case, we already shutdown the firewall, only the NAT. for such case, what should we configure the router ? what is your experience with your router ?? Thanks, George Lin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Cheng Sent: Wednesday, August 13, 2003 1:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP NAT question Hi George, Do you have qualify=yes set in sip.conf for your phones? When you check sip show peers, does it give you an OK (X ms) or does it say UNREACHABLE or UNMONITORED? If you enable qualify=yes or qualify=[some number] then Asterisk will poll the SIP UA every once in a while to make sure it is still reachable. This may or may not work. In some cases, if the UA doesn't support the SIP OPTIONS correctly, it will come back and Asterisk will think it is unreachable until it sends another register command. In other cases, it helps keep the ports open on the firewall. BTW, we have successfully tested NAT with multiple user agents as you describe with pretty much plug and play with Linksys, SMC, Shorewall/Linux and various other NAT router/fw devices with great success. Thus far, we've only had problems with DrayTek routers mangling the UDP packets. In those cases, the UAs registered successfully and all inbound calls worked, but outbound calls did not as the UDP/RTP streams weren't getting handled correctly by the router. They have an updated firmware that solves this problem, but we haven't finished testing it. On Wednesday, August 13, 2003, at 09:25 PM, Adams, Gavin wrote: From: George Lin [mailto:[EMAIL PROTECTED] I want to deploy multiple SIPs phone in our office. And we have shutdown the firewall at our office router(with ip 211.x.x.x). we have deployed the asterisk with IP 218.x.x.x. All SIP phones have 192.x.x.x. We have something similar George, * sits outside the firewall with a registered IP address, the SIP phones sit behind the firewall with 172.16.x.x addresses. When the SIP phone is power on, they are registered in the asterisk. we can check at asterisk side by issueing sip show peers, and all the phones are associated with 211.x.x.x:port-number. Sounds familiar. Question, do you hide all the phones behind a single IP address, or does each phone get a unique address? Also, what type of firewall? PRoblem: Now some times the sip can receive call, and some time it cannot recieve call. When we dumping the sip log, and see that asterisk tried to INVITE the specified SIP phone with the 211.x.x.x:port-number, and was failed after 5 times. But the call orginated from SIP phone is always OK. Yup, what we initially found. Basically, we started by attempting to hide all the phones behind a single IP address. In this case, make sure you uniquely assign the control port (by default UDP 5060) to something different for each phone. We use FireWall-1 (older version) that doesn't play nice with hide NAT. Basically, it would timeout UDP connections after 40 seconds of no activity. Not good unless you reduce the reregister time to something crazy like 30 seconds. Check to see how your firewall/NAT device handles [P]NAT translation. Questions are: 1. Does asterisk remember the mapping between 192.x.x.x AND 211.x.x.x:port-number ? It shouldn't. It might see the 192.x.x.x address in the SIP conversations, but even if it did, it would not be able to route the packets back. 2. When a call to a sip phone, is it asterisk responsiblility to map the 211.x.x.x:port-number to the 192.x.x.x, and send to the office router ? OR it is the office router to remeber all the mapping between each sip phone 192.x.x.x and 211.x.x.x:port-number, and asterisk juts sends the 211.x.x.x:port-number to the office router ?? Asterisk should attempt to contact the phone based upon
Re: [Asterisk-Users] SIP NAT question
Hi George, Do you have qualify=yes set in sip.conf for your phones? When you check sip show peers, does it give you an OK (X ms) or does it say UNREACHABLE or UNMONITORED? If you enable qualify=yes or qualify=[some number] then Asterisk will poll the SIP UA every once in a while to make sure it is still reachable. This may or may not work. In some cases, if the UA doesn't support the SIP OPTIONS correctly, it will come back and Asterisk will think it is unreachable until it sends another register command. In other cases, it helps keep the ports open on the firewall. BTW, we have successfully tested NAT with multiple user agents as you describe with pretty much plug and play with Linksys, SMC, Shorewall/Linux and various other NAT router/fw devices with great success. Thus far, we've only had problems with DrayTek routers mangling the UDP packets. In those cases, the UAs registered successfully and all inbound calls worked, but outbound calls did not as the UDP/RTP streams weren't getting handled correctly by the router. They have an updated firmware that solves this problem, but we haven't finished testing it. On Wednesday, August 13, 2003, at 09:25 PM, Adams, Gavin wrote: From: George Lin [mailto:[EMAIL PROTECTED] I want to deploy multiple SIPs phone in our office. And we have shutdown the firewall at our office router(with ip 211.x.x.x). we have deployed the asterisk with IP 218.x.x.x. All SIP phones have 192.x.x.x. We have something similar George, * sits outside the firewall with a registered IP address, the SIP phones sit behind the firewall with 172.16.x.x addresses. When the SIP phone is power on, they are registered in the asterisk. we can check at asterisk side by issueing sip show peers, and all the phones are associated with 211.x.x.x:port-number. Sounds familiar. Question, do you hide all the phones behind a single IP address, or does each phone get a unique address? Also, what type of firewall? PRoblem: Now some times the sip can receive call, and some time it cannot recieve call. When we dumping the sip log, and see that asterisk tried to INVITE the specified SIP phone with the 211.x.x.x:port-number, and was failed after 5 times. But the call orginated from SIP phone is always OK. Yup, what we initially found. Basically, we started by attempting to hide all the phones behind a single IP address. In this case, make sure you uniquely assign the control port (by default UDP 5060) to something different for each phone. We use FireWall-1 (older version) that doesn't play nice with hide NAT. Basically, it would timeout UDP connections after 40 seconds of no activity. Not good unless you reduce the reregister time to something crazy like 30 seconds. Check to see how your firewall/NAT device handles [P]NAT translation. Questions are: 1. Does asterisk remember the mapping between 192.x.x.x AND 211.x.x.x:port-number ? It shouldn't. It might see the 192.x.x.x address in the SIP conversations, but even if it did, it would not be able to route the packets back. 2. When a call to a sip phone, is it asterisk responsiblility to map the 211.x.x.x:port-number to the 192.x.x.x, and send to the office router ? OR it is the office router to remeber all the mapping between each sip phone 192.x.x.x and 211.x.x.x:port-number, and asterisk juts sends the 211.x.x.x:port-number to the office router ?? Asterisk should attempt to contact the phone based upon the IP and port seen during a 'sip show peers'. Network device responsible for any and all translations. 3. If it is the office router's responsiblity, what should we configure the office router even there is no firewall??? Unsure about this, I'd focus more on the NAT device. Can you describe the topology from the SIP phone to *? Regards, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inbound SIP DTMF detection
Hi, We have our * box configured to receive inbound SIP calls from FWD and enter into an autoattendant where someone can enter an extension directly. However, the inbound DTMF is not being correctly detected in most cases. Entering 8050 results in a correct detection, but enter 9003 results in only 90 being detected. Entering 700 results in 70 being detected. 799=79, etc. It appears that repeated digits do not get detected. 8750 works, but 999 results in only 9 being detected. Has anyone else experienced this problem? We are using X-lite as the client on the FWD side to test. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN
I've also got this problem over ISDN BRI using i4l. On Monday, August 4, 2003, at 09:17 AM, Stefano Finetti wrote: - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 03, 2003 5:52 PM Subject: Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN Are you experiencing it over PRI ? Can you send the pri debug span spanno trace ?Is your asterisk/libpri code very recent ? I'm experiencing this both over a PRI line (E1), with july CVS, and over a normal ISDN BRI line, with latest CVS sources (taken about a week ago). I'v tried to debug both SIP and using messages (/var/log/asterisk/messages) but i found no useful informations. It's quite important to solve this problem 'cause i'm not able to call some *very* important number used for my job (Telecom HelpDesk, and so on). Thanks, -- Stefano Finetti ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Paul Cheng Mátyás király ut 10 H-1121 Budapest HUNGARY [EMAIL PROTECTED] mobile: +36 30 381-9311 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need help with DTMF issues and IVR systems
Hi, With Asterisk and SIP phones (Cisco ATA186, Grandstream BT102), I'm having an issue with DTMF passing correctly to IVR systems like customer support phone numbers, voicemail, etc.: 1) If I set DTMF to SIP INFO, DTMF works for ISDN4LINUX calls to IVR systems with the CiscoATA186, but not with the Grandstream (this could be a bug) -- BUT the Asterisk Voicemail doesn't detect the DTMFs. SIP calls to gateways and other SIP phones work fine. 2) If I set DTMF to INBAND, there are problems with double-detection and it doesn't work with LBR codecs. However, in this mode, calls to IVR systems via I4L work with the Grandstream. 3) If I set DTMF to rfc2833, voicemail works, but ISDN4LINUX calls don't pass the DTMF to IVR systems from either the Cisco or the Grandstream. Has anyone had any luck getting one setting to work across all channels? The problem is not in the initial call, but usually after the call is connected, the DTMF doesn't get passed. What baffles me is why SIP INFO doesn't work with Asterisk AppVoicemail. Any hints, suggestions, experiences would be greately appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Strange behavior in latest CVS
Hi All, A week ago, we had Asterisk working very stably with SIP, I4L (ISDN BRI) with a passive Eicon card, chan_h323, g729, etc. However, if Asterisk ran for long periods of time without a restart, there would be a build-up of unterminated SIP channels without the ability to do a soft hangup. Restarting periodically would solve the problem. These artifact channels didn't really have an impact on Asterisk as far as we could tell. After we downloaded the latest CVS over the last two days, the sip artifacts don't seem to be there anymore, but if * runs for a day or two, the g729 codec support starts exhibiting strange behavior and even if there is a free licensed channel, it will refuse the call. Restarting fixes this problem. In addition, our chan_h323 no longer works. Inbound voice works, but outbound no longer works. Has anyone seen this behavior with the g729 codec or h323 with the latest CVS? We're using RH9 k2.4.18. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please help -- Syntax for dialing VoIP provider
Hi BK, Using your configuration info, I now have Nikotel working again. Other than the fromuser=, it appears that one also now needs the auth=md5 whereas before it was not necessary. To disable incoming calling, just delete the register - line for Nikotel. That way, no one can find you. You do not need to the register - line for outgoing calls. On Monday, July 7, 2003, at 11:16 AM, BK [address only for mailing lists] wrote: Hi Paul, thanks for your insights On Monday, July 7, 2003, at 03:59 PM, Paul Cheng wrote: To dial a PSTN number through Nikotel used to work from Asterisk, but they had a very serious security issue (you could make calls anytime anywhere and their billing wouldn't charge it) and after I informed them of this, they changed their authentication mechanism and since then I have not gotten it to work (they didn't even thank me!). This is what we have discovered last night. However, We have got it working now. I will document this in detail and make it available, but briefly here a quick summary ... First I had various glitches in my dial string. With the help of John Todd and some others on the IRC #asterisk channel I was able to fix those glitches. Thanks everybody who assisted. Then I tried a number of things I had already experimented with before. When I turned on SIP debug and watched the datagrams, I could see Nikotel's response account name does not match address of record. Together with the from part, this led me to fiddle with fromuser again and when I set it to the actual login name, it worked. Their tech people said it should work with a slight change: yes, we changed it yesterday. Now the user part of the From: address has to be the same as the username in the Proxy-Authentication line. I don't know if the Asterisk can do that. The ATA186 does it b[y] default. This CAN be done if you edit chan_sip.c, It would seem you can do it a lot simpler: in sip.conf --- -- register = myusername:[EMAIL PROTECTED] [nikotel] username=myusername fromuser=myusername ... --- -- but when I did this, it billed me a few times for unconnected calls Thanks for sharing this with us. I will watch this for a while and see if this happens here too. and I gave up trying to debug and switched to iConnect. iConnect is worse quality, but it is very easy to connect to. I had much better quality with calls via Nikotel than iConnect, but their support is non-existent/bad at best. I sent them 3-4 e-mails about their security issue before they even responded. Yes, support is not exactly their strength, is it?! FYI. Registering with Nikotel was futile anyways, because I never figured out how anyone could call into me. I don't want anybody to call in via Nikotel. Since they do not provide a telephone number for incoming calls, the only calls you could possibly get are from their public chat room. In the very best case you get a friendly test call from somebody who has just signed up and wants to try out the service, in the worst case you get prank calls in the middle of the night or indecent proposals and all the rest of it. I will have to find a way to disable incoming calls from Nikotel entirely. iConnect provides a PSTN-SIP dial in as an option, but I haven't tried it. Yes, I have seen that. And at $8.95/mth it would seem reasonably priced, too. Outbound calls do not require registering. I can provide examples of iConnect connection scripts if you contact me offline. Thanks, I will do that. again many thanks to everybody who has helped solving this riddle rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Paul Cheng Mátyás király ut 10 H-1121 Budapest HUNGARY [EMAIL PROTECTED] mobile: +36 30 381-9311 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXTEL toll-free
Hi, Has anyone been able to place a call via IAXTEL toll-free termination lately? I had it working at one time, but now it doesn't seem to work anymore. www.iaxtel.com also appears dead. Is this the server problem again or is it my config? Haven't been able to find any references in the list. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does anyone know how to get rid of this warning message?
Hi, I searched the archives about this, but didn't find any references. When I make an outbound SIP call, the call completes and everything is fine, but in the Asterisk console, I keep getting a huge stream of warning messages: WARNING[1200876848]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames I thought I saw this in a post earlier, but I don't get it in the search. Does anyone know what needs to be set to stop these? Thanks in advance. --- Paul Cheng Mátyás király ut 10 H-1121 Budapest HUNGARY [EMAIL PROTECTED] mobile: +36 30 381-9311 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
